From jmesquita at freeswitch.org  Fri Jun  1 00:10:04 2012
From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Thu, 31 May 2012 17:10:04 -0300
Subject: [Freeswitch-users] FSClient a Windows FreeSWITCH softphone
 Update Released
In-Reply-To: <CALn233rbxX1hS2KHRXh961ee5S7jOCogpsAwKrd8fi-i9i-FFQ@mail.gmail.com>
References: <CALn233rbxX1hS2KHRXh961ee5S7jOCogpsAwKrd8fi-i9i-FFQ@mail.gmail.com>
Message-ID: <C9477E2482C641E19003B37B1843DC52@freeswitch.org>

Mitch, I would like to express a big public congratulations to you.

You have pulled something that I failed (so far) to do. FSComm is still on my plans, it is just "on hold" and to be completed. Still, I know from the little bit I've done that it is not an easy task. Once again, congratulations man.

Regards,  

--  
Jo?o Mesquita
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Tuesday, May 29, 2012 at 11:28 PM, Mitch Capper wrote:

> An update had been posted for FSClient is a windows .NET softphone
> that uses libfreeswitch at the core.
> Info page at: http://wiki.freeswitch.org/wiki/FSClient
> Direct binary download:
> http://files.freeswitch.org/windows/installer/x86/FSClient.zip
> And source is in contrib repo under MitchCapper/FSClient
>  
> Some of the changes include:
> Built against FS 1.2 RC
> isac codec support
> for contacts you can now right click and choose what account to call from
> if sofia goes offline update the accounts status
> Jabra headset improvements to avoid some crash conditions
> background task improvements to avoid the phone breaking without you knowing
>  
>  
> Feedback always welcome. With ZRTP into core now the next version
> may have some ZRTP related features.
>  
>  
> ~Mitch
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> Join Us At ClueCon - Aug 7-9, 2012
>  
> 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
>  
>  


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From mytemike72 at gmail.com  Fri Jun  1 01:19:34 2012
From: mytemike72 at gmail.com (Michael Lutz)
Date: Thu, 31 May 2012 23:19:34 +0200
Subject: [Freeswitch-users] Problem with missing / duplicated DTMF
Message-ID: <CAPu0Fr04AvQSO0VZxtEQDFOqt2PEd+kv26X96AqwWY5mjsBbcQ@mail.gmail.com>

Hi,

Picking up on an older thread, I was wondering if this was ever
resolved and how, or if there are more people experiencing the same
issues (now or in the past)
I seem to be experiencing the same issue!

A lot of the times DTMF's are not recognized (and do not show up in
events or debug log) and apart from I have noticed recognition of
wrong dtmfs too.
Sometimes I even see 'recognized' dtmfs while no key is pressed at al!
(of which I am sure because it was my own test call)

Hope someone is able to help me with this! I was looking in the
combination of inband detection, because I use that for most of my
calls.
However reading this thread it appears that at least someone
experienced same behaviour without enabling inband detection using
start_dtmf.


Regards,
Mike.


>Anthony Minessale anthony.minessale at gmail.com
>Tue Feb 1 21:22:45 MSK 2011
>I dont think there any current dtmf issues open.
>It sounds like maybe you are going across the pstn and encountering
>some problems with transition from
>2833 to inband and back again or from hair-pinning the call.
>
On Tue, Feb 1, 2011 at 12:00 PM, Matt Stockton <mstockton at harqen.com> wrote:
> Sorry about not including the version. The version of freeswitch I am using
> is.
> FreeSWITCH Version 1.0.head (git-256a82d 2011-01-31 10-12-28 -0600)
>  I just updated to the latest yesterday to re-test it.
> On Tue, Feb 1, 2011 at 10:10 AM, Matt Stockton <mstockton at harqen.com> wrote:
>>
>> I have having trouble with both missing and duplicated DTMF in
>> Freeswitch.
>> Here are the steps of how I am using it:
>> 1. Leg A - I am calling out from my Freeswitch instance (through iCall),
>> and I am calling an iCall number that is also connected to the same
>> Freeswitch instance.
>> 2. Leg B - The above call is routed through iCall and then answered by the
>> same Freeswitch instance.
>> 3. On Leg B, I play a file and attempt to get DTMF in a lua script. Here
>> is the code of interest:
>>
>> callPin = session:playAndGetDigits(1, 10, 4, 30000, "#","/tmp/cw_17.wav",
>> "", "\\d+");
>>
>> 4. On Leg A, I send DTMF information in a lua script. Here is the code of
>> interest. I initiate a delay between each digit:
>>
>> local newPin = "";
>>
>> for i = 1, string.len(pin) do
>>
>>   newPin = newPin .. string.sub(pin, i, i) .. "W";
>>
>> end
>>
>> session:execute("send_dtmf", newPin .. "#@200");
>>
>> ** Note that there is a session:sleep on Leg A before I send the DTMF to
>> make sure i don't send it too early **
>>
>> The problem is that the recognized DTMF on Leg B is wrong about 30% of the
>> time. For example, if Leg A enters: 22063083, Leg B will get the DTMF digits
>> 222063083. This is an example of duplication, but I have also experienced
>> missing DTMF codes (and an occasional wrong code completely)
>>
>> I have messed with a bunch of DTMF settings in hopes of fixing this issue,
>> but I cannot seem to find something that is reliable 100% of the time.
>>
>> _____________________________
>>
>> Here are the DTMF settings I have looked at / messed with. I've tried
>> various values for the dtmf-duration in the config (and in the send_dtmf
>> command above)
>>
>>     <!-- This is in external.xml -->
>>
>>     <param name="dtmf-duration" value="500"/>
>>
>>     <param name="dtmf-type" value="rfc2833"/>
>>
>>     <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>>
>>     <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
>>
>>     <param name="inbound-codec-negotiation" value="greedy"/>
>>     <!-- This is in vars.xml -->
>>     <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU"/>
>>     <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU"/>
>> ________________________
>> I have run fs_cli with event logging and the DTMF events that Freeswitch
>> gets do correlate to the wrong value (e.g. the duplication / missing digits
>> is noticable in the Freeswitch events as well).
>> Also, I am not running any dtmf-related applications on the session before
>> I give control to the lua scripts (e.g. not running start_dtmf)
>> Has anyone experienced this type of issue? Or know what I can do to
>> resolve it? My next step was going to be trying this against another
>> provider besides iCall, but I figured I would see if anyone has encountered
>> a similar problem before. Any help is appreciated.
>> 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
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Fri Jun  1 02:15:21 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 31 May 2012 15:15:21 -0700
Subject: [Freeswitch-users] Problem with missing / duplicated DTMF
In-Reply-To: <CAPu0Fr04AvQSO0VZxtEQDFOqt2PEd+kv26X96AqwWY5mjsBbcQ@mail.gmail.com>
References: <CAPu0Fr04AvQSO0VZxtEQDFOqt2PEd+kv26X96AqwWY5mjsBbcQ@mail.gmail.com>
Message-ID: <CAKzWOxVQrP8Mzeo-86p0LOjbAV1zA1MYTx5DUAtOCwEeuU20mg@mail.gmail.com>

do you have a pcap w/ rtp and have you looked at it (and listened to it) in
wireshark?
-MC

On Thu, May 31, 2012 at 2:19 PM, Michael Lutz <mytemike72 at gmail.com> wrote:

> Hi,
>
> Picking up on an older thread, I was wondering if this was ever
> resolved and how, or if there are more people experiencing the same
> issues (now or in the past)
> I seem to be experiencing the same issue!
>
> A lot of the times DTMF's are not recognized (and do not show up in
> events or debug log) and apart from I have noticed recognition of
> wrong dtmfs too.
> Sometimes I even see 'recognized' dtmfs while no key is pressed at al!
> (of which I am sure because it was my own test call)
>
> Hope someone is able to help me with this! I was looking in the
> combination of inband detection, because I use that for most of my
> calls.
> However reading this thread it appears that at least someone
> experienced same behaviour without enabling inband detection using
> start_dtmf.
>
>
> Regards,
> Mike.
>
>
> >Anthony Minessale anthony.minessale at gmail.com
> >Tue Feb 1 21:22:45 MSK 2011
> >I dont think there any current dtmf issues open.
> >It sounds like maybe you are going across the pstn and encountering
> >some problems with transition from
> >2833 to inband and back again or from hair-pinning the call.
> >
> On Tue, Feb 1, 2011 at 12:00 PM, Matt Stockton <mstockton at harqen.com>
> wrote:
> > Sorry about not including the version. The version of freeswitch I am
> using
> > is.
> > FreeSWITCH Version 1.0.head (git-256a82d 2011-01-31 10-12-28 -0600)
> >  I just updated to the latest yesterday to re-test it.
> > On Tue, Feb 1, 2011 at 10:10 AM, Matt Stockton <mstockton at harqen.com>
> wrote:
> >>
> >> I have having trouble with both missing and duplicated DTMF in
> >> Freeswitch.
> >> Here are the steps of how I am using it:
> >> 1. Leg A - I am calling out from my Freeswitch instance (through iCall),
> >> and I am calling an iCall number that is also connected to the same
> >> Freeswitch instance.
> >> 2. Leg B - The above call is routed through iCall and then answered by
> the
> >> same Freeswitch instance.
> >> 3. On Leg B, I play a file and attempt to get DTMF in a lua script. Here
> >> is the code of interest:
> >>
> >> callPin = session:playAndGetDigits(1, 10, 4, 30000,
> "#","/tmp/cw_17.wav",
> >> "", "\\d+");
> >>
> >> 4. On Leg A, I send DTMF information in a lua script. Here is the code
> of
> >> interest. I initiate a delay between each digit:
> >>
> >> local newPin = "";
> >>
> >> for i = 1, string.len(pin) do
> >>
> >>   newPin = newPin .. string.sub(pin, i, i) .. "W";
> >>
> >> end
> >>
> >> session:execute("send_dtmf", newPin .. "#@200");
> >>
> >> ** Note that there is a session:sleep on Leg A before I send the DTMF to
> >> make sure i don't send it too early **
> >>
> >> The problem is that the recognized DTMF on Leg B is wrong about 30% of
> the
> >> time. For example, if Leg A enters: 22063083, Leg B will get the DTMF
> digits
> >> 222063083. This is an example of duplication, but I have also
> experienced
> >> missing DTMF codes (and an occasional wrong code completely)
> >>
> >> I have messed with a bunch of DTMF settings in hopes of fixing this
> issue,
> >> but I cannot seem to find something that is reliable 100% of the time.
> >>
> >> _____________________________
> >>
> >> Here are the DTMF settings I have looked at / messed with. I've tried
> >> various values for the dtmf-duration in the config (and in the send_dtmf
> >> command above)
> >>
> >>     <!-- This is in external.xml -->
> >>
> >>     <param name="dtmf-duration" value="500"/>
> >>
> >>     <param name="dtmf-type" value="rfc2833"/>
> >>
> >>     <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
> >>
> >>     <param name="outbound-codec-prefs"
> value="$${outbound_codec_prefs}"/>
> >>
> >>     <param name="inbound-codec-negotiation" value="greedy"/>
> >>     <!-- This is in vars.xml -->
> >>     <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU"/>
> >>     <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU"/>
> >> ________________________
> >> I have run fs_cli with event logging and the DTMF events that Freeswitch
> >> gets do correlate to the wrong value (e.g. the duplication / missing
> digits
> >> is noticable in the Freeswitch events as well).
> >> Also, I am not running any dtmf-related applications on the session
> before
> >> I give control to the lua scripts (e.g. not running start_dtmf)
> >> Has anyone experienced this type of issue? Or know what I can do to
> >> resolve it? My next step was going to be trying this against another
> >> provider besides iCall, but I figured I would see if anyone has
> encountered
> >> a similar problem before. Any help is appreciated.
> >> 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
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From frankjr at mcpeekdodge.com  Fri Jun  1 02:35:39 2012
From: frankjr at mcpeekdodge.com (Frank Busalacchi Jr)
Date: Thu, 31 May 2012 22:35:39 +0000
Subject: [Freeswitch-users] Freeswitch & FreeTDM question...
Message-ID: <C066384B0C1F1E4B9B870F050D7F1A9D401C2220@ADAMA.mcpeekdodge.local>

Hi, and thanks in advance for your help.

I run freeswitch with a SangomA A102 (FXO/FXS) card.  The freeswitch code base that it is running on is about 3 months old.  It is a production server, and taking it down other than in the middle of the night is problemsome...  It is running on Debian Lenny.

In the next couple of days I am going to be spending the night, doing a fresh install of the latest debian (squeeze) at which time I will do a fresh GIT, fresh sangoma wanpipe and be totally current on FreeSwitch.

My Question:

I am having intermittent problems with my FXO/FXS ports.  Basically I have fax machines attached.  Every so often (sometimes 5-10 days in between), the fax machine says that it has no dial tone.  I am able to resolve the problem by going to the CLI and unloading mod_freetdm and then reloading mod_freetdm.  When the "no dialtone problem" is happening, unloading mod_freetdm takes 10 seconds or so...I have even had it crash freeswitch during the unload...

I haven't attempted to narrow down the issue yet, and thought I might ask for some guidance before heading down that path.

One of my thoughts is that perhaps the Hardware is bad, but the fact that unloading/loading mod_freetdm fixes it makes me think it is something in mod_freetdm...

Your thoughts?

-Frank


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From chris at opencsta.org  Fri Jun  1 03:16:34 2012
From: chris at opencsta.org (Chris Mylonas)
Date: Fri, 1 Jun 2012 09:16:34 +1000
Subject: [Freeswitch-users] NAT issues and "best practices"
In-Reply-To: <036901cd3f58$bb857510$32905f30$@bizfocused.com>
References: <036901cd3f58$bb857510$32905f30$@bizfocused.com>
Message-ID: <1C0DB9B9-9542-455F-9FCF-2469C2974A12@opencsta.org>

NAT and SIP do not play nicely together easily all the time.
From chapter 9 of "building telephone systems with opensips" (a few years old)

NAT breaks SIP because SIP is a session establishment protocol and thus, belongs to the session layer of the OSI model. However, it includes network layer addresses in their headers. The NAT present in most routers, process only network (layer 3) headers and leaves the SIP headers unchanged.

I'm returning to the VoIP scene after 3 years of web sysadmin-ing and boat refitting, but in years gone by for example, a snom phone played well with a double NAT  ( phone-with-private-addy<----NAT-router>----Internet--->-----<---router--->----<---PBX-with private addy) whilst a polycom would barf.

Let's just say there are 10 points of failure in a voip call.  each of them relying on a hardware/firmware/software stack.
There are recipes that work very well...and some like the polycom of years ago that didn't.  I do not know what the status of the double-NAT polycom is these days but I would presume that it works in 2012.

I'm a FS n00b so cannot offer any specific advice.
There are different types of NAT-ing though: "full cone", "restricted cone", 1:1 (symmetrical) - one is IP:Port, one is IP, and the other is port forwarding.  



On 01/06/2012, at 4:10 AM, Sean Devoy wrote:

> HI All,
>  
> I have a customer location that has just been a nightmare to implement.  I am just learning that they ?may? have multiple NAT routers in sequence at their location.  I think I fully understand what NAT is trying to accomplish. There seem to be different levels and approaches. 
>  
> The most basic NAT setup (to me)  is a HOME LAN with multiple PCs where NAT allows multiple devices to share a single routable IP address on the WAN side from multiple local devices the LAN side.  Note I said OUTBOUND initiated connections.  Even FTP can have trouble with this level.  Almost all inbound traffic is blocked for security.
>  
> Clearly for FS we need the switch to be able to punch through from the WAN to specific local IPs on the LAN to reach specific phones.  This is INBOUND NAT and brings up many security issues for people.  Even on devices where you get this ?working? you may only be able to support one line per phone or a single inbound connection at a time.
>  
> I understand NAT has PMP and UPnP protocols and FS ?supports? both.  What I can?t find is where someone says ?Here is a great setup that works with cheap, available ?commodity? hardware from Cisco/Linksys that supports all the NAT you need for FS.?  I don?t care if it is PMP or UPnP and I might not even care why you pick one over the other, although it is probably a ?good read?.
>  
> Can someone just stand up say ?FS works GREAT with the XYZ router in ABC mode from MY COMPANY using NAT to Cisco phones??
>  
> I have seen some articles about Freeware/Shareware firmware in this devices, but as a novice I want to limit the unknowns until I get more up to speed.
>  
> My specific issue now is that I cannot get SCA to work at the NAT location.  I issued:
> sofia_contact 220 at mydomain.com
> sofia/external/sip:200@<ip address>:44234,sofia/external/sip:200@<same ip>:1024
>  
> Only one phone rings on inbound and the line indicator light does not change when either is picked up.  Same configuration is working on our LAN with the switch.
>  
> I am absolutely ready to by a router to fix these issues, I don?t want to lose this customer.
>  
> Thanks for your thoughts,
> Sean
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ash at url.net.au  Fri Jun  1 03:52:22 2012
From: ash at url.net.au (Ashley Breeden)
Date: Fri, 1 Jun 2012 09:52:22 +1000
Subject: [Freeswitch-users] NAT issues and "best practices"
In-Reply-To: <036901cd3f58$bb857510$32905f30$@bizfocused.com>
References: <036901cd3f58$bb857510$32905f30$@bizfocused.com>
Message-ID: <43E524DB-4519-4531-8CD4-4B01C95882B6@url.net.au>

Hi Sean,

Yes SIP and NAT do not play well together.  Most of our customers all use some form of NAT to access our FreeSWITCH servers, although not double NAT.  I have found that pretty much most of the current mainstream routers and phones work as long as you have the latest software.  One of the biggest things I find is that if you have more than one SIP device registered behind a NAT with SIP ALG you will get very mixed results.

It will also depend on how many devices you have behind the NAT, some routers seem to have issue processing NAT for more than 10 devices all originating on port 5060.

Here is a few things I will often do to get a customer to connect via a NAT device:

- Turn off SIP ALG on the router
- Look for a setting on your VoIP device called Rport, set this to enabled.  
- Try putting each phones local SIP port on its own unique port, e.g. extension 1001 = port 5001, 1002 = port 5002
- Try TCP as the transport protocol

Also have a look here I have been able to get some useful information from this article - http://wiki.freeswitch.org/wiki/ALG, especially the DG834's with DOS protection.

Personally I use the following routers with no issues:

- Billion 7401VGP
- Billion 7800NL
- Billion 7404
- Cisco 857/877
- MicroTik seemed to be quite reasonable as well

For Phones I have running:

- Yealink T22,T26,T28,T32,T38
- Snom 360
- Siemens Gigaset


Hope this helps.

Cheers,

Ash


On 01/06/2012, at 4:10 AM, Sean Devoy wrote:

> HI All,
>  
> I have a customer location that has just been a nightmare to implement.  I am just learning that they ?may? have multiple NAT routers in sequence at their location.  I think I fully understand what NAT is trying to accomplish. There seem to be different levels and approaches. 
>  
> The most basic NAT setup (to me)  is a HOME LAN with multiple PCs where NAT allows multiple devices to share a single routable IP address on the WAN side from multiple local devices the LAN side.  Note I said OUTBOUND initiated connections.  Even FTP can have trouble with this level.  Almost all inbound traffic is blocked for security.
>  
> Clearly for FS we need the switch to be able to punch through from the WAN to specific local IPs on the LAN to reach specific phones.  This is INBOUND NAT and brings up many security issues for people.  Even on devices where you get this ?working? you may only be able to support one line per phone or a single inbound connection at a time.
>  
> I understand NAT has PMP and UPnP protocols and FS ?supports? both.  What I can?t find is where someone says ?Here is a great setup that works with cheap, available ?commodity? hardware from Cisco/Linksys that supports all the NAT you need for FS.?  I don?t care if it is PMP or UPnP and I might not even care why you pick one over the other, although it is probably a ?good read?.
>  
> Can someone just stand up say ?FS works GREAT with the XYZ router in ABC mode from MY COMPANY using NAT to Cisco phones??
>  
> I have seen some articles about Freeware/Shareware firmware in this devices, but as a novice I want to limit the unknowns until I get more up to speed.
>  
> My specific issue now is that I cannot get SCA to work at the NAT location.  I issued:
> sofia_contact 220 at mydomain.com
> sofia/external/sip:200@<ip address>:44234,sofia/external/sip:200@<same ip>:1024
>  
> Only one phone rings on inbound and the line indicator light does not change when either is picked up.  Same configuration is working on our LAN with the switch.
>  
> I am absolutely ready to by a router to fix these issues, I don?t want to lose this customer.
>  
> Thanks for your thoughts,
> Sean
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From sherifomran2000 at yahoo.com  Fri Jun  1 04:36:29 2012
From: sherifomran2000 at yahoo.com (Sherif Omran)
Date: Thu, 31 May 2012 17:36:29 -0700 (PDT)
Subject: [Freeswitch-users] FSClient a Windows FreeSWITCH softphone
	Update Released
In-Reply-To: <C9477E2482C641E19003B37B1843DC52@freeswitch.org>
Message-ID: <1338510989.33281.YahooMailClassic@web110808.mail.gq1.yahoo.com>

It would be better if you don't need .net at all in order to install the application on cell phones and /or lite net books

regards
S.

--- On Thu, 5/31/12, Jo?o Mesquita <jmesquita at freeswitch.org> wrote:

From: Jo?o Mesquita <jmesquita at freeswitch.org>
Subject: Re: [Freeswitch-users] FSClient a Windows FreeSWITCH softphone Update Released
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Thursday, May 31, 2012, 11:10 PM


                Mitch, I would like to express a big public congratulations to you.
You have pulled something that I failed (so far) to do. FSComm is still on my plans, it is just "on hold" and to be completed. Still, I know from the little bit I've done that it is not an easy task. Once again, congratulations man.
Regards,
                
--?Jo?o MesquitaSent with Sparrow

                 
                On Tuesday, May 29, 2012 at 11:28 PM, Mitch Capper wrote:
                
                    An update had been posted for FSClient is a windows .NET softphonethat uses libfreeswitch at the core.Info page at: http://wiki.freeswitch.org/wiki/FSClientDirect binary download:http://files.freeswitch.org/windows/installer/x86/FSClient.zipAnd source is in contrib repo under MitchCapper/FSClient
 Some of the changes include:Built against FS 1.2 RCisac codec supportfor contacts you can now right click and choose what account to call fromif sofia goes offline update the accounts statusJabra headset improvements to avoid some crash conditionsbackground task improvements to avoid the phone breaking without you knowing

Feedback always welcome.   With ZRTP into core now the next versionmay have some ZRTP related features.

~Mitch
_________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com

Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
                 
                 
                 
                 
                
                 
                
                    

                
            
-----Inline Attachment Follows-----

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing 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 sdevoy at bizfocused.com  Fri Jun  1 05:32:41 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 31 May 2012 21:32:41 -0400
Subject: [Freeswitch-users] NAT issues and "best practices"
In-Reply-To: <43E524DB-4519-4531-8CD4-4B01C95882B6@url.net.au>
References: <036901cd3f58$bb857510$32905f30$@bizfocused.com>
	<43E524DB-4519-4531-8CD4-4B01C95882B6@url.net.au>
Message-ID: <052201cd3f96$6fe5ae20$4fb10a60$@bizfocused.com>

Thanks for the response.

 

At this point I can say I have  2 working and 1 not working configuration.
Also, thanks for the SIP ALG tip (3 hours too late!)  I turned it on and
things went to crap.

 

So my WOKING configa are:

<FS w/ local IP> to <FIOS 25MB router> to <Motorla Cable Modem> to <Cisco
RVS4000> to <CISCO SPA-504G phones>

<FS w/ local IP> to <FIOS 25MB router> to <Motorla Cable Modem> to <Linksys
home router ???? > to <CISCO SPA-504G phones>

Calls, MWI, SCA, etc across those 2 configs works PERFECTLY.

I left out <fs> on local lan to <Cisco 504g> is wonderful too.

 

Unfortunately, the customer setup is:

<FS w/ local IP> to <FIOS 25MB router> to <COMCAST Business Cable Modem> to
<Cisco E1200> to <CISCO SPA-504G phones>

Results are all over the place.  I don't think they can support 2 inbound
calls at the same time.  MWI is spotty, SCA is not working.  Turning on SIP
ALG on Cisco E1200 led to a small riot and has been banished from the
kingdom for every.

 

I think tomorrow we are getting them a Cisco RVS4000.

 

Sean

 

 

From: Ashley Breeden [mailto:ash at url.net.au] 
Sent: Thursday, May 31, 2012 7:52 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] NAT issues and "best practices"

 

Hi Sean,

 

Yes SIP and NAT do not play well together.  Most of our customers all use
some form of NAT to access our FreeSWITCH servers, although not double NAT.
I have found that pretty much most of the current mainstream routers and
phones work as long as you have the latest software.  One of the biggest
things I find is that if you have more than one SIP device registered behind
a NAT with SIP ALG you will get very mixed results.

 

It will also depend on how many devices you have behind the NAT, some
routers seem to have issue processing NAT for more than 10 devices all
originating on port 5060.

 

Here is a few things I will often do to get a customer to connect via a NAT
device:

 

- Turn off SIP ALG on the router

- Look for a setting on your VoIP device called Rport, set this to enabled.


- Try putting each phones local SIP port on its own unique port, e.g.
extension 1001 = port 5001, 1002 = port 5002

- Try TCP as the transport protocol

 

Also have a look here I have been able to get some useful information from
this article - http://wiki.freeswitch.org/wiki/ALG, especially the DG834's
with DOS protection.

 

Personally I use the following routers with no issues:

 

- Billion 7401VGP

- Billion 7800NL

- Billion 7404

- Cisco 857/877

- MicroTik seemed to be quite reasonable as well

 

For Phones I have running:

 

- Yealink T22,T26,T28,T32,T38

- Snom 360

- Siemens Gigaset

 

 

Hope this helps.

 

Cheers,

 

Ash

 

 

On 01/06/2012, at 4:10 AM, Sean Devoy wrote:





HI All,

 

I have a customer location that has just been a nightmare to implement.  I
am just learning that they "may" have multiple NAT routers in sequence at
their location.  I think I fully understand what NAT is trying to
accomplish. There seem to be different levels and approaches. 

 

The most basic NAT setup (to me)  is a HOME LAN with multiple PCs where NAT
allows multiple devices to share a single routable IP address on the WAN
side from multiple local devices the LAN side.  Note I said OUTBOUND
initiated connections.  Even FTP can have trouble with this level.  Almost
all inbound traffic is blocked for security.

 

Clearly for FS we need the switch to be able to punch through from the WAN
to specific local IPs on the LAN to reach specific phones.  This is INBOUND
NAT and brings up many security issues for people.  Even on devices where
you get this "working" you may only be able to support one line per phone or
a single inbound connection at a time.

 

I understand NAT has PMP and UPnP protocols and FS "supports" both.  What I
can't find is where someone says "Here is a great setup that works with
cheap, available "commodity" hardware from Cisco/Linksys that supports all
the NAT you need for FS."  I don't care if it is PMP or UPnP and I might not
even care why you pick one over the other, although it is probably a "good
read".

 

Can someone just stand up say "FS works GREAT with the XYZ router in ABC
mode from MY COMPANY using NAT to Cisco phones"?

 

I have seen some articles about Freeware/Shareware firmware in this devices,
but as a novice I want to limit the unknowns until I get more up to speed.

 

My specific issue now is that I cannot get SCA to work at the NAT location.
I issued:

sofia_contact 220 at mydomain.com

sofia/external/sip:200@<ip address>:44234,sofia/external/sip:200@<same
ip>:1024

 

Only one phone rings on inbound and the line indicator light does not change
when either is picked up.  Same configuration is working on our LAN with the
switch.

 

I am absolutely ready to by a router to fix these issues, I don't want to
lose this customer.

 

Thanks for your thoughts,

Sean

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

 

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From chris at opencsta.org  Fri Jun  1 05:59:49 2012
From: chris at opencsta.org (Chris Mylonas)
Date: Fri, 1 Jun 2012 11:59:49 +1000
Subject: [Freeswitch-users] calls to australian geo numbers for load test
Message-ID: <5EE7A1D0-78E9-45BD-B8E6-CDD28CC6BED5@opencsta.org>

hi fs users,

is it possible to bounce signalling from an overseas provider and keep media local?

i need to find reasonable rates for load testing to australian geo numbers - concurrent channels + rate per minute
not fussy about which country the calls originate.

i was thinking to work my way through this list of providers http://wiki.freeswitch.org/wiki/SIP_Provider_Examples to tally up costs
probably won't happen til july/august but just getting some notes down whilst it's raining and there's nothing for me to do except dev...


cheers
chris
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From jgallartm at gmail.com  Fri Jun  1 09:18:08 2012
From: jgallartm at gmail.com (Javier Gallart)
Date: Fri, 1 Jun 2012 07:18:08 +0200
Subject: [Freeswitch-users] Access to Cause filed in Reason header
	(Michael Collins)
Message-ID: <CACviLGZ+qSjkM-Ge6MfRXQm-o_5PUoKBy4mxh2z4PwdFo68XMw@mail.gmail.com>

Thanks Michael

I did exactly that, an "info" application is executied right ater the bride
and the variable is not there. I also tried firing an event when B-leg ends
and none of the headers showed the contents of the Reason Header.


Regards

Javi


> ---------- Forwarded message ----------
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Thu, 31 May 2012 11:16:46 -0700
> Subject: Re: [Freeswitch-users] Access to Cause filed in Reason header
> I believe that information is available in other channel variables. I'd do
> an info dump right after the bridge and see what shakes out.
> -MC
>
> On Thu, May 31, 2012 at 8:51 AM, Javier Gallart <jgallartm at gmail.com>wrote:
>
>> Hello list
>>
>> some of our providers include a Reason header in negative replies to an
>> INVITE. For instance, they might include a cause=65 inside a Sip 503 reply.
>> We need to react upon some of those ISDN causes: in this particular case
>> with if Reason header were absent we would simply do nothing but let FS to
>> send the response upstream; but if cause=65 in present we want to resend
>> the call to the same provider with a different set of offered codecs. Is
>> there any way to access to the content of that header?. Our applicatiion
>> listens to the event socket in inbound mode but I haven't found a way to do
>> this. Any help would be appreciated.
>>
>> Regards
>>
>> Javi
>>
>>
>
>
>
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From gamar at center.com  Fri Jun  1 02:15:57 2012
From: gamar at center.com (Gilbert Amar)
Date: Thu, 31 May 2012 15:15:57 -0700
Subject: [Freeswitch-users] How do you split bridged calls ?
In-Reply-To: <CAKzWOxXsOuDfcwttJEwjUEoaaYEcn8bb3pWkeE+z3pb96co-TQ@mail.gmail.com>
References: <CALNZuwkRLZ0avq-xaPLmrnRQYUUGHd4WDFKCaqA+H2N_rDTGKw@mail.gmail.com>	<20120523163913.b07558d1@mail.tritonwest.net>
	<CAKzWOxXsOuDfcwttJEwjUEoaaYEcn8bb3pWkeE+z3pb96co-TQ@mail.gmail.com>
Message-ID: <002701cd3f7a$f3dd50a0$db97f1e0$@center.com>

Hello,

 

I keep trying to split a bridged call.

Using playback wav I have audio on both leg, but in my IVR I will need
different message to be played to each leg.

I do not know if this is a bug or not:

I have audio only on one leg using uuid_transfer or uuid_dual_transfer for
phrase.

 

Test 1

         api uuid_transfer <uuid> -both 'm:^:phrase:queue_position,5^
inline'

 

Test 2:

         api uuid_dual_transfer <uuid> 'm:^:phrase:queue_position,5^ inline'
'm:^:phrase:queue_position,3^ inline'

 

notice queue_position 5 for the first and 3 for the second.

 

 

Please find attach a FS log showing for the second test that FS is only
looking to process the first leg.

 

Gilbert

 

 

Details regarding the application and log.

 

my dialplan

    <extension name="unpark">

      <condition field="destination_number" expression="^5100$">

         <action application="set" data="park_after_bridge=true"/>

         <action application="socket" data="192.168.1.100:8022 async full"/>

      </condition>

    </extension>

 

Why use phrase instead of files, mainly experience with IVR building, easier
to change, manage, can use parameters and language customization.

 

Application

Let's be optimistic and forget about any errors.

 

Leg 1 call fs ext 5100

         goes thru socket app

         api create_uuid for outgoing call

         execute playback phrase

         some other event initiate originate (in my case http request
result)

         bgapi originate
{origination_uuid=created_UUID}sofia/internal/1001%192.168.1.11 5100  //
this way leg 2 ends also on my socket app

 

Leg 2 outgoing call (ext 5100) 

         execute playback phrase

 

the 2 legs are there so now using DTMF handling on leg 1

 

         DTMF 1 bridge the two legs

         DTMF 2 broadcast msg on the 2 leg , this work but wasn't used in
the log provided

         DTMF 3 uuid_transfer or in our case uuid_dual_transfert

 

 

leg 1 = sofia/internal/1000 at 192.168.1.11

leg 2 = sofia/internal/1001 at 192.168.1.11

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, May 29, 2012 1:04 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How do you split bridged calls ?

 

Sorry for the late reply, but I just wanted to mention for the sake of
posterity/SEO that you could also use 'uuid_dual_transfer' if you want to
send each leg to a different destination. 
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_dual_transfer

-MC

On Wed, May 23, 2012 at 9:39 AM, Dave R. Kompel <drk at drkngs.net> wrote:

Or you  could use uuid_transfer, such as "uuid_transfer <uuid> -both park
inline". If you want to use dialplan apps in the transfer just use the
"inline" dialplan, and not XML.

 

--Dave

 

 

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From msc at freeswitch.org  Fri Jun  1 09:44:39 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 31 May 2012 22:44:39 -0700
Subject: [Freeswitch-users] Spanish prompt: "Para espanol,
	oprima el numero dos" or similar
Message-ID: <CAKzWOxVVMme_3dN-=t04=CmfDRf7DsaA6Pq5DTi1NhxitiM1Qw@mail.gmail.com>

Hey, does anyone have a nice little prompt in a female voice that says
something like "for Spanish, press..."? I'm working on some generic IVR
stuff and I'd love to be able to mix in something like that.

Thanks,
MC
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From msc at freeswitch.org  Fri Jun  1 09:50:57 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 31 May 2012 22:50:57 -0700
Subject: [Freeswitch-users] FSClient a Windows FreeSWITCH softphone
	Update Released
In-Reply-To: <1338510989.33281.YahooMailClassic@web110808.mail.gq1.yahoo.com>
References: <C9477E2482C641E19003B37B1843DC52@freeswitch.org>
	<1338510989.33281.YahooMailClassic@web110808.mail.gq1.yahoo.com>
Message-ID: <CAKzWOxUFda+-zhJE-FGH2ZecjRtGGcoPL7WQmDtQ4UdkrYMRFA@mail.gmail.com>

It would also be better if unicorns ate nuclear waste and pooped diamonds.
:)

First things first - let's get it working and stable on actual Windows
machines before we tackle the morass of cell phones and tablets.

-MC

On Thu, May 31, 2012 at 5:36 PM, Sherif Omran <sherifomran2000 at yahoo.com>wrote:

> It would be better if you don't need .net at all in order to install the
> application on cell phones and /or lite net books
>
> regards
> S.
>
> --- On *Thu, 5/31/12, Jo?o Mesquita <jmesquita at freeswitch.org>* wrote:
>
>
> From: Jo?o Mesquita <jmesquita at freeswitch.org>
> Subject: Re: [Freeswitch-users] FSClient a Windows FreeSWITCH softphone
> Update Released
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Thursday, May 31, 2012, 11:10 PM
>
> Mitch, I would like to express a big public congratulations to you.
>
> You have pulled something that I failed (so far) to do. FSComm is still on
> my plans, it is just "on hold" and to be completed. Still, I know from the
> little bit I've done that it is not an easy task. Once again,
> congratulations man.
>
> Regards,
>
> --
> Jo?o Mesquita
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Tuesday, May 29, 2012 at 11:28 PM, Mitch Capper wrote:
>
> An update had been posted for FSClient is a windows .NET softphone
> that uses libfreeswitch at the core.
> Info page at: http://wiki.freeswitch.org/wiki/FSClient
> Direct binary download:
> http://files.freeswitch.org/windows/installer/x86/FSClient.zip
> And source is in contrib repo under MitchCapper/FSClient
>
> Some of the changes include:
> Built against FS 1.2 RC
> isac codec support
> for contacts you can now right click and choose what account to call from
> if sofia goes offline update the accounts status
> Jabra headset improvements to avoid some crash conditions
> background task improvements to avoid the phone breaking without you
> knowing
>
>
> Feedback always welcome. With ZRTP into core now the next version
> may have some ZRTP related features.
>
>
> ~Mitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://mc/compose?to=consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at 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-----
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://mc/compose?to=consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cesar.bermudez at gmail.com  Fri Jun  1 10:08:22 2012
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Fri, 1 Jun 2012 03:08:22 -0300
Subject: [Freeswitch-users] Spanish prompt: "Para espanol,
 oprima el numero dos" or similar
In-Reply-To: <CAKzWOxVVMme_3dN-=t04=CmfDRf7DsaA6Pq5DTi1NhxitiM1Qw@mail.gmail.com>
References: <CAKzWOxVVMme_3dN-=t04=CmfDRf7DsaA6Pq5DTi1NhxitiM1Qw@mail.gmail.com>
Message-ID: <CADrcvV-LRZtaPMGiuvoUTBuRcBhdgegq5QyK_ou48YeTL-97Bg@mail.gmail.com>

a2billing sounds have it.

On Fri, Jun 1, 2012 at 2:44 AM, Michael Collins <msc at freeswitch.org> wrote:

> Hey, does anyone have a nice little prompt in a female voice that says
> something like "for Spanish, press..."? I'm working on some generic IVR
> stuff and I'd love to be able to mix in something like that.
>
> Thanks,
> MC
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun  1 10:10:31 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 31 May 2012 23:10:31 -0700
Subject: [Freeswitch-users] How do you split bridged calls ?
In-Reply-To: <002701cd3f7a$f3dd50a0$db97f1e0$@center.com>
References: <CALNZuwkRLZ0avq-xaPLmrnRQYUUGHd4WDFKCaqA+H2N_rDTGKw@mail.gmail.com>
	<20120523163913.b07558d1@mail.tritonwest.net>
	<CAKzWOxXsOuDfcwttJEwjUEoaaYEcn8bb3pWkeE+z3pb96co-TQ@mail.gmail.com>
	<002701cd3f7a$f3dd50a0$db97f1e0$@center.com>
Message-ID: <CAKzWOxWP6dmRBV-1SJhyS3V9+GqFd9zuaevfxOGM0N00YVsCJQ@mail.gmail.com>

I dropped your log file into our pastebin here:
http://pastebin.freeswitch.org/19209

The log shows that the issue is not that FS is only trying to process the
first leg, which can be seen by looking at lines 507 and 714. The problem
here is that your inline dialplan is not being parsed properly. What's
happening is that uuid_dual_transfer is literally trying to send the A leg
to destination number "m:^:phrase:queue_position,5^ inline" and the B leg
to destination number "m:^:phrase:queue_position,3^ inline".

What are you trying to do with each leg of the call? I see that you are
trying to announce the queue position, but what do you want to have happen
after that? As it stands your call legs would probably just end after the
announcement because there's nothing more to do.

-MC

On Thu, May 31, 2012 at 3:15 PM, Gilbert Amar <gamar at center.com> wrote:

> Hello,****
>
> ** **
>
> I keep trying to split a bridged call.****
>
> Using playback wav I have audio on both leg, but in my IVR I will need
> different message to be played to each leg.****
>
> I do not know if this is a bug or not:****
>
> I have audio only on one leg using uuid_transfer or uuid_dual_transfer for
> phrase.****
>
> ** **
>
> Test 1****
>
>          api uuid_transfer <uuid> -both 'm:^:phrase:queue_position,5^
> inline'****
>
> ** **
>
> Test 2:****
>
>          api uuid_dual_transfer <uuid> 'm:^:phrase:queue_position,5^
> inline' 'm:^:phrase:queue_position,3^ inline'****
>
> ** **
>
> notice queue_position 5 for the first and 3 for the second.****
>
> ** **
>
> ** **
>
> Please find attach a FS log showing for the second test that FS is only
> looking to process the first leg.****
>
> ** **
>
> Gilbert****
>
> ** **
>
> ** **
>
> Details regarding the application and log.****
>
> ** **
>
> my dialplan****
>
>     <extension name="unpark">****
>
>       <condition field="destination_number" expression="^5100$">****
>
>          <action application="set" data="park_after_bridge=true"/>****
>
>          <action application="socket" data="192.168.1.100:8022 async
> full"/>****
>
>       </condition>****
>
>     </extension>****
>
> ** **
>
> Why use phrase instead of files, mainly experience with IVR building,
> easier to change, manage, can use parameters and language customization.**
> **
>
> ** **
>
> Application****
>
> Let's be optimistic and forget about any errors.****
>
> ** **
>
> Leg 1 call fs ext 5100****
>
>          goes thru socket app****
>
>          api create_uuid for outgoing call****
>
>          execute playback phrase****
>
>          some other event initiate originate (in my case http request
> result)****
>
>          bgapi originate
> {origination_uuid=created_UUID}sofia/internal/1001%192.168.1.11 5100  //
> this way leg 2 ends also on my socket app****
>
> ** **
>
> Leg 2 outgoing call (ext 5100) ****
>
>          execute playback phrase****
>
> ** **
>
> the 2 legs are there so now using DTMF handling on leg 1****
>
> ** **
>
>          DTMF 1 bridge the two legs****
>
>          DTMF 2 broadcast msg on the 2 leg , this work but wasn't used in
> the log provided****
>
>          DTMF 3 uuid_transfer or in our case uuid_dual_transfert****
>
> ** **
>
> ** **
>
> leg 1 = sofia/internal/1000 at 192.168.1.11****
>
> leg 2 = sofia/internal/1001 at 192.168.1.11****
>
> ** **
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Tuesday, May 29, 2012 1:04 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] How do you split bridged calls ?****
>
> ** **
>
> Sorry for the late reply, but I just wanted to mention for the sake of
> posterity/SEO that you could also use 'uuid_dual_transfer' if you want to
> send each leg to a different destination.
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_dual_transfer
>
> -MC****
>
> On Wed, May 23, 2012 at 9:39 AM, Dave R. Kompel <drk at drkngs.net> wrote:***
> *
>
> Or you  could use uuid_transfer, such as "uuid_transfer <uuid> -both park
> inline". If you want to use dialplan apps in the transfer just use the
> "inline" dialplan, and not XML.****
>
>  ****
>
> --Dave****
>
> ** **
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun  1 10:10:51 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 31 May 2012 23:10:51 -0700
Subject: [Freeswitch-users] Spanish prompt: "Para espanol,
 oprima el numero dos" or similar
In-Reply-To: <CADrcvV-LRZtaPMGiuvoUTBuRcBhdgegq5QyK_ou48YeTL-97Bg@mail.gmail.com>
References: <CAKzWOxVVMme_3dN-=t04=CmfDRf7DsaA6Pq5DTi1NhxitiM1Qw@mail.gmail.com>
	<CADrcvV-LRZtaPMGiuvoUTBuRcBhdgegq5QyK_ou48YeTL-97Bg@mail.gmail.com>
Message-ID: <CAKzWOxU32=brojgytX6TnbCLhSBh2RMxsoQXcxh1eX01a0vc_Q@mail.gmail.com>

cool, thanks.
-MC

On Thu, May 31, 2012 at 11:08 PM, Cesar Bermudez
<cesar.bermudez at gmail.com>wrote:

> a2billing sounds have it.
>
> On Fri, Jun 1, 2012 at 2:44 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Hey, does anyone have a nice little prompt in a female voice that says
>> something like "for Spanish, press..."? I'm working on some generic IVR
>> stuff and I'd love to be able to mix in something like that.
>>
>> Thanks,
>> MC
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.villa at seletech.com  Fri Jun  1 11:38:19 2012
From: a.villa at seletech.com (alberto Villa)
Date: Fri, 01 Jun 2012 09:38:19 +0200
Subject: [Freeswitch-users] Custom Registration
In-Reply-To: <4FC7BB80.9050600@elder.hu>
References: <4FC71FD9.5010004@seletech.com> <4FC7BB80.9050600@elder.hu>
Message-ID: <4FC8716B.9080300@seletech.com>

Il 31/05/2012 20:42, Pusk?s Zsolt ha scritto:
> Hi.
>
> Even if you wrote an ESL script to watch registration events you can't
> refuse or miss a registration so you have to modify the sofia source
> code to achive such a function as far as I know.
>
> Are you using IP address based authentication for phones and afraid the
> user might change the username or why you want such a function ?
>
>
> 2012-05-31 09:38 keltez?ssel, alberto Villa ?rta:
>> Hello,
>>
>> I'm using FreeSWITCH in a very custom system and I need to setup this
>> kind of registration:
>>
>> There are two SIP phones A and B: when each SIP phone have its own
>> unique ID the registration to freeswitch goes as usual and no change is
>> needed, but when A and B share the same ID I would like to reject the
>> registration for the last SIP phone. So, if A and B have the same ID and
>> A is registered first, I would like to reject the subsequent
>> registration of B.
>>
>> I tried to change the values of some configuration parameters such as
>> "sip-allow-multiple-registrations" and "max-registrations-per-extension"
>> but it didn't work (by the way I'm not shure to have edited properly
>> this parameters).
>>
>> How can I achive this kind of registration? Thank you
>>
>> Alberto
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
Problem Solved!! I just wrote this line:

<param name="max-registrations-per-extension" value="1"/>

within the "params" tag when defining a new user in the 
~/freeswitch/conf/directory/default/ directory. Then issue "reloadxml" 
command and everything works as I wanted to: if Extension A is already 
registered with id ID_A and extension B tries to register with id ID_A, 
the registration is rejected ^_^

Thank you!

-- 
Dr. Villa Alberto
Sw Engineer

SeleTech srl
via Collodi, 8 20052 Monza (MI)
tel: +39 039 5962000
fax: +39 039 9716905
email: a.villa at seletech.com
web: www.seletech.com
      www.seletech.eu



From alex at digitalmail.com  Fri Jun  1 14:17:28 2012
From: alex at digitalmail.com (Alex Lake)
Date: Fri, 01 Jun 2012 11:17:28 +0100
Subject: [Freeswitch-users] att_xfer and loopback
Message-ID: <4FC896B8.9070102@digitalmail.com>

Got a lua script for a B-party "mid-call menu". Is it legitimate to do..
"session:execute("att_xfer", "loopback/"..destnum)"

I've tried it and it seems to start off doing the right things, but my 
A-party gets disconnected as soon as the call to the C-Party (the person 
I'm transferring the call to) answers the call.

Maybe better to try to orchestrate the entire affair from within the lua 
script? (Tricky for a beginner like me!)

Thanks,
Alex


From modesto at isimples.com.br  Fri Jun  1 15:46:06 2012
From: modesto at isimples.com.br (Antonio Modesto)
Date: Fri, 01 Jun 2012 08:46:06 -0300
Subject: [Freeswitch-users] FreeTDM CallerID Detection with DTMF
In-Reply-To: <CAKbxfG98vJ-wkLRZHYZ6RkDWR+yWg=2OduJdRRjb+epCaeOx=A@mail.gmail.com>
References: <1338403799.3112.43.camel@modesto.localdomain.net>
	<1338484250.23766.17.camel@modesto.localdomain.net>
	<CAKbxfG98vJ-wkLRZHYZ6RkDWR+yWg=2OduJdRRjb+epCaeOx=A@mail.gmail.com>
Message-ID: <1338551166.19779.20.camel@modesto.localdomain.net>

On Thu, 2012-05-31 at 14:20 -0500, Anthony Minessale wrote:
> I don't think it's implemented.
> You would have to ask Sangoma to implement that feature.

I've sent an email to sangoma asking them to verify this possibility. I
hope they can do something. Thank you very much.


Regards.

> 
> 
> 
> On Thu, May 31, 2012 at 12:10 PM, Antonio Modesto
> <modesto at isimples.com.br> wrote:
> > Hi,
> >
> >        Does anybody have any suggestions? I searched a lot about it but I was
> > not successful, I found that here in brazil the callerid is sent via
> > DTMF but without any warning or polarity reversal, it's a strange
> > implementation. Any hints would be appreciated.
> >
> > Regards.
> >
> >
> > On Wed, 2012-05-30 at 15:49 -0300, Antonio Modesto wrote:
> >> Hi,
> >>
> >>       I have a Digium TDM410P card, I am using it here in Brazil, where the
> >> signaling is not FSK, it's DTMF. It is working, though I am not
> >> receiving the callerid:
> >>
> >> Caller-Caller-ID-Number: [0000000000]
> >>
> >> Here is my freetdm.conf:
> >>
> >> [span zt FXO1]
> >> fxo-channel => 1
> >>
> >> [span zt FXO2]
> >> fxo-channel => 2
> >>
> >> [span zt FXO3]
> >> fxo-channel => 3
> >>
> >> Here is one section of my autoload_configs/freetdm.conf.xml:
> >>
> >>     <span name="FXO1">
> >>       <!--<param name="hold-music" value="$${moh_uri}"/>-->
> >>       <!--<param name="enable-analog-option" value="call-swap"/>-->
> >>       <!--<param name="enable-analog-option" value="3-way"/>-->
> >>       <param name="tonegroup" value="br"/>
> >>       <param name="digit-timeout" value="2000"/>
> >>       <param name="max-digits" value="11"/>
> >>       <param name="dialplan" value="XML"/>
> >>       <param name="context" value="public"/>
> >>       <param name="enable-callerid" value="true"/>
> >>       <!-- 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>
> >>
> >> Is it possible to enable the callerid detection in these conditions, or
> >> is it a hardware/driver limitation?
> >>
> >>
> >> Regards.
> >>
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From anita.hall at simmortel.com  Fri Jun  1 15:51:05 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Fri, 1 Jun 2012 17:21:05 +0530
Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS*
	device?
In-Reply-To: <4FBE0E4A.6090408@integrafin.co.uk>
References: <CBE0018C.34B31%krice@freeswitch.org>
	<4FBAA1CF.5060907@integrafin.co.uk>
	<CAKbxfG_csVT5A89DYnVirVKRkar3RaCBVAxT4gzCABvNgboRCA@mail.gmail.com>
	<CAKzWOxUj-q2b_NN08F0OD6Je5oJXkUa7VPR84q27BT4hGN1kEA@mail.gmail.com>
	<4FBAE7AE.1000606@coppice.org>
	<CAGKPtBst3pfELgXX1UoLZg_KxHizcwtfJefnibj2XhH6Ko7_wQ@mail.gmail.com>
	<4FBE0E4A.6090408@integrafin.co.uk>
Message-ID: <CAGKPtBt+WMHe9HA=2t0zq76ogToaXd4-8dEP7wYG7qDKw76K8A@mail.gmail.com>

Able to both receive and send Fax using emulated soft modem via mod_spandsp
and HylaFax.

Wikified at
http://wiki.freeswitch.org/wiki/HylaFax

A big thank you to everyone!


On Thu, May 24, 2012 at 4:02 PM, Alex Crow <acrow at integrafin.co.uk> wrote:

> On 24/05/12 09:07, Anita Hall wrote:
> > Many thanks to everybody!
> >
> > I will put this on wiki but I am stuck at the very beginning. I am
> > sorry for being such a noob but this is my first tryst with modems.
> >
> > I have modems appearing as /dev/FS[0-4] which link to /dev/pts/[4-7]
> >
> > # ls -l /dev/FS?
> > lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS0 -> /dev/pts/4
> > lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS1 -> /dev/pts/5
> > lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS2 -> /dev/pts/6
> > lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS3 -> /dev/pts/7
> > lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS4 -> /dev/pts/8
> >
> > But,
> > # cu -l /dev/FS0
> > cu: open (/dev/FS0): Permission denied
> > cu: /dev/FS0: Line in use
> >
> > I think this is because FreeSWITCH is using this device, but then how
> > does HylaFax or any other program like cu talk to it ?
> >
> > This page says that I should be able to check my modem before I
> > configure it with Hylafax.
> >
> http://www.hylafax.org/content/Handbook:Basic_Server_Configuration:Checking_your_Modem
> >
> >
> > regards,
> > Anita
> >
> >
>
> The permissions on the links are not relevant. It's the permissions on
> the /dev/pts/* devices that take effect.
>
> Alex
>
> --
> This message is intended only for the addressee and may contain
> confidential information.  Unless you are that person, you may not
> disclose its contents or use it in any way and are requested to delete
> the message along with any attachments and notify us immediately.
>
> "Transact" is operated by Integrated Financial Arrangements plc
> Domain House, 5-7 Singer Street, London  EC2A 4BQ
> Tel: (020) 7608 4900 Fax: (020) 7608 5300
> (Registered office: as above; Registered in England and Wales under
> number: 3727592)
> Authorised and regulated by the Financial Services Authority (entered on
> the FSA Register; number: 190856)
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From davidwaf at gmail.com  Fri Jun  1 16:01:25 2012
From: davidwaf at gmail.com (davidwaf at gmail.com)
Date: Fri, 1 Jun 2012 12:01:25 +0000
Subject: [Freeswitch-users] Spandsp mulated modems inbound/no
	/dev/FS*	device?
In-Reply-To: <CAGKPtBt+WMHe9HA=2t0zq76ogToaXd4-8dEP7wYG7qDKw76K8A@mail.gmail.com>
References: <CBE0018C.34B31%krice@freeswitch.org>
	<4FBAA1CF.5060907@integrafin.co.uk>
	<CAKbxfG_csVT5A89DYnVirVKRkar3RaCBVAxT4gzCABvNgboRCA@mail.gmail.com>
	<CAKzWOxUj-q2b_NN08F0OD6Je5oJXkUa7VPR84q27BT4hGN1kEA@mail.gmail.com>
	<4FBAE7AE.1000606@coppice.org>
	<CAGKPtBst3pfELgXX1UoLZg_KxHizcwtfJefnibj2XhH6Ko7_wQ@mail.gmail.com>
	<4FBE0E4A.6090408@integrafin.co.uk>
	<CAGKPtBt+WMHe9HA=2t0zq76ogToaXd4-8dEP7wYG7qDKw76K8A@mail.gmail.com>
Message-ID: <1994487283-1338552086-cardhu_decombobulator_blackberry.rim.net-234513875-@b26.c15.bise7.blackberry>

Sounds good to me. Cheers
Sent from my BlackBerry?

-----Original Message-----
From: Anita Hall <anita.hall at simmortel.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Fri, 1 Jun 2012 17:21:05 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS*
	device?

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


From dgarcia at anew.com.ve  Fri Jun  1 17:10:35 2012
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Fri, 01 Jun 2012 08:40:35 -0430
Subject: [Freeswitch-users] Freeswitch & FreeTDM question...
In-Reply-To: <C066384B0C1F1E4B9B870F050D7F1A9D401C2220@ADAMA.mcpeekdodge.local>
References: <C066384B0C1F1E4B9B870F050D7F1A9D401C2220@ADAMA.mcpeekdodge.local>
Message-ID: <4FC8BF4B.2060507@anew.com.ve>

Hi, Frank,

I just still novice with FS.

I don't think could be a problem in mod_freetdm. As far I see, when you 
do a unload/reload of freetdm, freetdm reincialize the hardware. Your 
hardware could getting stuck in a condition where the sangoma driver 
(wanpipe or dahdi) could not recover.

I suggest you try to narrow what is going on with your faxes. If you 
have a buttset (http://www.phonegeeks.com/buttsets1.html) , you could 
verify if the line is giving dial tone or not. Could be a voltage 
problem. Hardware incompatibility,  kernel issue, etc, etc, etc.

The second thing is update your hardware software to latest releases.

On 5/31/2012 6:05 PM, Frank Busalacchi Jr wrote:
>
> Hi, and thanks in advance for your help.
>
> I run freeswitch with a SangomA A102 (FXO/FXS) card.  The freeswitch 
> code base that it is running on is about 3 months old.  It is a 
> production server, and taking it down other than in the middle of the 
> night is problemsome...  It is running on Debian Lenny.
>
> In the next couple of days I am going to be spending the night, doing 
> a fresh install of the latest debian (squeeze) at which time I will do 
> a fresh GIT, fresh sangoma wanpipe and be totally current on FreeSwitch.
>
> My Question:
>
> I am having intermittent problems with my FXO/FXS ports.  Basically I 
> have fax machines attached.  Every so often (sometimes 5-10 days in 
> between), the fax machine says that it has no dial tone.  I am able to 
> resolve the problem by going to the CLI and unloading mod_freetdm and 
> then reloading mod_freetdm.  When the "no dialtone problem" is 
> happening, unloading mod_freetdm takes 10 seconds or so...I have even 
> had it crash freeswitch during the unload...
>
> I haven't attempted to narrow down the issue yet, and thought I might 
> ask for some guidance before heading down that path.
>
> One of my thoughts is that perhaps the Hardware is bad, but the fact 
> that unloading/loading mod_freetdm fixes it makes me think it is 
> something in mod_freetdm...
>
> Your thoughts?
>
> -Frank
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2178 / Virus Database: 2425/5038 - Release Date: 06/01/12
>


-- 
Atentamente,
*Dario Garc?a*
Consultor.

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Tel?fono: +58 212 9081842
Cel: +58 412 2221515
dgarcia at anew.com.ve
http://www.anew.com.ve
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From valernur at yahoo.com  Fri Jun  1 17:45:18 2012
From: valernur at yahoo.com (Valer Nur)
Date: Fri, 1 Jun 2012 06:45:18 -0700 (PDT)
Subject: [Freeswitch-users] Monitoring audio quality
In-Reply-To: <2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com>
References: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
	<2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com>
Message-ID: <1338558318.79806.YahooMailNeo@web120405.mail.ne1.yahoo.com>

Hi Mike,
If you are interested to achieve real-time improving of the quality (as opposed to passive monitoring) you can try PBXMate. This tool is *NOT* a passive monitoring tool but an active tool. It works like a Sip Proxy and if it encounters a quality issue during a call (e.g. noise and/or low volume) it will fix it in real-time. As a by product the PBXMate also provides statistics on the quality.? You can see one option for integration in the wiki http://wiki.freeswitch.org/wiki/PBXMate-FreeSWITCH-integration

Cheers,





________________________________
 From: Ash <ash at archerdrive.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Thursday, May 31, 2012 12:30 PM
Subject: Re: [Freeswitch-users] Monitoring audio quality
 
Hi Michael,

Not exactly an answer to your question, but? I have quite a lot of call traffic running over three servers.? I have found the best way to manage this is by using a monitoring tool.? The one I use is http://www.voipmonitor.org/, the actual daemon is open source but you can buy a webUI which uses rrdtool to graph the call quality and display the calls history.? Using a tool like this you can get an idea of call quality for each call.


Cheers,

Ash.


On 31/05/2012, at 7:19 PM, Michael Lutz wrote:

> Hi Guys,
> 
> I have a prety large system with a lot of traffic. Sometimes I get
> complains from customers about 'loud noises', 'not hearing audio', or
> just 'bad quality audio'.
> I can see a lot of info on the CDR's, no I have tried to search the
> wiki and groups for some explanaition about how to interpret them, but
> there is not much to find..
> 
> Is there someone who can tell me what these mean, and how to interpret
> them, could they help me in finding issues with audio?
> 
>? ? <rtp_audio_in_raw_bytes>573964</rtp_audio_in_raw_bytes>
>? ? <rtp_audio_in_media_bytes>572932</rtp_audio_in_media_bytes>
>? ? <rtp_audio_in_packet_count>3337</rtp_audio_in_packet_count>
>? ? <rtp_audio_in_media_packet_count>3331</rtp_audio_in_media_packet_count>
>? ? <rtp_audio_in_skip_packet_count>47</rtp_audio_in_skip_packet_count>
>? ? <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count>
>? ? <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count>
>? ? <rtp_audio_in_cng_packet_count>0</rtp_audio_in_cng_packet_count>
>? ? <rtp_audio_in_flush_packet_count>6</rtp_audio_in_flush_packet_count>
>? ? <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
>? ? <rtp_audio_out_raw_bytes>485556</rtp_audio_out_raw_bytes>
>? ? <rtp_audio_out_media_bytes>485556</rtp_audio_out_media_bytes>
>? ? <rtp_audio_out_packet_count>2823</rtp_audio_out_packet_count>
>? ? <rtp_audio_out_media_packet_count>2823</rtp_audio_out_media_packet_count>
>? ? <rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count>
>? ? <rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count>
>? ? <rtp_audio_out_cng_packet_count>0</rtp_audio_out_cng_packet_count>
>? ? <rtp_audio_rtcp_packet_count>0</rtp_audio_rtcp_packet_count>
>? ? <rtp_audio_rtcp_octet_count>0</rtp_audio_rtcp_octet_count>
> 
> Or are there other (better) methods of finding problems related to bad audio?
> 
> 
> Regards,
> Mike
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From nasida at live.ru  Fri Jun  1 20:07:21 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Fri, 1 Jun 2012 20:07:21 +0400
Subject: [Freeswitch-users] mod_soundtouch
Message-ID: <SNT127-W60745261C95B7D6208A5FBD5080@phx.gbl>


Hi guys.
I test mod_soundtouch. All work fine but how can I  reduce delay of transmiting of voice ? Now I wait about 5 sec and I would like to have realtime.
Please adviseThanks.
 		 	   		  
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From msc at freeswitch.org  Fri Jun  1 21:13:23 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Jun 2012 10:13:23 -0700
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <4FC896B8.9070102@digitalmail.com>
References: <4FC896B8.9070102@digitalmail.com>
Message-ID: <CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>

Why do you need to use loopback at all?
-MC

On Fri, Jun 1, 2012 at 3:17 AM, Alex Lake <alex at digitalmail.com> wrote:

> Got a lua script for a B-party "mid-call menu". Is it legitimate to do..
> "session:execute("att_xfer", "loopback/"..destnum)"
>
> I've tried it and it seems to start off doing the right things, but my
> A-party gets disconnected as soon as the call to the C-Party (the person
> I'm transferring the call to) answers the call.
>
> Maybe better to try to orchestrate the entire affair from within the lua
> script? (Tricky for a beginner like me!)
>
> Thanks,
> Alex
>
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From msc at freeswitch.org  Fri Jun  1 21:16:28 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Jun 2012 10:16:28 -0700
Subject: [Freeswitch-users] Custom Registration
In-Reply-To: <4FC8716B.9080300@seletech.com>
References: <4FC71FD9.5010004@seletech.com> <4FC7BB80.9050600@elder.hu>
	<4FC8716B.9080300@seletech.com>
Message-ID: <CAKzWOxUWTL4b5+EFR2P2SVCWfQD0z+wO9=fzii4y16ajA5QECQ@mail.gmail.com>

> Problem Solved!! I just wrote this line:
>
> <param name="max-registrations-per-extension" value="1"/>
>
> within the "params" tag when defining a new user in the
> ~/freeswitch/conf/directory/default/ directory. Then issue "reloadxml"
> command and everything works as I wanted to: if Extension A is already
> registered with id ID_A and extension B tries to register with id ID_A,
> the registration is rejected ^_^
>
> Thank you!
>
> --
> Dr. Villa Alberto
> Sw Engineer
>
>
The wiki page could use some love. Would you mind adding something useful
here:
http://wiki.freeswitch.org/wiki/Sofia.conf.xml#max-registrations-per-extension

Thanks,
MC
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From nasida at live.ru  Fri Jun  1 22:14:41 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Fri, 1 Jun 2012 22:14:41 +0400
Subject: [Freeswitch-users] mod_soundtouch
In-Reply-To: <SNT127-W60745261C95B7D6208A5FBD5080@phx.gbl>
References: <SNT127-W60745261C95B7D6208A5FBD5080@phx.gbl>
Message-ID: <SNT127-W262E04BF4A0CB78824EAD6D5080@phx.gbl>


Solved.

From: nasida at live.ru
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 1 Jun 2012 20:07:21 +0400
Subject: [Freeswitch-users] mod_soundtouch





Hi guys.
I test mod_soundtouch. All work fine but how can I  reduce delay of transmiting of voice ? Now I wait about 5 sec and I would like to have realtime.
Please adviseThanks.
 		 	   		  

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From alexanderjp at thinksimplicity.net  Fri Jun  1 22:42:35 2012
From: alexanderjp at thinksimplicity.net (Perovich Alexander)
Date: Fri, 1 Jun 2012 14:42:35 -0400
Subject: [Freeswitch-users] mod_directory and LUA for database users
In-Reply-To: <B2C68DE4-E421-4900-BAB7-1767E5D0DE88@thinksimplicity.net>
References: <B2C68DE4-E421-4900-BAB7-1767E5D0DE88@thinksimplicity.net>
Message-ID: <96004A14-ACA2-430E-AD9F-702ACEFFD554@thinksimplicity.net>


Hello,

I have been trying to get the mod_directory to query the users that are stored in a database and accessed using a lua script, however every attempt comes back match cannot be found.

Any ideas?

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From msc at freeswitch.org  Fri Jun  1 23:18:20 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Jun 2012 12:18:20 -0700
Subject: [Freeswitch-users] Sending arbitrary NOTIFY messages
Message-ID: <CAKzWOxVF_01Bf4=mKUjHeKtUict1uNO52nvuagZb5zQJcX6bXw@mail.gmail.com>

Has anyone gotten the magic formula down for doing this? I'm trying to send
a NOTIFY to a non-registered endpoint but I can't quite get the event
headers correct. I found this old commit that Tony did about 3 years ago
that suggests it's possible:
http://fisheye.freeswitch.org/changelog/freeswitch.git/?showid=1fa1e961e4587475a51dcbadd31765b5a06d3115

If you know what the necessary headers are, or better yet, if you have a
working example please let me know. And yes, I will gladly pay the wiki tax
on this one. ;)

Thanks,
MC
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From mario_fs at mgtech.com  Fri Jun  1 23:25:34 2012
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 1 Jun 2012 12:25:34 -0700
Subject: [Freeswitch-users] NAT issues and "best practices"
In-Reply-To: <052201cd3f96$6fe5ae20$4fb10a60$@bizfocused.com>
References: <036901cd3f58$bb857510$32905f30$@bizfocused.com>
	<43E524DB-4519-4531-8CD4-4B01C95882B6@url.net.au>
	<052201cd3f96$6fe5ae20$4fb10a60$@bizfocused.com>
Message-ID: <80CD8D38-3E1B-4B05-AC8E-7CA0C40EDB66@mgtech.com>

I had the RVS4000 and others. Got a Zyxel USG100 which has dual wan, and SIP ALG on it solved all my FreeSwitch NAT issues and handles fallover well. It took a year to find this router and I plan on a Wiki for it since so many people have NAT issues and this solved everything. I even have 1 static DSL and 1 dynamic and FS always works no matter what line is down. BTW, I have SPA962 and iPads and iPhones to the FS on a Mac Mini, then to 2 Westell DSL modems, then out to the phone lines. Take my word for it, USG100 vs RVS4000 is no contest and you'll never look back. My 21 cents.... from experience....
Mario G

On May 31, 2012, at 6:32 PM, Sean Devoy wrote:

> Thanks for the response.
>  
> At this point I can say I have  2 working and 1 not working configuration.  Also, thanks for the SIP ALG tip (3 hours too late!)  I turned it on and things went to crap.
>  
> So my WOKING configa are:
> <FS w/ local IP> to <FIOS 25MB router> to <Motorla Cable Modem> to <Cisco RVS4000> to <CISCO SPA-504G phones>
> <FS w/ local IP> to <FIOS 25MB router> to <Motorla Cable Modem> to <Linksys home router ???? > to <CISCO SPA-504G phones>
> Calls, MWI, SCA, etc across those 2 configs works PERFECTLY.
> I left out <fs> on local lan to <Cisco 504g> is wonderful too.
>  
> Unfortunately, the customer setup is:
> <FS w/ local IP> to <FIOS 25MB router> to <COMCAST Business Cable Modem> to <Cisco E1200> to <CISCO SPA-504G phones>
> Results are all over the place.  I don?t think they can support 2 inbound calls at the same time.  MWI is spotty, SCA is not working.  Turning on SIP ALG on Cisco E1200 led to a small riot and has been banished from the kingdom for every.
>  
> I think tomorrow we are getting them a Cisco RVS4000.
>  
> Sean
>  
>  
> From: Ashley Breeden [mailto:ash at url.net.au] 
> Sent: Thursday, May 31, 2012 7:52 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] NAT issues and "best practices"
>  
> Hi Sean,
>  
> Yes SIP and NAT do not play well together.  Most of our customers all use some form of NAT to access our FreeSWITCH servers, although not double NAT.  I have found that pretty much most of the current mainstream routers and phones work as long as you have the latest software.  One of the biggest things I find is that if you have more than one SIP device registered behind a NAT with SIP ALG you will get very mixed results.
>  
> It will also depend on how many devices you have behind the NAT, some routers seem to have issue processing NAT for more than 10 devices all originating on port 5060.
>  
> Here is a few things I will often do to get a customer to connect via a NAT device:
>  
> - Turn off SIP ALG on the router
> - Look for a setting on your VoIP device called Rport, set this to enabled.  
> - Try putting each phones local SIP port on its own unique port, e.g. extension 1001 = port 5001, 1002 = port 5002
> - Try TCP as the transport protocol
>  
> Also have a look here I have been able to get some useful information from this article - http://wiki.freeswitch.org/wiki/ALG, especially the DG834's with DOS protection.
>  
> Personally I use the following routers with no issues:
>  
> - Billion 7401VGP
> - Billion 7800NL
> - Billion 7404
> - Cisco 857/877
> - MicroTik seemed to be quite reasonable as well
>  
> For Phones I have running:
>  
> - Yealink T22,T26,T28,T32,T38
> - Snom 360
> - Siemens Gigaset
>  
>  
> Hope this helps.
>  
> Cheers,
>  
> Ash
>  
>  
> On 01/06/2012, at 4:10 AM, Sean Devoy wrote:
> 
> 
> HI All,
>  
> I have a customer location that has just been a nightmare to implement.  I am just learning that they ?may? have multiple NAT routers in sequence at their location.  I think I fully understand what NAT is trying to accomplish. There seem to be different levels and approaches. 
>  
> The most basic NAT setup (to me)  is a HOME LAN with multiple PCs where NAT allows multiple devices to share a single routable IP address on the WAN side from multiple local devices the LAN side.  Note I said OUTBOUND initiated connections.  Even FTP can have trouble with this level.  Almost all inbound traffic is blocked for security.
>  
> Clearly for FS we need the switch to be able to punch through from the WAN to specific local IPs on the LAN to reach specific phones.  This is INBOUND NAT and brings up many security issues for people.  Even on devices where you get this ?working? you may only be able to support one line per phone or a single inbound connection at a time.
>  
> I understand NAT has PMP and UPnP protocols and FS ?supports? both.  What I can?t find is where someone says ?Here is a great setup that works with cheap, available ?commodity? hardware from Cisco/Linksys that supports all the NAT you need for FS.?  I don?t care if it is PMP or UPnP and I might not even care why you pick one over the other, although it is probably a ?good read?.
>  
> Can someone just stand up say ?FS works GREAT with the XYZ router in ABC mode from MY COMPANY using NAT to Cisco phones??
>  
> I have seen some articles about Freeware/Shareware firmware in this devices, but as a novice I want to limit the unknowns until I get more up to speed.
>  
> My specific issue now is that I cannot get SCA to work at the NAT location.  I issued:
> sofia_contact 220 at mydomain.com
> sofia/external/sip:200@<ip address>:44234,sofia/external/sip:200@<same ip>:1024
>  
> Only one phone rings on inbound and the line indicator light does not change when either is picked up.  Same configuration is working on our LAN with the switch.
>  
> I am absolutely ready to by a router to fix these issues, I don?t want to lose this customer.
>  
> Thanks for your thoughts,
> Sean
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From cesar.bermudez at gmail.com  Fri Jun  1 23:35:31 2012
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Fri, 1 Jun 2012 16:35:31 -0300
Subject: [Freeswitch-users] mod_soundtouch
In-Reply-To: <SNT127-W262E04BF4A0CB78824EAD6D5080@phx.gbl>
References: <SNT127-W60745261C95B7D6208A5FBD5080@phx.gbl>
	<SNT127-W262E04BF4A0CB78824EAD6D5080@phx.gbl>
Message-ID: <CADrcvV8POvyUfSaWsA1LHB30uuU7yHY2BqwV=nW5ZH1tN2GTGA@mail.gmail.com>

maybe you can share or wikify that?

On Fri, Jun 1, 2012 at 3:14 PM, Yuriy Nasida <nasida at live.ru> wrote:

>  Solved.
>
> ------------------------------
> From: nasida at live.ru
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 1 Jun 2012 20:07:21 +0400
> Subject: [Freeswitch-users] mod_soundtouch
>
>
>  Hi guys.
>
> I test mod_soundtouch. All work fine but how can I  reduce delay of
> transmiting of voice ? Now I wait about 5 sec and I would like to have
> realtime.
>
> Please advise
> Thanks.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel
> Communication Server  Official FreeSWITCH Sites
> http://www.freeswitch.org http://wiki.freeswitch.org
> http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Sat Jun  2 04:02:57 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 01 Jun 2012 20:02:57 -0400
Subject: [Freeswitch-users] mod_soundtouch
In-Reply-To: <CADrcvV8POvyUfSaWsA1LHB30uuU7yHY2BqwV=nW5ZH1tN2GTGA@mail.gmail.com>
References: <SNT127-W60745261C95B7D6208A5FBD5080@phx.gbl>
	<SNT127-W262E04BF4A0CB78824EAD6D5080@phx.gbl>
	<CADrcvV8POvyUfSaWsA1LHB30uuU7yHY2BqwV=nW5ZH1tN2GTGA@mail.gmail.com>
Message-ID: <4FC95831.8000001@endigotech.com>

On Fri 01 Jun 2012 03:35:31 PM EDT, Cesar Bermudez wrote:
> maybe you can share or wikify that?
>
> On Fri, Jun 1, 2012 at 3:14 PM, Yuriy Nasida <nasida at live.ru
> <mailto:nasida at live.ru>> wrote:
>
>     Solved.
>
>     ------------------------------------------------------------------------
>     From: nasida at live.ru <mailto:nasida at live.ru>
>     To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     Date: Fri, 1 Jun 2012 20:07:21 +0400
>     Subject: [Freeswitch-users] mod_soundtouch
>
>
>     Hi guys.
>
>     I test mod_soundtouch. All work fine but how can I  reduce delay
>     of transmiting of voice ? Now I wait about 5 sec and I would like
>     to have realtime.
>
>     Please advise
>     Thanks.
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
>     CudaTel Communication Server  Official
>     FreeSWITCH Sites http://www.freeswitch.org
>     http://wiki.freeswitch.org http://www.cluecon.com Join Us At
>     ClueCon - Aug 7-9, 2012 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
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
Or at the very least, share on the mailing list for Google to index...

-BDF



From saami_mh at ymail.com  Sat Jun  2 13:02:21 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Sat, 2 Jun 2012 02:02:21 -0700 (PDT)
Subject: [Freeswitch-users] regular expression ?
Message-ID: <1338627741.80797.YahooMailNeo@web120106.mail.ne1.yahoo.com>

hi guys,
it possible to match *999*11# in regular expression in freeswitch ?
in the wiki the above example don't exis,
plz help thanks
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From covici at ccs.covici.com  Sat Jun  2 13:48:12 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 02 Jun 2012 05:48:12 -0400
Subject: [Freeswitch-users] regular expression ?
In-Reply-To: <1338627741.80797.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1338627741.80797.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <28858.1338630492@ccs.covici.com>

Just put a \ before each * and maybe the # as well and you should be
good to go.

Samira Mh <saami_mh at ymail.com> wrote:

> hi guys,
> it possible to match *999*11# in regular expression in freeswitch ?
> in the wiki the above example don't exis,
> plz help thanks
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From saami_mh at ymail.com  Sat Jun  2 14:11:57 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Sat, 2 Jun 2012 03:11:57 -0700 (PDT)
Subject: [Freeswitch-users] regular expression ?
In-Reply-To: <28858.1338630492@ccs.covici.com>
References: <1338627741.80797.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<28858.1338630492@ccs.covici.com>
Message-ID: <1338631917.10976.YahooMailNeo@web120106.mail.ne1.yahoo.com>

thanks so much for your reply,nut i think the below patter is true as well,?
^([*]999[*]11[#])$?
is that true?



________________________________
 From: "covici at ccs.covici.com" <covici at ccs.covici.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Saturday, June 2, 2012 2:18 PM
Subject: Re: [Freeswitch-users] regular expression ?
 
Just put a \ before each * and maybe the # as well and you should be
good to go.

Samira Mh <saami_mh at ymail.com> wrote:

> hi guys,
> it possible to match *999*11# in regular expression in freeswitch ?
> in the wiki the above example don't exis,
> plz help thanks
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 life is like a penny.? You're going to lose it.? The question is:
How do
you spend it?

? ? ? ?  John Covici
? ? ? ? covici at ccs.covici.com

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From nasida at live.ru  Sat Jun  2 15:23:09 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Sat, 2 Jun 2012 15:23:09 +0400
Subject: [Freeswitch-users] mod_soundtouch
In-Reply-To: <4FC95831.8000001@endigotech.com>
References: <SNT127-W60745261C95B7D6208A5FBD5080@phx.gbl>,
	<SNT127-W262E04BF4A0CB78824EAD6D5080@phx.gbl>,
	<CADrcvV8POvyUfSaWsA1LHB30uuU7yHY2BqwV=nW5ZH1tN2GTGA@mail.gmail.com>,
	<4FC95831.8000001@endigotech.com>
Message-ID: <SNT127-W49F5F7B55E84A51C470AC1D5090@phx.gbl>


Sure but in general there is nothing important for sharing. I just used default string from wiki<action application="soundtouch" data="hook_dtmf -0.3o 0.8r"/>0.8r - is not suitable. I had delay 5 sec with it. 
I used instead:
<action application="soundtouch" data="0.7o 0.99t"/>Really funny. The voice looks like as after ball with helium ;)
and
<action application="soundtouch" data="-0.7o 0.99t"/>Voice like Darth Vader :)
-- Yuriy
> Date: Fri, 1 Jun 2012 20:02:57 -0400
> From: bdfoster at endigotech.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_soundtouch
> 
> On Fri 01 Jun 2012 03:35:31 PM EDT, Cesar Bermudez wrote:
> > maybe you can share or wikify that?
> >
> > On Fri, Jun 1, 2012 at 3:14 PM, Yuriy Nasida <nasida at live.ru
> > <mailto:nasida at live.ru>> wrote:
> >
> >     Solved.
> >
> >     ------------------------------------------------------------------------
> >     From: nasida at live.ru <mailto:nasida at live.ru>
> >     To: freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>
> >     Date: Fri, 1 Jun 2012 20:07:21 +0400
> >     Subject: [Freeswitch-users] mod_soundtouch
> >
> >
> >     Hi guys.
> >
> >     I test mod_soundtouch. All work fine but how can I  reduce delay
> >     of transmiting of voice ? Now I wait about 5 sec and I would like
> >     to have realtime.
> >
> >     Please advise
> >     Thanks.
> >
> >
> >     _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
> >     CudaTel Communication Server  Official
> >     FreeSWITCH Sites http://www.freeswitch.org
> >     http://wiki.freeswitch.org http://www.cluecon.com Join Us At
> >     ClueCon - Aug 7-9, 2012 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
> >
> >     _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com
> >
> >     
> >     
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org
> >     http://wiki.freeswitch.org
> >     http://www.cluecon.com
> >
> >     Join Us At ClueCon - Aug 7-9, 2012
> >
> >     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
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> Or at the very least, share on the mailing list for Google to index...
> 
> -BDF
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From lazyvirus at gmx.com  Sat Jun  2 17:27:43 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Sat, 2 Jun 2012 15:27:43 +0200
Subject: [Freeswitch-users] regular expression ?
In-Reply-To: <1338627741.80797.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1338627741.80797.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <20120602152743.6c823e92@anubis.defcon1>

On Sat, 2 Jun 2012 02:02:21 -0700 (PDT)
Samira Mh <saami_mh at ymail.com> wrote:

> it possible to match *999*11# in regular expression in freeswitch ?
> in the wiki the above example don't exis,

You definitely should have a look at:
http://www.regular-expressions.info/

You could also use: http://www.quanetic.com/Regex
but _please_ read carefully the first one 'cos it will allow you to
build your own expressions without any help, so use this one _only_
to validate the result of your own work:)
[NB: There are plenty of other sites like this one].

My ?1+?1

JY
-- 
I can't think about that.  It doesn't go with HEDGES in the shape of
LITTLE LULU -- or ROBOTS making BRICKS ...


From nbhatti at gmail.com  Sat Jun  2 19:58:39 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Sat, 2 Jun 2012 18:58:39 +0300
Subject: [Freeswitch-users] Stuck calls
In-Reply-To: <CAKbxfG9dREahuS3_CqavBbWc4BdmyPGQUNaiGg4JPja3aJZ=aA@mail.gmail.com>
References: <CACak_yyYnKn12KdEQkOdeg0Lwzq=QoDjX_oBitSgKnFEQSGzMA@mail.gmail.com>
	<CAKbxfG_-zDrq1VZpE-b8y+5B7bucGOV4pAwUoKfMawhAjpB3Rg@mail.gmail.com>
	<CACak_yzD21VMiiPG48h1XsCyn5EhqkMPDREoAtnPTA_1ifmSZw@mail.gmail.com>
	<CAKzWOxVAcz+viU0oyUMPQ-gPXL4QosdSuGfAx3Woq-hmLnxFeA@mail.gmail.com>
	<CACak_yy8m=WBpUdPpwuG0BGTiJs4rz+462=mhuFDzaTRAuyvjg@mail.gmail.com>
	<CAKbxfG9dREahuS3_CqavBbWc4BdmyPGQUNaiGg4JPja3aJZ=aA@mail.gmail.com>
Message-ID: <CACak_yx77eR+dKWyeWz7-ix1PKisi_QWrOo6184z_+1fmPCOdg@mail.gmail.com>

Here we go again ..

freeswitch at internal> status
UP 0 years, 0 days, 18 hours, 13 minutes, 28 seconds, 780
milliseconds, 883 microseconds
FreeSWITCH is ready
51822 session(s) since startup
2000 session(s) 0/100
2000 session(s) max
min idle cpu 0.00/99.00

freeswitch at internal> show channels count

2000 total.

freeswitch at internal> show calls count

2000 total.

I am going to update to latest and try again and may open a JIRA. This
is now happening too often. I am on FreeSWITCH Version 1.1.beta1
(git-2a0ca29 2012-04-27 15-28-17 -0500).



On Fri, May 18, 2012 at 11:42 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> That spells out SQL error.
> The show channels is just a db keeping track of the channel status
> based on events.
>
> If you can reproduce it on git HEAD open a jira
>
>
> On Fri, May 18, 2012 at 3:01 PM, Muhammad Naseer Bhatti
> <nbhatti at gmail.com> wrote:
>> I was trying to dig some information out of it before a restart. There
>> is no traffic on this box right now, so I can restart/retest anything.
>> I'll update, grab some logs and will post back.
>>
>> Thanks.
>>
>> On Fri, May 18, 2012 at 10:50 PM, Michael Collins <msc at freeswitch.org> wrote:
>>> Can you turn debugging back on for a period of time and re-test? Also, any
>>> chance you can get to latest git for testing?
>>> -MC
>>>
>>>
>>> On Fri, May 18, 2012 at 12:18 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com>
>>> wrote:
>>>>
>>>> Show channels shows 46 calls, status shows 0. pb @
>>>> http://pastebin.freeswitch.org/19093
>>>> Logging was turned of from this production server to gain some disk
>>>> I/Os. Using mod_xml_cdr for cdr posting. Web server could possibly the
>>>> reason and ?if so, why the calls are in RINGING state? CDR is only
>>>> posted at the end of the call. Looks like we don't have much evidence.
>>>>
>>>> Goni
>>>>
>>>> On Fri, May 18, 2012 at 8:52 PM, Anthony Minessale
>>>> <anthony.minessale at gmail.com> wrote:
>>>> > compare "show channels" to "status" if status says there are 0 calls,
>>>> > look for a db error somewhere in fs log [or db specific log if you
>>>> > have odbc]
>>>> > cat freeswitch.log | grep CRIT
>>>> >
>>>> > if status sees channels too, see if you have a cdr module that is
>>>> > doing some post processing that could be stuck.
>>>> >
>>>> >
>>>> >
>>>> >
>>>> > On Fri, May 18, 2012 at 12:45 PM, Muhammad Naseer Bhatti
>>>> > <nbhatti at gmail.com> wrote:
>>>> >> I got like 46 calls stuck in FreeSWITCH. The switch is not doing any
>>>> >> traffic right now. It is running FreeSWITCH Version 1.2.0 (git-0709cc6
>>>> >> 2012-05-16 02-50-13 +0000). Should I file jira or look for something
>>>> >> else? They are mostly in CS_INIT,,,,,,,RINGING --
>>>> >> CS_ROUTING,,,,,,,DOWN and CS_CONSUME_MEDIA state.
>>>> >>
>>>> >> Thanks.
>>>> >>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From steveayre at gmail.com  Sat Jun  2 20:02:49 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 2 Jun 2012 17:02:49 +0100
Subject: [Freeswitch-users] DTMF Passthrough on different legs
In-Reply-To: <CAPu0Fr1bhfxXUiZuoWctVFi93pyB47pukgHG4RvP8pkrs5AVOA@mail.gmail.com>
References: <6ECAF1527329364583AB525CF34ABF950B31A67C@ms.kallback.com>
	<CAPu0Fr1bhfxXUiZuoWctVFi93pyB47pukgHG4RvP8pkrs5AVOA@mail.gmail.com>
Message-ID: <286B66B5-4234-4E09-883D-4E4EF901798F@gmail.com>

Re codec prefs... There is no such thing as 'G711' and it'll be ignored. It would be best to remove that from your config for clarity and to avoid problems later, eg you think you have 711 enabled but you don't because you have G711 but have removed PCMx. PCMA and PCMU are the correct definitions for G711.

-Steve

Sent from my iPad



On 31 May 2012, at 20:24, Michael Lutz <mytemike72 at gmail.com> wrote:

> According to CDRS's:
> 
> Session:A -> Inbound, PCMU-8000
> Session B -> FS to FS, PCMU-8000
> Session C -> Outbound, PMCA-8000
> 
> According to vars.xml:
>    <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G711,PCMU,PCMA"/>
>    <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G711,PCMU,PCMA"/>
> 
> 
> Regards,
> Mike.
> 
> 2012/5/31 JRichey <jrichey at itltd.net>:
>> What CODECs are being used?
>> 
>> In-band DTMF won't work correctly with compressed CODECs like G.729.
>> 
>> 
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of
>> Michael Lutz
>> Sent: Thursday, May 31, 2012 2:12 AM
>> To: FreeSWITCH Users Help
>> Subject: [Freeswitch-users] DTMF Passthrough on different legs
>> 
>> 
>> Hi Guys,
>> 
>> I am stuck with a bit of a problem I cannot figure out where it is going
>> wrong..
>> 
>> Here's my situation,
>> 
>> I have an inbound call, comming from an external provider, they don't
>> support RFC2833 so I turn on inband detection using 'start_dtmf' in
>> the dialplan.
>> This works ok, for the incomming channel and IVR provided (via Lua) to
>> the caller. (Though sometimes it seems to misinterpret and returns
>> wrong keys..) (Session:A)
>> 
>> After that, this call is bridged (via Lua to a a new inbound session
>> on the same switch) (without specifying start_dtmf) so RC2388 is used
>> by default (?) (Session:B)
>> 
>> This new sessions is playing soem audio (Lua) and is heard by the
>> original caller (the two sessions are bridged).
>> 
>> 
>> Seperatly I have an inbound ESL connection where I do an origanate
>> using a originate {execute_on_answer=start_dtmf, .... etc) &park() to
>> setup an new external call via the same gateway (they don't support
>> RFC2388 so I activate the inband detection on the b-leg to receive the
>> digits.
>> Let's call this leg Session:C.
>> 
>> After the orignated call is answered I bridge the parked call with the
>> previous session (Session:B), so at the end Session:A and Session:B
>> can talk to each other.
>> 
>> Al this works perfectly.
>> 
>> [Incommming call] Session:A ----- INBAND ------> Bridge(FS.Lua) --->
>> Session:B ---- RFC2388 ---->   FS.Bridge ---> Session:C --- INBAND
>> ------> ESL destination number
>> 
>> 
>> My symptoms are...
>> 
>> 1. I have a customer who had a IVR on his own side, (Session:C) and
>> requires input from Session:A. The weird thing is that it seems the
>> input is wrong.
>> Keys entered do not correspond to what his IVR is receiving. As if
>> newly dtmfs are generated somewhere.
>> When I switch to another inbound provider, making Session:A using
>> rfc2388 and not inband, but do dialout via the provider using inband
>> detection. It seems to work fine.
>> 
>> 
>> 2. Not being able to get the dtmf's from the ESL Destination number
>> (Session:C). I'm reading this by subscribing to events via ESL for
>> that particulair uuid.
>> When I have a connection (inbound) without inband detection it seems
>> to get the DTMF's from Session:C without any problems. Which makes it
>> weird for me.
>> 
>> note: I know the setup is a bit odd, but Session A cannot exit it's
>> Lua script, so I needed to do a workaround to be able to keep te lua
>> alive and being able to control the call (audio) via ESL.
>> 
>> I realy need some help on this one, becuase we have a lot of customers
>> complaining about not responding to dtmf's or receving invalid input
>> in the (FS) IVR.
>> 
>> 
>> Thanks for you help,
>> 
>> Regards,
>> Mike.
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From steveayre at gmail.com  Sat Jun  2 20:07:25 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 2 Jun 2012 17:07:25 +0100
Subject: [Freeswitch-users] DTMF Passthrough on different legs
In-Reply-To: <6ECAF1527329364583AB525CF34ABF950B31A67C@ms.kallback.com>
References: <6ECAF1527329364583AB525CF34ABF950B31A67C@ms.kallback.com>
Message-ID: <F58F409A-1615-453D-A1ED-0B35E9A90AB7@gmail.com>

> 1. I have a customer who had a IVR on his own side, (Session:C) and
> requires input from Session:A. The weird thing is that it seems the
> input is wrong.
> Keys entered do not correspond to what his IVR is receiving. As if
> newly dtmfs are generated somewhere.
> When I switch to another inbound provider, making Session:A using
> rfc2388 and not inband, but do dialout via the provider using inband
> detection. It seems to work fine.

start_dtmf adds a media bug that detects dtmf tones in the audio and generates the out of band rfc2833 messages. It's doesn't remove the inband ones though. If the receiving end accepts both it'll detect duplicated digits. This may be your issue. You may also be generating duplicate rfc2833 yourself if them sender is already sending both.

You should only use start_dtmf if the sender only sends inband and receiver only accepts outofband. It'll also have to decode all audio and examine the audio to detect tones - so running it only when required also decreases server load.

-Steve

Sent from my iPad



On 31 May 2012, at 18:16, JRichey <jrichey at itltd.net> wrote:

> 1. I have a customer who had a IVR on his own side, (Session:C) and
> requires input from Session:A. The weird thing is that it seems the
> input is wrong.
> Keys entered do not correspond to what his IVR is receiving. As if
> newly dtmfs are generated somewhere.
> When I switch to another inbound provider, making Session:A using
> rfc2388 and not inband, but do dialout via the provider using inband
> detection. It seems to work fine.


From nasida at live.ru  Sat Jun  2 20:18:44 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Sat, 2 Jun 2012 20:18:44 +0400
Subject: [Freeswitch-users] Playing of mms stream by means of FS
Message-ID: <SNT127-W22D38D5ECE9C422DFB0D06D5090@phx.gbl>


Hi guys
What the best way for the playing of mms stream like this  mms://live.rfn.ru/rcult_64  by means of FS ?I try to use mod_shout but without success. Do I have to use mplayer or vlc for this ?I saw this in wiki  http://wiki.freeswitch.org/wiki/Mod_vlc 
But... probably there is the more better way ?
Please adviseThanks. 		 	   		  
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From avi at avimarcus.net  Sat Jun  2 22:13:59 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 2 Jun 2012 21:13:59 +0300
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>
References: <4FC896B8.9070102@digitalmail.com>
	<CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>
Message-ID: <CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>

... because att_xfer seems to require a "sofia/$profile/$destination"
directive, and he just wants the call to hit the dialplan.

-Avi


On Fri, Jun 1, 2012 at 8:13 PM, Michael Collins <msc at freeswitch.org> wrote:

> Why do you need to use loopback at all?
> -MC
>
>
> On Fri, Jun 1, 2012 at 3:17 AM, Alex Lake <alex at digitalmail.com> wrote:
>
>> Got a lua script for a B-party "mid-call menu". Is it legitimate to do..
>> "session:execute("att_xfer", "loopback/"..destnum)"
>>
>> I've tried it and it seems to start off doing the right things, but my
>> A-party gets disconnected as soon as the call to the C-Party (the person
>> I'm transferring the call to) answers the call.
>>
>> Maybe better to try to orchestrate the entire affair from within the lua
>> script? (Tricky for a beginner like me!)
>>
>> Thanks,
>> Alex
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Jun  2 23:41:52 2012
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Sat, 2 Jun 2012 12:41:52 -0700 (PDT)
Subject: [Freeswitch-users] freeswitch segfault and crash
Message-ID: <1338666112.82144.YahooMailClassic@web160302.mail.bf1.yahoo.com>

Dear.

I am testing lcr and nibblebill and when i answer some call fs crash:

[ 2923.754712] freeswitch[2965]: segfault at 24 ip b72a27e2 sp b59f765c error 4 in libc-2.11.3.so[b7237000+140000]

My dialplan:


<extension name="Outbound to PSTN 11 Digits">
                <condition field="destination_number" expression="^(10[01][0-9])$">
                        <action application="set" data="hangup_after_bridge=false"/>
                        <action application="set" data="continue_on_fail=true"/>
                        <action application="lcr" data="$1"/>
                        <action application="set" data="nibble_account=${accountcode}"/>
                        <action application="set" data="nibble_rate=${lcr_rate_1}"/>
                        <action application="bridge" data="${lcr_auto_route}"/>
                </condition>
        </extension>

The log of call:

http://pastebin.freeswitch.org/19217

My switch.conf.xml say:

<param name="dump-cores" value="yes"/>

and my init script :


do_setlimits() {
        ulimit -c unlimited
        ulimit -d unlimited
        ulimit -f unlimited
        ulimit -i unlimited
        ulimit -n 999999
        ulimit -q unlimited
        ulimit -u unlimited
        ulimit -v unlimited
        ulimit -x unlimited
        ulimit -s 240
        ulimit -l unlimited
        return 0

Regards.


From errotan at elder.hu  Sat Jun  2 23:51:41 2012
From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=)
Date: Sat, 02 Jun 2012 21:51:41 +0200
Subject: [Freeswitch-users] freeswitch segfault and crash
In-Reply-To: <1338666112.82144.YahooMailClassic@web160302.mail.bf1.yahoo.com>
References: <1338666112.82144.YahooMailClassic@web160302.mail.bf1.yahoo.com>
Message-ID: <4FCA6ECD.7050508@elder.hu>

Hi.

Please read this article: http://wiki.freeswitch.org/wiki/Reporting_Bugs 
and make a bug report here: http://jira.freeswitch.org/secure/Dashboard.jspa


2012-06-02 21:41 keltez?ssel, FERNANDO VILLARROEL ?rta:
> Dear.
>
> I am testing lcr and nibblebill and when i answer some call fs crash:
>
> [ 2923.754712] freeswitch[2965]: segfault at 24 ip b72a27e2 sp b59f765c error 4 in libc-2.11.3.so[b7237000+140000]
>
> My dialplan:
>
>
> <extension name="Outbound to PSTN 11 Digits">
>                  <condition field="destination_number" expression="^(10[01][0-9])$">
>                          <action application="set" data="hangup_after_bridge=false"/>
>                          <action application="set" data="continue_on_fail=true"/>
>                          <action application="lcr" data="$1"/>
>                          <action application="set" data="nibble_account=${accountcode}"/>
>                          <action application="set" data="nibble_rate=${lcr_rate_1}"/>
>                          <action application="bridge" data="${lcr_auto_route}"/>
>                  </condition>
>          </extension>
>
> The log of call:
>
> http://pastebin.freeswitch.org/19217
>
> My switch.conf.xml say:
>
> <param name="dump-cores" value="yes"/>
>
> and my init script :
>
>
> do_setlimits() {
>          ulimit -c unlimited
>          ulimit -d unlimited
>          ulimit -f unlimited
>          ulimit -i unlimited
>          ulimit -n 999999
>          ulimit -q unlimited
>          ulimit -u unlimited
>          ulimit -v unlimited
>          ulimit -x unlimited
>          ulimit -s 240
>          ulimit -l unlimited
>          return 0
>
> Regards.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From nasida at live.ru  Sun Jun  3 01:20:27 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Sun, 3 Jun 2012 01:20:27 +0400
Subject: [Freeswitch-users] Playing of mms stream by means of FS
In-Reply-To: <SNT127-W22D38D5ECE9C422DFB0D06D5090@phx.gbl>
References: <SNT127-W22D38D5ECE9C422DFB0D06D5090@phx.gbl>
Message-ID: <SNT127-W50ED4868C4CDBF33B153B0D5090@phx.gbl>


I try to install mod_vlc but have many errors like this
make mod_vlc-install
/usr/src/freeswitch/src/mod/formats/mod_vlc/mod_vlc.c:42:21: error: vlc/vlc.h: No such file or directory/usr/src/freeswitch/src/mod/formats/mod_vlc/mod_vlc.c:43:37: error: vlc/libvlc_media_player.h: No such file or directory/usr/src/freeswitch/src/mod/formats/mod_vlc/mod_vlc.c:52: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token/usr/src/freeswitch/src/mod/formats/mod_vlc/mod_vlc.c:55: error: expected specifier-qualifier-list before ?libvlc_media_player_t?cc1: warnings being treated as errors

With mod_shout I can play HTTP stream only. Please correct me if I wrong.
How can I listen mms stream (mms://live.rfn.ru/rcult_64) ?
I still wait any advise.Thanks.

From: nasida at live.ru
To: freeswitch-users at lists.freeswitch.org
Date: Sat, 2 Jun 2012 20:18:44 +0400
Subject: [Freeswitch-users] Playing of mms stream by means of FS





Hi guys
What the best way for the playing of mms stream like this  mms://live.rfn.ru/rcult_64  by means of FS ?I try to use mod_shout but without success. Do I have to use mplayer or vlc for this ?I saw this in wiki  http://wiki.freeswitch.org/wiki/Mod_vlc 
But... probably there is the more better way ?
Please adviseThanks. 		 	   		  

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From sherifomran2000 at yahoo.com  Sun Jun  3 03:59:37 2012
From: sherifomran2000 at yahoo.com (Sherif Omran)
Date: Sat, 2 Jun 2012 16:59:37 -0700 (PDT)
Subject: [Freeswitch-users] acl deny - help
In-Reply-To: <SNT127-W50ED4868C4CDBF33B153B0D5090@phx.gbl>
Message-ID: <1338681577.32063.YahooMailClassic@web110813.mail.gq1.yahoo.com>

Hello guys,

I am trying to place a call and it says that 

2012-06-02 23:51:22.074382 [WARNING] sofia.c:7563 IP xx.xx.xx.xx Rejected by acl "domains"

Any body know how to list the acl or edit it? 

I am using vbilling which make use of xmlcurl and database

regards,
Sherif



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From bdfoster at endigotech.com  Sun Jun  3 04:12:50 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 2 Jun 2012 20:12:50 -0400
Subject: [Freeswitch-users] acl deny - help
In-Reply-To: <1338681577.32063.YahooMailClassic@web110813.mail.gq1.yahoo.com>
References: <SNT127-W50ED4868C4CDBF33B153B0D5090@phx.gbl>
	<1338681577.32063.YahooMailClassic@web110813.mail.gq1.yahoo.com>
Message-ID: <CAKV-jFxDhF0d8RWG+O+yJJbKKYqkJn-A8UX-j_qW-9XwZSkFrw@mail.gmail.com>

that's not an error message. Just means they have to authenticate.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 2, 2012 8:00 PM, "Sherif Omran" <sherifomran2000 at yahoo.com> wrote:

> Hello guys,
>
> I am trying to place a call and it says that
>
> 2012-06-02 23:51:22.074382 [WARNING] sofia.c:7563 IP xx.xx.xx.xx Rejected
> by acl "domains"
>
> Any body know how to list the acl or edit it?
>
> I am using vbilling which make use of xmlcurl and database
>
> regards,
> Sherif
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sherifomran2000 at yahoo.com  Sun Jun  3 04:23:41 2012
From: sherifomran2000 at yahoo.com (Sherif Omran)
Date: Sat, 2 Jun 2012 17:23:41 -0700 (PDT)
Subject: [Freeswitch-users] acl deny - help
In-Reply-To: <CAKV-jFxDhF0d8RWG+O+yJJbKKYqkJn-A8UX-j_qW-9XwZSkFrw@mail.gmail.com>
Message-ID: <1338683021.75038.YahooMailClassic@web110814.mail.gq1.yahoo.com>

Yes but to authenticate the IP?
It does not use the acl.conf.xml file where my authentication exists

Do you have an idea?


--- On Sun, 6/3/12, Brian Foster <bdfoster at endigotech.com> wrote:

From: Brian Foster <bdfoster at endigotech.com>
Subject: Re: [Freeswitch-users] acl deny - help
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Sunday, June 3, 2012, 3:12 AM

that's not an error message. Just means they have to authenticate.
Brian Foster

Endigo Computer LLC
Sent from a mobile device.
On Jun 2, 2012 8:00 PM, "Sherif Omran" <sherifomran2000 at yahoo.com> wrote:

Hello guys,

I am trying to place a call and it says that 

2012-06-02 23:51:22.074382 [WARNING] sofia.c:7563 IP xx.xx.xx.xx Rejected by acl "domains"


Any body know how to list the acl or edit it? 

I am using vbilling which make use of xmlcurl and database

regards,
Sherif




_________________________________________________________________________


Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com



Join Us At ClueCon - Aug 7-9, 2012



FreeSWITCH-users mailing list

FreeSWITCH-users at 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-----

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing 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 bdfoster at endigotech.com  Sun Jun  3 04:40:20 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 2 Jun 2012 20:40:20 -0400
Subject: [Freeswitch-users] acl deny - help
In-Reply-To: <1338683021.75038.YahooMailClassic@web110814.mail.gq1.yahoo.com>
References: <CAKV-jFxDhF0d8RWG+O+yJJbKKYqkJn-A8UX-j_qW-9XwZSkFrw@mail.gmail.com>
	<1338683021.75038.YahooMailClassic@web110814.mail.gq1.yahoo.com>
Message-ID: <CAKV-jFzckZGYoVNt3+6sA_pqXs8216=Rpoj=xx+7dURCMAM0vA@mail.gmail.com>

That's a vbilling specific issue. Either Goni will answer or your best bet
is to go and post on the vBilling forums if you haven't already done so.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 2, 2012 8:25 PM, "Sherif Omran" <sherifomran2000 at yahoo.com> wrote:

> Yes but to authenticate the IP?
> It does not use the acl.conf.xml file where my authentication exists
>
> Do you have an idea?
>
>
> --- On *Sun, 6/3/12, Brian Foster <bdfoster at endigotech.com>* wrote:
>
>
> From: Brian Foster <bdfoster at endigotech.com>
> Subject: Re: [Freeswitch-users] acl deny - help
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Sunday, June 3, 2012, 3:12 AM
>
> that's not an error message. Just means they have to authenticate.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 2, 2012 8:00 PM, "Sherif Omran" <sherifomran2000 at yahoo.com<http://mc/compose?to=sherifomran2000 at yahoo.com>>
> wrote:
>
> Hello guys,
>
> I am trying to place a call and it says that
>
> 2012-06-02 23:51:22.074382 [WARNING] sofia.c:7563 IP xx.xx.xx.xx Rejected
> by acl "domains"
>
> Any body know how to list the acl or edit it?
>
> I am using vbilling which make use of xmlcurl and database
>
> regards,
> Sherif
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://mc/compose?to=consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at 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-----
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://mc/compose?to=consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From drk at drkngs.net  Sun Jun  3 04:47:10 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Sat, 02 Jun 2012 17:47:10 -0700
Subject: [Freeswitch-users] acl deny - help
In-Reply-To: <1338683021.75038.YahooMailClassic@web110814.mail.gq1.yahoo.com>
Message-ID: <20120603004710.a25ae2ed@mail.tritonwest.net>

There is really no such thing on FS to "authenticate by IP" you can just allow calls w/o FS requesting the sender to authtnticate. The "domains" access list used in the example "internal" profile, is where it allows registrations from (authenticated).   
   
If you want to associate an IP address with a user, in your directory system, you would have to set your SIP profile to not authorize invites, and do it yourself in the dialplan.   
   
You could lookup the IP address if the call is not "${SIP_Authorized}" and if you found it in a database do a "set_user" and then transfer to the normal context, and if not, do a "respond 407/401" to make the far end authenticate.   
   
--Dave
      _____  

  From: Sherif Omran [mailto:sherifomran2000 at yahoo.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Sat, 02 Jun 2012 17:23:41 -0700
Subject: Re: [Freeswitch-users] acl deny - help

  
    
   Yes but to authenticate the IP?
It does not use the acl.conf.xml file where my authentication exists

Do you have an idea?


--- On Sun, 6/3/12, Brian Foster <bdfoster at endigotech.com> wrote:
  
From: Brian Foster <bdfoster at endigotech.com>
Subject: Re: [Freeswitch-users] acl deny - help
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Sunday, June 3, 2012, 3:12 AM

  
  

that's not an error message. Just means they have to authenticate.  

Brian Foster
Endigo Computer LLC  

Sent from a mobile device.  
On Jun 2, 2012 8:00 PM, "Sherif Omran" <sherifomran2000 at yahoo.com> wrote:
    
    
   Hello guys,

I am trying to place a call and it says that 

2012-06-02 23:51:22.074382 [WARNING] sofia.c:7563 IP xx.xx.xx.xx Rejected by acl "domains"

Any body know how to list the acl or edit it? 

I am using vbilling which make use of xmlcurl and database

regards,
Sherif





_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at 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-----

  
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

      
   
 
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From nbhatti at gmail.com  Sun Jun  3 04:51:44 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Sun, 3 Jun 2012 03:51:44 +0300
Subject: [Freeswitch-users] acl deny - help
In-Reply-To: <CAKV-jFzckZGYoVNt3+6sA_pqXs8216=Rpoj=xx+7dURCMAM0vA@mail.gmail.com>
References: <CAKV-jFxDhF0d8RWG+O+yJJbKKYqkJn-A8UX-j_qW-9XwZSkFrw@mail.gmail.com>
	<1338683021.75038.YahooMailClassic@web110814.mail.gq1.yahoo.com>
	<CAKV-jFzckZGYoVNt3+6sA_pqXs8216=Rpoj=xx+7dURCMAM0vA@mail.gmail.com>
Message-ID: <CACak_yzi4n3ffoC+=rAV24=Z_60Mx0S_BCVF9mPsX9YyamhJ7A@mail.gmail.com>

I did answer that already in the forum. You can not use xml files with
vBilling by default. All the incoming clients are authenticated against an
ACL. You would have to create a customer in vBilling and add the ip address
in ACL Nodes.

If you still like to use your xml files this requires some basic knowledge
of how xml_curl works. But eventually this will break the functionality of
vBilling. If the ip address is not found in the db, your call will not be
billed.

--
Sent from a mobile device.
On Jun 3, 2012 3:41 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> That's a vbilling specific issue. Either Goni will answer or your best bet
> is to go and post on the vBilling forums if you haven't already done so.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 2, 2012 8:25 PM, "Sherif Omran" <sherifomran2000 at yahoo.com> wrote:
>
>> Yes but to authenticate the IP?
>> It does not use the acl.conf.xml file where my authentication exists
>>
>> Do you have an idea?
>>
>>
>> --- On *Sun, 6/3/12, Brian Foster <bdfoster at endigotech.com>* wrote:
>>
>>
>> From: Brian Foster <bdfoster at endigotech.com>
>> Subject: Re: [Freeswitch-users] acl deny - help
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Date: Sunday, June 3, 2012, 3:12 AM
>>
>> that's not an error message. Just means they have to authenticate.
>>
>> Brian Foster
>> Endigo Computer LLC
>>
>> Sent from a mobile device.
>> On Jun 2, 2012 8:00 PM, "Sherif Omran" <sherifomran2000 at yahoo.com<http://mc/compose?to=sherifomran2000 at yahoo.com>>
>> wrote:
>>
>> Hello guys,
>>
>> I am trying to place a call and it says that
>>
>> 2012-06-02 23:51:22.074382 [WARNING] sofia.c:7563 IP xx.xx.xx.xx Rejected
>> by acl "domains"
>>
>> Any body know how to list the acl or edit it?
>>
>> I am using vbilling which make use of xmlcurl and database
>>
>> regards,
>> Sherif
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<http://mc/compose?to=consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at 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-----
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<http://mc/compose?to=consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sherifomran2000 at yahoo.com  Sun Jun  3 05:48:34 2012
From: sherifomran2000 at yahoo.com (Sherif Omran)
Date: Sat, 2 Jun 2012 18:48:34 -0700 (PDT)
Subject: [Freeswitch-users] acl deny - help
In-Reply-To: <CACak_yzi4n3ffoC+=rAV24=Z_60Mx0S_BCVF9mPsX9YyamhJ7A@mail.gmail.com>
Message-ID: <1338688114.9208.YahooMailClassic@web110806.mail.gq1.yahoo.com>

Thank you for? your reply. Creating a customer in vbilling needs creating a gateway and a chain of parameters to setup and this is not how freeswitch normally functions. I don't need to have a GW setup in FS to have ACL enabled or disabled for a profile.
I have FS working as SBC and in my case i don't need to bill such local routes, billing would be for other routes but not the local (proxy) one. There must be a way to bypass it. I added? acl-proxy parameter set to true and removed the acl-inbound-proxy. Now the call passes but there is a programming problem when acl-inbound-proxy is deleted, this causes FS not to find the profiles (for more details see the following link http://forum.vbilling.org/viewtopic.php?f=6&t=903&p=1204#p1204)

Why does FS produces invalid profile output? 
When i quit ./fs_cli and rerun ./fs_cli I see the profile again

kind regards
S.


--- On Sun, 6/3/12, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:

From: Muhammad Naseer Bhatti <nbhatti at gmail.com>
Subject: Re: [Freeswitch-users] acl deny - help
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Sunday, June 3, 2012, 3:51 AM

I did answer that already in the forum. You can not use xml files with vBilling by default. All the incoming clients are authenticated against an ACL. You would have to create a customer in vBilling and add the ip address in ACL Nodes. 

If you still like to use your xml files this requires some basic knowledge of how xml_curl works. But eventually this will break the functionality of vBilling. If the ip address is not found in the db, your call will not be billed.

--

Sent from a mobile device.
On Jun 3, 2012 3:41 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:

That's a vbilling specific issue. Either Goni will answer or your best bet is to go and post on the vBilling forums if you haven't already done so.
Brian Foster

Endigo Computer LLC
Sent from a mobile device.
On Jun 2, 2012 8:25 PM, "Sherif Omran" <sherifomran2000 at yahoo.com> wrote:


Yes but to authenticate the IP?
It does not use the acl.conf.xml file where my authentication exists

Do you have an idea?




--- On Sun, 6/3/12, Brian Foster <bdfoster at endigotech.com> wrote:



From: Brian Foster <bdfoster at endigotech.com>
Subject: Re: [Freeswitch-users] acl deny - help
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>


Date: Sunday, June 3, 2012, 3:12 AM

that's not an error message. Just means they have to authenticate.
Brian Foster

Endigo Computer LLC
Sent from a mobile device.
On Jun 2, 2012 8:00 PM, "Sherif Omran" <sherifomran2000 at yahoo.com> wrote:



Hello guys,

I am trying to place a call and it says that 

2012-06-02 23:51:22.074382 [WARNING] sofia.c:7563 IP xx.xx.xx.xx Rejected by acl "domains"




Any body know how to list the acl or edit it? 

I am using vbilling which make use of xmlcurl and database

regards,
Sherif




_________________________________________________________________________




Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com



Join Us At ClueCon - Aug 7-9, 2012



FreeSWITCH-users mailing list

FreeSWITCH-users at 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-----

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org


http://www.freeswitchsolutions.com






Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com



Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users


UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com



Join Us At ClueCon - Aug 7-9, 2012



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org




_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com



Join Us At ClueCon - Aug 7-9, 2012



FreeSWITCH-users mailing list

FreeSWITCH-users at 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-----

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing 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 sherifomran2000 at yahoo.com  Sun Jun  3 13:03:15 2012
From: sherifomran2000 at yahoo.com (Sherif Omran)
Date: Sun, 3 Jun 2012 02:03:15 -0700 (PDT)
Subject: [Freeswitch-users] profile alias
Message-ID: <1338714195.69230.YahooMailClassic@web110804.mail.gq1.yahoo.com>

Hello guys,

how can I create a profile alias from the sofia interface?

thanks
kind regards,
Sherif



From bdfoster at endigotech.com  Sun Jun  3 16:16:09 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 3 Jun 2012 08:16:09 -0400
Subject: [Freeswitch-users] profile alias
In-Reply-To: <1338714195.69230.YahooMailClassic@web110804.mail.gq1.yahoo.com>
References: <1338714195.69230.YahooMailClassic@web110804.mail.gq1.yahoo.com>
Message-ID: <CAKV-jFywRNHN1Yzj6BzjqR9yghnYZOZCT-ZmQz62HZeRQuPNNQ@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Mod_commands#alias

Please make sure you search the wiki before posting on the ML.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 3, 2012 5:06 AM, "Sherif Omran" <sherifomran2000 at yahoo.com> wrote:

> Hello guys,
>
> how can I create a profile alias from the sofia interface?
>
> thanks
> kind regards,
> Sherif
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bdfoster at endigotech.com  Sun Jun  3 16:18:52 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 3 Jun 2012 08:18:52 -0400
Subject: [Freeswitch-users] profile alias
In-Reply-To: <CAKV-jFywRNHN1Yzj6BzjqR9yghnYZOZCT-ZmQz62HZeRQuPNNQ@mail.gmail.com>
References: <1338714195.69230.YahooMailClassic@web110804.mail.gq1.yahoo.com>
	<CAKV-jFywRNHN1Yzj6BzjqR9yghnYZOZCT-ZmQz62HZeRQuPNNQ@mail.gmail.com>
Message-ID: <CAKV-jFxKx1S8s+Z73sL3sNuY7ACDkdVyGb4WEFbom1re9ZRSFQ@mail.gmail.com>

Crap... that wasn't the right alias...

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 3, 2012 8:16 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> http://wiki.freeswitch.org/wiki/Mod_commands#alias
>
> Please make sure you search the wiki before posting on the ML.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 3, 2012 5:06 AM, "Sherif Omran" <sherifomran2000 at yahoo.com> wrote:
>
>> Hello guys,
>>
>> how can I create a profile alias from the sofia interface?
>>
>> thanks
>> kind regards,
>> Sherif
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
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From bdfoster at endigotech.com  Sun Jun  3 16:24:14 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 3 Jun 2012 08:24:14 -0400
Subject: [Freeswitch-users] profile alias
In-Reply-To: <CAKV-jFxKx1S8s+Z73sL3sNuY7ACDkdVyGb4WEFbom1re9ZRSFQ@mail.gmail.com>
References: <1338714195.69230.YahooMailClassic@web110804.mail.gq1.yahoo.com>
	<CAKV-jFywRNHN1Yzj6BzjqR9yghnYZOZCT-ZmQz62HZeRQuPNNQ@mail.gmail.com>
	<CAKV-jFxKx1S8s+Z73sL3sNuY7ACDkdVyGb4WEFbom1re9ZRSFQ@mail.gmail.com>
Message-ID: <CAKV-jFySaBQAEYMVEHPqhfi-i18rUxrhybfux7=5sR3R6tye9Q@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Basic_settings

doesn't really look like you can add can alias via FS_cli.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 3, 2012 8:18 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> Crap... that wasn't the right alias...
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 3, 2012 8:16 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>
>> http://wiki.freeswitch.org/wiki/Mod_commands#alias
>>
>> Please make sure you search the wiki before posting on the ML.
>>
>> Brian Foster
>> Endigo Computer LLC
>>
>> Sent from a mobile device.
>> On Jun 3, 2012 5:06 AM, "Sherif Omran" <sherifomran2000 at yahoo.com> wrote:
>>
>>> Hello guys,
>>>
>>> how can I create a profile alias from the sofia interface?
>>>
>>> thanks
>>> kind regards,
>>> Sherif
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
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From neilp at cs.stanford.edu  Sun Jun  3 19:53:51 2012
From: neilp at cs.stanford.edu (Neil Patel)
Date: Sun, 3 Jun 2012 08:53:51 -0700
Subject: [Freeswitch-users] compiling ESL pymod in 32-bit
Message-ID: <CAAcoRq0tsSvEb5o1adqbAjtr3OUuSTY7iOWihV1P73rqwUQLDA@mail.gmail.com>

Hi All,

I'm trying to force ESL's pymod to compile in 32-bit mode to be compatible
with the rest of my setup (I am on 64-bit Mac OSX Lion, but using 32-bit
python). I tried doing the equivalent of "CFLAGS=-m32 CXXFLAGS=-m32
LDFLAGS=-m32" by adding that flag to the corresponding vars in
libs/esl/Makefile, but it still did not work for me.

Thanks in advance,
Neil
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From josefu at gmail.com  Sun Jun  3 19:47:39 2012
From: josefu at gmail.com (=?ISO-8859-1?Q?Jose_Fco=2E_Irles_Dur=E1?=)
Date: Sun, 3 Jun 2012 17:47:39 +0200
Subject: [Freeswitch-users] limit app problem with loopback channels
Message-ID: <CAJOPU6X24Ez5xxnDwZ_muaAAekuc6omnCzZ3E6NUPG8P9zCTmQ@mail.gmail.com>

Hello,

I'm trying to limit calls to one user with limit application, all
works fine when a user dials to the number user (the caller is
transfered to limit_exceeded extension)

But when a caller dials to the 2000 extensions (a "ring group" to
loopback channels) and one of the users (for example 1001) answer the
call, the limit app removes the limit count for this channel when the
loopback channel disappears (when the direct bridge occurs between the
caller channel and callee channel).

Where is the problem? How can I avoid that the limit app removes the
limit for this loopback channel?

There is the fragmet of my dialplan (is the vainilla dialplan from git
with the limit action added)
<context name="default">
...
 <extension name="Local_Extension">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="export" data="dialed_extension=$1"/>
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=30"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
       <action application="limit" data="hash callwaiting
${dialed_extension} 1"/>
        <action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="bridge"
data="loopback/app=voicemail:default ${domain_name}
${dialed_extension}"/>
      </condition>
    </extension>

    <extension name="group_dial_sales">
      <condition field="destination_number" expression="^2000$">
        <action application="bridge"
data="loopback/1000/default,loopback/1001/default"/>
      </condition>
    </extension>

...
</context>

-- 
Jose Fco. Irles Dur?


From msc at freeswitch.org  Mon Jun  4 00:15:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 3 Jun 2012 13:15:27 -0700
Subject: [Freeswitch-users] Stuck calls
In-Reply-To: <CACak_yx77eR+dKWyeWz7-ix1PKisi_QWrOo6184z_+1fmPCOdg@mail.gmail.com>
References: <CACak_yyYnKn12KdEQkOdeg0Lwzq=QoDjX_oBitSgKnFEQSGzMA@mail.gmail.com>
	<CAKbxfG_-zDrq1VZpE-b8y+5B7bucGOV4pAwUoKfMawhAjpB3Rg@mail.gmail.com>
	<CACak_yzD21VMiiPG48h1XsCyn5EhqkMPDREoAtnPTA_1ifmSZw@mail.gmail.com>
	<CAKzWOxVAcz+viU0oyUMPQ-gPXL4QosdSuGfAx3Woq-hmLnxFeA@mail.gmail.com>
	<CACak_yy8m=WBpUdPpwuG0BGTiJs4rz+462=mhuFDzaTRAuyvjg@mail.gmail.com>
	<CAKbxfG9dREahuS3_CqavBbWc4BdmyPGQUNaiGg4JPja3aJZ=aA@mail.gmail.com>
	<CACak_yx77eR+dKWyeWz7-ix1PKisi_QWrOo6184z_+1fmPCOdg@mail.gmail.com>
Message-ID: <CAKzWOxXK4JJVC+9kowxMbgnFXSvG-orjRmQxwkN7XhmO_w554Q@mail.gmail.com>

be sure to include configuration information, especially dialplan configs
so that we can see if there's a way to reproduce the symptoms you're
experiencing.
-MC

On Sat, Jun 2, 2012 at 8:58 AM, Muhammad Naseer Bhatti <nbhatti at gmail.com>wrote:

> Here we go again ..
>
> freeswitch at internal> status
> UP 0 years, 0 days, 18 hours, 13 minutes, 28 seconds, 780
> milliseconds, 883 microseconds
> FreeSWITCH is ready
> 51822 session(s) since startup
> 2000 session(s) 0/100
> 2000 session(s) max
> min idle cpu 0.00/99.00
>
> freeswitch at internal> show channels count
>
> 2000 total.
>
> freeswitch at internal> show calls count
>
> 2000 total.
>
> I am going to update to latest and try again and may open a JIRA. This
> is now happening too often. I am on FreeSWITCH Version 1.1.beta1
> (git-2a0ca29 2012-04-27 15-28-17 -0500).
>
>
>
> On Fri, May 18, 2012 at 11:42 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > That spells out SQL error.
> > The show channels is just a db keeping track of the channel status
> > based on events.
> >
> > If you can reproduce it on git HEAD open a jira
> >
> >
> > On Fri, May 18, 2012 at 3:01 PM, Muhammad Naseer Bhatti
> > <nbhatti at gmail.com> wrote:
> >> I was trying to dig some information out of it before a restart. There
> >> is no traffic on this box right now, so I can restart/retest anything.
> >> I'll update, grab some logs and will post back.
> >>
> >> Thanks.
> >>
> >> On Fri, May 18, 2012 at 10:50 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >>> Can you turn debugging back on for a period of time and re-test? Also,
> any
> >>> chance you can get to latest git for testing?
> >>> -MC
> >>>
> >>>
> >>> On Fri, May 18, 2012 at 12:18 PM, Muhammad Naseer Bhatti <
> nbhatti at gmail.com>
> >>> wrote:
> >>>>
> >>>> Show channels shows 46 calls, status shows 0. pb @
> >>>> http://pastebin.freeswitch.org/19093
> >>>> Logging was turned of from this production server to gain some disk
> >>>> I/Os. Using mod_xml_cdr for cdr posting. Web server could possibly the
> >>>> reason and  if so, why the calls are in RINGING state? CDR is only
> >>>> posted at the end of the call. Looks like we don't have much evidence.
> >>>>
> >>>> Goni
> >>>>
> >>>> On Fri, May 18, 2012 at 8:52 PM, Anthony Minessale
> >>>> <anthony.minessale at gmail.com> wrote:
> >>>> > compare "show channels" to "status" if status says there are 0
> calls,
> >>>> > look for a db error somewhere in fs log [or db specific log if you
> >>>> > have odbc]
> >>>> > cat freeswitch.log | grep CRIT
> >>>> >
> >>>> > if status sees channels too, see if you have a cdr module that is
> >>>> > doing some post processing that could be stuck.
> >>>> >
> >>>> >
> >>>> >
> >>>> >
> >>>> > On Fri, May 18, 2012 at 12:45 PM, Muhammad Naseer Bhatti
> >>>> > <nbhatti at gmail.com> wrote:
> >>>> >> I got like 46 calls stuck in FreeSWITCH. The switch is not doing
> any
> >>>> >> traffic right now. It is running FreeSWITCH Version 1.2.0
> (git-0709cc6
> >>>> >> 2012-05-16 02-50-13 +0000). Should I file jira or look for
> something
> >>>> >> else? They are mostly in CS_INIT,,,,,,,RINGING --
> >>>> >> CS_ROUTING,,,,,,,DOWN and CS_CONSUME_MEDIA state.
> >>>> >>
> >>>> >> Thanks.
> >>>> >>
> >>>
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> Join Us At ClueCon - Aug 7-9, 2012
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > 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
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nbhatti at gmail.com  Mon Jun  4 00:27:34 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Sun, 3 Jun 2012 23:27:34 +0300
Subject: [Freeswitch-users] Stuck calls
In-Reply-To: <CAKzWOxXK4JJVC+9kowxMbgnFXSvG-orjRmQxwkN7XhmO_w554Q@mail.gmail.com>
References: <CACak_yyYnKn12KdEQkOdeg0Lwzq=QoDjX_oBitSgKnFEQSGzMA@mail.gmail.com>
	<CAKbxfG_-zDrq1VZpE-b8y+5B7bucGOV4pAwUoKfMawhAjpB3Rg@mail.gmail.com>
	<CACak_yzD21VMiiPG48h1XsCyn5EhqkMPDREoAtnPTA_1ifmSZw@mail.gmail.com>
	<CAKzWOxVAcz+viU0oyUMPQ-gPXL4QosdSuGfAx3Woq-hmLnxFeA@mail.gmail.com>
	<CACak_yy8m=WBpUdPpwuG0BGTiJs4rz+462=mhuFDzaTRAuyvjg@mail.gmail.com>
	<CAKbxfG9dREahuS3_CqavBbWc4BdmyPGQUNaiGg4JPja3aJZ=aA@mail.gmail.com>
	<CACak_yx77eR+dKWyeWz7-ix1PKisi_QWrOo6184z_+1fmPCOdg@mail.gmail.com>
	<CAKzWOxXK4JJVC+9kowxMbgnFXSvG-orjRmQxwkN7XhmO_w554Q@mail.gmail.com>
Message-ID: <CACak_yxTNSFhgbO+DWO+SfgRmte32zdR9-BT+m8GP4GxQ2PE5Q@mail.gmail.com>

I just am doing a Jira. This happened again a few minutes back. I have
restarted the system to make it running, but got the core and trace.
Going to post JIRA now.

On Sun, Jun 3, 2012 at 11:15 PM, Michael Collins <msc at freeswitch.org> wrote:
> be sure to include configuration information, especially dialplan configs so
> that we can see if there's a way to reproduce the symptoms you're
> experiencing.
> -MC
>
>
> On Sat, Jun 2, 2012 at 8:58 AM, Muhammad Naseer Bhatti <nbhatti at gmail.com>
> wrote:
>>
>> Here we go again ..
>>
>> freeswitch at internal> status
>> UP 0 years, 0 days, 18 hours, 13 minutes, 28 seconds, 780
>> milliseconds, 883 microseconds
>> FreeSWITCH is ready
>> 51822 session(s) since startup
>> 2000 session(s) 0/100
>> 2000 session(s) max
>> min idle cpu 0.00/99.00
>>
>> freeswitch at internal> show channels count
>>
>> 2000 total.
>>
>> freeswitch at internal> show calls count
>>
>> 2000 total.
>>
>> I am going to update to latest and try again and may open a JIRA. This
>> is now happening too often. I am on FreeSWITCH Version 1.1.beta1
>> (git-2a0ca29 2012-04-27 15-28-17 -0500).
>>
>>
>>
>> On Fri, May 18, 2012 at 11:42 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > That spells out SQL error.
>> > The show channels is just a db keeping track of the channel status
>> > based on events.
>> >
>> > If you can reproduce it on git HEAD open a jira
>> >
>> >
>> > On Fri, May 18, 2012 at 3:01 PM, Muhammad Naseer Bhatti
>> > <nbhatti at gmail.com> wrote:
>> >> I was trying to dig some information out of it before a restart. There
>> >> is no traffic on this box right now, so I can restart/retest anything.
>> >> I'll update, grab some logs and will post back.
>> >>
>> >> Thanks.
>> >>
>> >> On Fri, May 18, 2012 at 10:50 PM, Michael Collins <msc at freeswitch.org>
>> >> wrote:
>> >>> Can you turn debugging back on for a period of time and re-test? Also,
>> >>> any
>> >>> chance you can get to latest git for testing?
>> >>> -MC
>> >>>
>> >>>
>> >>> On Fri, May 18, 2012 at 12:18 PM, Muhammad Naseer Bhatti
>> >>> <nbhatti at gmail.com>
>> >>> wrote:
>> >>>>
>> >>>> Show channels shows 46 calls, status shows 0. pb @
>> >>>> http://pastebin.freeswitch.org/19093
>> >>>> Logging was turned of from this production server to gain some disk
>> >>>> I/Os. Using mod_xml_cdr for cdr posting. Web server could possibly
>> >>>> the
>> >>>> reason and ?if so, why the calls are in RINGING state? CDR is only
>> >>>> posted at the end of the call. Looks like we don't have much
>> >>>> evidence.
>> >>>>
>> >>>> Goni
>> >>>>
>> >>>> On Fri, May 18, 2012 at 8:52 PM, Anthony Minessale
>> >>>> <anthony.minessale at gmail.com> wrote:
>> >>>> > compare "show channels" to "status" if status says there are 0
>> >>>> > calls,
>> >>>> > look for a db error somewhere in fs log [or db specific log if you
>> >>>> > have odbc]
>> >>>> > cat freeswitch.log | grep CRIT
>> >>>> >
>> >>>> > if status sees channels too, see if you have a cdr module that is
>> >>>> > doing some post processing that could be stuck.
>> >>>> >
>> >>>> >
>> >>>> >
>> >>>> >
>> >>>> > On Fri, May 18, 2012 at 12:45 PM, Muhammad Naseer Bhatti
>> >>>> > <nbhatti at gmail.com> wrote:
>> >>>> >> I got like 46 calls stuck in FreeSWITCH. The switch is not doing
>> >>>> >> any
>> >>>> >> traffic right now. It is running FreeSWITCH Version 1.2.0
>> >>>> >> (git-0709cc6
>> >>>> >> 2012-05-16 02-50-13 +0000). Should I file jira or look for
>> >>>> >> something
>> >>>> >> else? They are mostly in CS_INIT,,,,,,,RINGING --
>> >>>> >> CS_ROUTING,,,,,,,DOWN and CS_CONSUME_MEDIA state.
>> >>>> >>
>> >>>> >> Thanks.
>> >>>> >>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://wiki.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> Join Us At ClueCon - Aug 7-9, 2012
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> Join Us At ClueCon - Aug 7-9, 2012
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > 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
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > Join Us At ClueCon - Aug 7-9, 2012
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Mon Jun  4 00:33:46 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 3 Jun 2012 13:33:46 -0700
Subject: [Freeswitch-users] regular expression ?
In-Reply-To: <1338631917.10976.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1338627741.80797.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<28858.1338630492@ccs.covici.com>
	<1338631917.10976.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <CAKzWOxWrv6+QacBbwPQgKJsrbdAo+GKH4uwFmwOf-nPxtLR8rg@mail.gmail.com>

If you are trying to match the dialed digits "*999*11#" then yes, the
pattern you suggested below is not appropriate. Just putting special
characters inside of [ and ] is not enough to change their meaning. The
proper way to handle this is by doing what Covici suggested and use a \
(backslash) in front of the * characters. (The # char is not one of the
special characters in PCRE, therefore it does not need a backslash.)

The best pattern for your example is:

^(\*999\*11#)$

That would match "*999*11#" and absolutely nothing else. If you need to be
flexible on the digits that you match you could do something like this:

^(\*\d\d\d\*\d\d#)$

That would match * plus 3 digits plus * plus 2 digits. Example matches:
*123*45#
*876*99#
*333*33#

You could even grab the internal digit sequences and put them into $2 and
$3:

^(\*(\d\d\d)\*(\d\d)#)$

Notice what the special vars contain by using this little table:

Dialed:   $1:       $2:     $3:
*123*45#    *123*45#  123     45
*876*99#    *876*99#  876     99
*333*33#    *333*33#  333     33

I strongly recommend that you try it out. Put this in your dialplan and try
it out:

<extension name="tinkering with regexes">
  <condition field="destination_number"
expression="^(\*(\d\d\d)\*(\d\d)#)$">
      <action application="log" data="INFO \$1 is $1"/>
      <action application="log" data="INFO \$2 is $2"/>
      <action application="log" data="INFO \$3 is $3"/>
  </condition>
</extension>

Try it out - make a bunch of calls and see what happens. Change the regex
around and try dialing again. I promise you that once you start writing
your own regexes - and occasionally breaking them in the process - you will
learn them and use them to your advantage.

-MC

On Sat, Jun 2, 2012 at 3:11 AM, Samira Mh <saami_mh at ymail.com> wrote:

> thanks so much for your reply,nut i think the below patter is true as
> well,
> ^([*]999[*]11[#])$
> is that true?
>
>
>   ------------------------------
> *From:* "covici at ccs.covici.com" <covici at ccs.covici.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Saturday, June 2, 2012 2:18 PM
> *Subject:* Re: [Freeswitch-users] regular expression ?
>
> Just put a \ before each * and maybe the # as well and you should be
> good to go.
>
> Samira Mh <saami_mh at ymail.com> wrote:
>
> > hi guys,
> > it possible to match *999*11# in regular expression in freeswitch ?
> > in the wiki the above example don't exis,
> > plz help thanks
> > ----------------------------------------------------
> > Alternatives:
>
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From msc at freeswitch.org  Mon Jun  4 00:40:15 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 3 Jun 2012 13:40:15 -0700
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>
References: <4FC896B8.9070102@digitalmail.com>
	<CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>
	<CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>
Message-ID: <CAKzWOxUp5h2tMFP9yHJM4LaMmF1f+sXsOfDAcETwzaDG_2WgPA@mail.gmail.com>

Let me rephrase...

Since loopback is generally evil and should be avoided wherever possible,
what does loopback give you that you can't get from doing a normal
dialstring?
-MC

On Sat, Jun 2, 2012 at 11:13 AM, Avi Marcus <avi at avimarcus.net> wrote:

> ... because att_xfer seems to require a "sofia/$profile/$destination"
> directive, and he just wants the call to hit the dialplan.
>
> -Avi
>
>
> On Fri, Jun 1, 2012 at 8:13 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Why do you need to use loopback at all?
>> -MC
>>
>>
>> On Fri, Jun 1, 2012 at 3:17 AM, Alex Lake <alex at digitalmail.com> wrote:
>>
>>> Got a lua script for a B-party "mid-call menu". Is it legitimate to do..
>>> "session:execute("att_xfer", "loopback/"..destnum)"
>>>
>>> I've tried it and it seems to start off doing the right things, but my
>>> A-party gets disconnected as soon as the call to the C-Party (the person
>>> I'm transferring the call to) answers the call.
>>>
>>> Maybe better to try to orchestrate the entire affair from within the lua
>>> script? (Tricky for a beginner like me!)
>>>
>>> Thanks,
>>> Alex
>>>
>>
>>
>>
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From avi at avimarcus.net  Mon Jun  4 00:54:06 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 3 Jun 2012 23:54:06 +0300
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <CAKzWOxUp5h2tMFP9yHJM4LaMmF1f+sXsOfDAcETwzaDG_2WgPA@mail.gmail.com>
References: <4FC896B8.9070102@digitalmail.com>
	<CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>
	<CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>
	<CAKzWOxUp5h2tMFP9yHJM4LaMmF1f+sXsOfDAcETwzaDG_2WgPA@mail.gmail.com>
Message-ID: <CALNZuw=5W_k2MtJPxHn6HBZanCZNk83TXiMZTEGjZ9kRjb1G4w@mail.gmail.com>

... all the normal dialplan handling. Setting CID, options, LCR stuff,
billing controls.
-Avi


On Sun, Jun 3, 2012 at 11:40 PM, Michael Collins <msc at freeswitch.org> wrote:

> Let me rephrase...
>
> Since loopback is generally evil and should be avoided wherever possible,
> what does loopback give you that you can't get from doing a normal
> dialstring?
> -MC
>
>
> On Sat, Jun 2, 2012 at 11:13 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> ... because att_xfer seems to require a "sofia/$profile/$destination"
>> directive, and he just wants the call to hit the dialplan.
>>
>> -Avi
>>
>>
>> On Fri, Jun 1, 2012 at 8:13 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Why do you need to use loopback at all?
>>> -MC
>>>
>>>
>>> On Fri, Jun 1, 2012 at 3:17 AM, Alex Lake <alex at digitalmail.com> wrote:
>>>
>>>> Got a lua script for a B-party "mid-call menu". Is it legitimate to do..
>>>> "session:execute("att_xfer", "loopback/"..destnum)"
>>>>
>>>> I've tried it and it seems to start off doing the right things, but my
>>>> A-party gets disconnected as soon as the call to the C-Party (the person
>>>> I'm transferring the call to) answers the call.
>>>>
>>>> Maybe better to try to orchestrate the entire affair from within the lua
>>>> script? (Tricky for a beginner like me!)
>>>>
>>>> Thanks,
>>>> Alex
>>>>
>>>
>>>
>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun  4 00:54:08 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 3 Jun 2012 13:54:08 -0700
Subject: [Freeswitch-users] limit app problem with loopback channels
In-Reply-To: <CAJOPU6X24Ez5xxnDwZ_muaAAekuc6omnCzZ3E6NUPG8P9zCTmQ@mail.gmail.com>
References: <CAJOPU6X24Ez5xxnDwZ_muaAAekuc6omnCzZ3E6NUPG8P9zCTmQ@mail.gmail.com>
Message-ID: <CAKzWOxUb_mKTsL7g8W0c6Hq4ss=V=i+GBr2ThJna25Jemzm_rw@mail.gmail.com>

You can solve this issue by avoiding loopback. Don't use loopback unless
there is absolutely no other alternative. Try replacing the loopback calls
in the group definition to "user/xxx" instead:

   <extension name="group_dial_sales">
     <condition field="destination_number" expression="^2000$">
       <action application="bridge"
data="[execute_on_answer='limit hash callwaiting 1000 1']user/1000,
[execute_on_answer='limit hash callwaiting
1001 1']user/1001"/>
     </condition>
   </extension>

This is just off the top of my head, so please do the usual testing and
tinkering before you report back that it does not work. :)

-MC

On Sun, Jun 3, 2012 at 8:47 AM, Jose Fco. Irles Dur? <josefu at gmail.com>wrote:

> Hello,
>
> I'm trying to limit calls to one user with limit application, all
> works fine when a user dials to the number user (the caller is
> transfered to limit_exceeded extension)
>
> But when a caller dials to the 2000 extensions (a "ring group" to
> loopback channels) and one of the users (for example 1001) answer the
> call, the limit app removes the limit count for this channel when the
> loopback channel disappears (when the direct bridge occurs between the
> caller channel and callee channel).
>
> Where is the problem? How can I avoid that the limit app removes the
> limit for this loopback channel?
>
> There is the fragmet of my dialplan (is the vainilla dialplan from git
> with the limit action added)
> <context name="default">
> ...
>  <extension name="Local_Extension">
>      <condition field="destination_number" expression="^(10[01][0-9])$">
>        <action application="export" data="dialed_extension=$1"/>
>        <action application="set" data="ringback=${us-ring}"/>
>        <action application="set" data="transfer_ringback=$${hold_music}"/>
>        <action application="set" data="call_timeout=30"/>
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="set" data="continue_on_fail=true"/>
>       <action application="limit" data="hash callwaiting
> ${dialed_extension} 1"/>
>        <action application="bridge"
> data="user/${dialed_extension}@${domain_name}"/>
>        <action application="answer"/>
>        <action application="sleep" data="1000"/>
>        <action application="bridge"
> data="loopback/app=voicemail:default ${domain_name}
> ${dialed_extension}"/>
>      </condition>
>    </extension>
>
>    <extension name="group_dial_sales">
>      <condition field="destination_number" expression="^2000$">
>        <action application="bridge"
> data="loopback/1000/default,loopback/1001/default"/>
>      </condition>
>    </extension>
>
> ...
> </context>
>
> --
> Jose Fco. Irles Dur?
>
>
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From monemran at gmail.com  Mon Jun  4 01:00:47 2012
From: monemran at gmail.com (Mohammad Emran)
Date: Mon, 4 Jun 2012 03:00:47 +0600
Subject: [Freeswitch-users] How to play ivr on background?
Message-ID: <71E43EA5-B5C3-499C-8B5A-73A682398FA8@gmail.com>

Hi

I would like play minute announce ivr during  the process of call. Cause it will reduce connect time  on client side.

Pls advise how to do that.

Sent from my iPhone

From msc at freeswitch.org  Mon Jun  4 01:02:19 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 3 Jun 2012 14:02:19 -0700
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <CALNZuw=5W_k2MtJPxHn6HBZanCZNk83TXiMZTEGjZ9kRjb1G4w@mail.gmail.com>
References: <4FC896B8.9070102@digitalmail.com>
	<CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>
	<CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>
	<CAKzWOxUp5h2tMFP9yHJM4LaMmF1f+sXsOfDAcETwzaDG_2WgPA@mail.gmail.com>
	<CALNZuw=5W_k2MtJPxHn6HBZanCZNk83TXiMZTEGjZ9kRjb1G4w@mail.gmail.com>
Message-ID: <CAKzWOxWH2nM37ywvkyimobkLJqAaUX531w4YgE=3eqVU5q7y=Q@mail.gmail.com>

Au contraire mon frere!

You can do multiple things in a dialstring, like setting channel variables.
You can also use execute_on_ring/media/answer to execute the extension with
doing all the loopback overhead.

I propose an experiment: provide a dialplan and loopback dialstring and
we'll see if we can't give you a non-loopbackish alternative.

-MC

On Sun, Jun 3, 2012 at 1:54 PM, Avi Marcus <avi at avimarcus.net> wrote:

> ... all the normal dialplan handling. Setting CID, options, LCR stuff,
> billing controls.
> -Avi
>
>
> On Sun, Jun 3, 2012 at 11:40 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Let me rephrase...
>>
>> Since loopback is generally evil and should be avoided wherever possible,
>> what does loopback give you that you can't get from doing a normal
>> dialstring?
>> -MC
>>
>>
>> On Sat, Jun 2, 2012 at 11:13 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> ... because att_xfer seems to require a "sofia/$profile/$destination"
>>> directive, and he just wants the call to hit the dialplan.
>>>
>>> -Avi
>>>
>>>
>>> On Fri, Jun 1, 2012 at 8:13 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Why do you need to use loopback at all?
>>>> -MC
>>>>
>>>>
>>>> On Fri, Jun 1, 2012 at 3:17 AM, Alex Lake <alex at digitalmail.com> wrote:
>>>>
>>>>> Got a lua script for a B-party "mid-call menu". Is it legitimate to
>>>>> do..
>>>>> "session:execute("att_xfer", "loopback/"..destnum)"
>>>>>
>>>>> I've tried it and it seems to start off doing the right things, but my
>>>>> A-party gets disconnected as soon as the call to the C-Party (the
>>>>> person
>>>>> I'm transferring the call to) answers the call.
>>>>>
>>>>> Maybe better to try to orchestrate the entire affair from within the
>>>>> lua
>>>>> script? (Tricky for a beginner like me!)
>>>>>
>>>>> Thanks,
>>>>> Alex
>>>>>
>>>>
>>>>
>>>>
>>
>>
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From msc at freeswitch.org  Mon Jun  4 01:08:34 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 3 Jun 2012 14:08:34 -0700
Subject: [Freeswitch-users] How to play ivr on background?
In-Reply-To: <71E43EA5-B5C3-499C-8B5A-73A682398FA8@gmail.com>
References: <71E43EA5-B5C3-499C-8B5A-73A682398FA8@gmail.com>
Message-ID: <CAKzWOxVfqhbYi30LkVknqVtfwQhjvRv=KoOvZf0n9VtggkF1eg@mail.gmail.com>

Are you ignoring early media from the target leg? If so you can just set
the ringback variable to be the announcement (like
ivr/ivr-please_hold_while_your_party_is_being_reached.wav) and then
whatever ringing/music/etc. that you want them to hear while the call is
being connected.

You might want to use this !-separated list of sound files method to play
multiple sounds in the ringback:

http://wiki.freeswitch.org/wiki/Mod_file_string

-MC

On Sun, Jun 3, 2012 at 2:00 PM, Mohammad Emran <monemran at gmail.com> wrote:

> Hi
>
> I would like play minute announce ivr during  the process of call. Cause
> it will reduce connect time  on client side.
>
> Pls advise how to do that.
>
> Sent from my iPhone
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From avi at avimarcus.net  Mon Jun  4 01:15:14 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 4 Jun 2012 00:15:14 +0300
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <CAKzWOxWH2nM37ywvkyimobkLJqAaUX531w4YgE=3eqVU5q7y=Q@mail.gmail.com>
References: <4FC896B8.9070102@digitalmail.com>
	<CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>
	<CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>
	<CAKzWOxUp5h2tMFP9yHJM4LaMmF1f+sXsOfDAcETwzaDG_2WgPA@mail.gmail.com>
	<CALNZuw=5W_k2MtJPxHn6HBZanCZNk83TXiMZTEGjZ9kRjb1G4w@mail.gmail.com>
	<CAKzWOxWH2nM37ywvkyimobkLJqAaUX531w4YgE=3eqVU5q7y=Q@mail.gmail.com>
Message-ID: <CALNZuwkxN1JXVutNWuAysLgu_kTw40L50L37U-4tZYNV_Yadrg@mail.gmail.com>

I know you can do anything in the dialstring. But intended feature is to
allow the user to do an attended transfer to any number that they could
reach via the default calling. The default outbound path already has a LOT
of stuff set up and it would be impossible to duplicate that within a
SINGLE dialstring in a function call.
What is needed is for an att_xfer to be able to have leg C hit the dialplan
and bridged however a "normal" leg B to that number would be called.
Does this make sense?

-Avi



On Mon, Jun 4, 2012 at 12:02 AM, Michael Collins <msc at freeswitch.org> wrote:

> Au contraire mon frere!
>
> You can do multiple things in a dialstring, like setting channel
> variables. You can also use execute_on_ring/media/answer to execute the
> extension with doing all the loopback overhead.
>
> I propose an experiment: provide a dialplan and loopback dialstring and
> we'll see if we can't give you a non-loopbackish alternative.
>
> -MC
>
>
> On Sun, Jun 3, 2012 at 1:54 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> ... all the normal dialplan handling. Setting CID, options, LCR stuff,
>> billing controls.
>> -Avi
>>
>>
>> On Sun, Jun 3, 2012 at 11:40 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Let me rephrase...
>>>
>>> Since loopback is generally evil and should be avoided wherever
>>> possible, what does loopback give you that you can't get from doing a
>>> normal dialstring?
>>> -MC
>>>
>>>
>>> On Sat, Jun 2, 2012 at 11:13 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> ... because att_xfer seems to require a "sofia/$profile/$destination"
>>>> directive, and he just wants the call to hit the dialplan.
>>>>
>>>> -Avi
>>>>
>>>>
>>>> On Fri, Jun 1, 2012 at 8:13 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> Why do you need to use loopback at all?
>>>>> -MC
>>>>>
>>>>>
>>>>> On Fri, Jun 1, 2012 at 3:17 AM, Alex Lake <alex at digitalmail.com>wrote:
>>>>>
>>>>>> Got a lua script for a B-party "mid-call menu". Is it legitimate to
>>>>>> do..
>>>>>> "session:execute("att_xfer", "loopback/"..destnum)"
>>>>>>
>>>>>> I've tried it and it seems to start off doing the right things, but my
>>>>>> A-party gets disconnected as soon as the call to the C-Party (the
>>>>>> person
>>>>>> I'm transferring the call to) answers the call.
>>>>>>
>>>>>> Maybe better to try to orchestrate the entire affair from within the
>>>>>> lua
>>>>>> script? (Tricky for a beginner like me!)
>>>>>>
>>>>>> Thanks,
>>>>>> Alex
>>>>>>
>>>>>
>>>>>
>>>>>
>>>
>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From monemran at gmail.com  Mon Jun  4 01:17:42 2012
From: monemran at gmail.com (Mohammad Emran)
Date: Mon, 4 Jun 2012 03:17:42 +0600
Subject: [Freeswitch-users] How to play ivr on background?
In-Reply-To: <CAKzWOxVfqhbYi30LkVknqVtfwQhjvRv=KoOvZf0n9VtggkF1eg@mail.gmail.com>
References: <71E43EA5-B5C3-499C-8B5A-73A682398FA8@gmail.com>
	<CAKzWOxVfqhbYi30LkVknqVtfwQhjvRv=KoOvZf0n9VtggkF1eg@mail.gmail.com>
Message-ID: <0AB3DD47-BA77-49B5-B6DC-516812415001@gmail.com>

Thnkx for ur faster redponse.

Yes.i wanna ignore early media.

Actually my aim is to save time of playing ivr n when finished ivr clients will hear original ringtone of leg-b.

On the ringback, how i can call SAY command to announce minutes?

It would helpfull if i get an example.

Sent from my iPhone

On Jun 4, 2012, at 3:08 AM, Michael Collins <msc at freeswitch.org> wrote:

> Are you ignoring early media from the target leg? If so you can just set the ringback variable to be the announcement (like ivr/ivr-please_hold_while_your_party_is_being_reached.wav) and then whatever ringing/music/etc. that you want them to hear while the call is being connected.
> 
> You might want to use this !-separated list of sound files method to play multiple sounds in the ringback:
> 
> http://wiki.freeswitch.org/wiki/Mod_file_string
> 
> -MC
> 
> On Sun, Jun 3, 2012 at 2:00 PM, Mohammad Emran <monemran at gmail.com> wrote:
> Hi
> 
> I would like play minute announce ivr during  the process of call. Cause it will reduce connect time  on client side.
> 
> Pls advise how to do that.
> 
> Sent from my iPhone
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From drk at drkngs.net  Mon Jun  4 03:47:51 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Sun, 03 Jun 2012 16:47:51 -0700
Subject: [Freeswitch-users] limit app problem with loopback channels
In-Reply-To: <CAKzWOxUb_mKTsL7g8W0c6Hq4ss=V=i+GBr2ThJna25Jemzm_rw@mail.gmail.com>
Message-ID: <20120603234751.68a5e203@mail.tritonwest.net>

Change that to "execute_on_originate". On answer will blow the call off if it's over limits when the user answers.  
   
--Dave
      _____  

  From: Michael Collins [mailto:msc at freeswitch.org]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Sun, 03 Jun 2012 13:54:08 -0700
Subject: Re: [Freeswitch-users] limit app problem with loopback channels

You can solve this issue by avoiding loopback. Don't use loopback unless there is absolutely no other alternative. Try replacing the loopback calls in the group definition to "user/xxx" instead:

   <extension name="group_dial_sales">
     <condition field="destination_number" expression="^2000$">
       <action application="bridge"
data="[execute_on_answer='limit hash callwaiting 1000 1']user/1000, [execute_on_answer='limit hash callwaiting
1001 1']user/1001"/>
     </condition>
   </extension>
  
 This is just off the top of my head, so please do the usual testing and tinkering before you report back that it does not work. :)

-MC

  
On Sun, Jun 3, 2012 at 8:47 AM, Jose Fco. Irles Dur? <josefu at gmail.com> wrote:
  Hello,

I'm trying to limit calls to one user with limit application, all
works fine when a user dials to the number user (the caller is
transfered to limit_exceeded extension)

But when a caller dials to the 2000 extensions (a "ring group" to
loopback channels) and one of the users (for example 1001) answer the
call, the limit app removes the limit count for this channel when the
loopback channel disappears (when the direct bridge occurs between the
caller channel and callee channel).

Where is the problem? How can I avoid that the limit app removes the
limit for this loopback channel?

There is the fragmet of my dialplan (is the vainilla dialplan from git
with the limit action added)
<context name="default">
...
 <extension name="Local_Extension">
     <condition field="destination_number" expression="^(10[01][0-9])$">
       <action application="export" data="dialed_extension=$1"/>
       <action application="set" data="ringback=${us-ring}"/>
       <action application="set" data="transfer_ringback=$${hold_music}"/>
       <action application="set" data="call_timeout=30"/>
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="set" data="continue_on_fail=true"/>
      <action application="limit" data="hash callwaiting
${dialed_extension} 1"/>
       <action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>
       <action application="answer"/>
       <action application="sleep" data="1000"/>
       <action application="bridge"
data="loopback/app=voicemail:default ${domain_name}
${dialed_extension}"/>
     </condition>
   </extension>

   <extension name="group_dial_sales">
     <condition field="destination_number" expression="^2000$">
       <action application="bridge"
data="loopback/1000/default,loopback/1001/default"/>
     </condition>
   </extension>

...
</context>

--
Jose Fco. Irles Dur?


      
   
 
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From msc at freeswitch.org  Mon Jun  4 03:48:04 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 3 Jun 2012 16:48:04 -0700
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <CALNZuwkxN1JXVutNWuAysLgu_kTw40L50L37U-4tZYNV_Yadrg@mail.gmail.com>
References: <4FC896B8.9070102@digitalmail.com>
	<CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>
	<CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>
	<CAKzWOxUp5h2tMFP9yHJM4LaMmF1f+sXsOfDAcETwzaDG_2WgPA@mail.gmail.com>
	<CALNZuw=5W_k2MtJPxHn6HBZanCZNk83TXiMZTEGjZ9kRjb1G4w@mail.gmail.com>
	<CAKzWOxWH2nM37ywvkyimobkLJqAaUX531w4YgE=3eqVU5q7y=Q@mail.gmail.com>
	<CALNZuwkxN1JXVutNWuAysLgu_kTw40L50L37U-4tZYNV_Yadrg@mail.gmail.com>
Message-ID: <CAKzWOxVHSGrTqoqTXnF6__1q=_zEFsUzmBhYhb40pEOoVy8gjg@mail.gmail.com>

On Sun, Jun 3, 2012 at 2:15 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I know you can do anything in the dialstring. But intended feature is to
> allow the user to do an attended transfer to any number that they could
> reach via the default calling. The default outbound path already has a LOT
> of stuff set up and it would be impossible to duplicate that within a
> SINGLE dialstring in a function call.
> What is needed is for an att_xfer to be able to have leg C hit the
> dialplan and bridged however a "normal" leg B to that number would be
> called.
> Does this make sense?
>
Perhaps, but I remain unconvinced that this scenario is impossible without
loopback. How about the OP actually supply a sample Lua script and dialplan
and call log? I'd be willing to wager that the gurus could come up with a
non-evil alternative that actually works. Just because loopback seems like
a clean solution doesn't necessarily mean that it is. I'll leave it to
Anthony to give the technical reasons why loopback doesn't always work as
one would expect or why it should be avoided wherever possible.

-MC


> -Avi
>
>
>
> On Mon, Jun 4, 2012 at 12:02 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Au contraire mon frere!
>>
>> You can do multiple things in a dialstring, like setting channel
>> variables. You can also use execute_on_ring/media/answer to execute the
>> extension with doing all the loopback overhead.
>>
>> I propose an experiment: provide a dialplan and loopback dialstring and
>> we'll see if we can't give you a non-loopbackish alternative.
>>
>> -MC
>>
>>
>> On Sun, Jun 3, 2012 at 1:54 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> ... all the normal dialplan handling. Setting CID, options, LCR stuff,
>>> billing controls.
>>> -Avi
>>>
>>>
>>> On Sun, Jun 3, 2012 at 11:40 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Let me rephrase...
>>>>
>>>> Since loopback is generally evil and should be avoided wherever
>>>> possible, what does loopback give you that you can't get from doing a
>>>> normal dialstring?
>>>> -MC
>>>>
>>>>
>>>> On Sat, Jun 2, 2012 at 11:13 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>>>
>>>>> ... because att_xfer seems to require a "sofia/$profile/$destination"
>>>>> directive, and he just wants the call to hit the dialplan.
>>>>>
>>>>> -Avi
>>>>>
>>>>>
>>>>> On Fri, Jun 1, 2012 at 8:13 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> Why do you need to use loopback at all?
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> On Fri, Jun 1, 2012 at 3:17 AM, Alex Lake <alex at digitalmail.com>wrote:
>>>>>>
>>>>>>> Got a lua script for a B-party "mid-call menu". Is it legitimate to
>>>>>>> do..
>>>>>>> "session:execute("att_xfer", "loopback/"..destnum)"
>>>>>>>
>>>>>>> I've tried it and it seems to start off doing the right things, but
>>>>>>> my
>>>>>>> A-party gets disconnected as soon as the call to the C-Party (the
>>>>>>> person
>>>>>>> I'm transferring the call to) answers the call.
>>>>>>>
>>>>>>> Maybe better to try to orchestrate the entire affair from within the
>>>>>>> lua
>>>>>>> script? (Tricky for a beginner like me!)
>>>>>>>
>>>>>>> Thanks,
>>>>>>> Alex
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>
>>>>
>>
>>
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From msc at freeswitch.org  Mon Jun  4 03:54:43 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 3 Jun 2012 16:54:43 -0700
Subject: [Freeswitch-users] How to play ivr on background?
In-Reply-To: <0AB3DD47-BA77-49B5-B6DC-516812415001@gmail.com>
References: <71E43EA5-B5C3-499C-8B5A-73A682398FA8@gmail.com>
	<CAKzWOxVfqhbYi30LkVknqVtfwQhjvRv=KoOvZf0n9VtggkF1eg@mail.gmail.com>
	<0AB3DD47-BA77-49B5-B6DC-516812415001@gmail.com>
Message-ID: <CAKzWOxVsD8vXiCfd58GDpLS+UFrJ8QazAcvHc_PCQdD9HPLoOQ@mail.gmail.com>

You're getting into a tricky area here. You want to ignore early media, yet
you want your clients to hear original ring tone of the B leg after you
play an "IVR". (I honestly don't know what you mean by "IVR" here, but I'm
assuming you just mean some sort of recorded message.)

As far as using "say" or anything else like that you need to read up on
phrase macros. I recommend starting here:
http://wiki.freeswitch.org/wiki/Speech_Phrase_Management#Phrases_Section_Primer

Also, chapter 6 of the FS book has some good information on this subject.
It recommends that you look at sounds.xml in conf/lang/en/vm/ to get an
idea of some of the things you can do with phrase macros.

-MC

On Sun, Jun 3, 2012 at 2:17 PM, Mohammad Emran <monemran at gmail.com> wrote:

> Thnkx for ur faster redponse.
>
> Yes.i wanna ignore early media.
>
> Actually my aim is to save time of playing ivr n when finished ivr clients
> will hear original ringtone of leg-b.
>
> On the ringback, how i can call SAY command to announce minutes?
>
> It would helpfull if i get an example.
>
> Sent from my iPhone
>
> On Jun 4, 2012, at 3:08 AM, Michael Collins <msc at freeswitch.org> wrote:
>
> Are you ignoring early media from the target leg? If so you can just set
> the ringback variable to be the announcement (like
> ivr/ivr-please_hold_while_your_party_is_being_reached.wav) and then
> whatever ringing/music/etc. that you want them to hear while the call is
> being connected.
>
> You might want to use this !-separated list of sound files method to play
> multiple sounds in the ringback:
>
> http://wiki.freeswitch.org/wiki/Mod_file_string
>
> -MC
>
> On Sun, Jun 3, 2012 at 2:00 PM, Mohammad Emran <monemran at gmail.com> wrote:
>
>> Hi
>>
>> I would like play minute announce ivr during  the process of call. Cause
>> it will reduce connect time  on client side.
>>
>> Pls advise how to do that.
>>
>>
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From msc at freeswitch.org  Mon Jun  4 03:55:43 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 3 Jun 2012 16:55:43 -0700
Subject: [Freeswitch-users] limit app problem with loopback channels
In-Reply-To: <20120603234751.68a5e203@mail.tritonwest.net>
References: <CAKzWOxUb_mKTsL7g8W0c6Hq4ss=V=i+GBr2ThJna25Jemzm_rw@mail.gmail.com>
	<20120603234751.68a5e203@mail.tritonwest.net>
Message-ID: <CAKzWOxVVqCiRJMna8D8bAjtmJ=621Uk6HEtXH22OYK5JVXL=zQ@mail.gmail.com>

Good point! Thanks for the tip.
-MC

On Sun, Jun 3, 2012 at 4:47 PM, Dave R. Kompel <drk at drkngs.net> wrote:

> **
> Change that to "execute_on_originate". On answer will blow the call off if
> it's over limits when the user answers.
>
> --Dave
>
>  ------------------------------
> *From:* Michael Collins [mailto:msc at freeswitch.org]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> *Sent:* Sun, 03 Jun 2012 13:54:08 -0700
> *Subject:* Re: [Freeswitch-users] limit app problem with loopback channels
>
> You can solve this issue by avoiding loopback. Don't use loopback unless
> there is absolutely no other alternative. Try replacing the loopback calls
> in the group definition to "user/xxx" instead:
>
>    <extension name="group_dial_sales">
>      <condition field="destination_number" expression="^2000$">
>        <action application="bridge"
> data="[execute_on_answer='limit hash callwaiting 1000 1']user/1000,
> [execute_on_answer='limit hash callwaiting
> 1001 1']user/1001"/>
>      </condition>
>    </extension>
>
> This is just off the top of my head, so please do the usual testing and
> tinkering before you report back that it does not work. :)
>
> -MC
>
> On Sun, Jun 3, 2012 at 8:47 AM, Jose Fco. Irles Dur? <josefu at gmail.com>wrote:
>
>> Hello,
>>
>> I'm trying to limit calls to one user with limit application, all
>> works fine when a user dials to the number user (the caller is
>> transfered to limit_exceeded extension)
>>
>> But when a caller dials to the 2000 extensions (a "ring group" to
>> loopback channels) and one of the users (for example 1001) answer the
>> call, the limit app removes the limit count for this channel when the
>> loopback channel disappears (when the direct bridge occurs between the
>> caller channel and callee channel).
>>
>> Where is the problem? How can I avoid that the limit app removes the
>> limit for this loopback channel?
>>
>> There is the fragmet of my dialplan (is the vainilla dialplan from git
>> with the limit action added)
>> <context name="default">
>> ...
>>  <extension name="Local_Extension">
>>      <condition field="destination_number" expression="^(10[01][0-9])$">
>>        <action application="export" data="dialed_extension=$1"/>
>>        <action application="set" data="ringback=${us-ring}"/>
>>        <action application="set" data="transfer_ringback=$${hold_music}"/>
>>        <action application="set" data="call_timeout=30"/>
>>        <action application="set" data="hangup_after_bridge=true"/>
>>        <action application="set" data="continue_on_fail=true"/>
>>       <action application="limit" data="hash callwaiting
>> ${dialed_extension} 1"/>
>>        <action application="bridge"
>> data="user/${dialed_extension}@${domain_name}"/>
>>        <action application="answer"/>
>>        <action application="sleep" data="1000"/>
>>        <action application="bridge"
>> data="loopback/app=voicemail:default ${domain_name}
>> ${dialed_extension}"/>
>>      </condition>
>>    </extension>
>>
>>    <extension name="group_dial_sales">
>>      <condition field="destination_number" expression="^2000$">
>>        <action application="bridge"
>> data="loopback/1000/default,loopback/1001/default"/>
>>      </condition>
>>    </extension>
>>
>> ...
>> </context>
>>
>> --
>> Jose Fco. Irles Dur?
>>
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ocset at the800group.com  Mon Jun  4 09:30:48 2012
From: ocset at the800group.com (ocset)
Date: Mon, 04 Jun 2012 13:30:48 +0800
Subject: [Freeswitch-users] tones.conf for Brazil
In-Reply-To: <1338389701.3112.28.camel@modesto.localdomain.net>
References: <1338389701.3112.28.camel@modesto.localdomain.net>
Message-ID: <4FCC4808.5030005@the800group.com>

Is this useful?

http://www.google.com.au/url?sa=t&rct=j&q=&esrc=s&source=web&cd=1&ved=0CFQQFjAA&url=http%3A%2F%2Fwww.3cx.com%2Fdownloads%2Fmisc%2FGXPblog%2FCountryToneSetValues.pdf&ei=mXTHT-eVFqSSiAeqrZTjDg&usg=AFQjCNFbkJqxAa3T1KUCUC2EKZXDVHsbSQ 


Regards

On 30/05/12 22:55, Antonio Modesto wrote:
> Hi,
>
> 	I am testing FreeSWITCH/FreeTDM with a digium TDM410P card, I noticed
> that the default tones.conf has configuration for only a few contries,
> does anybody know how can I get or configure by myself tone
> configuration for brazil?
>
> Thanks
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From ocset at the800group.com  Mon Jun  4 09:36:02 2012
From: ocset at the800group.com (ocset)
Date: Mon, 04 Jun 2012 13:36:02 +0800
Subject: [Freeswitch-users] Capture media codes?
Message-ID: <4FCC4942.3000605@the800group.com>

Hi

Is there a way to capture and display the media codes being sent from a 
provider. I have been reading up about the various Early Media options 
but was hoping there was a way to log every tone being received by 
Freeswitch, to verify what tones a particular provider uses.

Thank
Dion.


From miha at softnet.si  Mon Jun  4 11:39:45 2012
From: miha at softnet.si (Miha)
Date: Mon, 04 Jun 2012 09:39:45 +0200
Subject: [Freeswitch-users] Multi-tenant dilplans
Message-ID: <4FCC6641.9060406@softnet.si>

Hi,

I set multi tenat for two domains which is forking well. Now I would 
like to seperate dialplans so that domain A will use default dialplan in 
directory dialplan/a and domain B will use default dialplan in directory 
dialplan/b.

I have tried few things but every time fs use default dialplan.

Thanks for help!

Regards,
Miha


From patrick at sunsus.net  Mon Jun  4 13:24:03 2012
From: patrick at sunsus.net (sunsus)
Date: Mon, 4 Jun 2012 02:24:03 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH TLS with StartSSL Certificate
Message-ID: <1338801843810-7579377.post@n2.nabble.com>

Hello 

To day I tried to add a Free StartSSL Class 1 Certificate to a FreeSWITCH
installation. Here I will share the script on how to generate the
Certificate Request: 


#!/bin/sh

CONFDIR=/usr/local/freeswitch/conf/ssl
DAYS=2190
KEY_SIZE=2048

TMPFILE="/tmp/fs-ca-$$-$(date +%Y%m%d%H%M%S)"

COMMON_NAME="FrwwSWICH VOIP"
ALT_NAME="DNS:sip.freeswitch.org"
ORG_NAME="FreeSWICHT"
OUTFILE="agent.pem"

umask 037

generate_request() {
    local val=""

    echo "Generating new request..."

    echo
    echo "--------------------------------------------------------"
    echo "CN: \"${COMMON_NAME}\""
    echo "ORG_NAME: \"${ORG_NAME}\""
    echo "ALT_NAME: \"${ALT_NAME}\""
    echo
    echo "Certificate filename \"${OUTFILE}\""
    echo
    echo "[Is this OK? (y/N)]"
    read val
    if [ "${val}" != "y" ] && [ "${val}" != "Y" ]; then
        echo "Aborted"
        return 2
    fi

    sed \
        -e "s|%CN%|$COMMON_NAME|" \
        -e "s|%ALTNAME%|$ALT_NAME|" \
        -e "s|%ORG%|$ORG_NAME|" \
        "${CONFDIR}/CA/config.tpl" \
        > "${TMPFILE}.cfg" || exit 1


    echo ${KEY_SIZE}
        openssl req -new -out "${TMPFILE}.req" \
            -newkey rsa:${KEY_SIZE} -keyout "${TMPFILE}.key" \
            -config "${TMPFILE}.cfg" -nodes -sha1 >/dev/null || exit 1
    
    echo
    cat ${TMPFILE}.req
    echo
    echo "go to http://www.startssl.com/ and generate a certificate"
    echo "past certificate:"
    while read LINE
        do
        echo $LINE >> ${TMPFILE}.crt
        if [ "$LINE" = "^A" ];then
            break
        fi
    done
    echo "other processing continues "

# openssl x509 -req -CAkey "${CONFDIR}/CA/cakey.pem" -CA
"${CONFDIR}/CA/cacert.pem" -CAcreateserial \
# -in "${TMPFILE}.req" -out "${TMPFILE}.crt" -extfile "${TMPFILE}.cfg" \
# -extensions "${EXTENSIONS}" -days ${DAYS} -sha1 >/dev/null || exit 1
    cat "${TMPFILE}.crt" "${TMPFILE}.key" > "${CONFDIR}/${OUTFILE}"

    wget http://www.startssl.com/certs/sub.class1.server.ca.pem
    wget http://www.startssl.com/certs/ca.pem
    cat sub.class1.server.ca.pem ca.pem >> ${CONFDIR}/cafile.pem
    rm -f sub.class1.server.ca.pem ca.pem
    rm "${TMPFILE}.cfg" "${TMPFILE}.crt" "${TMPFILE}.key" "${TMPFILE}.req"

    echo "DONE"
}


remove_startssl() {
	echo "Removing StartSSL"

	if [ -d "${CONFDIR}/agent.pem" ]; then
		rm "${CONFDIR}/agent.pem"
	fi

	echo "DONE"
}
OUTFILESET="0"
command="$1"
shift

while [ $# -gt 0 ]; do
	case $1 in
		-cn)
			shift
			COMMON_NAME="$1"
			;;
		-alt)
			shift
			ALT_NAME="$1"
			;;
		-org)
			shift
			ORG_NAME="$1"
			;;
		-out)
			shift
			OUTFILE="$1"
			OUTFILESET="1"
			;;
		-days)
			shift
			DAYS="$1"
			;;
	esac
	shift
done


case ${command} in
    create_request)
        EXTENSIONS="request"
        generate_request
        ;;

	remove)
		echo "Are you sure you want to delete the StartSSL Certificate? [YES to
delete]"
		read val
		if [ "${val}" = "YES" ]; then
			remove_startssl
		else
			echo "Not deleting CA"
		fi
		;;

	*)
		cat <<-EOF
		$0 <create_request|remove> [options]

		  * commands:

		    remove - Remove StartSSL

            create_request - Create a new certificate request for startSSL

		  * options:

		   -cn       Set common name
		   -alt      Set alternative name (use prefix 'DNS:' or 'URI:')
		   -org      Set organization name
		   -out      Filename for new certificate (create only)
		   -days     Certificate expires in X days (default: 365)

		EOF
		exit 1
		;;
esac


Everything seams to work, expect the validation of a SNOM phone. Does any
one know how to tell FreeSWITCH to publish the correct ca bundel and
certificate track. Because the CA Certificate of Start SSL is included in
the SNOM:

regards

Patrick 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-TLS-with-StartSSL-Certificate-tp7579377.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From chrisbware at yahoo.it  Mon Jun  4 13:54:58 2012
From: chrisbware at yahoo.it (Chris B. Ware)
Date: Mon, 4 Jun 2012 10:54:58 +0100 (BST)
Subject: [Freeswitch-users] Music on hold while I add someone to the
	conference
Message-ID: <1338803698.17011.YahooMailNeo@web132306.mail.ird.yahoo.com>

Hi,

I'm testing example reported here:

http://wiki.freeswitch.org/wiki/Conference_Add_Call_Example



I'd like to modify extension?"Add new OB call to conference" in order to play music on hold to the other conference members , while moderator digit a new member number to add. In my case we have a conference with A (moderator) and B (first member) and A try to add C. That's my conf:

?<extension name="Add new OB call to conference">
? ? ? <condition field="destination_number" expression="^ADD_THIRD$">
? ? ? ? <action application="set" data="res=${bgapi uuid_hold ${uuid_A} }" />

? ? ? ???<!-- ask caller for a number + #, collect into ${target_num} variable -->
? ? ? ? <action application="play_and_get_digits" data="4 11 1 5000 # ivr/ivr-enter_destination_telephone_number.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/>
? ? ? ?<action application="set" data="res=${bgapi uuid_hold off ${uuid_A} }" />
? ? ? ? <!-- add this call to the conference -->
? ? ? ? <action application="execute_extension" data="ADD_CALL_TO_CONF_${target_num}"/>
? ? ? </condition>
? ? </extension>


Everything works except B can't hear any music. Where's the error?


Thank you
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From monemran at gmail.com  Mon Jun  4 14:13:51 2012
From: monemran at gmail.com (M.Emran)
Date: Mon, 4 Jun 2012 16:13:51 +0600
Subject: [Freeswitch-users] How to play ivr on background?
In-Reply-To: <CAKzWOxVsD8vXiCfd58GDpLS+UFrJ8QazAcvHc_PCQdD9HPLoOQ@mail.gmail.com>
References: <71E43EA5-B5C3-499C-8B5A-73A682398FA8@gmail.com>
	<CAKzWOxVfqhbYi30LkVknqVtfwQhjvRv=KoOvZf0n9VtggkF1eg@mail.gmail.com>
	<0AB3DD47-BA77-49B5-B6DC-516812415001@gmail.com>
	<CAKzWOxVsD8vXiCfd58GDpLS+UFrJ8QazAcvHc_PCQdD9HPLoOQ@mail.gmail.com>
Message-ID: <CAMbKPxaQ5tSMEtVSxL4KOOygMAgxbsSaSGQ=2GahHDZkH+4spQ@mail.gmail.com>

IVR means to play a native sound file i.e. "You have 10 minutes"

I tried to set ringback data on dialplan. i got some noise or some tone.
here is my settings.

<action application="set" data="continue_on_fail=true"/>
<action application="set" data="call_timeout=30"/>
<action application="set" data="session_in_hangup_hook=true"/>
<action application="set" data="bypass_media=false"/>
<action application="set" data="ignore_early_media=true"/>
<action application="ring_ready"/>
<action application="set" data="ringback=voicemail/vm-you_have.G729"/>

Am i doing any wrong ?

On Mon, Jun 4, 2012 at 5:54 AM, Michael Collins <msc at freeswitch.org> wrote:

> You're getting into a tricky area here. You want to ignore early media,
> yet you want your clients to hear original ring tone of the B leg after you
> play an "IVR". (I honestly don't know what you mean by "IVR" here, but I'm
> assuming you just mean some sort of recorded message.)
>
> As far as using "say" or anything else like that you need to read up on
> phrase macros. I recommend starting here:
>
> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management#Phrases_Section_Primer
>
> Also, chapter 6 of the FS book has some good information on this subject.
> It recommends that you look at sounds.xml in conf/lang/en/vm/ to get an
> idea of some of the things you can do with phrase macros.
>
> -MC
>
>
> On Sun, Jun 3, 2012 at 2:17 PM, Mohammad Emran <monemran at gmail.com> wrote:
>
>> Thnkx for ur faster redponse.
>>
>> Yes.i wanna ignore early media.
>>
>> Actually my aim is to save time of playing ivr n when finished ivr
>> clients will hear original ringtone of leg-b.
>>
>> On the ringback, how i can call SAY command to announce minutes?
>>
>> It would helpfull if i get an example.
>>
>> Sent from my iPhone
>>
>> On Jun 4, 2012, at 3:08 AM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> Are you ignoring early media from the target leg? If so you can just set
>> the ringback variable to be the announcement (like
>> ivr/ivr-please_hold_while_your_party_is_being_reached.wav) and then
>> whatever ringing/music/etc. that you want them to hear while the call is
>> being connected.
>>
>> You might want to use this !-separated list of sound files method to play
>> multiple sounds in the ringback:
>>
>> http://wiki.freeswitch.org/wiki/Mod_file_string
>>
>> -MC
>>
>> On Sun, Jun 3, 2012 at 2:00 PM, Mohammad Emran <monemran at gmail.com>wrote:
>>
>>> Hi
>>>
>>> I would like play minute announce ivr during  the process of call. Cause
>>> it will reduce connect time  on client side.
>>>
>>> Pls advise how to do that.
>>>
>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards
----------
M Emran
www.e-softbilling.com
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From ocset at the800group.com  Mon Jun  4 14:13:52 2012
From: ocset at the800group.com (ocset)
Date: Mon, 04 Jun 2012 18:13:52 +0800
Subject: [Freeswitch-users] Sequential endpoint bridge question.
Message-ID: <4FCC8A60.7040509@the800group.com>

Hi

I have a setup where I am using a GXW4104 FXO gateway to access a GSM 
endpoint. With a simple dialplan, making calls works great.

However, I am trying to setup a "Sequential Call Routing" where 
Freeswitch first tries the GSM endpoint and then uses a PSTN line 
instead. With the GSM line, there is always a 5 second delay before the 
call is made, which does not cause an issue with a simple dialplan, but 
when I try the a sequential

Here is an example of the simple dialplan which works

<extension name="Standard Call">
<condition field="destination_number" expression="^(\d{10})$">
<action application="bridge" data="sofia/internal/$1 at 192.168.0.160:5064" />
</condition>
</extension>

<extension name="Standard Call - Seq">
<condition field="destination_number" expression="^(\d{10})$">
<action application="bridge" 
data="{ignore_early_media=true}sofia/internal/$1 at 192.168.0.160:5064 | 
sofia/internal/$1 at 192.168.0.160:5060" />
</condition>
</extension>

What can I do to get Freeswitch to wait longer for the first endpoint or 
is there a better way to achieve this? Should I have two dialplans, one 
after another, that will execute the second if the GSM one fails?

Thanks in advance
O






From miha at softnet.si  Mon Jun  4 14:44:39 2012
From: miha at softnet.si (Miha)
Date: Mon, 04 Jun 2012 12:44:39 +0200
Subject: [Freeswitch-users] Multi-tenant dilplans
In-Reply-To: <4FCC6641.9060406@softnet.si>
References: <4FCC6641.9060406@softnet.si>
Message-ID: <4FCC9197.1010101@softnet.si>

On 6/4/2012 9:39 AM, Miha wrote:
> Hi,
>
> I set multi tenat for two domains which is forking well. Now I would
> like to seperate dialplans so that domain A will use default dialplan in
> directory dialplan/a and domain B will use default dialplan in directory
> dialplan/b.
>
> I have tried few things but every time fs use default dialplan.
>
> Thanks for help!
>
> Regards,
> Miha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 was having set wrong <variable name="user_context" value="domain" /> .

THanks!

Miha


From nasida at live.ru  Mon Jun  4 16:02:00 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Mon, 4 Jun 2012 16:02:00 +0400
Subject: [Freeswitch-users] Playing of mms stream by means of FS
In-Reply-To: <SNT127-W50ED4868C4CDBF33B153B0D5090@phx.gbl>
References: <SNT127-W22D38D5ECE9C422DFB0D06D5090@phx.gbl>,
	<SNT127-W50ED4868C4CDBF33B153B0D5090@phx.gbl>
Message-ID: <SNT127-W30E999B7D45BB27AADDD6BD50F0@phx.gbl>


Up.

From: nasida at live.ru
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 3 Jun 2012 01:20:27 +0400
Subject: Re: [Freeswitch-users] Playing of mms stream by means of FS





I try to install mod_vlc but have many errors like this
make mod_vlc-install
/usr/src/freeswitch/src/mod/formats/mod_vlc/mod_vlc.c:42:21: error: vlc/vlc.h: No such file or directory/usr/src/freeswitch/src/mod/formats/mod_vlc/mod_vlc.c:43:37: error: vlc/libvlc_media_player.h: No such file or directory/usr/src/freeswitch/src/mod/formats/mod_vlc/mod_vlc.c:52: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token/usr/src/freeswitch/src/mod/formats/mod_vlc/mod_vlc.c:55: error: expected specifier-qualifier-list before ?libvlc_media_player_t?cc1: warnings being treated as errors

With mod_shout I can play HTTP stream only. Please correct me if I wrong.
How can I listen mms stream (mms://live.rfn.ru/rcult_64) ?
I still wait any advise.Thanks.

From: nasida at live.ru
To: freeswitch-users at lists.freeswitch.org
Date: Sat, 2 Jun 2012 20:18:44 +0400
Subject: [Freeswitch-users] Playing of mms stream by means of FS





Hi guys
What the best way for the playing of mms stream like this  mms://live.rfn.ru/rcult_64  by means of FS ?I try to use mod_shout but without success. Do I have to use mplayer or vlc for this ?I saw this in wiki  http://wiki.freeswitch.org/wiki/Mod_vlc 
But... probably there is the more better way ?
Please adviseThanks. 		 	   		  

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From kbdfck at gmail.com  Mon Jun  4 18:53:01 2012
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Mon, 4 Jun 2012 18:53:01 +0400
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <CAKzWOxVHSGrTqoqTXnF6__1q=_zEFsUzmBhYhb40pEOoVy8gjg@mail.gmail.com>
References: <4FC896B8.9070102@digitalmail.com>
	<CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>
	<CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>
	<CAKzWOxUp5h2tMFP9yHJM4LaMmF1f+sXsOfDAcETwzaDG_2WgPA@mail.gmail.com>
	<CALNZuw=5W_k2MtJPxHn6HBZanCZNk83TXiMZTEGjZ9kRjb1G4w@mail.gmail.com>
	<CAKzWOxWH2nM37ywvkyimobkLJqAaUX531w4YgE=3eqVU5q7y=Q@mail.gmail.com>
	<CALNZuwkxN1JXVutNWuAysLgu_kTw40L50L37U-4tZYNV_Yadrg@mail.gmail.com>
	<CAKzWOxVHSGrTqoqTXnF6__1q=_zEFsUzmBhYhb40pEOoVy8gjg@mail.gmail.com>
Message-ID: <CA+GmxbYZ0fVusbdqDNeLUxwi94aDaK8SjPRU+py-V+JRL6TUqQ@mail.gmail.com>

I asked this question many times on mailing list, and now I'm sure this
can't be really done with loopback.
The only alternative for loopback is to re-inject call into FS via some
separate Sofia profile, and specify that profile in string for att_xfer.
This brings up large amount of troubles including DTMF transcoding,
sequential att_xfer attempt recognition and overall voice/dtmf delay
introduced by chained channels. Maybe some channels can be moved out of
scene by using 'simplify' api on correct channels, but this needs tests.

Anyway, loopback channel in FS is completely unusable, so we do need to
have some  best practices on how to do things without it in FS wiki...
Maybe I have time and will describe our experience soon.

2012/6/4 Michael Collins <msc at freeswitch.org>

>
>
> On Sun, Jun 3, 2012 at 2:15 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I know you can do anything in the dialstring. But intended feature is to
>> allow the user to do an attended transfer to any number that they could
>> reach via the default calling. The default outbound path already has a LOT
>> of stuff set up and it would be impossible to duplicate that within a
>> SINGLE dialstring in a function call.
>> What is needed is for an att_xfer to be able to have leg C hit the
>> dialplan and bridged however a "normal" leg B to that number would be
>> called.
>> Does this make sense?
>>
> Perhaps, but I remain unconvinced that this scenario is impossible without
> loopback. How about the OP actually supply a sample Lua script and dialplan
> and call log? I'd be willing to wager that the gurus could come up with a
> non-evil alternative that actually works. Just because loopback seems like
> a clean solution doesn't necessarily mean that it is. I'll leave it to
> Anthony to give the technical reasons why loopback doesn't always work as
> one would expect or why it should be avoided wherever possible.
>
> -MC
>
>
>>  -Avi
>>
>>
>>
>> On Mon, Jun 4, 2012 at 12:02 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Au contraire mon frere!
>>>
>>> You can do multiple things in a dialstring, like setting channel
>>> variables. You can also use execute_on_ring/media/answer to execute the
>>> extension with doing all the loopback overhead.
>>>
>>> I propose an experiment: provide a dialplan and loopback dialstring and
>>> we'll see if we can't give you a non-loopbackish alternative.
>>>
>>> -MC
>>>
>>>
>>> On Sun, Jun 3, 2012 at 1:54 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> ... all the normal dialplan handling. Setting CID, options, LCR stuff,
>>>> billing controls.
>>>> -Avi
>>>>
>>>>
>>>> On Sun, Jun 3, 2012 at 11:40 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> Let me rephrase...
>>>>>
>>>>> Since loopback is generally evil and should be avoided wherever
>>>>> possible, what does loopback give you that you can't get from doing a
>>>>> normal dialstring?
>>>>> -MC
>>>>>
>>>>>
>>>>> On Sat, Jun 2, 2012 at 11:13 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>>>>
>>>>>> ... because att_xfer seems to require a "sofia/$profile/$destination"
>>>>>> directive, and he just wants the call to hit the dialplan.
>>>>>>
>>>>>> -Avi
>>>>>>
>>>>>>
>>>>>> On Fri, Jun 1, 2012 at 8:13 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>
>>>>>>> Why do you need to use loopback at all?
>>>>>>> -MC
>>>>>>>
>>>>>>>
>>>>>>> On Fri, Jun 1, 2012 at 3:17 AM, Alex Lake <alex at digitalmail.com>wrote:
>>>>>>>
>>>>>>>> Got a lua script for a B-party "mid-call menu". Is it legitimate to
>>>>>>>> do..
>>>>>>>> "session:execute("att_xfer", "loopback/"..destnum)"
>>>>>>>>
>>>>>>>> I've tried it and it seems to start off doing the right things, but
>>>>>>>> my
>>>>>>>> A-party gets disconnected as soon as the call to the C-Party (the
>>>>>>>> person
>>>>>>>> I'm transferring the call to) answers the call.
>>>>>>>>
>>>>>>>> Maybe better to try to orchestrate the entire affair from within
>>>>>>>> the lua
>>>>>>>> script? (Tricky for a beginner like me!)
>>>>>>>>
>>>>>>>> Thanks,
>>>>>>>> Alex
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>
>>>>>
>>>
>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From josefu at gmail.com  Mon Jun  4 14:21:28 2012
From: josefu at gmail.com (=?ISO-8859-1?Q?Jose_Fco=2E_Irles_Dur=E1?=)
Date: Mon, 4 Jun 2012 12:21:28 +0200
Subject: [Freeswitch-users] limit app problem with loopback channels
In-Reply-To: <CAKzWOxVVqCiRJMna8D8bAjtmJ=621Uk6HEtXH22OYK5JVXL=zQ@mail.gmail.com>
References: <CAKzWOxUb_mKTsL7g8W0c6Hq4ss=V=i+GBr2ThJna25Jemzm_rw@mail.gmail.com>
	<20120603234751.68a5e203@mail.tritonwest.net>
	<CAKzWOxVVqCiRJMna8D8bAjtmJ=621Uk6HEtXH22OYK5JVXL=zQ@mail.gmail.com>
Message-ID: <CAJOPU6X4dPV1an6g6Cy7w7bMvaZo2qAihVAGxzxQZs6s0i1v_A@mail.gmail.com>

I tried with "execute_on_originate" and with one "member" in the ring
group works, but when I have more than one member doesn't work.

This is a log with one memeber on the ring group (user 100), this works:
http://pastebin.com/9rdeQhDf

And this is with two members (user 100 and 171), don't work:
http://pastebin.com/aTdTTnnH

My bridge args is:
[execute_on_originate=limit hash callwaiting 100
1]user/100 at 128.23.0.1,[execute_on_originate=limit hash callwaiting 171
1]user/171 at 128.23.0.1 for the second case.


I'm thinking that I need some event processing for doing this.


2012/6/4 Michael Collins <msc at freeswitch.org>:
> Good point! Thanks for the tip.
> -MC
>
> On Sun, Jun 3, 2012 at 4:47 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>>
>> Change that to "execute_on_originate". On answer will blow the call off if
>> it's over limits when the user answers.
>>
>> --Dave
>>
>> ________________________________
>> From: Michael Collins [mailto:msc at freeswitch.org]
>> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
>> Sent: Sun, 03 Jun 2012 13:54:08 -0700
>> Subject: Re: [Freeswitch-users] limit app problem with loopback channels
>>
>> You can solve this issue by avoiding loopback. Don't use loopback unless
>> there is absolutely no other alternative. Try replacing the loopback calls
>> in the group definition to "user/xxx" instead:
>>
>> ?? <extension name="group_dial_sales">
>> ? ? ?<condition field="destination_number" expression="^2000$">
>> ? ? ? ?<action application="bridge"
>> data="[execute_on_answer='limit hash callwaiting 1000 1']user/1000,
>> [execute_on_answer='limit hash callwaiting
>> 1001 1']user/1001"/>
>> ? ?? </condition>
>> ?? </extension>
>>
>> This is just off the top of my head, so please do the usual testing and
>> tinkering before you report back that it does not work. :)
>>
>> -MC
>>
>> On Sun, Jun 3, 2012 at 8:47 AM, Jose Fco. Irles Dur? <josefu at gmail.com>
>> wrote:
>>>
>>> Hello,
>>>
>>> I'm trying to limit calls to one user with limit application, all
>>> works fine when a user dials to the number user (the caller is
>>> transfered to limit_exceeded extension)
>>>
>>> But when a caller dials to the 2000 extensions (a "ring group" to
>>> loopback channels) and one of the users (for example 1001) answer the
>>> call, the limit app removes the limit count for this channel when the
>>> loopback channel disappears (when the direct bridge occurs between the
>>> caller channel and callee channel).
>>>
>>> Where is the problem? How can I avoid that the limit app removes the
>>> limit for this loopback channel?
>>>
>>> There is the fragmet of my dialplan (is the vainilla dialplan from git
>>> with the limit action added)
>>> <context name="default">
>>> ...
>>> ?<extension name="Local_Extension">
>>> ? ? ?<condition field="destination_number" expression="^(10[01][0-9])$">
>>> ? ? ? ?<action application="export" data="dialed_extension=$1"/>
>>> ? ? ? ?<action application="set" data="ringback=${us-ring}"/>
>>> ? ? ? ?<action application="set"
>>> data="transfer_ringback=$${hold_music}"/>
>>> ? ? ? ?<action application="set" data="call_timeout=30"/>
>>> ? ? ? ?<action application="set" data="hangup_after_bridge=true"/>
>>> ? ? ? ?<action application="set" data="continue_on_fail=true"/>
>>> ? ? ? <action application="limit" data="hash callwaiting
>>> ${dialed_extension} 1"/>
>>> ? ? ? ?<action application="bridge"
>>> data="user/${dialed_extension}@${domain_name}"/>
>>> ? ? ? ?<action application="answer"/>
>>> ? ? ? ?<action application="sleep" data="1000"/>
>>> ? ? ? ?<action application="bridge"
>>> data="loopback/app=voicemail:default ${domain_name}
>>> ${dialed_extension}"/>
>>> ? ? ?</condition>
>>> ? ?</extension>
>>>
>>> ? ?<extension name="group_dial_sales">
>>> ? ? ?<condition field="destination_number" expression="^2000$">
>>> ? ? ? ?<action application="bridge"
>>> data="loopback/1000/default,loopback/1001/default"/>
>>> ? ? ?</condition>
>>> ? ?</extension>
>>>
>>> ...
>>> </context>
>>>
>>> --
>>> Jose Fco. Irles Dur?
>>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jose Fco. Irles Dur?


From msc at freeswitch.org  Mon Jun  4 19:12:15 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Jun 2012 08:12:15 -0700
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <CA+GmxbYZ0fVusbdqDNeLUxwi94aDaK8SjPRU+py-V+JRL6TUqQ@mail.gmail.com>
References: <4FC896B8.9070102@digitalmail.com>
	<CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>
	<CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>
	<CAKzWOxUp5h2tMFP9yHJM4LaMmF1f+sXsOfDAcETwzaDG_2WgPA@mail.gmail.com>
	<CALNZuw=5W_k2MtJPxHn6HBZanCZNk83TXiMZTEGjZ9kRjb1G4w@mail.gmail.com>
	<CAKzWOxWH2nM37ywvkyimobkLJqAaUX531w4YgE=3eqVU5q7y=Q@mail.gmail.com>
	<CALNZuwkxN1JXVutNWuAysLgu_kTw40L50L37U-4tZYNV_Yadrg@mail.gmail.com>
	<CAKzWOxVHSGrTqoqTXnF6__1q=_zEFsUzmBhYhb40pEOoVy8gjg@mail.gmail.com>
	<CA+GmxbYZ0fVusbdqDNeLUxwi94aDaK8SjPRU+py-V+JRL6TUqQ@mail.gmail.com>
Message-ID: <CAKzWOxUmazNg3OwEEm5e3H7t2=-r4106WwWC66Pnpuaq9c6d1w@mail.gmail.com>

Comments inline...
On Mon, Jun 4, 2012 at 7:53 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> I asked this question many times on mailing list, and now I'm sure this
> can't be really done with loopback.
> The only alternative for loopback is to re-inject call into FS via some
> separate Sofia profile, and specify that profile in string for att_xfer.
>
Maybe there's a more elegant way to do this.

This brings up large amount of troubles including DTMF transcoding,
> sequential att_xfer attempt recognition and overall voice/dtmf delay
> introduced by chained channels. Maybe some channels can be moved out of
> scene by using 'simplify' api on correct channels, but this needs tests.

This makes me wonder if att_xfer is the right tool.


> Anyway, loopback channel in FS is completely unusable, so we do need to
> have some  best practices on how to do things without it in FS wiki...
> Maybe I have time and will describe our experience soon.
>
Please do. I get the feeling that we (the community attempting to help)
have just enough information to be frustrated. I suspect that if we knew
the big picture we could probably offer some more useful suggestions.

-MC
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From admin at blindi.net  Mon Jun  4 19:27:54 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Mon, 4 Jun 2012 17:27:54 +0200 (CEST)
Subject: [Freeswitch-users] opensource project voicechat for freeswitch
In-Reply-To: <1338558318.79806.YahooMailNeo@web120405.mail.ne1.yahoo.com>
References: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
	<2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com>
	<1338558318.79806.YahooMailNeo@web120405.mail.ne1.yahoo.com>
Message-ID: <Pine.LNX.4.64.1206041700001.22234@tomback.antagus.de>

Hi all,

I like to plan a opensourceproject.
Open_voicechat is the first open source voice chat worldwide.
Freeswitch and other voip-systems don.t have a big application.

Unfortunately, I can not even work, programming.
I am looking for people to help me, and dominate one of the following 
programming languages:
perl lua java php python or ruby.
There is no specific programming language requested.

I have setup a big mysql database for all modular features of a nice chat.
Mailboxes, news forums, line managment and more!
An exact documented functionplan  has already been added.
A german demo voicefile has ben included to the archive.

You can download my long work at:
http://www.blindi.net/downloads/open_vocechat.tar.gz
I have 2 Languages german and english added to the archive.


I would be happy if someone help me.
The advantage of FreeSWITCH is the flexibility.

Thanks for your nice help




---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From bdfoster at endigotech.com  Mon Jun  4 19:31:45 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 04 Jun 2012 11:31:45 -0400
Subject: [Freeswitch-users] opensource project voicechat for freeswitch
In-Reply-To: <Pine.LNX.4.64.1206041700001.22234@tomback.antagus.de>
References: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
	<2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com>
	<1338558318.79806.YahooMailNeo@web120405.mail.ne1.yahoo.com>
	<Pine.LNX.4.64.1206041700001.22234@tomback.antagus.de>
Message-ID: <4FCCD4E1.9050703@endigotech.com>

What kind of voice chat are we talking about? There are plenty of voice 
chat type things out there (skype for one). Specify your demographic 
and what the actual purpose is, and how it's going to be different from 
any other provider.

-BDF

On Mon 04 Jun 2012 11:27:54 AM EDT, Thomas Hoellriegel wrote:
> Hi all,
>
> I like to plan a opensourceproject.
> Open_voicechat is the first open source voice chat worldwide.
> Freeswitch and other voip-systems don.t have a big application.
>
> Unfortunately, I can not even work, programming.
> I am looking for people to help me, and dominate one of the following
> programming languages:
> perl lua java php python or ruby.
> There is no specific programming language requested.
>
> I have setup a big mysql database for all modular features of a nice
> chat.
> Mailboxes, news forums, line managment and more!
> An exact documented functionplan has already been added.
> A german demo voicefile has ben included to the archive.
>
> You can download my long work at:
> http://www.blindi.net/downloads/open_vocechat.tar.gz
> I have 2 Languages german and english added to the archive.
>
>
> I would be happy if someone help me.
> The advantage of FreeSWITCH is the flexibility.
>
> Thanks for your nice help
>
>
>
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From sdevoy at bizfocused.com  Mon Jun  4 19:39:40 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 4 Jun 2012 11:39:40 -0400
Subject: [Freeswitch-users] Multi-tenant dilplans
In-Reply-To: <4FCC9197.1010101@softnet.si>
References: <4FCC6641.9060406@softnet.si> <4FCC9197.1010101@softnet.si>
Message-ID: <03e101cd4268$41651cf0$c42f56d0$@bizfocused.com>

HI Miha,

I can almost never answer a questi9on here, but this one I have working
perfectly.

In the user directory where you define you extensions, you must set the
"context" for that user's/extension's dialplan:
<variable name="user_context" value="from-COMPANY_A"/>  pick your own value,
but match it in the dialplan.

In the dialplan, you start with that context, and group your dialplan
directives under inside it.  In this example someone from a Company_A
extension dials extension 200.  You cann see how the context allows Company
A users to dial 200 and Company B users to dial extension 200 and get
different users.

<context name="from-COMPANY_A">
     <extension name="200" >
        <condition field="destination_number" expression="^200$">
              <action application="set" data="ringback=${us-ring}"/>
              <action application="set"
data="effective_caller_id_number=${internal_caller_id_number}"/>
              <action application="set"
data="effective_caller_id_name=${internal_caller_id_name}"/>
              <action application="set" data="hangup_after_bridge=true"/>
              <action application="set" data="call_timeout=20"/>
             <action application="bridge"
data="{sip_invite_domain=company_a.com}user/200 at company_a.com"  />
             <action application="answer"/>
             <action application="voicemail" data="default
voicemail_COMPANY_A 200"/>
             <action application="hangup"/>
          </condition>
      </extension>
...
</context>

Just FYI, here is my full directory entry for extension 200 (which also has
a Shared Call Appearance that you may not want/need).  I have changed the
domain names and password:
    <user id="200">
        <variables>
          <variable name="outbound_caller_id_name" value="Our Company
name"/>
          <variable name="outbound_caller_id_number" value="Our External
Number"/>
          <variable name="internal_caller_id_name" value="Shirley"/>
          <variable name="internal_caller_id_number" value="200"/>
          <variable name="user_context" value="from-COMPANY_A"/>
          <variable name="user_originated" value="true"/>
          <variable name="toll_allow" value="domestic"/>
          <variable name="accountcode" value="200"/>
          <variable name="mwi-account" value="200 at voicemail_COMPANY_A"/>
        </variables>
        <params>
          <param name="password"  value="SomePasswordHere"/>
          <param name="manage-shared-appearance" value-="true" />"
          <param name="dial-string"
value="{presence_id=200 at company_a.com}${sofia_contact(200 at company_a.com)}"/>
          <param name="mwi-account" value="200 at voicemail_company_a"/>
        </params>
      </user>

I hope that helps.

Sean
-----Original Message-----
From: Miha [mailto:miha at softnet.si] 
Sent: Monday, June 04, 2012 6:45 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Multi-tenant dilplans

On 6/4/2012 9:39 AM, Miha wrote:
> Hi,
>
> I set multi tenat for two domains which is forking well. Now I would 
> like to seperate dialplans so that domain A will use default dialplan 
> in directory dialplan/a and domain B will use default dialplan in 
> directory dialplan/b.
>
> I have tried few things but every time fs use default dialplan.
>
> Thanks for help!
>
> Regards,
> Miha
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users 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
>
Hi,

I was having set wrong <variable name="user_context" value="domain" /> .

THanks!

Miha






From msc at freeswitch.org  Mon Jun  4 20:09:20 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Jun 2012 09:09:20 -0700
Subject: [Freeswitch-users] limit app problem with loopback channels
In-Reply-To: <CAJOPU6X4dPV1an6g6Cy7w7bMvaZo2qAihVAGxzxQZs6s0i1v_A@mail.gmail.com>
References: <CAKzWOxUb_mKTsL7g8W0c6Hq4ss=V=i+GBr2ThJna25Jemzm_rw@mail.gmail.com>
	<20120603234751.68a5e203@mail.tritonwest.net>
	<CAKzWOxVVqCiRJMna8D8bAjtmJ=621Uk6HEtXH22OYK5JVXL=zQ@mail.gmail.com>
	<CAJOPU6X4dPV1an6g6Cy7w7bMvaZo2qAihVAGxzxQZs6s0i1v_A@mail.gmail.com>
Message-ID: <CAKzWOxX6+s3b3hvfRq_FwttruoqQKNQjht=9_oYG30t-JqXCww@mail.gmail.com>

Hmm, not entirely sure why it's doing that. It looks like the limit is
being called twice for each dialstring. Just for kicks, can you change
execute_on_originate to execute_on_media? If that doesn't work, try
execute_on_ring as well. I'm just hoping to get some empirical data on what
happens in these scenarios. Once we iron it out we'll get it all wikified.

Thanks,
MC

On Mon, Jun 4, 2012 at 3:21 AM, Jose Fco. Irles Dur? <josefu at gmail.com>wrote:

> I tried with "execute_on_originate" and with one "member" in the ring
> group works, but when I have more than one member doesn't work.
>
> This is a log with one memeber on the ring group (user 100), this works:
> http://pastebin.com/9rdeQhDf
>
> And this is with two members (user 100 and 171), don't work:
> http://pastebin.com/aTdTTnnH
>
> My bridge args is:
> [execute_on_originate=limit hash callwaiting 100
> 1]user/100 at 128.23.0.1,[execute_on_originate=limit hash callwaiting 171
> 1]user/171 at 128.23.0.1 for the second case.
>
>
> I'm thinking that I need some event processing for doing this.
>
>
> 2012/6/4 Michael Collins <msc at freeswitch.org>:
> > Good point! Thanks for the tip.
> > -MC
> >
> > On Sun, Jun 3, 2012 at 4:47 PM, Dave R. Kompel <drk at drkngs.net> wrote:
> >>
> >> Change that to "execute_on_originate". On answer will blow the call off
> if
> >> it's over limits when the user answers.
> >>
> >> --Dave
> >>
> >> ________________________________
> >> From: Michael Collins [mailto:msc at freeswitch.org]
> >> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org
> ]
> >> Sent: Sun, 03 Jun 2012 13:54:08 -0700
> >> Subject: Re: [Freeswitch-users] limit app problem with loopback channels
> >>
> >> You can solve this issue by avoiding loopback. Don't use loopback unless
> >> there is absolutely no other alternative. Try replacing the loopback
> calls
> >> in the group definition to "user/xxx" instead:
> >>
> >>    <extension name="group_dial_sales">
> >>      <condition field="destination_number" expression="^2000$">
> >>        <action application="bridge"
> >> data="[execute_on_answer='limit hash callwaiting 1000 1']user/1000,
> >> [execute_on_answer='limit hash callwaiting
> >> 1001 1']user/1001"/>
> >>      </condition>
> >>    </extension>
> >>
> >> This is just off the top of my head, so please do the usual testing and
> >> tinkering before you report back that it does not work. :)
> >>
> >> -MC
> >>
> >> On Sun, Jun 3, 2012 at 8:47 AM, Jose Fco. Irles Dur? <josefu at gmail.com>
> >> wrote:
> >>>
> >>> Hello,
> >>>
> >>> I'm trying to limit calls to one user with limit application, all
> >>> works fine when a user dials to the number user (the caller is
> >>> transfered to limit_exceeded extension)
> >>>
> >>> But when a caller dials to the 2000 extensions (a "ring group" to
> >>> loopback channels) and one of the users (for example 1001) answer the
> >>> call, the limit app removes the limit count for this channel when the
> >>> loopback channel disappears (when the direct bridge occurs between the
> >>> caller channel and callee channel).
> >>>
> >>> Where is the problem? How can I avoid that the limit app removes the
> >>> limit for this loopback channel?
> >>>
> >>> There is the fragmet of my dialplan (is the vainilla dialplan from git
> >>> with the limit action added)
> >>> <context name="default">
> >>> ...
> >>>  <extension name="Local_Extension">
> >>>      <condition field="destination_number"
> expression="^(10[01][0-9])$">
> >>>        <action application="export" data="dialed_extension=$1"/>
> >>>        <action application="set" data="ringback=${us-ring}"/>
> >>>        <action application="set"
> >>> data="transfer_ringback=$${hold_music}"/>
> >>>        <action application="set" data="call_timeout=30"/>
> >>>        <action application="set" data="hangup_after_bridge=true"/>
> >>>        <action application="set" data="continue_on_fail=true"/>
> >>>       <action application="limit" data="hash callwaiting
> >>> ${dialed_extension} 1"/>
> >>>        <action application="bridge"
> >>> data="user/${dialed_extension}@${domain_name}"/>
> >>>        <action application="answer"/>
> >>>        <action application="sleep" data="1000"/>
> >>>        <action application="bridge"
> >>> data="loopback/app=voicemail:default ${domain_name}
> >>> ${dialed_extension}"/>
> >>>      </condition>
> >>>    </extension>
> >>>
> >>>    <extension name="group_dial_sales">
> >>>      <condition field="destination_number" expression="^2000$">
> >>>        <action application="bridge"
> >>> data="loopback/1000/default,loopback/1001/default"/>
> >>>      </condition>
> >>>    </extension>
> >>>
> >>> ...
> >>> </context>
> >>>
> >>> --
> >>> Jose Fco. Irles Dur?
> >>>
> >>
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Jose Fco. Irles Dur?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jeremyc at ssimicro.com  Mon Jun  4 19:57:00 2012
From: jeremyc at ssimicro.com (Jeremy Childs)
Date: Mon, 04 Jun 2012 09:57:00 -0600
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <CA+GmxbYZ0fVusbdqDNeLUxwi94aDaK8SjPRU+py-V+JRL6TUqQ@mail.gmail.com>
References: <4FC896B8.9070102@digitalmail.com>
	<CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>
	<CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>
	<CAKzWOxUp5h2tMFP9yHJM4LaMmF1f+sXsOfDAcETwzaDG_2WgPA@mail.gmail.com>
	<CALNZuw=5W_k2MtJPxHn6HBZanCZNk83TXiMZTEGjZ9kRjb1G4w@mail.gmail.com>
	<CAKzWOxWH2nM37ywvkyimobkLJqAaUX531w4YgE=3eqVU5q7y=Q@mail.gmail.com>
	<CALNZuwkxN1JXVutNWuAysLgu_kTw40L50L37U-4tZYNV_Yadrg@mail.gmail.com>
	<CAKzWOxVHSGrTqoqTXnF6__1q=_zEFsUzmBhYhb40pEOoVy8gjg@mail.gmail.com>
	<CA+GmxbYZ0fVusbdqDNeLUxwi94aDaK8SjPRU+py-V+JRL6TUqQ@mail.gmail.com>
Message-ID: <4FCCDACC.5070304@ssimicro.com>

I'm also very interested in some answers! I've been bitten by att_xfer 
in the past.

Is getting dialplan processing working within the att_xfer function 
possible, and just a low priority for implementation, or is there some 
technical reason why this is not feasible?

Lastly, is this somewhere that execute_extension can help? I've never 
been able to get it to work inside att_xfer.


On 12-06-04 8:53 AM, Dmitry Sytchev wrote:
> I asked this question many times on mailing list, and now I'm sure 
> this can't be really done with loopback.
> The only alternative for loopback is to re-inject call into FS via 
> some separate Sofia profile, and specify that profile in string for 
> att_xfer.
> This brings up large amount of troubles including DTMF transcoding, 
> sequential att_xfer attempt recognition and overall voice/dtmf delay 
> introduced by chained channels. Maybe some channels can be moved out 
> of scene by using 'simplify' api on correct channels, but this needs 
> tests.
>
> Anyway, loopback channel in FS is completely unusable, so we do need 
> to have some  best practices on how to do things without it in FS 
> wiki... Maybe I have time and will describe our experience soon.
>
> 2012/6/4 Michael Collins <msc at freeswitch.org <mailto:msc at freeswitch.org>>
>
>
>
>     On Sun, Jun 3, 2012 at 2:15 PM, Avi Marcus <avi at avimarcus.net
>     <mailto:avi at avimarcus.net>> wrote:
>
>         I know you can do anything in the dialstring. But intended
>         feature is to allow the user to do an attended transfer to any
>         number that they could reach via the default calling. The
>         default outbound path already has a LOT of stuff set up and it
>         would be impossible to duplicate that within a SINGLE
>         dialstring in a function call.
>         What is needed is for an att_xfer to be able to have leg C hit
>         the dialplan and bridged however a "normal" leg B to that
>         number would be called.
>         Does this make sense?
>
>     Perhaps, but I remain unconvinced that this scenario is impossible
>     without loopback. How about the OP actually supply a sample Lua
>     script and dialplan and call log? I'd be willing to wager that the
>     gurus could come up with a non-evil alternative that actually
>     works. Just because loopback seems like a clean solution doesn't
>     necessarily mean that it is. I'll leave it to Anthony to give the
>     technical reasons why loopback doesn't always work as one would
>     expect or why it should be avoided wherever possible.
>
>     -MC
>
>
>         -Avi
>
>
>
>         On Mon, Jun 4, 2012 at 12:02 AM, Michael Collins
>         <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>
>             Au contraire mon frere!
>
>             You can do multiple things in a dialstring, like setting
>             channel variables. You can also use
>             execute_on_ring/media/answer to execute the extension with
>             doing all the loopback overhead.
>
>             I propose an experiment: provide a dialplan and loopback
>             dialstring and we'll see if we can't give you a
>             non-loopbackish alternative.
>
>             -MC
>
>
>             On Sun, Jun 3, 2012 at 1:54 PM, Avi Marcus
>             <avi at avimarcus.net <mailto:avi at avimarcus.net>> wrote:
>
>                 ... all the normal dialplan handling. Setting CID,
>                 options, LCR stuff, billing controls.
>                 -Avi
>
>
>                 On Sun, Jun 3, 2012 at 11:40 PM, Michael Collins
>                 <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>
>                     Let me rephrase...
>
>                     Since loopback is generally evil and should be
>                     avoided wherever possible, what does loopback give
>                     you that you can't get from doing a normal dialstring?
>                     -MC
>
>
>                     On Sat, Jun 2, 2012 at 11:13 AM, Avi Marcus
>                     <avi at avimarcus.net <mailto:avi at avimarcus.net>> wrote:
>
>                         ... because att_xfer seems to require a
>                         "sofia/$profile/$destination" directive, and
>                         he just wants the call to hit the dialplan.
>
>                         -Avi
>
>
>                         On Fri, Jun 1, 2012 at 8:13 PM, Michael
>                         Collins <msc at freeswitch.org
>                         <mailto:msc at freeswitch.org>> wrote:
>
>                             Why do you need to use loopback at all?
>                             -MC
>
>
>                             On Fri, Jun 1, 2012 at 3:17 AM, Alex Lake
>                             <alex at digitalmail.com
>                             <mailto:alex at digitalmail.com>> wrote:
>
>                                 Got a lua script for a B-party
>                                 "mid-call menu". Is it legitimate to do..
>                                 "session:execute("att_xfer",
>                                 "loopback/"..destnum)"
>
>                                 I've tried it and it seems to start
>                                 off doing the right things, but my
>                                 A-party gets disconnected as soon as
>                                 the call to the C-Party (the person
>                                 I'm transferring the call to) answers
>                                 the call.
>
>                                 Maybe better to try to orchestrate the
>                                 entire affair from within the lua
>                                 script? (Tricky for a beginner like me!)
>
>                                 Thanks,
>                                 Alex
>
>
>
>
>
>
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
>
> -- 
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Jun  4 20:55:22 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 4 Jun 2012 09:55:22 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly News And Notes
Message-ID: <CAKzWOxU2woeZjMeHkHnkMjUejU1Huwe3Ttf7059sV1ndSjWuLw@mail.gmail.com>

Happy June to all!

We had a busy month of May but last Wednesday (May 30) we took a break from
the formal presentations on the FreeSWITCH conference call and instead
focused some time and energy on topics of general interest to our
community. Feel free to
download<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls?cc12-0604>last
week's conference call recording and listen to Travis Cross and myself
discuss the pros and cons of licensing for content other than source code,
such as the wiki documentation and the sounds we've recorded over the
years.

Another topic of interest is the original FreeSWITCH
book<http://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book?cc12-0604>,
a.k.a. the "bridge book" because of the cover photograph. Can you believe
that this summer it will be two years since the book was released? The
FreeSWITCH team and a number of interested community members have been
discussing what a second edition would look like - what changes would be
made, what new content could be added, etc. We have no definitive plans at
the moment, so now is the time to talk about what you would like to see. Do
you have an idea on how to make the second edition of the FreeSWITCH book
even better? If so, please let us know. Perhaps by ClueCon 2013 we'll have
a brand new book on the shelves!

Speaking of ClueCon <http://www.cluecon.com?cc12-0604>, we would like to
let everyone know that the plans for this year's event are coming along
nicely. Our speaker <http://www.cluecon.com/speakers?cc12-0604> list is
growing and the schedule is filling out. We still have a few openings, so
please get your speaking proposals in to us right away. We are looking
forward to seeing everyone again this August, so please
register<http://www.cluecon.com/register?cc12-0604>now, that way we
can make plans for ancillary events like the welcome
reception.

Thanks for being such a great community! Talk to you next week.

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
cc12-0604
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From admin at blindi.net  Mon Jun  4 21:00:23 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Mon, 4 Jun 2012 19:00:23 +0200 (CEST)
Subject: [Freeswitch-users] opensource project voicechat for freeswitch
In-Reply-To: <4FCCD4E1.9050703@endigotech.com>
References: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
	<2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com>
	<1338558318.79806.YahooMailNeo@web120405.mail.ne1.yahoo.com>
	<Pine.LNX.4.64.1206041700001.22234@tomback.antagus.de>
	<4FCCD4E1.9050703@endigotech.com>
Message-ID: <Pine.LNX.4.64.1206041758490.22616@tomback.antagus.de>

Hi Brian,
> What kind of voice chat are we talking about? There are plenty of voice
> chat type things out there (skype for one).

This is a Freeswitch-interface.
This chat can be setup as a sipbased Freeswitch.

Skype is proprietary and not opensource.
You can.t setup skype to a simpli root server on textbase only system.

>Specify your demographic

Sorry, i.m. blind. grafic???? in a voicechat?? I don.t understand!

Skype is not operable on simple text based systems.
the installation of freeswitch on skype is very difficult.


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From anthony.minessale at gmail.com  Mon Jun  4 21:03:39 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Jun 2012 12:03:39 -0500
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <4FCCDACC.5070304@ssimicro.com>
References: <4FC896B8.9070102@digitalmail.com>
	<CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>
	<CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>
	<CAKzWOxUp5h2tMFP9yHJM4LaMmF1f+sXsOfDAcETwzaDG_2WgPA@mail.gmail.com>
	<CALNZuw=5W_k2MtJPxHn6HBZanCZNk83TXiMZTEGjZ9kRjb1G4w@mail.gmail.com>
	<CAKzWOxWH2nM37ywvkyimobkLJqAaUX531w4YgE=3eqVU5q7y=Q@mail.gmail.com>
	<CALNZuwkxN1JXVutNWuAysLgu_kTw40L50L37U-4tZYNV_Yadrg@mail.gmail.com>
	<CAKzWOxVHSGrTqoqTXnF6__1q=_zEFsUzmBhYhb40pEOoVy8gjg@mail.gmail.com>
	<CA+GmxbYZ0fVusbdqDNeLUxwi94aDaK8SjPRU+py-V+JRL6TUqQ@mail.gmail.com>
	<4FCCDACC.5070304@ssimicro.com>
Message-ID: <CAKbxfG_QaKgU3Cw=HgQrYLvEi0dvoypgUfjqt3--S4Bd118Rag@mail.gmail.com>

This thread is all FUD now, please disregard above.


att_xfer and loopback endpoint are both crutches but they work at the
cost of elegance since you are doing emulated behaviors.

Your problem probably comes from the loopback bow-out happening too
soon that tries to cut its way out of the call..

You should be looking at logs and finding the root of the problem not
guessing at things.

My recommendation:

1) Make sure you are on latest GIT we have fixed a few issues in both
in the recent past.
2) try {ignore_early_media=ring_ready,loopback_bowout_on_execute=true}loopback/foo
in your dialstring  --- This may eliminate the loopback right off the
bat.
3) try {ignore_early_media=ring_ready}}loopback/foo ---- it may just
need to wait for the call to be answered.
4) try {loopback_bowout=false}loopback/foo --- the bummer here is it
will never eliminate loopback


On Mon, Jun 4, 2012 at 10:57 AM, Jeremy Childs <jeremyc at ssimicro.com> wrote:
> I'm also very interested in some answers! I've been bitten by att_xfer in
> the past.
>
> Is getting dialplan processing working within the att_xfer function
> possible, and just a low priority for implementation, or is there some
> technical reason why this is not feasible?
>
> Lastly, is this somewhere that execute_extension can help? I've never been
> able to get it to work inside att_xfer.
>
>
>
> On 12-06-04 8:53 AM, Dmitry Sytchev wrote:
>
> I asked this question many times on mailing list, and now I'm sure this
> can't be really done with loopback.
> The only alternative for loopback is to re-inject call into FS via some
> separate Sofia profile, and specify that profile in string for att_xfer.
> This brings up large amount of troubles including DTMF transcoding,
> sequential att_xfer attempt recognition and overall voice/dtmf delay
> introduced by chained channels. Maybe some channels can be moved out of
> scene by using 'simplify' api on correct channels, but this needs tests.
>
> Anyway, loopback channel in FS is completely unusable, so we do need to have
> some ?best practices on how to do things without it in FS wiki... Maybe I
> have time and will describe our experience soon.
>
> 2012/6/4 Michael Collins <msc at freeswitch.org>
>>
>>
>>
>> On Sun, Jun 3, 2012 at 2:15 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>> I know you can do anything in the dialstring. But intended feature is to
>>> allow the user to do an attended transfer to any number that they could
>>> reach via the default calling. The default outbound path already has a LOT
>>> of stuff set up and it would be impossible to duplicate that within a SINGLE
>>> dialstring in a function call.
>>> What is needed is for an att_xfer to be able to have leg C hit the
>>> dialplan and bridged however a "normal" leg B to that number would be
>>> called.
>>> Does this make sense?
>>
>> Perhaps, but I remain unconvinced that this scenario is impossible without
>> loopback. How about the OP actually supply a sample Lua script and dialplan
>> and call log? I'd be willing to wager that the gurus could come up with a
>> non-evil alternative that actually works. Just because loopback seems like a
>> clean solution doesn't necessarily mean that it is. I'll leave it to Anthony
>> to give the technical reasons why loopback doesn't always work as one would
>> expect or why it should be avoided wherever possible.
>>
>> -MC
>>
>>>
>>> -Avi
>>>
>>>
>>>
>>> On Mon, Jun 4, 2012 at 12:02 AM, Michael Collins <msc at freeswitch.org>
>>> wrote:
>>>>
>>>> Au contraire mon frere!
>>>>
>>>> You can do multiple things in a dialstring, like setting channel
>>>> variables. You can also use execute_on_ring/media/answer to execute the
>>>> extension with doing all the loopback overhead.
>>>>
>>>> I propose an experiment: provide a dialplan and loopback dialstring and
>>>> we'll see if we can't give you a non-loopbackish alternative.
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Sun, Jun 3, 2012 at 1:54 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>>>
>>>>> ... all the normal dialplan handling. Setting CID, options, LCR stuff,
>>>>> billing controls.
>>>>> -Avi
>>>>>
>>>>>
>>>>> On Sun, Jun 3, 2012 at 11:40 PM, Michael Collins <msc at freeswitch.org>
>>>>> wrote:
>>>>>>
>>>>>> Let me rephrase...
>>>>>>
>>>>>> Since loopback is generally evil and should be avoided wherever
>>>>>> possible, what does loopback give you that you can't get from doing a normal
>>>>>> dialstring?
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> On Sat, Jun 2, 2012 at 11:13 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>>>>>>
>>>>>>> ... because att_xfer seems to require a "sofia/$profile/$destination"
>>>>>>> directive, and he just wants the call to hit the dialplan.
>>>>>>>
>>>>>>> -Avi
>>>>>>>
>>>>>>>
>>>>>>> On Fri, Jun 1, 2012 at 8:13 PM, Michael Collins <msc at freeswitch.org>
>>>>>>> wrote:
>>>>>>>>
>>>>>>>> Why do you need to use loopback at all?
>>>>>>>> -MC
>>>>>>>>
>>>>>>>>
>>>>>>>> On Fri, Jun 1, 2012 at 3:17 AM, Alex Lake <alex at digitalmail.com>
>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>> Got a lua script for a B-party "mid-call menu". Is it legitimate to
>>>>>>>>> do..
>>>>>>>>> "session:execute("att_xfer", "loopback/"..destnum)"
>>>>>>>>>
>>>>>>>>> I've tried it and it seems to start off doing the right things, but
>>>>>>>>> my
>>>>>>>>> A-party gets disconnected as soon as the call to the C-Party (the
>>>>>>>>> person
>>>>>>>>> I'm transferring the call to) answers the call.
>>>>>>>>>
>>>>>>>>> Maybe better to try to orchestrate the entire affair from within
>>>>>>>>> the lua
>>>>>>>>> script? (Tricky for a beginner like me!)
>>>>>>>>>
>>>>>>>>> Thanks,
>>>>>>>>> Alex
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>
>>>>>>
>>>>
>>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From freeswitch at scottisheyes.com  Mon Jun  4 21:34:49 2012
From: freeswitch at scottisheyes.com (James)
Date: Mon, 4 Jun 2012 10:34:49 -0700
Subject: [Freeswitch-users] opensource project voicechat for freeswitch
In-Reply-To: <Pine.LNX.4.64.1206041758490.22616@tomback.antagus.de>
References: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
	<2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com>
	<1338558318.79806.YahooMailNeo@web120405.mail.ne1.yahoo.com>
	<Pine.LNX.4.64.1206041700001.22234@tomback.antagus.de>
	<4FCCD4E1.9050703@endigotech.com>
	<Pine.LNX.4.64.1206041758490.22616@tomback.antagus.de>
Message-ID: <CALfx8k3mHtDs-EC_m9gD_JqVQFde=7ynsgPhLTfqBTYAZ_TM5g@mail.gmail.com>

baresip is a console-based open source SIP client.

http://www.creytiv.com/baresip.html

The documentation could be better, but it's text-based (assume that running
in the console or terminal is what you meant by text-based).

On Mon, Jun 4, 2012 at 10:00 AM, Thomas Hoellriegel <admin at blindi.net>wrote:

> Hi Brian,
>
>  What kind of voice chat are we talking about? There are plenty of voice
>> chat type things out there (skype for one).
>>
>
> This is a Freeswitch-interface.
> This chat can be setup as a sipbased Freeswitch.
>
> Skype is proprietary and not opensource.
> You can.t setup skype to a simpli root server on textbase only system.
>
>  Specify your demographic
>>
>
> Sorry, i.m. blind. grafic???? in a voicechat?? I don.t understand!
>
> Skype is not operable on simple text based systems.
> the installation of freeswitch on skype is very difficult.
>
>
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/**listinfo/blinde-misc<http://www.blindi.net/mailman/listinfo/blinde-misc>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From josefu at gmail.com  Mon Jun  4 21:45:46 2012
From: josefu at gmail.com (=?ISO-8859-1?Q?Jose_Fco=2E_Irles_Dur=E1?=)
Date: Mon, 4 Jun 2012 19:45:46 +0200
Subject: [Freeswitch-users] limit app problem with loopback channels
In-Reply-To: <CAKzWOxX6+s3b3hvfRq_FwttruoqQKNQjht=9_oYG30t-JqXCww@mail.gmail.com>
References: <CAKzWOxUb_mKTsL7g8W0c6Hq4ss=V=i+GBr2ThJna25Jemzm_rw@mail.gmail.com>
	<20120603234751.68a5e203@mail.tritonwest.net>
	<CAKzWOxVVqCiRJMna8D8bAjtmJ=621Uk6HEtXH22OYK5JVXL=zQ@mail.gmail.com>
	<CAJOPU6X4dPV1an6g6Cy7w7bMvaZo2qAihVAGxzxQZs6s0i1v_A@mail.gmail.com>
	<CAKzWOxX6+s3b3hvfRq_FwttruoqQKNQjht=9_oYG30t-JqXCww@mail.gmail.com>
Message-ID: <CAJOPU6WkkUwn3h2yOfYGeLdCH7HxEyb=r0fBM3DN9mqnT+UQAg@mail.gmail.com>

> Hmm, not entirely sure why it's doing that. It looks like the limit is being
> called twice for each dialstring. Just for kicks, can you change
> execute_on_originate to execute_on_media? If that doesn't work, try
> execute_on_ring as well. I'm just hoping to get some empirical data on what
> happens in these scenarios. Once we iron it out we'll get it all wikified.

With execute_on_ring works, with execute_on_media or
execute_on_originate doesn't work.

I also tried another approach with a lua script that starts with
freeswitch, receive the "CHANNEL_CALLSTATE" events,
increment/drecrement my counters and insert the uuid of the calls that
i need for pickup/spy later.


-- 
Jose Fco. Irles Dur?


From babak.freeswitch at gmail.com  Mon Jun  4 21:54:41 2012
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Mon, 4 Jun 2012 22:24:41 +0430
Subject: [Freeswitch-users] help with lua and callcenter
Message-ID: <CAEVfcabeKTdbaVks+VMupvGFF_F+X9AS4Tjut5ihsSoh3PF1Eg@mail.gmail.com>

Hi
In the following lua function, I want to invoke VMX(another function) after
execution of callcenter is completed. but it seems when  agent answers the
call from callcenter, execution continues from the line after
"session:execute("callcenter","tehccq");", I mean it is not waiting till
the dialog between agent and member ends. Is it normal (or I'm doing
something wrong)? if it is normal is there anyway I can get this working
the way I need?
thanx

function DialCallCenter()
setLastState('CALLCENTER');
session:execute('answer');
current_call_add(customerId,callerNumber,firstname,lastname);
session:execute("callcenter","tehccq");
local agent_found = session:getVariable('cc_agent_found');
freeswitch.consoleLog("info","hellophone:cc agent"..(agent_found or ""));
if agent_found == nil or agent_found ~= 'true' then
VMX('no-answer-leave-a-message');
end
return true;
end
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From bdfoster at endigotech.com  Mon Jun  4 22:21:27 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 4 Jun 2012 14:21:27 -0400
Subject: [Freeswitch-users] opensource project voicechat for freeswitch
In-Reply-To: <Pine.LNX.4.64.1206041758490.22616@tomback.antagus.de>
References: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
	<2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com>
	<1338558318.79806.YahooMailNeo@web120405.mail.ne1.yahoo.com>
	<Pine.LNX.4.64.1206041700001.22234@tomback.antagus.de>
	<4FCCD4E1.9050703@endigotech.com>
	<Pine.LNX.4.64.1206041758490.22616@tomback.antagus.de>
Message-ID: <CAKV-jFzHrdyJ8oOYtfgCTZGk==RJyfNbHgtx6Aq537uYgcdZ_w@mail.gmail.com>

I don't understand what you are trying to accomplish. You said you were
blind, which is why it should be text based, but what exactly are you
trying to accomplish? I downloaded your files but that doesn't tell me much.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 4, 2012 1:01 PM, "Thomas Hoellriegel" <admin at blindi.net> wrote:

> Hi Brian,
>
>> What kind of voice chat are we talking about? There are plenty of voice
>> chat type things out there (skype for one).
>>
>
> This is a Freeswitch-interface.
> This chat can be setup as a sipbased Freeswitch.
>
> Skype is proprietary and not opensource.
> You can.t setup skype to a simpli root server on textbase only system.
>
>  Specify your demographic
>>
>
> Sorry, i.m. blind. grafic???? in a voicechat?? I don.t understand!
>
> Skype is not operable on simple text based systems.
> the installation of freeswitch on skype is very difficult.
>
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/**listinfo/blinde-misc<http://www.blindi.net/mailman/listinfo/blinde-misc>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vvenkatar at gmail.com  Mon Jun  4 22:42:21 2012
From: vvenkatar at gmail.com (Venkatesh)
Date: Mon, 4 Jun 2012 11:42:21 -0700
Subject: [Freeswitch-users] Setting custom SIP headers on transferring call
	leg.
Message-ID: <CAP_69mvXXp7GEZmkH580yWsre_otoKU8h_QH_aZCjwpViHXBTQ@mail.gmail.com>

Hi !

Apologies if this question has already been answered. I am developing a
simple IVR application using JS. The application simply answers the call;
does a bunch of stuff. One of the things it does is to transfer the call to
an extension. I want to be able to set some custom SIP headers on the
"outbound" call leg. I tried using
session.setVariable("variable_sip_h_X-My-Header", "my_value"), but don't
see the header in the outbound invite. I suspect that this is acting on the
session object of the incoming call and not on the outbound. Wanted to know
if freeswitch provides this capability and if yes, the semantics for the
same when used via JS. Any help would be great.

Regards,
Venkatesh
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From modesto at isimples.com.br  Mon Jun  4 23:05:24 2012
From: modesto at isimples.com.br (Antonio Modesto)
Date: Mon, 04 Jun 2012 16:05:24 -0300
Subject: [Freeswitch-users] Doubts with att_xfer
Message-ID: <1338836724.3534.12.camel@modesto.localdomain.net>

Hi,


I am trying to enable att_xfer for calls that come from pstn, these call
are being routed to an internal registered SIP Phone. In asterisk I had
those options "Tt" in the Dial application to enable transfer to the
caller party and to the calling party. The question is that for internal
call I want both legs be able to transfer to whoever they want, but for
external calls I want just the internal leg be able to transfer calls.
How can I do that?


Regards.



From msc at freeswitch.org  Mon Jun  4 23:05:38 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Jun 2012 12:05:38 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call June 6
Message-ID: <CAKzWOxUcaUug0qZEGqebddaFgFDoE379T6OzfOU-4XNPHfjk8Q@mail.gmail.com>

Hello all!

Just an update: We've rescheduled Darren's SIP 101 discussion for June
20th. Also, this Wednesday, June 6th yours truly will be doing an
introduction to mod_httapi! We will be keeping it simple this week: just
getting your server up and running, trying out the sample code, and maybe
we'll do some what-if's from the audience. The agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_06

Keep in mind that mod_httapi allows for *your* FreeSWITCH server to call *my
* web server to get dialplan instructions, so be sure to compile and enable
mod_httapi <http://wiki.freeswitch.org/wiki/Mod_httapi> on your systems.

Talk to you Wednesday!

-Michael
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From msc at freeswitch.org  Mon Jun  4 23:27:28 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Jun 2012 12:27:28 -0700
Subject: [Freeswitch-users] How to play ivr on background?
In-Reply-To: <CAMbKPxaQ5tSMEtVSxL4KOOygMAgxbsSaSGQ=2GahHDZkH+4spQ@mail.gmail.com>
References: <71E43EA5-B5C3-499C-8B5A-73A682398FA8@gmail.com>
	<CAKzWOxVfqhbYi30LkVknqVtfwQhjvRv=KoOvZf0n9VtggkF1eg@mail.gmail.com>
	<0AB3DD47-BA77-49B5-B6DC-516812415001@gmail.com>
	<CAKzWOxVsD8vXiCfd58GDpLS+UFrJ8QazAcvHc_PCQdD9HPLoOQ@mail.gmail.com>
	<CAMbKPxaQ5tSMEtVSxL4KOOygMAgxbsSaSGQ=2GahHDZkH+4spQ@mail.gmail.com>
Message-ID: <CAKzWOxVUp6oz3PBMgcuT_FtVaJPMLD90OBV70pe1ZSaf+5__fg@mail.gmail.com>

Try setting ringback variable before executing the ring_ready app.
-MC

On Mon, Jun 4, 2012 at 3:13 AM, M.Emran <monemran at gmail.com> wrote:

>
> IVR means to play a native sound file i.e. "You have 10 minutes"
>
> I tried to set ringback data on dialplan. i got some noise or some tone.
> here is my settings.
>
> <action application="set" data="continue_on_fail=true"/>
> <action application="set" data="call_timeout=30"/>
> <action application="set" data="session_in_hangup_hook=true"/>
> <action application="set" data="bypass_media=false"/>
> <action application="set" data="ignore_early_media=true"/>
> <action application="ring_ready"/>
> <action application="set" data="ringback=voicemail/vm-you_have.G729"/>
>
> Am i doing any wrong ?
>
> On Mon, Jun 4, 2012 at 5:54 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> You're getting into a tricky area here. You want to ignore early media,
>> yet you want your clients to hear original ring tone of the B leg after you
>> play an "IVR". (I honestly don't know what you mean by "IVR" here, but I'm
>> assuming you just mean some sort of recorded message.)
>>
>> As far as using "say" or anything else like that you need to read up on
>> phrase macros. I recommend starting here:
>>
>> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management#Phrases_Section_Primer
>>
>> Also, chapter 6 of the FS book has some good information on this subject.
>> It recommends that you look at sounds.xml in conf/lang/en/vm/ to get an
>> idea of some of the things you can do with phrase macros.
>>
>> -MC
>>
>>
>> On Sun, Jun 3, 2012 at 2:17 PM, Mohammad Emran <monemran at gmail.com>wrote:
>>
>>> Thnkx for ur faster redponse.
>>>
>>> Yes.i wanna ignore early media.
>>>
>>> Actually my aim is to save time of playing ivr n when finished ivr
>>> clients will hear original ringtone of leg-b.
>>>
>>> On the ringback, how i can call SAY command to announce minutes?
>>>
>>> It would helpfull if i get an example.
>>>
>>> Sent from my iPhone
>>>
>>> On Jun 4, 2012, at 3:08 AM, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>> Are you ignoring early media from the target leg? If so you can just set
>>> the ringback variable to be the announcement (like
>>> ivr/ivr-please_hold_while_your_party_is_being_reached.wav) and then
>>> whatever ringing/music/etc. that you want them to hear while the call is
>>> being connected.
>>>
>>> You might want to use this !-separated list of sound files method to
>>> play multiple sounds in the ringback:
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_file_string
>>>
>>> -MC
>>>
>>> On Sun, Jun 3, 2012 at 2:00 PM, Mohammad Emran <monemran at gmail.com>wrote:
>>>
>>>> Hi
>>>>
>>>> I would like play minute announce ivr during  the process of call.
>>>> Cause it will reduce connect time  on client side.
>>>>
>>>> Pls advise how to do that.
>>>>
>>>>
>>
>>
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From admin at blindi.net  Mon Jun  4 23:33:26 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Mon, 4 Jun 2012 21:33:26 +0200 (CEST)
Subject: [Freeswitch-users] opensource project voicechat for freeswitch
In-Reply-To: <CAKV-jFzHrdyJ8oOYtfgCTZGk==RJyfNbHgtx6Aq537uYgcdZ_w@mail.gmail.com>
References: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
	<2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com>
	<1338558318.79806.YahooMailNeo@web120405.mail.ne1.yahoo.com>
	<Pine.LNX.4.64.1206041700001.22234@tomback.antagus.de>
	<4FCCD4E1.9050703@endigotech.com>
	<Pine.LNX.4.64.1206041758490.22616@tomback.antagus.de>
	<CAKV-jFzHrdyJ8oOYtfgCTZGk==RJyfNbHgtx6Aq537uYgcdZ_w@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1206042109160.23485@tomback.antagus.de>


Hi Brain,

> I don't understand what you are trying to accomplish.
>You said you were
> blind, which is why it should be text based, but what exactly are you
> trying to accomplish?

>ou meant something of a graphics demo.
I can.t look any grafics.

Mmy intention is:
I like to create a Unified Messaging chat.
Since there are many blind people, they rely on language.
  One can make with the help of these forums people information easily
accessible.
  Since the functions I have a very well thought out,    you could also use 
this system well in the bussiness areas.


I downloaded your files but that doesn't tell me much.

I means i have a idea, but i.m. not a programmer.
I have setup the database, and i documented the features to progamm a 
professional voicechat for free.

Sip is open and very flexible.

>

---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From msc at freeswitch.org  Mon Jun  4 23:54:02 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Jun 2012 12:54:02 -0700
Subject: [Freeswitch-users] Music on hold while I add someone to the
	conference
In-Reply-To: <1338803698.17011.YahooMailNeo@web132306.mail.ird.yahoo.com>
References: <1338803698.17011.YahooMailNeo@web132306.mail.ird.yahoo.com>
Message-ID: <CAKzWOxVszUZJKHJE4ej4HgppoMFN20Mg8Ua5FyC6K11zgyqz8g@mail.gmail.com>

Chris,

Wow, that's a really well-written wiki page! My hat's off to whomever wrote
all that. :)

Okay, my guess is that you just need to have the "wait-mod" flag set when
sending people into the conference. I'd modify both occurrences of the
conference app to add the flag. In the "normal" extension ("46\d\d" in the
wiki example) do this:

<action application="conference" data="$1 at simple+flags{wait-mod}"/>

In the "moderator" extension ("*46\d\d" in the wiki example) do this:

<action application="conference" data="$1 at simple+flags{wait-mod|moderator}"/>


In my wiki example I didn't actually make the person who dials *46xx an
actual "moderator" per se; I simply gave them the BDA controls. I'll update
the wiki to make *46xx set the moderator flag.

-MC

On Mon, Jun 4, 2012 at 2:54 AM, Chris B. Ware <chrisbware at yahoo.it> wrote:

> Hi,
>
> I'm testing example reported here:
>
> http://wiki.freeswitch.org/wiki/Conference_Add_Call_Example
>
>
> I'd like to modify extension "Add new OB call to conference" in order to
> play music on hold to the other conference members , while moderator digit
> a new member number to add. In my case we have a conference with A
> (moderator) and B (first member) and A try to add C. That's my conf:
>
>  <extension name="Add new OB call to conference">
>       <condition field="destination_number" expression="^ADD_THIRD$">
>         <action application="set" data="res=${bgapi uuid_hold ${uuid_A} }"
> />
>          <!-- ask caller for a number + #, collect into ${target_num}
> variable -->
>         <action application="play_and_get_digits" data="4 11 1 5000 #
> ivr/ivr-enter_destination_telephone_number.wav
> ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/>
>        <action application="set" data="res=${bgapi uuid_hold off
> ${uuid_A} }" />
>         <!-- add this call to the conference -->
>         <action application="execute_extension"
> data="ADD_CALL_TO_CONF_${target_num}"/>
>       </condition>
>     </extension>
>
>
> Everything works except B can't hear any music. Where's the error?
>
>
> Thank you
>
>
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From msc at freeswitch.org  Tue Jun  5 00:01:43 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Jun 2012 13:01:43 -0700
Subject: [Freeswitch-users] opensource project voicechat for freeswitch
In-Reply-To: <Pine.LNX.4.64.1206042109160.23485@tomback.antagus.de>
References: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
	<2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com>
	<1338558318.79806.YahooMailNeo@web120405.mail.ne1.yahoo.com>
	<Pine.LNX.4.64.1206041700001.22234@tomback.antagus.de>
	<4FCCD4E1.9050703@endigotech.com>
	<Pine.LNX.4.64.1206041758490.22616@tomback.antagus.de>
	<CAKV-jFzHrdyJ8oOYtfgCTZGk==RJyfNbHgtx6Aq537uYgcdZ_w@mail.gmail.com>
	<Pine.LNX.4.64.1206042109160.23485@tomback.antagus.de>
Message-ID: <CAKzWOxVjT8m5Ri6P8K1qi0xXb8K95RAmzMdhYVXwKRMa9ZxU7Q@mail.gmail.com>

Hi Thomas,

I appreciate your effort in communicating here and in trying to make
something the helps other blind and visually-impaired users.

When Brian Foster asked about "demographics" he was using an English word
that really means "target audience" or something like that. (
www.freetranslation.com translates it into German as "Demografisch" if that
helps.) What he was asking is, "Who would use the software that you are
wanting to build?" You've already answered part of that question: people
who use screen readers and need a text-based setup.

I believe another person mentioned Baresip in this email thread. If you
haven't checked that out you might want to. It could give you some ideas on
how to proceed.

Thanks,
Michael


On Mon, Jun 4, 2012 at 12:33 PM, Thomas Hoellriegel <admin at blindi.net>wrote:

>
> Hi Brain,
>
>  I don't understand what you are trying to accomplish.
>> You said you were
>> blind, which is why it should be text based, but what exactly are you
>> trying to accomplish?
>>
>
>  ou meant something of a graphics demo.
>>
> I can.t look any grafics.
>
> Mmy intention is:
> I like to create a Unified Messaging chat.
> Since there are many blind people, they rely on language.
>  One can make with the help of these forums people information easily
> accessible.
>  Since the functions I have a very well thought out,    you could also use
> this system well in the bussiness areas.
>
>
> I downloaded your files but that doesn't tell me much.
>
> I means i have a idea, but i.m. not a programmer.
> I have setup the database, and i documented the features to progamm a
> professional voicechat for free.
>
> Sip is open and very flexible.
>
>
>>
>
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From monemran at gmail.com  Tue Jun  5 00:12:57 2012
From: monemran at gmail.com (Mohammad Emran)
Date: Tue, 5 Jun 2012 02:12:57 +0600
Subject: [Freeswitch-users] How to play ivr on background?
In-Reply-To: <CAKzWOxVUp6oz3PBMgcuT_FtVaJPMLD90OBV70pe1ZSaf+5__fg@mail.gmail.com>
References: <71E43EA5-B5C3-499C-8B5A-73A682398FA8@gmail.com>
	<CAKzWOxVfqhbYi30LkVknqVtfwQhjvRv=KoOvZf0n9VtggkF1eg@mail.gmail.com>
	<0AB3DD47-BA77-49B5-B6DC-516812415001@gmail.com>
	<CAKzWOxVsD8vXiCfd58GDpLS+UFrJ8QazAcvHc_PCQdD9HPLoOQ@mail.gmail.com>
	<CAMbKPxaQ5tSMEtVSxL4KOOygMAgxbsSaSGQ=2GahHDZkH+4spQ@mail.gmail.com>
	<CAKzWOxVUp6oz3PBMgcuT_FtVaJPMLD90OBV70pe1ZSaf+5__fg@mail.gmail.com>
Message-ID: <D74F51C7-A604-4115-8485-0676628CA31A@gmail.com>

It works...great! 

Freeswitch rocks!!!!

It seems sound file played faster.is there any variable to play file slowly???



Sent from my iPad

On Jun 5, 2012, at 1:27 AM, Michael Collins <msc at freeswitch.org> wrote:

> Try setting ringback variable before executing the ring_ready app.
> -MC
> 
> On Mon, Jun 4, 2012 at 3:13 AM, M.Emran <monemran at gmail.com> wrote:
> 
> IVR means to play a native sound file i.e. "You have 10 minutes" 
> 
> I tried to set ringback data on dialplan. i got some noise or some tone. here is my settings.
> 
> <action application="set" data="continue_on_fail=true"/>
> <action application="set" data="call_timeout=30"/>
> <action application="set" data="session_in_hangup_hook=true"/>
> <action application="set" data="bypass_media=false"/>
> <action application="set" data="ignore_early_media=true"/>
> <action application="ring_ready"/>
> <action application="set" data="ringback=voicemail/vm-you_have.G729"/>
> 
> Am i doing any wrong ?
> 
> On Mon, Jun 4, 2012 at 5:54 AM, Michael Collins <msc at freeswitch.org> wrote:
> You're getting into a tricky area here. You want to ignore early media, yet you want your clients to hear original ring tone of the B leg after you play an "IVR". (I honestly don't know what you mean by "IVR" here, but I'm assuming you just mean some sort of recorded message.)
> 
> As far as using "say" or anything else like that you need to read up on phrase macros. I recommend starting here:
> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management#Phrases_Section_Primer
> 
> Also, chapter 6 of the FS book has some good information on this subject. It recommends that you look at sounds.xml in conf/lang/en/vm/ to get an idea of some of the things you can do with phrase macros.
> 
> -MC
> 
> 
> On Sun, Jun 3, 2012 at 2:17 PM, Mohammad Emran <monemran at gmail.com> wrote:
> Thnkx for ur faster redponse.
> 
> Yes.i wanna ignore early media.
> 
> Actually my aim is to save time of playing ivr n when finished ivr clients will hear original ringtone of leg-b.
> 
> On the ringback, how i can call SAY command to announce minutes?
> 
> It would helpfull if i get an example.
> 
> Sent from my iPhone
> 
> On Jun 4, 2012, at 3:08 AM, Michael Collins <msc at freeswitch.org> wrote:
> 
>> Are you ignoring early media from the target leg? If so you can just set the ringback variable to be the announcement (like ivr/ivr-please_hold_while_your_party_is_being_reached.wav) and then whatever ringing/music/etc. that you want them to hear while the call is being connected.
>> 
>> You might want to use this !-separated list of sound files method to play multiple sounds in the ringback:
>> 
>> http://wiki.freeswitch.org/wiki/Mod_file_string
>> 
>> -MC
>> 
>> On Sun, Jun 3, 2012 at 2:00 PM, Mohammad Emran <monemran at gmail.com> wrote:
>> Hi
>> 
>> I would like play minute announce ivr during  the process of call. Cause it will reduce connect time  on client side.
>> 
>> Pls advise how to do that.
>> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun  5 00:19:46 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Jun 2012 13:19:46 -0700
Subject: [Freeswitch-users] How to play ivr on background?
In-Reply-To: <D74F51C7-A604-4115-8485-0676628CA31A@gmail.com>
References: <71E43EA5-B5C3-499C-8B5A-73A682398FA8@gmail.com>
	<CAKzWOxVfqhbYi30LkVknqVtfwQhjvRv=KoOvZf0n9VtggkF1eg@mail.gmail.com>
	<0AB3DD47-BA77-49B5-B6DC-516812415001@gmail.com>
	<CAKzWOxVsD8vXiCfd58GDpLS+UFrJ8QazAcvHc_PCQdD9HPLoOQ@mail.gmail.com>
	<CAMbKPxaQ5tSMEtVSxL4KOOygMAgxbsSaSGQ=2GahHDZkH+4spQ@mail.gmail.com>
	<CAKzWOxVUp6oz3PBMgcuT_FtVaJPMLD90OBV70pe1ZSaf+5__fg@mail.gmail.com>
	<D74F51C7-A604-4115-8485-0676628CA31A@gmail.com>
Message-ID: <CAKzWOxUQ_c=cpFxpw9R9qJTwuj8dN_pB6NMZ1TVEEkzcQh6QAg@mail.gmail.com>

That's a totally different question. I would check the file encoding, try
different files, use sox to modify the source file, etc.

In any case, I'm glad you got it working.

-MC

On Mon, Jun 4, 2012 at 1:12 PM, Mohammad Emran <monemran at gmail.com> wrote:

> It works...great!
>
> Freeswitch rocks!!!!
>
> It seems sound file played faster.is there any variable to play file
> slowly???
>
>
>
> Sent from my iPad
>
> On Jun 5, 2012, at 1:27 AM, Michael Collins <msc at freeswitch.org> wrote:
>
> Try setting ringback variable before executing the ring_ready app.
> -MC
>
> On Mon, Jun 4, 2012 at 3:13 AM, M.Emran < <monemran at gmail.com>
> monemran at gmail.com> wrote:
>
>>
>> IVR means to play a native sound file i.e. "You have 10 minutes"
>>
>> I tried to set ringback data on dialplan. i got some noise or some tone.
>> here is my settings.
>>
>> <action application="set" data="continue_on_fail=true"/>
>> <action application="set" data="call_timeout=30"/>
>> <action application="set" data="session_in_hangup_hook=true"/>
>> <action application="set" data="bypass_media=false"/>
>> <action application="set" data="ignore_early_media=true"/>
>> <action application="ring_ready"/>
>> <action application="set" data="ringback=voicemail/vm-you_have.G729"/>
>>
>> Am i doing any wrong ?
>>
>> On Mon, Jun 4, 2012 at 5:54 AM, Michael Collins < <msc at freeswitch.org>
>> msc at freeswitch.org> wrote:
>>
>>> You're getting into a tricky area here. You want to ignore early media,
>>> yet you want your clients to hear original ring tone of the B leg after you
>>> play an "IVR". (I honestly don't know what you mean by "IVR" here, but I'm
>>> assuming you just mean some sort of recorded message.)
>>>
>>> As far as using "say" or anything else like that you need to read up on
>>> phrase macros. I recommend starting here:
>>>
>>> <http://wiki.freeswitch.org/wiki/Speech_Phrase_Management#Phrases_Section_Primer>
>>> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management#Phrases_Section_Primer
>>>
>>> Also, chapter 6 of the FS book has some good information on this
>>> subject. It recommends that you look at sounds.xml in conf/lang/en/vm/ to
>>> get an idea of some of the things you can do with phrase macros.
>>>
>>> -MC
>>>
>>>
>>> On Sun, Jun 3, 2012 at 2:17 PM, Mohammad Emran < <monemran at gmail.com>
>>> monemran at gmail.com> wrote:
>>>
>>>> Thnkx for ur faster redponse.
>>>>
>>>> Yes.i wanna ignore early media.
>>>>
>>>> Actually my aim is to save time of playing ivr n when finished ivr
>>>> clients will hear original ringtone of leg-b.
>>>>
>>>> On the ringback, how i can call SAY command to announce minutes?
>>>>
>>>> It would helpfull if i get an example.
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Jun 4, 2012, at 3:08 AM, Michael Collins < <msc at freeswitch.org>
>>>> msc at freeswitch.org> wrote:
>>>>
>>>> Are you ignoring early media from the target leg? If so you can just
>>>> set the ringback variable to be the announcement (like
>>>> ivr/ivr-please_hold_while_your_party_is_being_reached.wav) and then
>>>> whatever ringing/music/etc. that you want them to hear while the call is
>>>> being connected.
>>>>
>>>> You might want to use this !-separated list of sound files method to
>>>> play multiple sounds in the ringback:
>>>>
>>>> <http://wiki.freeswitch.org/wiki/Mod_file_string>
>>>> http://wiki.freeswitch.org/wiki/Mod_file_string
>>>>
>>>> -MC
>>>>
>>>> On Sun, Jun 3, 2012 at 2:00 PM, Mohammad Emran < <monemran at gmail.com>
>>>> monemran at gmail.com> wrote:
>>>>
>>>>> Hi
>>>>>
>>>>> I would like play minute announce ivr during  the process of call.
>>>>> Cause it will reduce connect time  on client side.
>>>>>
>>>>> Pls advise how to do that.
>>>>>
>>>>>
>>>
>>>
>
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From steveayre at gmail.com  Tue Jun  5 03:09:15 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 5 Jun 2012 00:09:15 +0100
Subject: [Freeswitch-users] Setting custom SIP headers on transferring
	call leg.
In-Reply-To: <CAP_69mvXXp7GEZmkH580yWsre_otoKU8h_QH_aZCjwpViHXBTQ@mail.gmail.com>
References: <CAP_69mvXXp7GEZmkH580yWsre_otoKU8h_QH_aZCjwpViHXBTQ@mail.gmail.com>
Message-ID: <F2EDDFBD-68BD-4C1E-906E-3C4696F69ADA@gmail.com>

The variable should be named sip_h_X-My-Header

The variable_ prefix you see in the info app output is to differentiate variables from caller profile fields, but is not part of the variable name.

If that fails to work, try using the export app (see the wiki)

Sent from my iPad



On 4 Jun 2012, at 19:42, Venkatesh <vvenkatar at gmail.com> wrote:

> Hi !
> 
> Apologies if this question has already been answered. I am developing a simple IVR application using JS. The application simply answers the call; does a bunch of stuff. One of the things it does is to transfer the call to an extension. I want to be able to set some custom SIP headers on the "outbound" call leg. I tried using session.setVariable("variable_sip_h_X-My-Header", "my_value"), but don't see the header in the outbound invite. I suspect that this is acting on the session object of the incoming call and not on the outbound. Wanted to know if freeswitch provides this capability and if yes, the semantics for the same when used via JS. Any help would be great.
> 
> Regards,
> Venkatesh
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From vvenkatar at gmail.com  Tue Jun  5 03:16:48 2012
From: vvenkatar at gmail.com (Venkatesh)
Date: Mon, 4 Jun 2012 16:16:48 -0700
Subject: [Freeswitch-users] Setting custom SIP headers on transferring
 call leg.
In-Reply-To: <F2EDDFBD-68BD-4C1E-906E-3C4696F69ADA@gmail.com>
References: <CAP_69mvXXp7GEZmkH580yWsre_otoKU8h_QH_aZCjwpViHXBTQ@mail.gmail.com>
	<F2EDDFBD-68BD-4C1E-906E-3C4696F69ADA@gmail.com>
Message-ID: <CAP_69mtD0A6Fgxeido6KDG-+eEgk1sNYaefV-BFNB8H=sqS+=A@mail.gmail.com>

Thank you very much for your response. I was able to get this working
correctly after fixing this in my script....

Venkatesh

On Mon, Jun 4, 2012 at 4:09 PM, Steven Ayre <steveayre at gmail.com> wrote:

> The variable should be named sip_h_X-My-Header
>
> The variable_ prefix you see in the info app output is to differentiate
> variables from caller profile fields, but is not part of the variable name.
>
> If that fails to work, try using the export app (see the wiki)
>
> Sent from my iPad
>
>
>
> On 4 Jun 2012, at 19:42, Venkatesh <vvenkatar at gmail.com> wrote:
>
> > Hi !
> >
> > Apologies if this question has already been answered. I am developing a
> simple IVR application using JS. The application simply answers the call;
> does a bunch of stuff. One of the things it does is to transfer the call to
> an extension. I want to be able to set some custom SIP headers on the
> "outbound" call leg. I tried using
> session.setVariable("variable_sip_h_X-My-Header", "my_value"), but don't
> see the header in the outbound invite. I suspect that this is acting on the
> session object of the incoming call and not on the outbound. Wanted to know
> if freeswitch provides this capability and if yes, the semantics for the
> same when used via JS. Any help would be great.
> >
> > Regards,
> > Venkatesh
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gamar at center.com  Tue Jun  5 03:18:03 2012
From: gamar at center.com (Gilbert Amar)
Date: Mon, 4 Jun 2012 16:18:03 -0700
Subject: [Freeswitch-users] How do you split bridged calls ?
In-Reply-To: <CAKzWOxWP6dmRBV-1SJhyS3V9+GqFd9zuaevfxOGM0N00YVsCJQ@mail.gmail.com>
References: <CALNZuwkRLZ0avq-xaPLmrnRQYUUGHd4WDFKCaqA+H2N_rDTGKw@mail.gmail.com>	<20120523163913.b07558d1@mail.tritonwest.net>	<CAKzWOxXsOuDfcwttJEwjUEoaaYEcn8bb3pWkeE+z3pb96co-TQ@mail.gmail.com>	<002701cd3f7a$f3dd50a0$db97f1e0$@center.com>
	<CAKzWOxWP6dmRBV-1SJhyS3V9+GqFd9zuaevfxOGM0N00YVsCJQ@mail.gmail.com>
Message-ID: <000301cd42a8$4a82fdd0$df88f970$@center.com>

Hello 

 

After several mistakes I find out that if you want to break a bridge, from a
socket application, and keep managing the two legs in your application the
solution is:

 

uuid_transfer <uuid> -both park inline //the inline does the magic here.

 

then you can play phrases,get dtmf... and even re-bridge the two legs.

 

Thanks to those who point me to the right direction.

 

Gilbert

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, May 31, 2012 11:11 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How do you split bridged calls ?

 

I dropped your log file into our pastebin here:
http://pastebin.freeswitch.org/19209

The log shows that the issue is not that FS is only trying to process the
first leg, which can be seen by looking at lines 507 and 714. The problem
here is that your inline dialplan is not being parsed properly. What's
happening is that uuid_dual_transfer is literally trying to send the A leg
to destination number "m:^:phrase:queue_position,5^ inline" and the B leg to
destination number "m:^:phrase:queue_position,3^ inline". 

What are you trying to do with each leg of the call? I see that you are
trying to announce the queue position, but what do you want to have happen
after that? As it stands your call legs would probably just end after the
announcement because there's nothing more to do.

-MC

On Thu, May 31, 2012 at 3:15 PM, Gilbert Amar <gamar at center.com> wrote:

Hello,

 

I keep trying to split a bridged call.

Using playback wav I have audio on both leg, but in my IVR I will need
different message to be played to each leg.

I do not know if this is a bug or not:

I have audio only on one leg using uuid_transfer or uuid_dual_transfer for
phrase.

 

Test 1

         api uuid_transfer <uuid> -both 'm:^:phrase:queue_position,5^
inline'

 

Test 2:

         api uuid_dual_transfer <uuid> 'm:^:phrase:queue_position,5^ inline'
'm:^:phrase:queue_position,3^ inline'

 

notice queue_position 5 for the first and 3 for the second.

 

 

Please find attach a FS log showing for the second test that FS is only
looking to process the first leg.

 

Gilbert

 

 

Details regarding the application and log.

 

my dialplan

    <extension name="unpark">

      <condition field="destination_number" expression="^5100$">

         <action application="set" data="park_after_bridge=true"/>

         <action application="socket" data="192.168.1.100:8022 async full"/>

      </condition>

    </extension>

 

Why use phrase instead of files, mainly experience with IVR building, easier
to change, manage, can use parameters and language customization.

 

Application

Let's be optimistic and forget about any errors.

 

Leg 1 call fs ext 5100

         goes thru socket app

         api create_uuid for outgoing call

         execute playback phrase

         some other event initiate originate (in my case http request
result)

         bgapi originate
{origination_uuid=created_UUID}sofia/internal/1001%192.168.1.11 5100  //
this way leg 2 ends also on my socket app

 

Leg 2 outgoing call (ext 5100) 

         execute playback phrase

 

the 2 legs are there so now using DTMF handling on leg 1

 

         DTMF 1 bridge the two legs

         DTMF 2 broadcast msg on the 2 leg , this work but wasn't used in
the log provided

         DTMF 3 uuid_transfer or in our case uuid_dual_transfert

 

 

leg 1 = sofia/internal/1000 at 192.168.1.11

leg 2 = sofia/internal/1001 at 192.168.1.11

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, May 29, 2012 1:04 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How do you split bridged calls ?

 

Sorry for the late reply, but I just wanted to mention for the sake of
posterity/SEO that you could also use 'uuid_dual_transfer' if you want to
send each leg to a different destination. 
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_dual_transfer

-MC

On Wed, May 23, 2012 at 9:39 AM, Dave R. Kompel <drk at drkngs.net> wrote:

Or you  could use uuid_transfer, such as "uuid_transfer <uuid> -both park
inline". If you want to use dialplan apps in the transfer just use the
"inline" dialplan, and not XML.

 

--Dave

 

 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
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From msc at freeswitch.org  Tue Jun  5 03:45:40 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Jun 2012 16:45:40 -0700
Subject: [Freeswitch-users] How do you split bridged calls ?
In-Reply-To: <000301cd42a8$4a82fdd0$df88f970$@center.com>
References: <CALNZuwkRLZ0avq-xaPLmrnRQYUUGHd4WDFKCaqA+H2N_rDTGKw@mail.gmail.com>
	<20120523163913.b07558d1@mail.tritonwest.net>
	<CAKzWOxXsOuDfcwttJEwjUEoaaYEcn8bb3pWkeE+z3pb96co-TQ@mail.gmail.com>
	<002701cd3f7a$f3dd50a0$db97f1e0$@center.com>
	<CAKzWOxWP6dmRBV-1SJhyS3V9+GqFd9zuaevfxOGM0N00YVsCJQ@mail.gmail.com>
	<000301cd42a8$4a82fdd0$df88f970$@center.com>
Message-ID: <CAKzWOxXPC7A0y2tC6gGOaeRXMz_F45LKcWMGEMkGoP4v_uJ6pQ@mail.gmail.com>

Thanks for posting news of your success! Search engines all over the world
send you their thanks. :P
-MC

On Mon, Jun 4, 2012 at 4:18 PM, Gilbert Amar <gamar at center.com> wrote:

> Hello ****
>
> ** **
>
> After several mistakes I find out that if you want to break a bridge, from
> a socket application, and keep managing the two legs in your application
> the solution is:****
>
> ** **
>
> uuid_transfer <uuid> -both park inline //the inline does the magic here.**
> **
>
> ** **
>
> then you can play phrases,get dtmf... and even re-bridge the two legs.****
>
> ** **
>
> Thanks to those who point me to the right direction.****
>
> ** **
>
> Gilbert****
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, May 31, 2012 11:11 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] How do you split bridged calls ?****
>
> ** **
>
> I dropped your log file into our pastebin here:
> http://pastebin.freeswitch.org/19209
>
> The log shows that the issue is not that FS is only trying to process the
> first leg, which can be seen by looking at lines 507 and 714. The problem
> here is that your inline dialplan is not being parsed properly. What's
> happening is that uuid_dual_transfer is literally trying to send the A leg
> to destination number "m:^:phrase:queue_position,5^ inline" and the B leg
> to destination number "m:^:phrase:queue_position,3^ inline".
>
> What are you trying to do with each leg of the call? I see that you are
> trying to announce the queue position, but what do you want to have happen
> after that? As it stands your call legs would probably just end after the
> announcement because there's nothing more to do.
>
> -MC****
>
> On Thu, May 31, 2012 at 3:15 PM, Gilbert Amar <gamar at center.com> wrote:***
> *
>
> Hello,****
>
>  ****
>
> I keep trying to split a bridged call.****
>
> Using playback wav I have audio on both leg, but in my IVR I will need
> different message to be played to each leg.****
>
> I do not know if this is a bug or not:****
>
> I have audio only on one leg using uuid_transfer or uuid_dual_transfer for
> phrase.****
>
>  ****
>
> Test 1****
>
>          api uuid_transfer <uuid> -both 'm:^:phrase:queue_position,5^
> inline'****
>
>  ****
>
> Test 2:****
>
>          api uuid_dual_transfer <uuid> 'm:^:phrase:queue_position,5^
> inline' 'm:^:phrase:queue_position,3^ inline'****
>
>  ****
>
> notice queue_position 5 for the first and 3 for the second.****
>
>  ****
>
>  ****
>
> Please find attach a FS log showing for the second test that FS is only
> looking to process the first leg.****
>
>  ****
>
> Gilbert****
>
>  ****
>
>  ****
>
> Details regarding the application and log.****
>
>  ****
>
> my dialplan****
>
>     <extension name="unpark">****
>
>       <condition field="destination_number" expression="^5100$">****
>
>          <action application="set" data="park_after_bridge=true"/>****
>
>          <action application="socket" data="192.168.1.100:8022 async
> full"/>****
>
>       </condition>****
>
>     </extension>****
>
>  ****
>
> Why use phrase instead of files, mainly experience with IVR building,
> easier to change, manage, can use parameters and language customization.**
> **
>
>  ****
>
> Application****
>
> Let's be optimistic and forget about any errors.****
>
>  ****
>
> Leg 1 call fs ext 5100****
>
>          goes thru socket app****
>
>          api create_uuid for outgoing call****
>
>          execute playback phrase****
>
>          some other event initiate originate (in my case http request
> result)****
>
>          bgapi originate
> {origination_uuid=created_UUID}sofia/internal/1001%192.168.1.11 5100  //
> this way leg 2 ends also on my socket app****
>
>  ****
>
> Leg 2 outgoing call (ext 5100) ****
>
>          execute playback phrase****
>
>  ****
>
> the 2 legs are there so now using DTMF handling on leg 1****
>
>  ****
>
>          DTMF 1 bridge the two legs****
>
>          DTMF 2 broadcast msg on the 2 leg , this work but wasn't used in
> the log provided****
>
>          DTMF 3 uuid_transfer or in our case uuid_dual_transfert****
>
>  ****
>
>  ****
>
> leg 1 = sofia/internal/1000 at 192.168.1.11****
>
> leg 2 = sofia/internal/1001 at 192.168.1.11****
>
>  ****
>
>  ****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Tuesday, May 29, 2012 1:04 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] How do you split bridged calls ?****
>
>  ****
>
> Sorry for the late reply, but I just wanted to mention for the sake of
> posterity/SEO that you could also use 'uuid_dual_transfer' if you want to
> send each leg to a different destination.
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_dual_transfer
>
> -MC****
>
> On Wed, May 23, 2012 at 9:39 AM, Dave R. Kompel <drk at drkngs.net> wrote:***
> *
>
> Or you  could use uuid_transfer, such as "uuid_transfer <uuid> -both park
> inline". If you want to use dialplan apps in the transfer just use the
> "inline" dialplan, and not XML.****
>
>  ****
>
> --Dave****
>
>  ****
>
>
>
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From fvillarroel at yahoo.com  Tue Jun  5 06:10:03 2012
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Mon, 4 Jun 2012 19:10:03 -0700 (PDT)
Subject: [Freeswitch-users] Transfer hangup but b-leg ring
Message-ID: <1338862203.33799.YahooMailClassic@web160304.mail.bf1.yahoo.com>

Hi.

I am trying to use nibblebill and lcr.

If originator (example 1006) account have zero balance and trying to call to 1004, the call hangup but the phone 1004 ring anyway.

My dialplan:


  <context name="default">

        <extension name="Outbound to PSTN 11 Digits">
                <condition field="destination_number" expression="^(10[01][0-9])$">
                        <action application="set" data="hangup_after_bridge=false"/>
                        <action application="set" data="continue_on_fail=true"/>
                        <action application="lcr" data="$1 pg_prefix2"/>
                        <action application="set" data="nibble_account=${accountcode}"/>
                        <action application="set" data="nibble_rate=${lcr_rate_1}"/>
                        <action application="bridge" data="${lcr_auto_route}"/>

                </condition>
        </extension>

        <extension name="hangup">
                <condition field="destination_number" expression="^(hangup)$">
                        <action application="hangup"/>
                </condition>
        </extension>


The log of call:

http://pastebin.freeswitch.org/19228

How i can do?


From miha at softnet.si  Tue Jun  5 10:34:57 2012
From: miha at softnet.si (Miha)
Date: Tue, 05 Jun 2012 08:34:57 +0200
Subject: [Freeswitch-users] Multi-tenant dilplans
In-Reply-To: <03e101cd4268$41651cf0$c42f56d0$@bizfocused.com>
References: <4FCC6641.9060406@softnet.si> <4FCC9197.1010101@softnet.si>
	<03e101cd4268$41651cf0$c42f56d0$@bizfocused.com>
Message-ID: <4FCDA891.7070302@softnet.si>

Hi @Sean,

thanks:)

Regards,
Miha

On 6/4/2012 5:39 PM, Sean Devoy wrote:
> HI Miha,
>
> I can almost never answer a questi9on here, but this one I have working
> perfectly.
>
> In the user directory where you define you extensions, you must set the
> "context" for that user's/extension's dialplan:
> <variable name="user_context" value="from-COMPANY_A"/>   pick your own value,
> but match it in the dialplan.
>
> In the dialplan, you start with that context, and group your dialplan
> directives under inside it.  In this example someone from a Company_A
> extension dials extension 200.  You cann see how the context allows Company
> A users to dial 200 and Company B users to dial extension 200 and get
> different users.
>
> <context name="from-COMPANY_A">
>       <extension name="200">
>          <condition field="destination_number" expression="^200$">
>                <action application="set" data="ringback=${us-ring}"/>
>                <action application="set"
> data="effective_caller_id_number=${internal_caller_id_number}"/>
>                <action application="set"
> data="effective_caller_id_name=${internal_caller_id_name}"/>
>                <action application="set" data="hangup_after_bridge=true"/>
>                <action application="set" data="call_timeout=20"/>
>               <action application="bridge"
> data="{sip_invite_domain=company_a.com}user/200 at company_a.com"  />
>               <action application="answer"/>
>               <action application="voicemail" data="default
> voicemail_COMPANY_A 200"/>
>               <action application="hangup"/>
>            </condition>
>        </extension>
> ...
> </context>
>
> Just FYI, here is my full directory entry for extension 200 (which also has
> a Shared Call Appearance that you may not want/need).  I have changed the
> domain names and password:
>      <user id="200">
>          <variables>
>            <variable name="outbound_caller_id_name" value="Our Company
> name"/>
>            <variable name="outbound_caller_id_number" value="Our External
> Number"/>
>            <variable name="internal_caller_id_name" value="Shirley"/>
>            <variable name="internal_caller_id_number" value="200"/>
>            <variable name="user_context" value="from-COMPANY_A"/>
>            <variable name="user_originated" value="true"/>
>            <variable name="toll_allow" value="domestic"/>
>            <variable name="accountcode" value="200"/>
>            <variable name="mwi-account" value="200 at voicemail_COMPANY_A"/>
>          </variables>
>          <params>
>            <param name="password"  value="SomePasswordHere"/>
>            <param name="manage-shared-appearance" value-="true" />"
>            <param name="dial-string"
> value="{presence_id=200 at company_a.com}${sofia_contact(200 at company_a.com)}"/>
>            <param name="mwi-account" value="200 at voicemail_company_a"/>
>          </params>
>        </user>
>
> I hope that helps.
>
> Sean
> -----Original Message-----
> From: Miha [mailto:miha at softnet.si]
> Sent: Monday, June 04, 2012 6:45 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Multi-tenant dilplans
>
> On 6/4/2012 9:39 AM, Miha wrote:
>> Hi,
>>
>> I set multi tenat for two domains which is forking well. Now I would
>> like to seperate dialplans so that domain A will use default dialplan
>> in directory dialplan/a and domain B will use default dialplan in
>> directory dialplan/b.
>>
>> I have tried few things but every time fs use default dialplan.
>>
>> Thanks for help!
>>
>> Regards,
>> Miha
>>
>> ______________________________________________________________________
>> ___ Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users 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
>>
> Hi,
>
> I was having set wrong<variable name="user_context" value="domain" />  .
>
> THanks!
>
> Miha
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From miha at softnet.si  Tue Jun  5 10:42:06 2012
From: miha at softnet.si (Miha)
Date: Tue, 05 Jun 2012 08:42:06 +0200
Subject: [Freeswitch-users] internal/external dialing between different
	domains
Message-ID: <4FCDAA3E.1010305@softnet.si>

Hello,

what is the best way to dial between domains (multi tentat) on same FS?

Is better to route it internal or external (sbc send it back and use 
public dialplan to send it to right domain)?

What would be the best way to route it internally, so that dialplan 
would not change if I add more domains?

Users are now registered like this: if domain is enterprise.test.com, 
user will be number.enterprise.

Thanks!

BR;
Miha


From chrisbware at yahoo.it  Tue Jun  5 14:42:55 2012
From: chrisbware at yahoo.it (Chris B. Ware)
Date: Tue, 5 Jun 2012 11:42:55 +0100 (BST)
Subject: [Freeswitch-users] Music on hold while I add someone to the
	conference
Message-ID: <1338892975.53314.YahooMailNeo@web132303.mail.ird.yahoo.com>

Thank you Michael, I'll test it.

In the meanwhile I've found this workaround:

<extension name="Add new OB call to conference">
? ? ? <condition field="destination_number" expression="^ADD_THIRD$">
? ? ? ? <action application="set" data="res=${bgapi uuid_broadcast ${uuid_B} local_stream://moh}" />
? ? ? ??<!-- ask caller for a number + #, collect into ${target_num} variable -->?
? ? ? ? <action application="play_and_get_digits" data="4 11 1 5000 # ivr/ivr-enter_destination_telephone_number.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/>

? ? ? ? <action application="set" data="res=${bgapi uuid_break ${uuid_B} }" />
? ? ? ? <!-- add this call to the conference -->
? ? ? ? <action application="execute_extension" data="ADD_CALL_TO_CONF_${target_num}"/>
? ? ? </condition>
? ? </extension>


uuid_broadcast and uuid_break do the trick!
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From modesto at isimples.com.br  Tue Jun  5 16:26:51 2012
From: modesto at isimples.com.br (Antonio Modesto)
Date: Tue, 05 Jun 2012 09:26:51 -0300
Subject: [Freeswitch-users] Doubts with att_xfer
In-Reply-To: <1338836724.3534.12.camel@modesto.localdomain.net>
References: <1338836724.3534.12.camel@modesto.localdomain.net>
Message-ID: <1338899211.20521.5.camel@modesto.localdomain.net>

On Mon, 2012-06-04 at 16:05 -0300, Antonio Modesto wrote:
> Hi,
> 
> 
> I am trying to enable att_xfer for calls that come from pstn, these call
> are being routed to an internal registered SIP Phone. In asterisk I had
> those options "Tt" in the Dial application to enable transfer to the
> caller party and to the calling party. The question is that for internal
> call I want both legs be able to transfer to whoever they want, but for
> external calls I want just the internal leg be able to transfer calls.
> How can I do that?
> 
> 
> Regards.


I've tried to change this line in the Local_Extension, but it still
doesn't work, only the originator of the call can do transfers:

	        <action application="bind_meta_app" data="4 ab s
execute_extension::att_xfer XML features"/>



Has anybody any clue of what I've to do to get the transfers working on
both legs?


Thanks.


> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From miconda at gmail.com  Tue Jun  5 16:41:44 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 05 Jun 2012 14:41:44 +0200
Subject: [Freeswitch-users] unallocated number error after integrating
 FS with Kamailio
In-Reply-To: <1336984433.38446.YahooMailClassic@web110806.mail.gq1.yahoo.com>
References: <1336984433.38446.YahooMailClassic@web110806.mail.gq1.yahoo.com>
Message-ID: <4FCDFE88.9020102@gmail.com>

Hello,

On 5/14/12 10:33 AM, Sherif Omran wrote:
> Hello Guys,
>
> I have Kamailio + FS running on the same server and integrated 
> together according to this tutorial as SBC + Media Services
>
> http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc
>
> Issues:
> 1- 2 Sip softphones are registered. I can see them online in kamailio. 
> When I call FS Echo service, it works. However, when I call the other 
> softphone number, it gives unallocated number.
> 2- If user is unreachable, I should get to his voice mail. But I get 
> also unknown Invalid application voice mail.
>
> Any body figures this error?
>
do you get the INVITE out of FreeSwitch, back to Kamailio?

You can use ngrep to watch the sip traffic on the network, be sure you 
use the option '-d any' to listen on all interfaces. You can send the 
ngrep output here if you don't figure out yourself.

If the call is not sent out, then something is wrong with your dialplan, 
not intercepting the call to be sent out, but probably terminating it 
due to now explicit rule for it.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



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From ahe.sanath at gmail.com  Tue Jun  5 16:48:16 2012
From: ahe.sanath at gmail.com (Sanath Prasanna)
Date: Tue, 5 Jun 2012 18:18:16 +0530
Subject: [Freeswitch-users] "originate" Command not found in fs_cli
Message-ID: <CALQZz0xQA0mP31=kbszjYB55it5gMU1mnUj5bNWuTR4FmSFuVQ@mail.gmail.com>

Hi,
when I run originate command in fs_cli following error coming. Pls advice
to resolve that.

freeswitch at internal> originate sofia/external/ivr/9179123456 9179123458
-ERR originate  sofia/external/ivr/9179123456 9179123458 Command not found!

Br,
Sanath
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From saami_mh at ymail.com  Tue Jun  5 17:07:58 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Tue, 5 Jun 2012 06:07:58 -0700 (PDT)
Subject: [Freeswitch-users] "originate" Command not found in fs_cli
In-Reply-To: <CALQZz0xQA0mP31=kbszjYB55it5gMU1mnUj5bNWuTR4FmSFuVQ@mail.gmail.com>
References: <CALQZz0xQA0mP31=kbszjYB55it5gMU1mnUj5bNWuTR4FmSFuVQ@mail.gmail.com>
Message-ID: <1338901678.1847.YahooMailNeo@web120105.mail.ne1.yahoo.com>

hi,

If you have a phone registered to your FreeSWITCH server, then use the originate command from fs_cli. The basic syntax of
originate is:
originate <dialstring> <destination number>
Try the following and see what happens. Replace "1000" with the extension number of your phone:
originate loopback/9664 1000
originate user/1000 9664
originate error/USER_BUSY 1000
originate loopback/9192 1000
originate loopback/4000 1000
?
?
?
?

________________________________
 From: Sanath Prasanna <ahe.sanath at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Tuesday, June 5, 2012 5:18 PM
Subject: [Freeswitch-users] "originate" Command not found in fs_cli
 

Hi,
when I run originate command in fs_cli following error coming. Pls advice to resolve that.


freeswitch at internal> originate sofia/external/ivr/9179123456?9179123458
-ERR originate?
sofia/external/ivr/9179123456?9179123458?Command not found!

Br,
Sanath
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing 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 odermann at googlemail.com  Tue Jun  5 17:17:58 2012
From: odermann at googlemail.com (Dennis)
Date: Tue, 5 Jun 2012 15:17:58 +0200
Subject: [Freeswitch-users] Missing event for "180 Ringing"?
Message-ID: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>

Hi,

for us it seems as if there is an event missing for 180 Ringing. Is
there a way to receive/activate this event?

We are using ESL and are getting nearly all events we need. For "183
Session Progress SDP" we get "channel_progress_media" and for "200 OK
SDP" we get "channel_answer". So we only need one more event...


Thanks for your help
Dennis


From kamminthang.nengzalam at a-cti.com  Tue Jun  5 18:26:07 2012
From: kamminthang.nengzalam at a-cti.com (Kamminthang Nengzalam)
Date: Tue, 5 Jun 2012 19:56:07 +0530
Subject: [Freeswitch-users] Freeswitch + open vox G400P
In-Reply-To: <mailman.11174.1338905758.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.11174.1338905758.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAL8RuBONbdxEWy9z5oTTuBZ_7B8hE4oNday=SBtDm6oaz2d8Zg@mail.gmail.com>

Hi Guys,

i want to configure openvox G400P in my freeswitch.
Can anyone lemme kno how i should do it?
will mod_gsmopen work and if so/not how a should make it work.
Pliz and thank you in advance for helping.....

Thanks,
Kam
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From gmaruzz at gmail.com  Tue Jun  5 18:37:18 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 5 Jun 2012 16:37:18 +0200
Subject: [Freeswitch-users] Freeswitch + open vox G400P
In-Reply-To: <CAL8RuBONbdxEWy9z5oTTuBZ_7B8hE4oNday=SBtDm6oaz2d8Zg@mail.gmail.com>
References: <mailman.11174.1338905758.13443.freeswitch-users@lists.freeswitch.org>
	<CAL8RuBONbdxEWy9z5oTTuBZ_7B8hE4oNday=SBtDm6oaz2d8Zg@mail.gmail.com>
Message-ID: <CALXCt0oBODJP=O_KLMKKA8_FEK7h8LkTV=pvJYiFyOW-S8ou6Q@mail.gmail.com>

On Tue, Jun 5, 2012 at 4:26 PM, Kamminthang Nengzalam <
kamminthang.nengzalam at a-cti.com> wrote:

>
> i want to configure openvox G400P in my freeswitch.
> Can anyone lemme kno how i should do it?
> will mod_gsmopen work and if so/not how a should make it work.
>

mod_gsmopen will work on any server that has an USB port. You will need to
use compatible GSM dongles.
please read http://wiki.freeswitch.org/wiki/Gsmopen#What_Is_GSMopen and
that entire wiki page.

-giovanni





> Pliz and thank you in advance for helping.....
>
> Thanks,
> Kam
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From b2m at a-cti.com  Tue Jun  5 18:49:56 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Tue, 5 Jun 2012 20:19:56 +0530
Subject: [Freeswitch-users] Freeswitch + open vox G400P
In-Reply-To: <CALXCt0oBODJP=O_KLMKKA8_FEK7h8LkTV=pvJYiFyOW-S8ou6Q@mail.gmail.com>
References: <mailman.11174.1338905758.13443.freeswitch-users@lists.freeswitch.org>
	<CAL8RuBONbdxEWy9z5oTTuBZ_7B8hE4oNday=SBtDm6oaz2d8Zg@mail.gmail.com>
	<CALXCt0oBODJP=O_KLMKKA8_FEK7h8LkTV=pvJYiFyOW-S8ou6Q@mail.gmail.com>
Message-ID: <CAP2fbbcbDJGXyHrJSYU8rfG3qhVqd=kC=0URDrX=49+DcGXiaw@mail.gmail.com>

This particular module is actually on PCI, so guess it may not work.

Thanks,
Bala

On Tue, Jun 5, 2012 at 8:07 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> On Tue, Jun 5, 2012 at 4:26 PM, Kamminthang Nengzalam <
> kamminthang.nengzalam at a-cti.com> wrote:
>
>>
>> i want to configure openvox G400P in my freeswitch.
>> Can anyone lemme kno how i should do it?
>> will mod_gsmopen work and if so/not how a should make it work.
>>
>
> mod_gsmopen will work on any server that has an USB port. You will need to
> use compatible GSM dongles.
> please read http://wiki.freeswitch.org/wiki/Gsmopen#What_Is_GSMopen and
> that entire wiki page.
>
> -giovanni
>
>
>
>
>
>> Pliz and thank you in advance for helping.....
>>
>> Thanks,
>> Kam
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Adam.Lappe at qsc.de  Tue Jun  5 18:51:55 2012
From: Adam.Lappe at qsc.de (Lappe, Adam)
Date: Tue, 5 Jun 2012 16:51:55 +0200
Subject: [Freeswitch-users] Freeswitch send 423 (Interval Too Brief)
Message-ID: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57E7E@CGNMSG01.oc.qsc.de>

Hi all,

is there a way to make freeswitch send a 423 (Interval Too Brief) during a SIP REGISTER request?
I want to prevent clients to (re)register with a low expire-time and thus spam our network.
The "sip-force-expires"-variable would work but don't seem to be SIP-compliant to me.

>From RFC 3261:



               10.3 Processing REGISTER Requests


        When receiving a REGISTER request, a registrar follows these steps:



        (...)



        7. The registrar now processes each contact address in the Contact

         header field in turn.  For each address, it determines the

         expiration interval as follows:



         -  If the field value has an "expires" parameter, that value

            MUST be taken as the requested expiration.



         -  If there is no such parameter, but the request has an

            Expires header field, that value MUST be taken as the

            requested expiration.



         -  If there is neither, a locally-configured default value MUST

            be taken as the requested expiration.



         The registrar MAY choose an expiration less than the requested

         expiration interval.  If and only if the requested expiration

         interval is greater than zero AND smaller than one hour AND

         less than a registrar-configured minimum, the registrar MAY

         reject the registration with a response of 423 (Interval Too

         Brief).  This response MUST contain a Min-Expires header field

         that states the minimum expiration interval the registrar is

         willing to honor.  It then skips the remaining steps.





Thanks in Advance,

Adam

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From cristian.re.work at gmail.com  Tue Jun  5 19:33:59 2012
From: cristian.re.work at gmail.com (cristian re)
Date: Tue, 5 Jun 2012 17:33:59 +0200
Subject: [Freeswitch-users] how to: get session variable after
	api_hangup_hook
Message-ID: <CABx_eV3-KAYSTaDC09uXpH-wWc1adJhgiAiDxg7yaRCwos73fg@mail.gmail.com>

Hello,

I write a custom cpp module for freeswitch that exposes an API:

SWITCH_ADD_API(commands_api_interface, "my_hangup", "my_hangup",
 my_hangup_api , MY_HANGUP_USAGE);

I want to call this api from dialplan (after bridge hangup) for reading the
variable "billmsec".
I put this into my dialplan:

<action application="set" data="session_in_hangup_hook=true"/>
<action application="set" data="api_hangup_hook=my_hangup_api"/>

After hangup freeswitch call correctly my API but I have not figured out
how to get the variable.
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From kamminthang.nengzalam at a-cti.com  Tue Jun  5 20:06:55 2012
From: kamminthang.nengzalam at a-cti.com (Kamminthang Nengzalam)
Date: Tue, 5 Jun 2012 21:36:55 +0530
Subject: [Freeswitch-users]  Freeswitch + open vox G400P
In-Reply-To: <CAP2fbbf2VP452CgXRBbvXkqsA5grpQZGTcWApJyBfctzHoKomA@mail.gmail.com>
References: <mailman.11174.1338905758.13443.freeswitch-users@lists.freeswitch.org>
	<CAL8RuBONbdxEWy9z5oTTuBZ_7B8hE4oNday=SBtDm6oaz2d8Zg@mail.gmail.com>
	<CAP2fbbf2VP452CgXRBbvXkqsA5grpQZGTcWApJyBfctzHoKomA@mail.gmail.com>
Message-ID: <CAL8RuBN0VqGNHzvyW=NKKhe_oZVECfJ3GmqpbbG5r8-xGtaVAA@mail.gmail.com>

i already have configure GSMopen for E1550 and it is working ok but the
audio quality is not very good.
Is there a way i can boost it or use openvox instead...

Thanks,
Kam

On Tue, Jun 5, 2012 at 7:56 PM, Kamminthang Nengzalam <
kamminthang.nengzalam at a-cti.com> wrote:

>
> Hi Guys,
>
> i want to configure openvox G400P in my freeswitch.
> Can anyone lemme kno how i should do it?
> will mod_gsmopen work and if so/not how a should make it work.
> Pliz and thank you in advance for helping.....
>
> Thanks,
> Kam
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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://staging.originateglobal.com/ClickToCall?id=%2244Rf0r6tLOxU8APXNT92IIxh93OzvT2clF0mEJCf8schNn%22>
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From steveu at coppice.org  Tue Jun  5 20:14:51 2012
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 06 Jun 2012 00:14:51 +0800
Subject: [Freeswitch-users] Freeswitch + open vox G400P
In-Reply-To: <CAL8RuBN0VqGNHzvyW=NKKhe_oZVECfJ3GmqpbbG5r8-xGtaVAA@mail.gmail.com>
References: <mailman.11174.1338905758.13443.freeswitch-users@lists.freeswitch.org>
	<CAL8RuBONbdxEWy9z5oTTuBZ_7B8hE4oNday=SBtDm6oaz2d8Zg@mail.gmail.com>
	<CAP2fbbf2VP452CgXRBbvXkqsA5grpQZGTcWApJyBfctzHoKomA@mail.gmail.com>
	<CAL8RuBN0VqGNHzvyW=NKKhe_oZVECfJ3GmqpbbG5r8-xGtaVAA@mail.gmail.com>
Message-ID: <4FCE307B.30801@coppice.org>

An E1550 is perfectly capable of giving the maximum audio quality GSM 
permits. If you are getting bad results perhaps you have the device in 
some dark corner with a weak signal.

Steve

On 06/06/2012 12:06 AM, Kamminthang Nengzalam wrote:
> i already have configure GSMopen for E1550 and it is working ok but 
> the audio quality is not very good.
> Is there a way i can boost it or use openvox instead...
>
> Thanks,
> Kam
>
> On Tue, Jun 5, 2012 at 7:56 PM, Kamminthang Nengzalam 
> <kamminthang.nengzalam at a-cti.com 
> <mailto:kamminthang.nengzalam at a-cti.com>> wrote:
>
>
>     Hi Guys,
>
>     i want to configure openvox G400P in my freeswitch.
>     Can anyone lemme kno how i should do it?
>     will mod_gsmopen work and if so/not how a should make it work.
>     Pliz and thank you in advance for helping.....
>
>     Thanks,
>     Kam
>
>



From kamminthang.nengzalam at a-cti.com  Tue Jun  5 20:31:11 2012
From: kamminthang.nengzalam at a-cti.com (Kamminthang Nengzalam)
Date: Tue, 5 Jun 2012 22:01:11 +0530
Subject: [Freeswitch-users] Freeswitch + open vox G400P
In-Reply-To: <CAL8RuBN0VqGNHzvyW=NKKhe_oZVECfJ3GmqpbbG5r8-xGtaVAA@mail.gmail.com>
References: <mailman.11174.1338905758.13443.freeswitch-users@lists.freeswitch.org>
	<CAL8RuBONbdxEWy9z5oTTuBZ_7B8hE4oNday=SBtDm6oaz2d8Zg@mail.gmail.com>
	<CAP2fbbf2VP452CgXRBbvXkqsA5grpQZGTcWApJyBfctzHoKomA@mail.gmail.com>
	<CAL8RuBN0VqGNHzvyW=NKKhe_oZVECfJ3GmqpbbG5r8-xGtaVAA@mail.gmail.com>
Message-ID: <CAL8RuBN0ikW_mLW-LDVyoXH13K30KqyFUooWHVd4HCpgZTnNBg@mail.gmail.com>

i followed the wiki correctly. And i'm getting no error. Can you lemme kno
how to get better sound quality......
Do i need to provide anything pliz lemme kno....

Thanks,
Kam

On Tue, Jun 5, 2012 at 9:36 PM, Kamminthang Nengzalam <
kamminthang.nengzalam at a-cti.com> wrote:

> i already have configure GSMopen for E1550 and it is working ok but the
> audio quality is not very good.
> Is there a way i can boost it or use openvox instead...
>
> Thanks,
> Kam
>
>
> On Tue, Jun 5, 2012 at 7:56 PM, Kamminthang Nengzalam <
> kamminthang.nengzalam at a-cti.com> wrote:
>
>>
>> Hi Guys,
>>
>> i want to configure openvox G400P in my freeswitch.
>> Can anyone lemme kno how i should do it?
>> will mod_gsmopen work and if so/not how a should make it work.
>> Pliz and thank you in advance for helping.....
>>
>> Thanks,
>> Kam
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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://staging.originateglobal.com/ClickToCall?id=%2244Rf0r6tLOxU8APXNT92IIxh93OzvT2clF0mEJCf8schNn%22>
>
>
>
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From gmaruzz at gmail.com  Tue Jun  5 20:54:28 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 5 Jun 2012 18:54:28 +0200
Subject: [Freeswitch-users] Freeswitch + open vox G400P
In-Reply-To: <CAL8RuBN0VqGNHzvyW=NKKhe_oZVECfJ3GmqpbbG5r8-xGtaVAA@mail.gmail.com>
References: <mailman.11174.1338905758.13443.freeswitch-users@lists.freeswitch.org>
	<CAL8RuBONbdxEWy9z5oTTuBZ_7B8hE4oNday=SBtDm6oaz2d8Zg@mail.gmail.com>
	<CAP2fbbf2VP452CgXRBbvXkqsA5grpQZGTcWApJyBfctzHoKomA@mail.gmail.com>
	<CAL8RuBN0VqGNHzvyW=NKKhe_oZVECfJ3GmqpbbG5r8-xGtaVAA@mail.gmail.com>
Message-ID: <CALXCt0rF5hJfcrVu==-Z+mZYzSK5z_YXA+Pxn6LUZfYjXcr=WA@mail.gmail.com>

On Tue, Jun 5, 2012 at 6:06 PM, Kamminthang Nengzalam <
kamminthang.nengzalam at a-cti.com> wrote:

> i already have configure GSMopen for E1550 and it is working ok but the
> audio quality is not very good.
>


In which way audio is not good (it's straight from the digital stream, so
it's exactly how the carrier give it to you) ? Can you elaborate on this?



> Is there a way i can boost it or use openvox instead...
>
>
definitely not


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From bdfoster at endigotech.com  Tue Jun  5 21:03:09 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 5 Jun 2012 13:03:09 -0400
Subject: [Freeswitch-users] Freeswitch + open vox G400P
In-Reply-To: <CAL8RuBN0VqGNHzvyW=NKKhe_oZVECfJ3GmqpbbG5r8-xGtaVAA@mail.gmail.com>
References: <mailman.11174.1338905758.13443.freeswitch-users@lists.freeswitch.org>
	<CAL8RuBONbdxEWy9z5oTTuBZ_7B8hE4oNday=SBtDm6oaz2d8Zg@mail.gmail.com>
	<CAP2fbbf2VP452CgXRBbvXkqsA5grpQZGTcWApJyBfctzHoKomA@mail.gmail.com>
	<CAL8RuBN0VqGNHzvyW=NKKhe_oZVECfJ3GmqpbbG5r8-xGtaVAA@mail.gmail.com>
Message-ID: <CAKV-jFxfpprurtcLf=YcrG80pV+7YuLknQO7dkEmbSveWyv0kw@mail.gmail.com>

Maybe you've gotten used to the superior sound quality freeswitch provides
:-)

Your audio quality is dependent on the equipment you are using, and the
E1550 is a good piece of equipment. Check your signal stregnth and make
sure you are giving the equipment the best chance of working. If that
device doesn't have an external Jack then USB port extenders work wonders.
Just remember that GSM isn't a high quality codec.

If your phones take GSM (most IP phones do) then make sure GSM is being
used end to end (I.e. no transcoding). That might yield better sound
quality, and you should be doing this to decrease the load being placed on
your system.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 5, 2012 12:08 PM, "Kamminthang Nengzalam" <
kamminthang.nengzalam at a-cti.com> wrote:

> i already have configure GSMopen for E1550 and it is working ok but the
> audio quality is not very good.
> Is there a way i can boost it or use openvox instead...
>
> Thanks,
> Kam
>
> On Tue, Jun 5, 2012 at 7:56 PM, Kamminthang Nengzalam <
> kamminthang.nengzalam at a-cti.com> wrote:
>
>>
>> Hi Guys,
>>
>> i want to configure openvox G400P in my freeswitch.
>> Can anyone lemme kno how i should do it?
>> will mod_gsmopen work and if so/not how a should make it work.
>> Pliz and thank you in advance for helping.....
>>
>> Thanks,
>> Kam
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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://staging.originateglobal.com/ClickToCall?id=%2244Rf0r6tLOxU8APXNT92IIxh93OzvT2clF0mEJCf8schNn%22>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Tue Jun  5 21:16:47 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 5 Jun 2012 13:16:47 -0400
Subject: [Freeswitch-users] "originate" Command not found in fs_cli
In-Reply-To: <CALQZz0xQA0mP31=kbszjYB55it5gMU1mnUj5bNWuTR4FmSFuVQ@mail.gmail.com>
References: <CALQZz0xQA0mP31=kbszjYB55it5gMU1mnUj5bNWuTR4FmSFuVQ@mail.gmail.com>
Message-ID: <CAKV-jFztkkESpAXZbrB-htebmzNnf2JAormnSChPrn0-_kHiAQ@mail.gmail.com>

Your not understanding the dialstring aspect...

Example:

originate sofia/gateway/flowroute/13175551212 user/1000 at 192.168.1.79

Where sofia is the channel type, gateway is a keyword, flowroute is what I
named my gateway, 13175551212 is the number I'm calling through my gateway
flowroute, and user/1000 is the extension to send it to and 192.168.1.79 is
the domain that user is attached to.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 5, 2012 8:49 AM, "Sanath Prasanna" <ahe.sanath at gmail.com> wrote:

> Hi,
> when I run originate command in fs_cli following error coming. Pls advice
> to resolve that.
>
> freeswitch at internal> originate sofia/external/ivr/9179123456 9179123458
> -ERR originate  sofia/external/ivr/9179123456 9179123458 Command not
> found!
>
> Br,
> Sanath
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 gmail.com  Tue Jun  5 21:17:03 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 5 Jun 2012 19:17:03 +0200
Subject: [Freeswitch-users] Freeswitch + open vox G400P
In-Reply-To: <CAKV-jFxfpprurtcLf=YcrG80pV+7YuLknQO7dkEmbSveWyv0kw@mail.gmail.com>
References: <mailman.11174.1338905758.13443.freeswitch-users@lists.freeswitch.org>
	<CAL8RuBONbdxEWy9z5oTTuBZ_7B8hE4oNday=SBtDm6oaz2d8Zg@mail.gmail.com>
	<CAP2fbbf2VP452CgXRBbvXkqsA5grpQZGTcWApJyBfctzHoKomA@mail.gmail.com>
	<CAL8RuBN0VqGNHzvyW=NKKhe_oZVECfJ3GmqpbbG5r8-xGtaVAA@mail.gmail.com>
	<CAKV-jFxfpprurtcLf=YcrG80pV+7YuLknQO7dkEmbSveWyv0kw@mail.gmail.com>
Message-ID: <CALXCt0qD4Ev=GaeJYG=8O8Nt-DyNz=XZyQUGVJRvhsu4+sezwQ@mail.gmail.com>

On Tue, Jun 5, 2012 at 7:03 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> Maybe you've gotten used to the superior sound quality freeswitch provides
> :-)
>
> Your audio quality is dependent on the equipment you are using, and the
> E1550 is a good piece of equipment. Check your signal stregnth and make
> sure you are giving the equipment the best chance of working. If that
> device doesn't have an external Jack then USB port extenders work wonders.
> Just remember that GSM isn't a high quality codec.
>
> If your phones take GSM (most IP phones do) then make sure GSM is being
> used end to end (I.e. no transcoding). That might yield better sound
> quality, and you should be doing this to decrease the load being placed on
> your system.
>

No, there is no transcoding to/from GSM codec with mod_gsmopen.

mod_gsmopen takes a digital stream in signed linear (SL16) at 16 bit, 8 khz
directly from the modem. It's the modem chipset that digitally converts
from GSM codec to signed linear. So, mod_gsmopen passes to FreeSWITCH the
SL16 stream 8khz, without touching it, in its purest form.

But it's true that if you're used to use FreeSWITCH with 16khz (or
superior) codecs, then you can be spoiled ;).

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From bdfoster at endigotech.com  Tue Jun  5 21:31:46 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 5 Jun 2012 13:31:46 -0400
Subject: [Freeswitch-users] Freeswitch + open vox G400P
In-Reply-To: <CALXCt0qD4Ev=GaeJYG=8O8Nt-DyNz=XZyQUGVJRvhsu4+sezwQ@mail.gmail.com>
References: <mailman.11174.1338905758.13443.freeswitch-users@lists.freeswitch.org>
	<CAL8RuBONbdxEWy9z5oTTuBZ_7B8hE4oNday=SBtDm6oaz2d8Zg@mail.gmail.com>
	<CAP2fbbf2VP452CgXRBbvXkqsA5grpQZGTcWApJyBfctzHoKomA@mail.gmail.com>
	<CAL8RuBN0VqGNHzvyW=NKKhe_oZVECfJ3GmqpbbG5r8-xGtaVAA@mail.gmail.com>
	<CAKV-jFxfpprurtcLf=YcrG80pV+7YuLknQO7dkEmbSveWyv0kw@mail.gmail.com>
	<CALXCt0qD4Ev=GaeJYG=8O8Nt-DyNz=XZyQUGVJRvhsu4+sezwQ@mail.gmail.com>
Message-ID: <CAKV-jFxmg-NZt3udvAYCKrMVB=AWypQBucew=2gdQeERkzsF+Q@mail.gmail.com>

Sorry, I misunderstood how that worked :-)

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 5, 2012 1:18 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:

> On Tue, Jun 5, 2012 at 7:03 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> Maybe you've gotten used to the superior sound quality freeswitch
>> provides :-)
>>
>> Your audio quality is dependent on the equipment you are using, and the
>> E1550 is a good piece of equipment. Check your signal stregnth and make
>> sure you are giving the equipment the best chance of working. If that
>> device doesn't have an external Jack then USB port extenders work wonders.
>> Just remember that GSM isn't a high quality codec.
>>
>> If your phones take GSM (most IP phones do) then make sure GSM is being
>> used end to end (I.e. no transcoding). That might yield better sound
>> quality, and you should be doing this to decrease the load being placed on
>> your system.
>>
>
> No, there is no transcoding to/from GSM codec with mod_gsmopen.
>
> mod_gsmopen takes a digital stream in signed linear (SL16) at 16 bit, 8
> khz directly from the modem. It's the modem chipset that digitally converts
> from GSM codec to signed linear. So, mod_gsmopen passes to FreeSWITCH the
> SL16 stream 8khz, without touching it, in its purest form.
>
> But it's true that if you're used to use FreeSWITCH with 16khz (or
> superior) codecs, then you can be spoiled ;).
>
> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cliff at develix.com  Tue Jun  5 21:36:16 2012
From: cliff at develix.com (Cliff Wells)
Date: Tue, 05 Jun 2012 10:36:16 -0700
Subject: [Freeswitch-users] Max handles 50 exceeded, blocking....
Message-ID: <1338917776.1915.19.camel@portable-evil>

I've been getting this recently:

2012-06-05 10:31:44.579059 [ERR] mod_commands.c:4063 Error connecting
2012-06-05 10:31:44.619059 [ERR] sofia_glue.c:6464 Error connecting
2012-06-05 10:31:44.619059 [ERR] sofia_glue.c:6504 Error Opening DB
2012-06-05 10:31:44.619059 [WARNING] sofia_glue.c:6464 Max handles 50 exceeded, blocking....

Is there a workaround?  I see runtime.max_db_handles = 50; in
switch_core.c, so I assume there's no setting I can adjust to help.
Would switching core to use ODBC solve the issue?  Or something simpler?

Regards,
Cliff



From cliff at develix.com  Tue Jun  5 21:47:42 2012
From: cliff at develix.com (Cliff Wells)
Date: Tue, 05 Jun 2012 10:47:42 -0700
Subject: [Freeswitch-users] Max handles 50 exceeded, blocking....
In-Reply-To: <1338917776.1915.19.camel@portable-evil>
References: <1338917776.1915.19.camel@portable-evil>
Message-ID: <1338918462.1915.20.camel@portable-evil>

On Tue, 2012-06-05 at 10:36 -0700, Cliff Wells wrote:
> I've been getting this recently:
> 
> 2012-06-05 10:31:44.579059 [ERR] mod_commands.c:4063 Error connecting
> 2012-06-05 10:31:44.619059 [ERR] sofia_glue.c:6464 Error connecting
> 2012-06-05 10:31:44.619059 [ERR] sofia_glue.c:6504 Error Opening DB
> 2012-06-05 10:31:44.619059 [WARNING] sofia_glue.c:6464 Max handles 50 exceeded, blocking....
> 
> Is there a workaround?  I see runtime.max_db_handles = 50; in
> switch_core.c, so I assume there's no setting I can adjust to help.
> Would switching core to use ODBC solve the issue?  Or something simpler?
> 

... and in classic fashion, two minutes after posting I find the answer
in autoload_configs/switch.conf.xml.

Sorry for the noise.

Cliff



From bdfoster at endigotech.com  Tue Jun  5 21:51:12 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 5 Jun 2012 13:51:12 -0400
Subject: [Freeswitch-users] Max handles 50 exceeded, blocking....
In-Reply-To: <1338917776.1915.19.camel@portable-evil>
References: <1338917776.1915.19.camel@portable-evil>
Message-ID: <CAKV-jFwxb5hNX_9qj8Eo3TeMFPFimSRSh=aZ2KkM+3T=fEw3LA@mail.gmail.com>

I'm not in front of a server right now but check in
/usr/local/freeswitch/conf/autoload_confocal/switch.conf.xml and see if
there's a variable along the lines of Max DB handles. If it just refers to
odbc stuff you have your answer.

Before you go much further, what was your call volume? It could have thrown
that error because something is locking the DB, so there could be something
wrong with it.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 5, 2012 1:36 PM, "Cliff Wells" <cliff at develix.com> wrote:

> I've been getting this recently:
>
> 2012-06-05 10:31:44.579059 [ERR] mod_commands.c:4063 Error connecting
> 2012-06-05 10:31:44.619059 [ERR] sofia_glue.c:6464 Error connecting
> 2012-06-05 10:31:44.619059 [ERR] sofia_glue.c:6504 Error Opening DB
> 2012-06-05 10:31:44.619059 [WARNING] sofia_glue.c:6464 Max handles 50
> exceeded, blocking....
>
> Is there a workaround?  I see runtime.max_db_handles = 50; in
> switch_core.c, so I assume there's no setting I can adjust to help.
> Would switching core to use ODBC solve the issue?  Or something simpler?
>
> Regards,
> Cliff
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun  5 21:56:41 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jun 2012 10:56:41 -0700
Subject: [Freeswitch-users] how to: get session variable after
	api_hangup_hook
In-Reply-To: <CABx_eV3-KAYSTaDC09uXpH-wWc1adJhgiAiDxg7yaRCwos73fg@mail.gmail.com>
References: <CABx_eV3-KAYSTaDC09uXpH-wWc1adJhgiAiDxg7yaRCwos73fg@mail.gmail.com>
Message-ID: <CAKzWOxUuNAjtuGwoTeud+_=ZHAtVshmqyZTxKfeNL961N6cm8w@mail.gmail.com>

Are you setting session_in_hangup_hook?
http://wiki.freeswitch.org/wiki/Channel_Variables#session_in_hangup_hook
-MC

On Tue, Jun 5, 2012 at 8:33 AM, cristian re <cristian.re.work at gmail.com>wrote:

> Hello,
>
> I write a custom cpp module for freeswitch that exposes an API:
>
> SWITCH_ADD_API(commands_api_interface, "my_hangup", "my_hangup",
>  my_hangup_api , MY_HANGUP_USAGE);
>
> I want to call this api from dialplan (after bridge hangup) for reading
> the variable "billmsec".
> I put this into my dialplan:
>
> <action application="set" data="session_in_hangup_hook=true"/>
> <action application="set" data="api_hangup_hook=my_hangup_api"/>
>
> After hangup freeswitch call correctly my API but I have not figured out
> how to get the variable.
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cliff at develix.com  Tue Jun  5 22:24:57 2012
From: cliff at develix.com (Cliff Wells)
Date: Tue, 05 Jun 2012 11:24:57 -0700
Subject: [Freeswitch-users] Max handles 50 exceeded, blocking....
In-Reply-To: <CAKV-jFwxb5hNX_9qj8Eo3TeMFPFimSRSh=aZ2KkM+3T=fEw3LA@mail.gmail.com>
References: <1338917776.1915.19.camel@portable-evil>
	<CAKV-jFwxb5hNX_9qj8Eo3TeMFPFimSRSh=aZ2KkM+3T=fEw3LA@mail.gmail.com>
Message-ID: <1338920697.1915.33.camel@portable-evil>

On Tue, 2012-06-05 at 13:51 -0400, Brian Foster wrote:
> I'm not in front of a server right now but check
> in /usr/local/freeswitch/conf/autoload_confocal/switch.conf.xml and
> see if there's a variable along the lines of Max DB handles. If it
> just refers to odbc stuff you have your answer.

There was just such a setting.  Thanks.

> 
> Before you go much further, what was your call volume? It could have
> thrown that error because something is locking the DB, so there could
> be something wrong with it.
> 

When total channels exceeds 400ish (~65% inbound, ~35% outbound), system
load suddenly spikes (under 350 or so, it's usually under 2 or 3, when
we hit 400+, it will jump as high as 80).  At that point the system
appears to be queueing calls as the volume will sometimes shoot up to as
high as 900, even though that's not really possible (hard limit due to
T1 count), so it seems that it might be blocking on some resource. 

Also caught this error:

2012-06-05 11:23:21.419535 [CRIT] switch_core_session.c:1861 Throttle Error! 1342
2012-06-05 11:23:21.520601 [CRIT] switch_time.c:957 Over Session Rate of 30!

So I adjusted that upwards as well.


16 cores Xeon 3GHz, 16GB RAM, 6 drive SAS RAID 10
Scientific Linux 6.1
Freeswitch git pull from yesterday.

Regards,
Cliff






From avi at avimarcus.net  Tue Jun  5 22:35:21 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 5 Jun 2012 21:35:21 +0300
Subject: [Freeswitch-users] Max handles 50 exceeded, blocking....
In-Reply-To: <1338920697.1915.33.camel@portable-evil>
References: <1338917776.1915.19.camel@portable-evil>
	<CAKV-jFwxb5hNX_9qj8Eo3TeMFPFimSRSh=aZ2KkM+3T=fEw3LA@mail.gmail.com>
	<1338920697.1915.33.camel@portable-evil>
Message-ID: <CALNZuwkq4d=_RJHaUcNFF4RaPd42CmJfniCnmjY-2rdLX26FUA@mail.gmail.com>

If you're writing the channels to the DB, you can either skip it with
-nosql or you can ramcache the SQL DBs.
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#FreeSWITCH.27s_core.db_I.2FO_bottleneck

-Avi



On Tue, Jun 5, 2012 at 9:24 PM, Cliff Wells <cliff at develix.com> wrote:

> On Tue, 2012-06-05 at 13:51 -0400, Brian Foster wrote:
> > I'm not in front of a server right now but check
> > in /usr/local/freeswitch/conf/autoload_confocal/switch.conf.xml and
> > see if there's a variable along the lines of Max DB handles. If it
> > just refers to odbc stuff you have your answer.
>
> There was just such a setting.  Thanks.
>
> >
> > Before you go much further, what was your call volume? It could have
> > thrown that error because something is locking the DB, so there could
> > be something wrong with it.
> >
>
> When total channels exceeds 400ish (~65% inbound, ~35% outbound), system
> load suddenly spikes (under 350 or so, it's usually under 2 or 3, when
> we hit 400+, it will jump as high as 80).  At that point the system
> appears to be queueing calls as the volume will sometimes shoot up to as
> high as 900, even though that's not really possible (hard limit due to
> T1 count), so it seems that it might be blocking on some resource.
>
> Also caught this error:
>
> 2012-06-05 11:23:21.419535 [CRIT] switch_core_session.c:1861 Throttle
> Error! 1342
> 2012-06-05 11:23:21.520601 [CRIT] switch_time.c:957 Over Session Rate of
> 30!
>
> So I adjusted that upwards as well.
>
>
> 16 cores Xeon 3GHz, 16GB RAM, 6 drive SAS RAID 10
> Scientific Linux 6.1
> Freeswitch git pull from yesterday.
>
> Regards,
> Cliff
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From acrow at integrafin.co.uk  Tue Jun  5 22:38:13 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Tue, 05 Jun 2012 19:38:13 +0100
Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS*
 device?
In-Reply-To: <1994487283-1338552086-cardhu_decombobulator_blackberry.rim.net-234513875-@b26.c15.bise7.blackberry>
References: <CBE0018C.34B31%krice@freeswitch.org>
	<4FBAA1CF.5060907@integrafin.co.uk>
	<CAKbxfG_csVT5A89DYnVirVKRkar3RaCBVAxT4gzCABvNgboRCA@mail.gmail.com>
	<CAKzWOxUj-q2b_NN08F0OD6Je5oJXkUa7VPR84q27BT4hGN1kEA@mail.gmail.com>
	<4FBAE7AE.1000606@coppice.org>
	<CAGKPtBst3pfELgXX1UoLZg_KxHizcwtfJefnibj2XhH6Ko7_wQ@mail.gmail.com>
	<4FBE0E4A.6090408@integrafin.co.uk>
	<CAGKPtBt+WMHe9HA=2t0zq76ogToaXd4-8dEP7wYG7qDKw76K8A@mail.gmail.com>
	<1994487283-1338552086-cardhu_decombobulator_blackberry.rim.net-234513875-@b26.c15.bise7.blackberry>
Message-ID: <4FCE5215.9080006@integrafin.co.uk>

Hi,

I spent many hours playing with udev rules, as, sensibly, I don't run 
FreeSWITCH as root.

I could not get any devices created, not even those in /dev/tts/<ttsX> 
(running on Debian Squeeze)

Has anyone got this working as a non-root user? Cutting out t38modem is 
one layer less to go wrong and I'd like to use it.

All ideas received gratefully.

Cheers

Alex

On 01/06/12 13:01, davidwaf at gmail.com wrote:
> Sounds good to me. Cheers
> Sent from my BlackBerry?
>
> -----Original Message-----
> From: Anita Hall<anita.hall at simmortel.com>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Fri, 1 Jun 2012 17:21:05
> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS*
> 	device?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


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From jeremyc at ssimicro.com  Tue Jun  5 23:44:29 2012
From: jeremyc at ssimicro.com (Jeremy Childs)
Date: Tue, 05 Jun 2012 13:44:29 -0600
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <CAKbxfG_QaKgU3Cw=HgQrYLvEi0dvoypgUfjqt3--S4Bd118Rag@mail.gmail.com>
References: <4FC896B8.9070102@digitalmail.com>
	<CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>
	<CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>
	<CAKzWOxUp5h2tMFP9yHJM4LaMmF1f+sXsOfDAcETwzaDG_2WgPA@mail.gmail.com>
	<CALNZuw=5W_k2MtJPxHn6HBZanCZNk83TXiMZTEGjZ9kRjb1G4w@mail.gmail.com>
	<CAKzWOxWH2nM37ywvkyimobkLJqAaUX531w4YgE=3eqVU5q7y=Q@mail.gmail.com>
	<CALNZuwkxN1JXVutNWuAysLgu_kTw40L50L37U-4tZYNV_Yadrg@mail.gmail.com>
	<CAKzWOxVHSGrTqoqTXnF6__1q=_zEFsUzmBhYhb40pEOoVy8gjg@mail.gmail.com>
	<CA+GmxbYZ0fVusbdqDNeLUxwi94aDaK8SjPRU+py-V+JRL6TUqQ@mail.gmail.com>
	<4FCCDACC.5070304@ssimicro.com>
	<CAKbxfG_QaKgU3Cw=HgQrYLvEi0dvoypgUfjqt3--S4Bd118Rag@mail.gmail.com>
Message-ID: <4FCE619D.2090501@ssimicro.com>

Updated to the latest git, so I can verify:

{ignore_early_media=ring_ready,loopback_bowout_on_execute=true}loopback/foo

does the right thing


{ignore_early_media=ring_ready}loopback/foo

Bridges all 3 legs together immediately (not the right thing).



On 12-06-04 11:03 AM, Anthony Minessale wrote:
> This thread is all FUD now, please disregard above.
>
>
> att_xfer and loopback endpoint are both crutches but they work at the
> cost of elegance since you are doing emulated behaviors.
>
> Your problem probably comes from the loopback bow-out happening too
> soon that tries to cut its way out of the call..
>
> You should be looking at logs and finding the root of the problem not
> guessing at things.
>
> My recommendation:
>
> 1) Make sure you are on latest GIT we have fixed a few issues in both
> in the recent past.
> 2) try {ignore_early_media=ring_ready,loopback_bowout_on_execute=true}loopback/foo
> in your dialstring  --- This may eliminate the loopback right off the
> bat.
> 3) try {ignore_early_media=ring_ready}}loopback/foo ---- it may just
> need to wait for the call to be answered.
> 4) try {loopback_bowout=false}loopback/foo --- the bummer here is it
> will never eliminate loopback
>
>
> On Mon, Jun 4, 2012 at 10:57 AM, Jeremy Childs<jeremyc at ssimicro.com>  wrote:
>> I'm also very interested in some answers! I've been bitten by att_xfer in
>> the past.
>>
>> Is getting dialplan processing working within the att_xfer function
>> possible, and just a low priority for implementation, or is there some
>> technical reason why this is not feasible?
>>
>> Lastly, is this somewhere that execute_extension can help? I've never been
>> able to get it to work inside att_xfer.
>>
>>
>>
>> On 12-06-04 8:53 AM, Dmitry Sytchev wrote:
>>
>> I asked this question many times on mailing list, and now I'm sure this
>> can't be really done with loopback.
>> The only alternative for loopback is to re-inject call into FS via some
>> separate Sofia profile, and specify that profile in string for att_xfer.
>> This brings up large amount of troubles including DTMF transcoding,
>> sequential att_xfer attempt recognition and overall voice/dtmf delay
>> introduced by chained channels. Maybe some channels can be moved out of
>> scene by using 'simplify' api on correct channels, but this needs tests.
>>
>> Anyway, loopback channel in FS is completely unusable, so we do need to have
>> some  best practices on how to do things without it in FS wiki... Maybe I
>> have time and will describe our experience soon.
>>
>> 2012/6/4 Michael Collins<msc at freeswitch.org>
>>>
>>>
>>> On Sun, Jun 3, 2012 at 2:15 PM, Avi Marcus<avi at avimarcus.net>  wrote:
>>>> I know you can do anything in the dialstring. But intended feature is to
>>>> allow the user to do an attended transfer to any number that they could
>>>> reach via the default calling. The default outbound path already has a LOT
>>>> of stuff set up and it would be impossible to duplicate that within a SINGLE
>>>> dialstring in a function call.
>>>> What is needed is for an att_xfer to be able to have leg C hit the
>>>> dialplan and bridged however a "normal" leg B to that number would be
>>>> called.
>>>> Does this make sense?
>>> Perhaps, but I remain unconvinced that this scenario is impossible without
>>> loopback. How about the OP actually supply a sample Lua script and dialplan
>>> and call log? I'd be willing to wager that the gurus could come up with a
>>> non-evil alternative that actually works. Just because loopback seems like a
>>> clean solution doesn't necessarily mean that it is. I'll leave it to Anthony
>>> to give the technical reasons why loopback doesn't always work as one would
>>> expect or why it should be avoided wherever possible.
>>>
>>> -MC
>>>
>>>> -Avi
>>>>
>>>>
>>>>
>>>> On Mon, Jun 4, 2012 at 12:02 AM, Michael Collins<msc at freeswitch.org>
>>>> wrote:
>>>>> Au contraire mon frere!
>>>>>
>>>>> You can do multiple things in a dialstring, like setting channel
>>>>> variables. You can also use execute_on_ring/media/answer to execute the
>>>>> extension with doing all the loopback overhead.
>>>>>
>>>>> I propose an experiment: provide a dialplan and loopback dialstring and
>>>>> we'll see if we can't give you a non-loopbackish alternative.
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>>> On Sun, Jun 3, 2012 at 1:54 PM, Avi Marcus<avi at avimarcus.net>  wrote:
>>>>>> ... all the normal dialplan handling. Setting CID, options, LCR stuff,
>>>>>> billing controls.
>>>>>> -Avi
>>>>>>
>>>>>>
>>>>>> On Sun, Jun 3, 2012 at 11:40 PM, Michael Collins<msc at freeswitch.org>
>>>>>> wrote:
>>>>>>> Let me rephrase...
>>>>>>>
>>>>>>> Since loopback is generally evil and should be avoided wherever
>>>>>>> possible, what does loopback give you that you can't get from doing a normal
>>>>>>> dialstring?
>>>>>>> -MC
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Jun 2, 2012 at 11:13 AM, Avi Marcus<avi at avimarcus.net>  wrote:
>>>>>>>> ... because att_xfer seems to require a "sofia/$profile/$destination"
>>>>>>>> directive, and he just wants the call to hit the dialplan.
>>>>>>>>
>>>>>>>> -Avi
>>>>>>>>
>>>>>>>>
>>>>>>>> On Fri, Jun 1, 2012 at 8:13 PM, Michael Collins<msc at freeswitch.org>
>>>>>>>> wrote:
>>>>>>>>> Why do you need to use loopback at all?
>>>>>>>>> -MC
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Fri, Jun 1, 2012 at 3:17 AM, Alex Lake<alex at digitalmail.com>
>>>>>>>>> wrote:
>>>>>>>>>> Got a lua script for a B-party "mid-call menu". Is it legitimate to
>>>>>>>>>> do..
>>>>>>>>>> "session:execute("att_xfer", "loopback/"..destnum)"
>>>>>>>>>>
>>>>>>>>>> I've tried it and it seems to start off doing the right things, but
>>>>>>>>>> my
>>>>>>>>>> A-party gets disconnected as soon as the call to the C-Party (the
>>>>>>>>>> person
>>>>>>>>>> I'm transferring the call to) answers the call.
>>>>>>>>>>
>>>>>>>>>> Maybe better to try to orchestrate the entire affair from within
>>>>>>>>>> the lua
>>>>>>>>>> script? (Tricky for a beginner like me!)
>>>>>>>>>>
>>>>>>>>>> Thanks,
>>>>>>>>>> Alex
>>>>>>>>>
>>>>>>>>>
>>>>>>>
>>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.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 Jun  5 23:54:31 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jun 2012 12:54:31 -0700
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <4FCE619D.2090501@ssimicro.com>
References: <4FC896B8.9070102@digitalmail.com>
	<CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>
	<CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>
	<CAKzWOxUp5h2tMFP9yHJM4LaMmF1f+sXsOfDAcETwzaDG_2WgPA@mail.gmail.com>
	<CALNZuw=5W_k2MtJPxHn6HBZanCZNk83TXiMZTEGjZ9kRjb1G4w@mail.gmail.com>
	<CAKzWOxWH2nM37ywvkyimobkLJqAaUX531w4YgE=3eqVU5q7y=Q@mail.gmail.com>
	<CALNZuwkxN1JXVutNWuAysLgu_kTw40L50L37U-4tZYNV_Yadrg@mail.gmail.com>
	<CAKzWOxVHSGrTqoqTXnF6__1q=_zEFsUzmBhYhb40pEOoVy8gjg@mail.gmail.com>
	<CA+GmxbYZ0fVusbdqDNeLUxwi94aDaK8SjPRU+py-V+JRL6TUqQ@mail.gmail.com>
	<4FCCDACC.5070304@ssimicro.com>
	<CAKbxfG_QaKgU3Cw=HgQrYLvEi0dvoypgUfjqt3--S4Bd118Rag@mail.gmail.com>
	<4FCE619D.2090501@ssimicro.com>
Message-ID: <CAKzWOxXEuQpGCDWK6zv_usOSxbUc_NncbrBYPOX_DUUjvG9xZA@mail.gmail.com>

We appreciate the validation. Any chance you could find a place on the wiki
to add this hard-earned knowledge?
-MC

On Tue, Jun 5, 2012 at 12:44 PM, Jeremy Childs <jeremyc at ssimicro.com> wrote:

> Updated to the latest git, so I can verify:
>
> {ignore_early_media=ring_ready,loopback_bowout_on_execute=true}loopback/foo
>
> does the right thing
>
>
> {ignore_early_media=ring_ready}loopback/foo
>
> Bridges all 3 legs together immediately (not the right thing).
>
>
>
> On 12-06-04 11:03 AM, Anthony Minessale wrote:
> > This thread is all FUD now, please disregard above.
> >
> >
> > att_xfer and loopback endpoint are both crutches but they work at the
> > cost of elegance since you are doing emulated behaviors.
> >
> > Your problem probably comes from the loopback bow-out happening too
> > soon that tries to cut its way out of the call..
> >
> > You should be looking at logs and finding the root of the problem not
> > guessing at things.
> >
> > My recommendation:
> >
> > 1) Make sure you are on latest GIT we have fixed a few issues in both
> > in the recent past.
> > 2) try
> {ignore_early_media=ring_ready,loopback_bowout_on_execute=true}loopback/foo
> > in your dialstring  --- This may eliminate the loopback right off the
> > bat.
> > 3) try {ignore_early_media=ring_ready}}loopback/foo ---- it may just
> > need to wait for the call to be answered.
> > 4) try {loopback_bowout=false}loopback/foo --- the bummer here is it
> > will never eliminate loopback
> >
> >
> > On Mon, Jun 4, 2012 at 10:57 AM, Jeremy Childs<jeremyc at ssimicro.com>
>  wrote:
> >> I'm also very interested in some answers! I've been bitten by att_xfer
> in
> >> the past.
> >>
> >> Is getting dialplan processing working within the att_xfer function
> >> possible, and just a low priority for implementation, or is there some
> >> technical reason why this is not feasible?
> >>
> >> Lastly, is this somewhere that execute_extension can help? I've never
> been
> >> able to get it to work inside att_xfer.
> >>
> >>
> >>
> >> On 12-06-04 8:53 AM, Dmitry Sytchev wrote:
> >>
> >> I asked this question many times on mailing list, and now I'm sure this
> >> can't be really done with loopback.
> >> The only alternative for loopback is to re-inject call into FS via some
> >> separate Sofia profile, and specify that profile in string for att_xfer.
> >> This brings up large amount of troubles including DTMF transcoding,
> >> sequential att_xfer attempt recognition and overall voice/dtmf delay
> >> introduced by chained channels. Maybe some channels can be moved out of
> >> scene by using 'simplify' api on correct channels, but this needs tests.
> >>
> >> Anyway, loopback channel in FS is completely unusable, so we do need to
> have
> >> some  best practices on how to do things without it in FS wiki... Maybe
> I
> >> have time and will describe our experience soon.
> >>
> >> 2012/6/4 Michael Collins<msc at freeswitch.org>
> >>>
> >>>
> >>> On Sun, Jun 3, 2012 at 2:15 PM, Avi Marcus<avi at avimarcus.net>  wrote:
> >>>> I know you can do anything in the dialstring. But intended feature is
> to
> >>>> allow the user to do an attended transfer to any number that they
> could
> >>>> reach via the default calling. The default outbound path already has
> a LOT
> >>>> of stuff set up and it would be impossible to duplicate that within a
> SINGLE
> >>>> dialstring in a function call.
> >>>> What is needed is for an att_xfer to be able to have leg C hit the
> >>>> dialplan and bridged however a "normal" leg B to that number would be
> >>>> called.
> >>>> Does this make sense?
> >>> Perhaps, but I remain unconvinced that this scenario is impossible
> without
> >>> loopback. How about the OP actually supply a sample Lua script and
> dialplan
> >>> and call log? I'd be willing to wager that the gurus could come up
> with a
> >>> non-evil alternative that actually works. Just because loopback seems
> like a
> >>> clean solution doesn't necessarily mean that it is. I'll leave it to
> Anthony
> >>> to give the technical reasons why loopback doesn't always work as one
> would
> >>> expect or why it should be avoided wherever possible.
> >>>
> >>> -MC
> >>>
> >>>> -Avi
> >>>>
> >>>>
> >>>>
> >>>> On Mon, Jun 4, 2012 at 12:02 AM, Michael Collins<msc at freeswitch.org>
> >>>> wrote:
> >>>>> Au contraire mon frere!
> >>>>>
> >>>>> You can do multiple things in a dialstring, like setting channel
> >>>>> variables. You can also use execute_on_ring/media/answer to execute
> the
> >>>>> extension with doing all the loopback overhead.
> >>>>>
> >>>>> I propose an experiment: provide a dialplan and loopback dialstring
> and
> >>>>> we'll see if we can't give you a non-loopbackish alternative.
> >>>>>
> >>>>> -MC
> >>>>>
> >>>>>
> >>>>> On Sun, Jun 3, 2012 at 1:54 PM, Avi Marcus<avi at avimarcus.net>
>  wrote:
> >>>>>> ... all the normal dialplan handling. Setting CID, options, LCR
> stuff,
> >>>>>> billing controls.
> >>>>>> -Avi
> >>>>>>
> >>>>>>
> >>>>>> On Sun, Jun 3, 2012 at 11:40 PM, Michael Collins<msc at freeswitch.org
> >
> >>>>>> wrote:
> >>>>>>> Let me rephrase...
> >>>>>>>
> >>>>>>> Since loopback is generally evil and should be avoided wherever
> >>>>>>> possible, what does loopback give you that you can't get from
> doing a normal
> >>>>>>> dialstring?
> >>>>>>> -MC
> >>>>>>>
> >>>>>>>
> >>>>>>> On Sat, Jun 2, 2012 at 11:13 AM, Avi Marcus<avi at avimarcus.net>
>  wrote:
> >>>>>>>> ... because att_xfer seems to require a
> "sofia/$profile/$destination"
> >>>>>>>> directive, and he just wants the call to hit the dialplan.
> >>>>>>>>
> >>>>>>>> -Avi
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> On Fri, Jun 1, 2012 at 8:13 PM, Michael Collins<
> msc at freeswitch.org>
> >>>>>>>> wrote:
> >>>>>>>>> Why do you need to use loopback at all?
> >>>>>>>>> -MC
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>> On Fri, Jun 1, 2012 at 3:17 AM, Alex Lake<alex at digitalmail.com>
> >>>>>>>>> wrote:
> >>>>>>>>>> Got a lua script for a B-party "mid-call menu". Is it
> legitimate to
> >>>>>>>>>> do..
> >>>>>>>>>> "session:execute("att_xfer", "loopback/"..destnum)"
> >>>>>>>>>>
> >>>>>>>>>> I've tried it and it seems to start off doing the right things,
> but
> >>>>>>>>>> my
> >>>>>>>>>> A-party gets disconnected as soon as the call to the C-Party
> (the
> >>>>>>>>>> person
> >>>>>>>>>> I'm transferring the call to) answers the call.
> >>>>>>>>>>
> >>>>>>>>>> Maybe better to try to orchestrate the entire affair from within
> >>>>>>>>>> the lua
> >>>>>>>>>> script? (Tricky for a beginner like me!)
> >>>>>>>>>>
> >>>>>>>>>> Thanks,
> >>>>>>>>>> Alex
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>
> >>>>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> Join Us At ClueCon - Aug 7-9, 2012
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> --
> >> Best regards,
> >>
> >> Dmitry Sytchev,
> >> IT Engineer
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jeremyc at ssimicro.com  Tue Jun  5 23:59:45 2012
From: jeremyc at ssimicro.com (Jeremy Childs)
Date: Tue, 05 Jun 2012 13:59:45 -0600
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <CAKzWOxXEuQpGCDWK6zv_usOSxbUc_NncbrBYPOX_DUUjvG9xZA@mail.gmail.com>
References: <4FC896B8.9070102@digitalmail.com>
	<CAKzWOxVZesFbHR8ojtB6kBOtHsSDu7kCOrT1-CfpaPPwziD7-g@mail.gmail.com>
	<CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>
	<CAKzWOxUp5h2tMFP9yHJM4LaMmF1f+sXsOfDAcETwzaDG_2WgPA@mail.gmail.com>
	<CALNZuw=5W_k2MtJPxHn6HBZanCZNk83TXiMZTEGjZ9kRjb1G4w@mail.gmail.com>
	<CAKzWOxWH2nM37ywvkyimobkLJqAaUX531w4YgE=3eqVU5q7y=Q@mail.gmail.com>
	<CALNZuwkxN1JXVutNWuAysLgu_kTw40L50L37U-4tZYNV_Yadrg@mail.gmail.com>
	<CAKzWOxVHSGrTqoqTXnF6__1q=_zEFsUzmBhYhb40pEOoVy8gjg@mail.gmail.com>
	<CA+GmxbYZ0fVusbdqDNeLUxwi94aDaK8SjPRU+py-V+JRL6TUqQ@mail.gmail.com>
	<4FCCDACC.5070304@ssimicro.com>
	<CAKbxfG_QaKgU3Cw=HgQrYLvEi0dvoypgUfjqt3--S4Bd118Rag@mail.gmail.com>
	<4FCE619D.2090501@ssimicro.com>
	<CAKzWOxXEuQpGCDWK6zv_usOSxbUc_NncbrBYPOX_DUUjvG9xZA@mail.gmail.com>
Message-ID: <4FCE6531.7050600@ssimicro.com>

I'd love to document this (and more). Can I contact you off-list to work 
out the details?

On 12-06-05 1:54 PM, Michael Collins wrote:
> We appreciate the validation. Any chance you could find a place on the 
> wiki to add this hard-earned knowledge?
> -MC
>
> On Tue, Jun 5, 2012 at 12:44 PM, Jeremy Childs <jeremyc at ssimicro.com 
> <mailto:jeremyc at ssimicro.com>> wrote:
>
>     Updated to the latest git, so I can verify:
>
>     {ignore_early_media=ring_ready,loopback_bowout_on_execute=true}loopback/foo
>
>     does the right thing
>
>
>     {ignore_early_media=ring_ready}loopback/foo
>
>     Bridges all 3 legs together immediately (not the right thing).
>
>
>
>     On 12-06-04 11:03 AM, Anthony Minessale wrote:
>     > This thread is all FUD now, please disregard above.
>     >
>     >
>     > att_xfer and loopback endpoint are both crutches but they work
>     at the
>     > cost of elegance since you are doing emulated behaviors.
>     >
>     > Your problem probably comes from the loopback bow-out happening too
>     > soon that tries to cut its way out of the call..
>     >
>     > You should be looking at logs and finding the root of the
>     problem not
>     > guessing at things.
>     >
>     > My recommendation:
>     >
>     > 1) Make sure you are on latest GIT we have fixed a few issues in
>     both
>     > in the recent past.
>     > 2) try
>     {ignore_early_media=ring_ready,loopback_bowout_on_execute=true}loopback/foo
>     > in your dialstring  --- This may eliminate the loopback right
>     off the
>     > bat.
>     > 3) try {ignore_early_media=ring_ready}}loopback/foo ---- it may just
>     > need to wait for the call to be answered.
>     > 4) try {loopback_bowout=false}loopback/foo --- the bummer here is it
>     > will never eliminate loopback
>     >
>     >
>     > On Mon, Jun 4, 2012 at 10:57 AM, Jeremy
>     Childs<jeremyc at ssimicro.com <mailto:jeremyc at ssimicro.com>>  wrote:
>     >> I'm also very interested in some answers! I've been bitten by
>     att_xfer in
>     >> the past.
>     >>
>     >> Is getting dialplan processing working within the att_xfer function
>     >> possible, and just a low priority for implementation, or is
>     there some
>     >> technical reason why this is not feasible?
>     >>
>     >> Lastly, is this somewhere that execute_extension can help? I've
>     never been
>     >> able to get it to work inside att_xfer.
>     >>
>     >>
>     >>
>     >> On 12-06-04 8:53 AM, Dmitry Sytchev wrote:
>     >>
>     >> I asked this question many times on mailing list, and now I'm
>     sure this
>     >> can't be really done with loopback.
>     >> The only alternative for loopback is to re-inject call into FS
>     via some
>     >> separate Sofia profile, and specify that profile in string for
>     att_xfer.
>     >> This brings up large amount of troubles including DTMF transcoding,
>     >> sequential att_xfer attempt recognition and overall voice/dtmf
>     delay
>     >> introduced by chained channels. Maybe some channels can be
>     moved out of
>     >> scene by using 'simplify' api on correct channels, but this
>     needs tests.
>     >>
>     >> Anyway, loopback channel in FS is completely unusable, so we do
>     need to have
>     >> some  best practices on how to do things without it in FS
>     wiki... Maybe I
>     >> have time and will describe our experience soon.
>     >>
>     >> 2012/6/4 Michael Collins<msc at freeswitch.org
>     <mailto:msc at freeswitch.org>>
>     >>>
>     >>>
>     >>> On Sun, Jun 3, 2012 at 2:15 PM, Avi Marcus<avi at avimarcus.net
>     <mailto:avi at avimarcus.net>>  wrote:
>     >>>> I know you can do anything in the dialstring. But intended
>     feature is to
>     >>>> allow the user to do an attended transfer to any number that
>     they could
>     >>>> reach via the default calling. The default outbound path
>     already has a LOT
>     >>>> of stuff set up and it would be impossible to duplicate that
>     within a SINGLE
>     >>>> dialstring in a function call.
>     >>>> What is needed is for an att_xfer to be able to have leg C
>     hit the
>     >>>> dialplan and bridged however a "normal" leg B to that number
>     would be
>     >>>> called.
>     >>>> Does this make sense?
>     >>> Perhaps, but I remain unconvinced that this scenario is
>     impossible without
>     >>> loopback. How about the OP actually supply a sample Lua script
>     and dialplan
>     >>> and call log? I'd be willing to wager that the gurus could
>     come up with a
>     >>> non-evil alternative that actually works. Just because
>     loopback seems like a
>     >>> clean solution doesn't necessarily mean that it is. I'll leave
>     it to Anthony
>     >>> to give the technical reasons why loopback doesn't always work
>     as one would
>     >>> expect or why it should be avoided wherever possible.
>     >>>
>     >>> -MC
>     >>>
>     >>>> -Avi
>     >>>>
>     >>>>
>     >>>>
>     >>>> On Mon, Jun 4, 2012 at 12:02 AM, Michael
>     Collins<msc at freeswitch.org <mailto:msc at freeswitch.org>>
>     >>>> wrote:
>     >>>>> Au contraire mon frere!
>     >>>>>
>     >>>>> You can do multiple things in a dialstring, like setting channel
>     >>>>> variables. You can also use execute_on_ring/media/answer to
>     execute the
>     >>>>> extension with doing all the loopback overhead.
>     >>>>>
>     >>>>> I propose an experiment: provide a dialplan and loopback
>     dialstring and
>     >>>>> we'll see if we can't give you a non-loopbackish alternative.
>     >>>>>
>     >>>>> -MC
>     >>>>>
>     >>>>>
>     >>>>> On Sun, Jun 3, 2012 at 1:54 PM, Avi Marcus<avi at avimarcus.net
>     <mailto:avi at avimarcus.net>>  wrote:
>     >>>>>> ... all the normal dialplan handling. Setting CID, options,
>     LCR stuff,
>     >>>>>> billing controls.
>     >>>>>> -Avi
>     >>>>>>
>     >>>>>>
>     >>>>>> On Sun, Jun 3, 2012 at 11:40 PM, Michael
>     Collins<msc at freeswitch.org <mailto:msc at freeswitch.org>>
>     >>>>>> wrote:
>     >>>>>>> Let me rephrase...
>     >>>>>>>
>     >>>>>>> Since loopback is generally evil and should be avoided
>     wherever
>     >>>>>>> possible, what does loopback give you that you can't get
>     from doing a normal
>     >>>>>>> dialstring?
>     >>>>>>> -MC
>     >>>>>>>
>     >>>>>>>
>     >>>>>>> On Sat, Jun 2, 2012 at 11:13 AM, Avi
>     Marcus<avi at avimarcus.net <mailto:avi at avimarcus.net>>  wrote:
>     >>>>>>>> ... because att_xfer seems to require a
>     "sofia/$profile/$destination"
>     >>>>>>>> directive, and he just wants the call to hit the dialplan.
>     >>>>>>>>
>     >>>>>>>> -Avi
>     >>>>>>>>
>     >>>>>>>>
>     >>>>>>>> On Fri, Jun 1, 2012 at 8:13 PM, Michael
>     Collins<msc at freeswitch.org <mailto:msc at freeswitch.org>>
>     >>>>>>>> wrote:
>     >>>>>>>>> Why do you need to use loopback at all?
>     >>>>>>>>> -MC
>     >>>>>>>>>
>     >>>>>>>>>
>     >>>>>>>>> On Fri, Jun 1, 2012 at 3:17 AM, Alex
>     Lake<alex at digitalmail.com <mailto:alex at digitalmail.com>>
>     >>>>>>>>> wrote:
>     >>>>>>>>>> Got a lua script for a B-party "mid-call menu". Is it
>     legitimate to
>     >>>>>>>>>> do..
>     >>>>>>>>>> "session:execute("att_xfer", "loopback/"..destnum)"
>     >>>>>>>>>>
>     >>>>>>>>>> I've tried it and it seems to start off doing the right
>     things, but
>     >>>>>>>>>> my
>     >>>>>>>>>> A-party gets disconnected as soon as the call to the
>     C-Party (the
>     >>>>>>>>>> person
>     >>>>>>>>>> I'm transferring the call to) answers the call.
>     >>>>>>>>>>
>     >>>>>>>>>> Maybe better to try to orchestrate the entire affair
>     from within
>     >>>>>>>>>> the lua
>     >>>>>>>>>> script? (Tricky for a beginner like me!)
>     >>>>>>>>>>
>     >>>>>>>>>> Thanks,
>     >>>>>>>>>> Alex
>     >>>>>>>>>
>     >>>>>>>>>
>     >>>>>>>
>     >>>>>
>     >>>
>     >>>
>     _________________________________________________________________________
>     >>> Professional FreeSWITCH Consulting Services:
>     >>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     >>> http://www.freeswitchsolutions.com
>     >>>
>     >>> 
>     >>> 
>     >>>
>     >>> Official FreeSWITCH Sites
>     >>> http://www.freeswitch.org
>     >>> http://wiki.freeswitch.org
>     >>> http://www.cluecon.com
>     >>>
>     >>> Join Us At ClueCon - Aug 7-9, 2012
>     >>>
>     >>> 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
>     >>>
>     >>
>     >>
>     >> --
>     >> Best regards,
>     >>
>     >> Dmitry Sytchev,
>     >> IT Engineer
>     >>
>     >>
>     >>
>     _________________________________________________________________________
>     >> Professional FreeSWITCH Consulting Services:
>     >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     >> http://www.freeswitchsolutions.com
>     >>
>     >> 
>     >> 
>     >>
>     >> Official FreeSWITCH Sites
>     >> http://www.freeswitch.org
>     >> http://wiki.freeswitch.org
>     >> http://www.cluecon.com
>     >>
>     >> Join Us At ClueCon - Aug 7-9, 2012
>     >>
>     >> 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
>     >>
>     >>
>     >>
>     >>
>     _________________________________________________________________________
>     >> Professional FreeSWITCH Consulting Services:
>     >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     >> http://www.freeswitchsolutions.com
>     >>
>     >> 
>     >> 
>     >>
>     >> Official FreeSWITCH Sites
>     >> http://www.freeswitch.org
>     >> http://wiki.freeswitch.org
>     >> http://www.cluecon.com
>     >>
>     >> Join Us At ClueCon - Aug 7-9, 2012
>     >>
>     >> 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
>     >>
>     >
>     >
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From bdfoster at endigotech.com  Wed Jun  6 01:15:14 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 5 Jun 2012 17:15:14 -0400
Subject: [Freeswitch-users] att_xfer and loopback
In-Reply-To: <4FCE6531.7050600@ssimicro.com>
References: <4FC896B8.9070102@digitalmail.com>
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	<CALNZuw=JzsumSAOZAPDRexM2GZ4cF8JNFQg=vBcj1PFsjWOXhg@mail.gmail.com>
	<CAKzWOxUp5h2tMFP9yHJM4LaMmF1f+sXsOfDAcETwzaDG_2WgPA@mail.gmail.com>
	<CALNZuw=5W_k2MtJPxHn6HBZanCZNk83TXiMZTEGjZ9kRjb1G4w@mail.gmail.com>
	<CAKzWOxWH2nM37ywvkyimobkLJqAaUX531w4YgE=3eqVU5q7y=Q@mail.gmail.com>
	<CALNZuwkxN1JXVutNWuAysLgu_kTw40L50L37U-4tZYNV_Yadrg@mail.gmail.com>
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	<CA+GmxbYZ0fVusbdqDNeLUxwi94aDaK8SjPRU+py-V+JRL6TUqQ@mail.gmail.com>
	<4FCCDACC.5070304@ssimicro.com>
	<CAKbxfG_QaKgU3Cw=HgQrYLvEi0dvoypgUfjqt3--S4Bd118Rag@mail.gmail.com>
	<4FCE619D.2090501@ssimicro.com>
	<CAKzWOxXEuQpGCDWK6zv_usOSxbUc_NncbrBYPOX_DUUjvG9xZA@mail.gmail.com>
	<4FCE6531.7050600@ssimicro.com>
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Sign up for an account on the Wiki at wiki.freeswitch.org and go from there.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 5, 2012 4:00 PM, "Jeremy Childs" <jeremyc at ssimicro.com> wrote:

>  I'd love to document this (and more). Can I contact you off-list to work
> out the details?
>
> On 12-06-05 1:54 PM, Michael Collins wrote:
>
> We appreciate the validation. Any chance you could find a place on the
> wiki to add this hard-earned knowledge?
> -MC
>
> On Tue, Jun 5, 2012 at 12:44 PM, Jeremy Childs <jeremyc at ssimicro.com>wrote:
>
>> Updated to the latest git, so I can verify:
>>
>>
>> {ignore_early_media=ring_ready,loopback_bowout_on_execute=true}loopback/foo
>>
>> does the right thing
>>
>>
>> {ignore_early_media=ring_ready}loopback/foo
>>
>> Bridges all 3 legs together immediately (not the right thing).
>>
>>
>>
>> On 12-06-04 11:03 AM, Anthony Minessale wrote:
>> > This thread is all FUD now, please disregard above.
>> >
>> >
>> > att_xfer and loopback endpoint are both crutches but they work at the
>> > cost of elegance since you are doing emulated behaviors.
>> >
>> > Your problem probably comes from the loopback bow-out happening too
>> > soon that tries to cut its way out of the call..
>> >
>> > You should be looking at logs and finding the root of the problem not
>> > guessing at things.
>> >
>> > My recommendation:
>> >
>> > 1) Make sure you are on latest GIT we have fixed a few issues in both
>> > in the recent past.
>> > 2) try
>> {ignore_early_media=ring_ready,loopback_bowout_on_execute=true}loopback/foo
>> > in your dialstring  --- This may eliminate the loopback right off the
>> > bat.
>> > 3) try {ignore_early_media=ring_ready}}loopback/foo ---- it may just
>> > need to wait for the call to be answered.
>> > 4) try {loopback_bowout=false}loopback/foo --- the bummer here is it
>> > will never eliminate loopback
>> >
>> >
>> > On Mon, Jun 4, 2012 at 10:57 AM, Jeremy Childs<jeremyc at ssimicro.com>
>>  wrote:
>> >> I'm also very interested in some answers! I've been bitten by att_xfer
>> in
>> >> the past.
>> >>
>> >> Is getting dialplan processing working within the att_xfer function
>> >> possible, and just a low priority for implementation, or is there some
>> >> technical reason why this is not feasible?
>> >>
>> >> Lastly, is this somewhere that execute_extension can help? I've never
>> been
>> >> able to get it to work inside att_xfer.
>> >>
>> >>
>> >>
>> >> On 12-06-04 8:53 AM, Dmitry Sytchev wrote:
>> >>
>> >> I asked this question many times on mailing list, and now I'm sure this
>> >> can't be really done with loopback.
>> >> The only alternative for loopback is to re-inject call into FS via some
>> >> separate Sofia profile, and specify that profile in string for
>> att_xfer.
>> >> This brings up large amount of troubles including DTMF transcoding,
>> >> sequential att_xfer attempt recognition and overall voice/dtmf delay
>> >> introduced by chained channels. Maybe some channels can be moved out of
>> >> scene by using 'simplify' api on correct channels, but this needs
>> tests.
>> >>
>> >> Anyway, loopback channel in FS is completely unusable, so we do need
>> to have
>> >> some  best practices on how to do things without it in FS wiki...
>> Maybe I
>> >> have time and will describe our experience soon.
>> >>
>> >> 2012/6/4 Michael Collins<msc at freeswitch.org>
>> >>>
>> >>>
>> >>> On Sun, Jun 3, 2012 at 2:15 PM, Avi Marcus<avi at avimarcus.net>  wrote:
>> >>>> I know you can do anything in the dialstring. But intended feature
>> is to
>> >>>> allow the user to do an attended transfer to any number that they
>> could
>> >>>> reach via the default calling. The default outbound path already has
>> a LOT
>> >>>> of stuff set up and it would be impossible to duplicate that within
>> a SINGLE
>> >>>> dialstring in a function call.
>> >>>> What is needed is for an att_xfer to be able to have leg C hit the
>> >>>> dialplan and bridged however a "normal" leg B to that number would be
>> >>>> called.
>> >>>> Does this make sense?
>> >>> Perhaps, but I remain unconvinced that this scenario is impossible
>> without
>> >>> loopback. How about the OP actually supply a sample Lua script and
>> dialplan
>> >>> and call log? I'd be willing to wager that the gurus could come up
>> with a
>> >>> non-evil alternative that actually works. Just because loopback seems
>> like a
>> >>> clean solution doesn't necessarily mean that it is. I'll leave it to
>> Anthony
>> >>> to give the technical reasons why loopback doesn't always work as one
>> would
>> >>> expect or why it should be avoided wherever possible.
>> >>>
>> >>> -MC
>> >>>
>> >>>> -Avi
>> >>>>
>> >>>>
>> >>>>
>> >>>> On Mon, Jun 4, 2012 at 12:02 AM, Michael Collins<msc at freeswitch.org>
>> >>>> wrote:
>> >>>>> Au contraire mon frere!
>> >>>>>
>> >>>>> You can do multiple things in a dialstring, like setting channel
>> >>>>> variables. You can also use execute_on_ring/media/answer to execute
>> the
>> >>>>> extension with doing all the loopback overhead.
>> >>>>>
>> >>>>> I propose an experiment: provide a dialplan and loopback dialstring
>> and
>> >>>>> we'll see if we can't give you a non-loopbackish alternative.
>> >>>>>
>> >>>>> -MC
>> >>>>>
>> >>>>>
>> >>>>> On Sun, Jun 3, 2012 at 1:54 PM, Avi Marcus<avi at avimarcus.net>
>>  wrote:
>> >>>>>> ... all the normal dialplan handling. Setting CID, options, LCR
>> stuff,
>> >>>>>> billing controls.
>> >>>>>> -Avi
>> >>>>>>
>> >>>>>>
>> >>>>>> On Sun, Jun 3, 2012 at 11:40 PM, Michael Collins<
>> msc at freeswitch.org>
>> >>>>>> wrote:
>> >>>>>>> Let me rephrase...
>> >>>>>>>
>> >>>>>>> Since loopback is generally evil and should be avoided wherever
>> >>>>>>> possible, what does loopback give you that you can't get from
>> doing a normal
>> >>>>>>> dialstring?
>> >>>>>>> -MC
>> >>>>>>>
>> >>>>>>>
>> >>>>>>> On Sat, Jun 2, 2012 at 11:13 AM, Avi Marcus<avi at avimarcus.net>
>>  wrote:
>> >>>>>>>> ... because att_xfer seems to require a
>> "sofia/$profile/$destination"
>> >>>>>>>> directive, and he just wants the call to hit the dialplan.
>> >>>>>>>>
>> >>>>>>>> -Avi
>> >>>>>>>>
>> >>>>>>>>
>> >>>>>>>> On Fri, Jun 1, 2012 at 8:13 PM, Michael Collins<
>> msc at freeswitch.org>
>> >>>>>>>> wrote:
>> >>>>>>>>> Why do you need to use loopback at all?
>> >>>>>>>>> -MC
>> >>>>>>>>>
>> >>>>>>>>>
>> >>>>>>>>> On Fri, Jun 1, 2012 at 3:17 AM, Alex Lake<alex at digitalmail.com>
>> >>>>>>>>> wrote:
>> >>>>>>>>>> Got a lua script for a B-party "mid-call menu". Is it
>> legitimate to
>> >>>>>>>>>> do..
>> >>>>>>>>>> "session:execute("att_xfer", "loopback/"..destnum)"
>> >>>>>>>>>>
>> >>>>>>>>>> I've tried it and it seems to start off doing the right
>> things, but
>> >>>>>>>>>> my
>> >>>>>>>>>> A-party gets disconnected as soon as the call to the C-Party
>> (the
>> >>>>>>>>>> person
>> >>>>>>>>>> I'm transferring the call to) answers the call.
>> >>>>>>>>>>
>> >>>>>>>>>> Maybe better to try to orchestrate the entire affair from
>> within
>> >>>>>>>>>> the lua
>> >>>>>>>>>> script? (Tricky for a beginner like me!)
>> >>>>>>>>>>
>> >>>>>>>>>> Thanks,
>> >>>>>>>>>> Alex
>> >>>>>>>>>
>> >>>>>>>>>
>> >>>>>>>
>> >>>>>
>> >>>
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://wiki.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> Join Us At ClueCon - Aug 7-9, 2012
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> --
>> >> Best regards,
>> >>
>> >> Dmitry Sytchev,
>> >> IT Engineer
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> Join Us At ClueCon - Aug 7-9, 2012
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> Join Us At ClueCon - Aug 7-9, 2012
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cliff at develix.com  Wed Jun  6 02:22:49 2012
From: cliff at develix.com (Cliff Wells)
Date: Tue, 05 Jun 2012 15:22:49 -0700
Subject: [Freeswitch-users] Max handles 50 exceeded, blocking....
In-Reply-To: <CALNZuwkq4d=_RJHaUcNFF4RaPd42CmJfniCnmjY-2rdLX26FUA@mail.gmail.com>
References: <1338917776.1915.19.camel@portable-evil>
	<CAKV-jFwxb5hNX_9qj8Eo3TeMFPFimSRSh=aZ2KkM+3T=fEw3LA@mail.gmail.com>
	<1338920697.1915.33.camel@portable-evil>
	<CALNZuwkq4d=_RJHaUcNFF4RaPd42CmJfniCnmjY-2rdLX26FUA@mail.gmail.com>
Message-ID: <1338934969.1915.35.camel@portable-evil>

Looks like one of our profiles was getting flooded with too many CPS
(300+).  Added more profiles and it seems okay now (although won't know
for sure until peak hours tomorrow).

Thanks for all the tips.

Cliff


On Tue, 2012-06-05 at 21:35 +0300, Avi Marcus wrote:
> If you're writing the channels to the DB, you can either skip it with
> -nosql or you can ramcache the SQL DBs.
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#FreeSWITCH.27s_core.db_I.2FO_bottleneck
> 
> -Avi 
> 
> 
> 
> On Tue, Jun 5, 2012 at 9:24 PM, Cliff Wells <cliff at develix.com> wrote:
>         On Tue, 2012-06-05 at 13:51 -0400, Brian Foster wrote:
>         > I'm not in front of a server right now but check
>         >
>         in /usr/local/freeswitch/conf/autoload_confocal/switch.conf.xml and
>         > see if there's a variable along the lines of Max DB handles.
>         If it
>         > just refers to odbc stuff you have your answer.
>         
>         
>         There was just such a setting.  Thanks.
>         
>         >
>         > Before you go much further, what was your call volume? It
>         could have
>         > thrown that error because something is locking the DB, so
>         there could
>         > be something wrong with it.
>         >
>         
>         
>         When total channels exceeds 400ish (~65% inbound, ~35%
>         outbound), system
>         load suddenly spikes (under 350 or so, it's usually under 2 or
>         3, when
>         we hit 400+, it will jump as high as 80).  At that point the
>         system
>         appears to be queueing calls as the volume will sometimes
>         shoot up to as
>         high as 900, even though that's not really possible (hard
>         limit due to
>         T1 count), so it seems that it might be blocking on some
>         resource.
>         
>         Also caught this error:
>         
>         2012-06-05 11:23:21.419535 [CRIT] switch_core_session.c:1861
>         Throttle Error! 1342
>         2012-06-05 11:23:21.520601 [CRIT] switch_time.c:957 Over
>         Session Rate of 30!
>         
>         So I adjusted that upwards as well.
>         
>         
>         16 cores Xeon 3GHz, 16GB RAM, 6 drive SAS RAID 10
>         Scientific Linux 6.1
>         Freeswitch git pull from yesterday.
>         
>         Regards,
>         Cliff
>         
>         
>         
>         
>         
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org
>         http://www.freeswitchsolutions.com
>         
>         
>         
>         
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>         
>         Join Us At ClueCon - Aug 7-9, 2012
>         
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From drk at drkngs.net  Wed Jun  6 03:33:09 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Tue, 05 Jun 2012 16:33:09 -0700
Subject: [Freeswitch-users]
	=?iso-8859-1?q?how_to=3A_get_session_variable_?=
	=?iso-8859-1?q?after=09api=5Fhangup=5Fhook?=
In-Reply-To: <CABx_eV3-KAYSTaDC09uXpH-wWc1adJhgiAiDxg7yaRCwos73fg@mail.gmail.com>
Message-ID: <20120605233309.c52d99a3@mail.tritonwest.net>

Session is one of the parameters that get passed to API calls. If the API was invoked by a session, and not out of band, like form the console or event_socket, or any other way to invoke it, the argument will not be null. You can then get variables from the session handle just like you would do anywhere else.  
   
--Dave
      _____  

  From: cristian re [mailto:cristian.re.work at gmail.com]
To: freeswitch-users at lists.freeswitch.org
Sent: Tue, 05 Jun 2012 08:33:59 -0700
Subject: [Freeswitch-users] how to: get session variable after api_hangup_hook

  
  
  
Hello,  

  
I write a custom cpp module for freeswitch that exposes an API:  

  
SWITCH_ADD_API(commands_api_interface, "my_hangup", "my_hangup",  my_hangup_api , MY_HANGUP_USAGE);  

  
I want to call this api from dialplan (after bridge hangup) for reading the variable "billmsec".  
I put this into my dialplan:  

  
<action application="set" data="session_in_hangup_hook=true"/>  
<action application="set" data="api_hangup_hook=my_hangup_api"/>  

  
After hangup freeswitch call correctly my API but I have not figured out how to get the variable.  

  

  

  

  

      
   
 
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From bdfoster at endigotech.com  Wed Jun  6 04:10:32 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 5 Jun 2012 20:10:32 -0400
Subject: [Freeswitch-users] Max handles 50 exceeded, blocking....
In-Reply-To: <1338934969.1915.35.camel@portable-evil>
References: <1338917776.1915.19.camel@portable-evil>
	<CAKV-jFwxb5hNX_9qj8Eo3TeMFPFimSRSh=aZ2KkM+3T=fEw3LA@mail.gmail.com>
	<1338920697.1915.33.camel@portable-evil>
	<CALNZuwkq4d=_RJHaUcNFF4RaPd42CmJfniCnmjY-2rdLX26FUA@mail.gmail.com>
	<1338934969.1915.35.camel@portable-evil>
Message-ID: <CAKV-jFzS4yx+qXqrtVfm4sf=Ri58wmRDEteYh5NedyXJqFUh1A@mail.gmail.com>

If you're running that kind of call volume, you might consider taking Avi's
advice and go for a ramcache setup. I think your testing really the limits
of what you can handle with the DB being on a disk. Maybe not, it depends
on what all you are doing with the DB. I would look into that possibility
though, it may give you better results.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 5, 2012 6:23 PM, "Cliff Wells" <cliff at develix.com> wrote:

> Looks like one of our profiles was getting flooded with too many CPS
> (300+).  Added more profiles and it seems okay now (although won't know
> for sure until peak hours tomorrow).
>
> Thanks for all the tips.
>
> Cliff
>
>
> On Tue, 2012-06-05 at 21:35 +0300, Avi Marcus wrote:
> > If you're writing the channels to the DB, you can either skip it with
> > -nosql or you can ramcache the SQL DBs.
> >
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#FreeSWITCH.27s_core.db_I.2FO_bottleneck
> >
> > -Avi
> >
> >
> >
> > On Tue, Jun 5, 2012 at 9:24 PM, Cliff Wells <cliff at develix.com> wrote:
> >         On Tue, 2012-06-05 at 13:51 -0400, Brian Foster wrote:
> >         > I'm not in front of a server right now but check
> >         >
> >         in /usr/local/freeswitch/conf/autoload_confocal/switch.conf.xml
> and
> >         > see if there's a variable along the lines of Max DB handles.
> >         If it
> >         > just refers to odbc stuff you have your answer.
> >
> >
> >         There was just such a setting.  Thanks.
> >
> >         >
> >         > Before you go much further, what was your call volume? It
> >         could have
> >         > thrown that error because something is locking the DB, so
> >         there could
> >         > be something wrong with it.
> >         >
> >
> >
> >         When total channels exceeds 400ish (~65% inbound, ~35%
> >         outbound), system
> >         load suddenly spikes (under 350 or so, it's usually under 2 or
> >         3, when
> >         we hit 400+, it will jump as high as 80).  At that point the
> >         system
> >         appears to be queueing calls as the volume will sometimes
> >         shoot up to as
> >         high as 900, even though that's not really possible (hard
> >         limit due to
> >         T1 count), so it seems that it might be blocking on some
> >         resource.
> >
> >         Also caught this error:
> >
> >         2012-06-05 11:23:21.419535 [CRIT] switch_core_session.c:1861
> >         Throttle Error! 1342
> >         2012-06-05 11:23:21.520601 [CRIT] switch_time.c:957 Over
> >         Session Rate of 30!
> >
> >         So I adjusted that upwards as well.
> >
> >
> >         16 cores Xeon 3GHz, 16GB RAM, 6 drive SAS RAID 10
> >         Scientific Linux 6.1
> >         Freeswitch git pull from yesterday.
> >
> >         Regards,
> >         Cliff
> >
> >
> >
> >
> >
> >
> _________________________________________________________________________
> >         Professional FreeSWITCH Consulting Services:
> >         consulting at freeswitch.org
> >         http://www.freeswitchsolutions.com
> >
> >         
> >         
> >
> >         Official FreeSWITCH Sites
> >         http://www.freeswitch.org
> >         http://wiki.freeswitch.org
> >         http://www.cluecon.com
> >
> >         Join Us At ClueCon - Aug 7-9, 2012
> >
> >         FreeSWITCH-users mailing list
> >         FreeSWITCH-users at lists.freeswitch.org
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From cliff at develix.com  Wed Jun  6 04:59:07 2012
From: cliff at develix.com (Cliff Wells)
Date: Tue, 05 Jun 2012 17:59:07 -0700
Subject: [Freeswitch-users] Max handles 50 exceeded, blocking....
In-Reply-To: <CAKV-jFzS4yx+qXqrtVfm4sf=Ri58wmRDEteYh5NedyXJqFUh1A@mail.gmail.com>
References: <1338917776.1915.19.camel@portable-evil>
	<CAKV-jFwxb5hNX_9qj8Eo3TeMFPFimSRSh=aZ2KkM+3T=fEw3LA@mail.gmail.com>
	<1338920697.1915.33.camel@portable-evil>
	<CALNZuwkq4d=_RJHaUcNFF4RaPd42CmJfniCnmjY-2rdLX26FUA@mail.gmail.com>
	<1338934969.1915.35.camel@portable-evil>
	<CAKV-jFzS4yx+qXqrtVfm4sf=Ri58wmRDEteYh5NedyXJqFUh1A@mail.gmail.com>
Message-ID: <1338944347.2103.0.camel@portable-evil>

On Tue, 2012-06-05 at 20:10 -0400, Brian Foster wrote:
> If you're running that kind of call volume, you might consider taking
> Avi's advice and go for a ramcache setup. I think your testing really
> the limits of what you can handle with the DB being on a disk. Maybe
> not, it depends on what all you are doing with the DB. I would look
> into that possibility though, it may give you better results.

I did that as well.  

Regards,
Cliff





From albert_nguyen16 at hotmail.com  Wed Jun  6 05:29:02 2012
From: albert_nguyen16 at hotmail.com (Albert Nguyen)
Date: Wed, 6 Jun 2012 01:29:02 +0000
Subject: [Freeswitch-users] when to use stop_dtmf
In-Reply-To: <3820475734403626844@unknownmsgid>
References: <20120527181143.62d2a620@mail.tritonwest.net>,
	<3820475734403626844@unknownmsgid>
Message-ID: <BAY170-W5845762FB9A832B9BC8F9E950D0@phx.gbl>


 
Hi,
 
I am using start_DTMF to convert inband DTMF to RFC 2833 for calls. Just wanted to make sure :-), when a call is ended by FS (i.e [CS_DESTROY]), would this will also automatically stop the DTMF conversion function or do I have to explicitly using stop_DTMF ? 
 
Thanks in advance.
 
Al 		 	   		  
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From ahe.sanath at gmail.com  Wed Jun  6 08:31:24 2012
From: ahe.sanath at gmail.com (Sanath Prasanna)
Date: Wed, 6 Jun 2012 10:01:24 +0530
Subject: [Freeswitch-users] "originate" Command not found in fs_cli
In-Reply-To: <CAKV-jFztkkESpAXZbrB-htebmzNnf2JAormnSChPrn0-_kHiAQ@mail.gmail.com>
References: <CALQZz0xQA0mP31=kbszjYB55it5gMU1mnUj5bNWuTR4FmSFuVQ@mail.gmail.com>
	<CAKV-jFztkkESpAXZbrB-htebmzNnf2JAormnSChPrn0-_kHiAQ@mail.gmail.com>
Message-ID: <CALQZz0xacCGTQ7sijfCakPYWc44AP+UDpYNrip53VK6sOEefjA@mail.gmail.com>

Hi
Tx for advice. Here is the my profile
vi sip_profiles/external/ivr.xml
<include>
  <gateway name="ivr">
    <!-- Replace these value with your iCall Carrier Services username and
password. -->
    <!-- Even if you use an IP-based sub-account, FreeSWITCH needs these
values -->
    <param name="password" value="1234" />
    <param name="proxy" value="$${operator_sip_ip}" />
    <param name="realm" value="$${operator_realm}" />
    <!-- Set to "false" for IP-based accounts, or "true" for registration
based -->
   <!-- Here I need  IP-based account.Because operator is not provided
registration based sip account.-->
    <param name="register" value="false" />
    <!-- <param name="ping" value="30" /> -->
  </gateway>
</include>

 vi dialplan/outbound.xml

<include>
  <context name="outbound">
   <extension name="outbound">
      <condition field="destination_number" expression="^(54\d{9})$">
        <action application="bridge" data="sofia/gateway/ivr/$1"/>
      </condition>
    </extension>
  </context>
</include>
~

still same problem come
===================
freeswitch at internal> originate sofia/gateway/ivr/14722796816 54722796817
-ERR originate sofia/gateway/dtl/ 14722796816   54722796817 Command not
found!

also when run reloadxml
freeswitch at internal> reloadxml
-ERR reloadxml Command not found!
So when change conf, need to restart service every time.

Pls advice to resolve this problems.
Br,
Sanath


On Tue, Jun 5, 2012 at 10:46 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> Your not understanding the dialstring aspect...
>
> Example:
>
> originate sofia/gateway/flowroute/13175551212 user/1000 at 192.168.1.79
>
> Where sofia is the channel type, gateway is a keyword, flowroute is what I
> named my gateway, 13175551212 is the number I'm calling through my
> gateway flowroute, and user/1000 is the extension to send it to and
> 192.168.1.79 is the domain that user is attached to.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 5, 2012 8:49 AM, "Sanath Prasanna" <ahe.sanath at gmail.com> wrote:
>
>> Hi,
>> when I run originate command in fs_cli following error coming. Pls advice
>> to resolve that.
>>
>> freeswitch at internal> originate sofia/external/ivr/9179123456 9179123458
>> -ERR originate  sofia/external/ivr/9179123456 9179123458 Command not
>> found!
>>
>> Br,
>> Sanath
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ahe.sanath at gmail.com  Wed Jun  6 08:34:34 2012
From: ahe.sanath at gmail.com (Sanath Prasanna)
Date: Wed, 6 Jun 2012 10:04:34 +0530
Subject: [Freeswitch-users] opensource project voicechat for freeswitch
In-Reply-To: <Pine.LNX.4.64.1206041700001.22234@tomback.antagus.de>
References: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
	<2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com>
	<1338558318.79806.YahooMailNeo@web120405.mail.ne1.yahoo.com>
	<Pine.LNX.4.64.1206041700001.22234@tomback.antagus.de>
Message-ID: <CALQZz0zLpRBnFu3jce07Ycq=mn_OnAGrVFZgm-ibY_3eGg93Xw@mail.gmail.com>

Hi Thomas,
I like to help as LUA programmer. Pls let me know the which task need to
doing in LUA
Br,
Sanath

On Mon, Jun 4, 2012 at 8:57 PM, Thomas Hoellriegel <admin at blindi.net> wrote:

> Hi all,
>
> I like to plan a opensourceproject.
> Open_voicechat is the first open source voice chat worldwide.
> Freeswitch and other voip-systems don.t have a big application.
>
> Unfortunately, I can not even work, programming.
> I am looking for people to help me, and dominate one of the following
> programming languages:
> perl lua java php python or ruby.
> There is no specific programming language requested.
>
> I have setup a big mysql database for all modular features of a nice chat.
> Mailboxes, news forums, line managment and more!
> An exact documented functionplan  has already been added.
> A german demo voicefile has ben included to the archive.
>
> You can download my long work at:
> http://www.blindi.net/**downloads/open_vocechat.tar.gz<http://www.blindi.net/downloads/open_vocechat.tar.gz>
> I have 2 Languages german and english added to the archive.
>
>
> I would be happy if someone help me.
> The advantage of FreeSWITCH is the flexibility.
>
> Thanks for your nice help
>
>
>
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/**listinfo/blinde-misc<http://www.blindi.net/mailman/listinfo/blinde-misc>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Jun  6 09:06:33 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jun 2012 22:06:33 -0700
Subject: [Freeswitch-users] when to use stop_dtmf
In-Reply-To: <BAY170-W5845762FB9A832B9BC8F9E950D0@phx.gbl>
References: <20120527181143.62d2a620@mail.tritonwest.net>
	<3820475734403626844@unknownmsgid>
	<BAY170-W5845762FB9A832B9BC8F9E950D0@phx.gbl>
Message-ID: <CAKzWOxXoW5hPTi9hWn-SrZsuYALUm_3L8TvpqMyZgtMAJMQ7rA@mail.gmail.com>

If you are using the dialplan application "start_dtmf" then it applies only
to the current, existing channel. So no, you don't have to explicitly call
stop_dtmf, especially after the call has ended! :)

-MC

On Tue, Jun 5, 2012 at 6:29 PM, Albert Nguyen
<albert_nguyen16 at hotmail.com>wrote:

>
> Hi,
>
> I am using start_DTMF to convert inband DTMF to RFC 2833 for calls. Just
> wanted to make sure :-), when a call is ended by FS (i.e [CS_DESTROY]),
> would this will also automatically stop the DTMF conversion function or do
> I have to explicitly using stop_DTMF ?
>
> Thanks in advance.
>
> Al
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Jun  6 09:08:58 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jun 2012 22:08:58 -0700
Subject: [Freeswitch-users] "originate" Command not found in fs_cli
In-Reply-To: <CALQZz0xacCGTQ7sijfCakPYWc44AP+UDpYNrip53VK6sOEefjA@mail.gmail.com>
References: <CALQZz0xQA0mP31=kbszjYB55it5gMU1mnUj5bNWuTR4FmSFuVQ@mail.gmail.com>
	<CAKV-jFztkkESpAXZbrB-htebmzNnf2JAormnSChPrn0-_kHiAQ@mail.gmail.com>
	<CALQZz0xacCGTQ7sijfCakPYWc44AP+UDpYNrip53VK6sOEefjA@mail.gmail.com>
Message-ID: <CAKzWOxXiuP-s8ewW4gH5kEuuNYWE0tp7vbt7tVS+m_5FOMP4zg@mail.gmail.com>

This is a bigger problem than originate. It looks like none of your
commands are working. I would shut down FS and restart it in the
foreground, i.e. don't use the "-nc" flag. Watch the console and see what
errors pop up. I'm curious to know if there's an issue with mod_commands or
something like that.

-MC

On Tue, Jun 5, 2012 at 9:31 PM, Sanath Prasanna <ahe.sanath at gmail.com>wrote:

> Hi
> Tx for advice. Here is the my profile
> vi sip_profiles/external/ivr.xml
> <include>
>   <gateway name="ivr">
>     <!-- Replace these value with your iCall Carrier Services username and
> password. -->
>     <!-- Even if you use an IP-based sub-account, FreeSWITCH needs these
> values -->
>     <param name="password" value="1234" />
>     <param name="proxy" value="$${operator_sip_ip}" />
>     <param name="realm" value="$${operator_realm}" />
>     <!-- Set to "false" for IP-based accounts, or "true" for registration
> based -->
>    <!-- Here I need  IP-based account.Because operator is not provided
> registration based sip account.-->
>     <param name="register" value="false" />
>     <!-- <param name="ping" value="30" /> -->
>   </gateway>
> </include>
>
>  vi dialplan/outbound.xml
>
> <include>
>   <context name="outbound">
>    <extension name="outbound">
>       <condition field="destination_number" expression="^(54\d{9})$">
>         <action application="bridge" data="sofia/gateway/ivr/$1"/>
>       </condition>
>     </extension>
>   </context>
> </include>
> ~
>
> still same problem come
> ===================
> freeswitch at internal> originate sofia/gateway/ivr/14722796816 54722796817
> -ERR originate sofia/gateway/dtl/ 14722796816   54722796817 Command not
> found!
>
> also when run reloadxml
> freeswitch at internal> reloadxml
> -ERR reloadxml Command not found!
> So when change conf, need to restart service every time.
>
> Pls advice to resolve this problems.
> Br,
> Sanath
>
>
> On Tue, Jun 5, 2012 at 10:46 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> Your not understanding the dialstring aspect...
>>
>> Example:
>>
>> originate sofia/gateway/flowroute/13175551212 user/1000 at 192.168.1.79
>>
>> Where sofia is the channel type, gateway is a keyword, flowroute is what
>> I named my gateway, 13175551212 is the number I'm calling through my
>> gateway flowroute, and user/1000 is the extension to send it to and
>> 192.168.1.79 is the domain that user is attached to.
>>
>> Brian Foster
>> Endigo Computer LLC
>>
>> Sent from a mobile device.
>> On Jun 5, 2012 8:49 AM, "Sanath Prasanna" <ahe.sanath at gmail.com> wrote:
>>
>>> Hi,
>>> when I run originate command in fs_cli following error coming. Pls
>>> advice to resolve that.
>>>
>>> freeswitch at internal> originate sofia/external/ivr/9179123456 9179123458
>>> -ERR originate  sofia/external/ivr/9179123456 9179123458 Command not
>>> found!
>>>
>>> Br,
>>> Sanath
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From engineerzuhairraza at gmail.com  Wed Jun  6 12:41:28 2012
From: engineerzuhairraza at gmail.com (Zohair Raza)
Date: Wed, 6 Jun 2012 12:41:28 +0400
Subject: [Freeswitch-users] NibbleBill query
Message-ID: <CAF_PTet-NJC3_s87_y4NJsFgHqvqMdd9FZ9arKQo21Kn_tBqAw@mail.gmail.com>

Hi All,

Looking for suggestions to implement billing slabs A,B and C as in
a2billing using xml_curl

Lets say for first five minutes, call is charged $0.1 per minute,
afterwards $0.05 per minute for next five minutes and then $.005 per
minute for next rest of call

Regards,
Zohair Raza


From awais-nazeer at hotmail.com  Wed Jun  6 12:49:33 2012
From: awais-nazeer at hotmail.com (awais nazir)
Date: Wed, 6 Jun 2012 13:49:33 +0500
Subject: [Freeswitch-users] (no subject)
Message-ID: <BLU162-W1643A87AE4B56840F71257880D0@phx.gbl>


Greetings freeswitchers,

 See if you can help in this.If a freeswitch crashes abnormally
 , can we somehow retrieve the CDR before its death.   A single abnormal crash in a month gives us some loss as I run around 1200 CC.

-- 
waisee

 		 	   		  
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From awais-nazeer at hotmail.com  Wed Jun  6 13:11:50 2012
From: awais-nazeer at hotmail.com (awais nazir)
Date: Wed, 6 Jun 2012 14:11:50 +0500
Subject: [Freeswitch-users] CDR recovery on crash
Message-ID: <BLU162-W4036FF37BC59F586695152880D0@phx.gbl>


Greetings freeswitchers,

 See if you can help in this.If a 
freeswitch crashes abnormally
 , can we somehow retrieve the CDR before its death.   A single abnormal
 crash in a month gives us some loss as I run around 1200 CC.

-- 
waisee 		 	   		  
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From torstein.knutsen at gmail.com  Wed Jun  6 14:17:52 2012
From: torstein.knutsen at gmail.com (Torstein Knutsen)
Date: Wed, 6 Jun 2012 12:17:52 +0200
Subject: [Freeswitch-users] Controlling caller_id_name
Message-ID: <CAMnjH4QKpTbDw+pqwV0ZH3G4zS82AVBU_zs=rLFNDTRD+gh-0g@mail.gmail.com>

Hi

I'm trying to control the "caller_id_name" variable used in conferencing.
My dialplan looks like this :

<extension name="public_extensions">
<condition field="destination_number" expression="^(0meet)$">
<action application="conference" data="72013 at webconf"/>

When I call into the conference, I do an API "conference
xml_list" which gives details of the conferencees ..
I want to be able to change the caller_id_name in the listing, but all my
efforts for doing this via dialplan fails.

Is this even possible somehow ?

I tried to add this to the dialplan before executing the conference :
<action application="export" data="caller_id_name=Hobbes" />
<action application="set" data="caller_id_name=Hobbes" />
<action application="export" data="effective_caller_id_name=Hobbes" />
<action application="export" data="effective_caller_id_name=Hobbes" />
<action application="set" data="origination_caller_id_name=Hobbes" />
<action application="export" data="origination_caller_id_name=Hobbes" />

But to no avail...

Any pointers would be appreciated.

Thank you
Torstein
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From avi at avimarcus.net  Wed Jun  6 14:30:39 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 6 Jun 2012 13:30:39 +0300
Subject: [Freeswitch-users] CDR recovery on crash
In-Reply-To: <BLU162-W4036FF37BC59F586695152880D0@phx.gbl>
References: <BLU162-W4036FF37BC59F586695152880D0@phx.gbl>
Message-ID: <CALNZuwnZoHBUFXrnYHfEicr8VL89XGL6ZdQ7hkkYCE3+TUN5Cw@mail.gmail.com>

Hey, can you do the actual math on that? 1200 calls X 3? Minute acd X
$0.01cost per minute? = $36 which in the grand scheme of things isn't that
much. Your numbers may be different...
Alternatively there is "track calls" to be used in some manner.

-Avi
(This message was painstakingly thumbed out on my mobile, so apologies for
brevity and errors.)

On Jun 6, 2012 12:13 PM, "awais nazir" <awais-nazeer at hotmail.com> wrote:

 Greetings freeswitchers,

 See if you can help in this.If a freeswitch crashes abnormally , can we
somehow retrieve the CDR before its death.   A single abnormal crash in a
month gives us some loss as I run around 1200 CC.

-- 
waisee

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From saami_mh at ymail.com  Wed Jun  6 15:58:57 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Wed, 6 Jun 2012 04:58:57 -0700 (PDT)
Subject: [Freeswitch-users] execute_extension coudn't worked correctly !!
Message-ID: <1338983937.89947.YahooMailNeo@web120103.mail.ne1.yahoo.com>

hi guys,
in the path:/usr/loca/freeswitch/conf/default/
i have created two .xml file named:
001_luacallduration.xml ?and ?01_ratelist.xml
so the content of ?the above files are as follow:

<include>
? <extension name="mytest">
? ? ? ? <condition field="destination_number" expression="^788" break="never">
? ? ? ? ? ? ? ? <action application="odbc_query" data="select cash as cashvalue from accounts where id like '${nibble_account}';"/>
? ? ? ? ? ? ? ? <action application="log" data="INFO The value of cashvalue is ${cashvalue}" />
? ? ? ? ? ? ? ? <!-- Check Cash -->
? ? ? ? ? ? ? ? <action application="lua" data="checkcash.lua ${cashvalue}" />
? ? ? ? ? ? ? ? <!-- Check Nibble_rate -->
? ? ? ? ? ? ? ? <action application="log" data="INFO The value of nibble_account is ?${nibble_account}"/>
? ? ? ? ? ? ? ? <action application="log" data="INFO The value of nibble_rate [before] is ?${nibble_rate}"/>
? ? ? ? ? ? ? ? <action application="execute_extension" ?data="mainratelist ?XML ?default "/>
</condition>
?</extension>
</include>

01_ratelist.xml?:


<extension name="mainratelist">
<condition field="destination_number" expression="^788" break="on-true">

<action application="set" data="nibble_rate=1570"/>

</condition>

</condition>

</extension>


i want to set nibble_rate using execute_extemsion?? application and return 1570 to mytest extension, but execute_extension ?don't work and nibble_rate left empty and coldn't return back any things :(


plz help ,thanks?
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From albert_nguyen16 at hotmail.com  Wed Jun  6 16:02:30 2012
From: albert_nguyen16 at hotmail.com (Albert Nguyen)
Date: Wed, 6 Jun 2012 12:02:30 +0000
Subject: [Freeswitch-users] when to use stop_dtmf
In-Reply-To: <CAKzWOxXoW5hPTi9hWn-SrZsuYALUm_3L8TvpqMyZgtMAJMQ7rA@mail.gmail.com>
References: <20120527181143.62d2a620@mail.tritonwest.net>,
	<3820475734403626844@unknownmsgid>,
	<BAY170-W5845762FB9A832B9BC8F9E950D0@phx.gbl>,
	<CAKzWOxXoW5hPTi9hWn-SrZsuYALUm_3L8TvpqMyZgtMAJMQ7rA@mail.gmail.com>
Message-ID: <BAY170-W10213C121AB0AB841C1EEFF950D0@phx.gbl>



Hi Michael,

That's great. Thanks very much for your help.

Al
Date: Tue, 5 Jun 2012 22:06:33 -0700
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] when to use stop_dtmf

If you are using the dialplan application "start_dtmf" then it applies only to the current, existing channel. So no, you don't have to explicitly call stop_dtmf, especially after the call has ended! :)


-MC

On Tue, Jun 5, 2012 at 6:29 PM, Albert Nguyen <albert_nguyen16 at hotmail.com> wrote:





 

Hi,

 

I am using start_DTMF to convert inband DTMF to RFC 2833 for calls. Just wanted to make sure :-), when a call is ended by FS (i.e [CS_DESTROY]), would this will also automatically stop the DTMF conversion function or do I have to explicitly using stop_DTMF ? 


 

Thanks in advance.

 

Al
 		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com



Join Us At ClueCon - Aug 7-9, 2012



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 gmail.com  Wed Jun  6 16:40:08 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 6 Jun 2012 14:40:08 +0200
Subject: [Freeswitch-users] warm and fuzzy
Message-ID: <CALXCt0pVu=zj=4K8gMaAiZXgBWb30KyyX_1A70Zb5HJVFaRKFw@mail.gmail.com>

(2:35:49 PM) vipkilla: on hold with Time warner cable... .ahhh the FS hold
music...
(2:36:25 PM) gmaruzz: :) yay!
(2:36:46 PM) blee: haha thats awesome


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From Bill.Ryder at millicorp.com  Wed Jun  6 18:26:51 2012
From: Bill.Ryder at millicorp.com (Bill Ryder)
Date: Wed, 6 Jun 2012 14:26:51 +0000
Subject: [Freeswitch-users] =?windows-1252?q?Freeswitch_=93real=94_respons?=
 =?windows-1252?q?e_to_api_chat=3F__Having_a_problem_determining_result_vi?=
 =?windows-1252?q?a_esl=2E?=
Message-ID: <5ABA6F8E7A50AB4B84651BF8EF5B557457DA0F@MAILBOX.millicorp.com>

I'm sending an sms message like this over esl (telnet, java client, various methods all yield the same results)(numbers are all dummies):

api chat sip|13215555555 at 6.50.120.201|internal/2395555555 at 6.50.120.200|test message

This works fine with valid numbers. If I put in an invalid number somewhere I may see something like this in the sip trace indicating a problem:

U +0.745829 6.50.120.200:5060 -> 6.50.120.201:5060 SIP/2.0 484 Address Incomplete. ...

...however over the esl socket I get the same response event no matter what, just "sent". Also, when I bind to the esl port and attempt to spit out every message possible (no filters), we see most of what comes through a sip trace, but again this bad result sip packet seems not to generate an esl event.

Is there a way over esl to probe raw sip messages so that we can test for such a packet manually, or some other way of seeking the final async result?

Thanks in advance!

Sincerely,
William Ryder
Network Engineer, Millicorp
12748 University Drive, Fort Myers, FL 33907
Direct: 239-443-5892
Fax: 239-245-9087

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From kamminthang.nengzalam at a-cti.com  Wed Jun  6 18:47:49 2012
From: kamminthang.nengzalam at a-cti.com (Kamminthang Nengzalam)
Date: Wed, 6 Jun 2012 20:17:49 +0530
Subject: [Freeswitch-users] Freeswitch + open vox G400P
In-Reply-To: <CAL8RuBN0ikW_mLW-LDVyoXH13K30KqyFUooWHVd4HCpgZTnNBg@mail.gmail.com>
References: <mailman.11174.1338905758.13443.freeswitch-users@lists.freeswitch.org>
	<CAL8RuBONbdxEWy9z5oTTuBZ_7B8hE4oNday=SBtDm6oaz2d8Zg@mail.gmail.com>
	<CAP2fbbf2VP452CgXRBbvXkqsA5grpQZGTcWApJyBfctzHoKomA@mail.gmail.com>
	<CAL8RuBN0VqGNHzvyW=NKKhe_oZVECfJ3GmqpbbG5r8-xGtaVAA@mail.gmail.com>
	<CAL8RuBN0ikW_mLW-LDVyoXH13K30KqyFUooWHVd4HCpgZTnNBg@mail.gmail.com>
Message-ID: <CAL8RuBPOK23EHNMobk2VT63UDMhY1FahHWn3M-nY96vgzcbDXw@mail.gmail.com>

Yep freeswitch sound quality is awesome n i am loving it. :-)
Well regarding my gsmopen,
i run it in my local desktop ubuntu.
and login to the gsm using x-lite and a made sure the x-lite have gsm codec.
after logging to the x-lite a dialout to my cell phone  and sometime a will
not hear the ringtone on my x-lite and some times no audio. The same goes
for when a call my gsm which i configure to ring in my x-lite.
Hope this helps. Thank you guys.

Kam

On Tue, Jun 5, 2012 at 10:01 PM, Kamminthang Nengzalam <
kamminthang.nengzalam at a-cti.com> wrote:

> i followed the wiki correctly. And i'm getting no error. Can you lemme kno
> how to get better sound quality......
> Do i need to provide anything pliz lemme kno....
>
> Thanks,
> Kam
>
>
> On Tue, Jun 5, 2012 at 9:36 PM, Kamminthang Nengzalam <
> kamminthang.nengzalam at a-cti.com> wrote:
>
>> i already have configure GSMopen for E1550 and it is working ok but the
>> audio quality is not very good.
>> Is there a way i can boost it or use openvox instead...
>>
>> Thanks,
>> Kam
>>
>>
>> On Tue, Jun 5, 2012 at 7:56 PM, Kamminthang Nengzalam <
>> kamminthang.nengzalam at a-cti.com> wrote:
>>
>>>
>>> Hi Guys,
>>>
>>> i want to configure openvox G400P in my freeswitch.
>>> Can anyone lemme kno how i should do it?
>>> will mod_gsmopen work and if so/not how a should make it work.
>>> Pliz and thank you in advance for helping.....
>>>
>>> Thanks,
>>> Kam
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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://staging.originateglobal.com/ClickToCall?id=%2244Rf0r6tLOxU8APXNT92IIxh93OzvT2clF0mEJCf8schNn%22>
>>
>>
>>
>
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From bdfoster at endigotech.com  Wed Jun  6 19:04:54 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 6 Jun 2012 11:04:54 -0400
Subject: [Freeswitch-users] Controlling caller_id_name
In-Reply-To: <CAMnjH4QKpTbDw+pqwV0ZH3G4zS82AVBU_zs=rLFNDTRD+gh-0g@mail.gmail.com>
References: <CAMnjH4QKpTbDw+pqwV0ZH3G4zS82AVBU_zs=rLFNDTRD+gh-0g@mail.gmail.com>
Message-ID: <CAKV-jFwssLbwcFHOchkkwzFLckAzt6cmDsy3vY_wpAES4GehyA@mail.gmail.com>

You need to be setting effective_caller_id_name before the bridge.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 6, 2012 6:19 AM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
wrote:

> Hi
>
> I'm trying to control the "caller_id_name" variable used in conferencing.
> My dialplan looks like this :
>
> <extension name="public_extensions">
> <condition field="destination_number" expression="^(0meet)$">
> <action application="conference" data="72013 at webconf"/>
>
> When I call into the conference, I do an API "conference
> xml_list" which gives details of the conferencees ..
> I want to be able to change the caller_id_name in the listing, but all my
> efforts for doing this via dialplan fails.
>
> Is this even possible somehow ?
>
> I tried to add this to the dialplan before executing the conference :
> <action application="export" data="caller_id_name=Hobbes" />
> <action application="set" data="caller_id_name=Hobbes" />
> <action application="export" data="effective_caller_id_name=Hobbes" />
> <action application="export" data="effective_caller_id_name=Hobbes" />
> <action application="set" data="origination_caller_id_name=Hobbes" />
> <action application="export" data="origination_caller_id_name=Hobbes" />
>
> But to no avail...
>
> Any pointers would be appreciated.
>
> Thank you
> Torstein
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jchavanton at gmail.com  Wed Jun  6 19:10:03 2012
From: jchavanton at gmail.com (Julien Chavanton)
Date: Wed, 6 Jun 2012 11:10:03 -0400
Subject: [Freeswitch-users] scope of channel variables,
	when are they accessible ?
Message-ID: <CAKmcL2=nuVhuzsOqWgduC7VSFAkqhXkXNpp2ZPvUvS0vgnmHaw@mail.gmail.com>

I do not understand the scope of channel variables, when are they
accessible ?

For example, I am looking for the sip username of the caller call leg.

In this example ${sip_auth_username} is empty when logging or calling
sf_update.pl but it is correct when calling sf_hangup.pl

<include>
        <extension name="outdialing">
                <condition field="destination_number" expression=".*)">
                        <action application="answer"/>
                        <action application="log" data="VARIABLES
[${sip_auth_username}][${Caller-Username}][${sip_from_user}][${effective_caller_id_name}][${effective_caller_id_number}]"/>
                        <action application="perl" data="/var/fs_hook/
sf_update.pl ${sip_auth_username} ${sip_to_user}"/>
                        <action application="set"
data="api_hangup_hook=perl /var/fs_hook/sf_hangup.pl  ${sip_auth_username}
${sip_to_user} "/>
                        <action application="bridge"
 data="{absolute_codec_string='PCMA,PCMU,G729',call_leg=B,session_in_hangup_hook=true'}sofia/gateway/XXXX/XXXX$1"/>

                 </condition>
</extension>
</include>
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From admin at blindi.net  Wed Jun  6 19:52:04 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 6 Jun 2012 17:52:04 +0200 (CEST)
Subject: [Freeswitch-users] opensource project voicechat for freeswitch
In-Reply-To: <CALQZz0zLpRBnFu3jce07Ycq=mn_OnAGrVFZgm-ibY_3eGg93Xw@mail.gmail.com>
References: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
	<2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com>
	<1338558318.79806.YahooMailNeo@web120405.mail.ne1.yahoo.com>
	<Pine.LNX.4.64.1206041700001.22234@tomback.antagus.de>
	<CALQZz0zLpRBnFu3jce07Ycq=mn_OnAGrVFZgm-ibY_3eGg93Xw@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1206061742390.4294@tomback.antagus.de>

Hi Sanath

Thanks for your helpfull message


I suggest: we do everything step by step.

First: the voice mail creation and login.

for example:
press 1 for login.
Press 2 to create your onw mailbox.
  It is important that the mailbox already exists.
If the Mailbox exists, play a error wavfile for example:
Sorry this mailbox is allready exists, and go back to the creation option 
again.

Thanks.


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From moises.silva at gmail.com  Wed Jun  6 20:01:33 2012
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 6 Jun 2012 12:01:33 -0400
Subject: [Freeswitch-users] Freeswitch & FreeTDM question...
In-Reply-To: <C066384B0C1F1E4B9B870F050D7F1A9D401C2220@ADAMA.mcpeekdodge.local>
References: <C066384B0C1F1E4B9B870F050D7F1A9D401C2220@ADAMA.mcpeekdodge.local>
Message-ID: <CAA4nhyABsyoN42JxQ=jVp3cqCrN_+J5p7L7J3JHaKQSr=WoyJQ@mail.gmail.com>

Hello Frank,

On Thu, May 31, 2012 at 6:35 PM, Frank Busalacchi Jr <
frankjr at mcpeekdodge.com> wrote:

> I am having intermittent problems with my FXO/FXS ports.  Basically I have
> fax machines attached.  Every so often (sometimes 5-10 days in between),
> the fax machine says that it has no dial tone.  I am able to resolve the
> problem by going to the CLI and unloading mod_freetdm and then reloading
> mod_freetdm.  When the ?no dialtone problem? is happening, unloading
> mod_freetdm takes 10 seconds or so?I have even had it crash freeswitch
> during the unload?
>
> **
>

Whenever this problem happens, you need to enable debug logging in
FreeSWITCH. Dialtone is provided by freetdm not the card, but if the card
does not detect the fax going offhook, then no dialtone will be provided.
First thing is to determine whether FreeTDM receives the offhook event from
the card/drivers, that can be seen if debug logging is enabled (some
message saying something like "wanpipe event x received").



*Moises Silva
**Manager, Software Engineering***

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



**<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>

Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`|
| YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
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From jaasmailing at gmail.com  Wed Jun  6 20:13:58 2012
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Wed, 06 Jun 2012 18:13:58 +0200
Subject: [Freeswitch-users] RTP NAT issue
Message-ID: <4FCF81C6.4010004@gmail.com>

Hi all,

I have a problem with RTP and NAT.
The scenario is Hosted PBX and Natted phones (yealink):

Phones (192.168.0.x) - NAT -> FS (public IP)

When I call FS (for example an IVR) from a Phone, FS send the RTP to the 
private address (192.168.0.x) instead to the public NAT IP.
The registration is ok:

freeswitch at internal> sofia status profile tenant1.bs.dev.voip.clio.it reg
1
Registrations:
=================================================================================================
Call-ID:        488014850 at 192.168.0.100
User:           202 at tenant1.test.com
Contact:        "Test 202" 
<sip:202 at 192.168.0.100:5062;fs_nat=yes;fs_path=sip%3A202%40<NAT_IP>%3A37710>
Agent:          Yealink SIP-T20P 9.61.0.70
Status:         Registered(UDP-NAT)(unknown) EXP(2012-06-06 19:01:55) 
EXPSECS(3232)
Host:           localhost.localdomain
IP: <NAT_IP>
Port:           37710
Auth-User:      202
Auth-Realm: tenant1.test.com
MWI-Account:    202 at tenant1.test.com


How I can tell FS to send the RTP Packets to the right address? I think 
is needed a "comedia mode" like in Asterisk (or RTPProxy in openser)...


Best regards,


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From bdfoster at endigotech.com  Wed Jun  6 20:28:50 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 6 Jun 2012 12:28:50 -0400
Subject: [Freeswitch-users] RTP NAT issue
In-Reply-To: <4FCF81C6.4010004@gmail.com>
References: <4FCF81C6.4010004@gmail.com>
Message-ID: <CAKV-jFxUFOtY6EzmKMv-3pZyvBf5WuyvFJOafh7=rWnWg=iW-w@mail.gmail.com>

The issue is more likely the phone, as the phone is responsible for handing
FS the correct IP. There is however a way to force this on the FS side but
may break other devices. Please take a look at NDLB (No Device Left Behind)
parameters for Sofia on the wiki.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 6, 2012 12:15 PM, "Carlo Dimaggio" <jaasmailing at gmail.com> wrote:

>  Hi all,
>
> I have a problem with RTP and NAT.
> The scenario is Hosted PBX and Natted phones (yealink):
>
> Phones (192.168.0.x) - NAT -> FS (public IP)
>
> When I call FS (for example an IVR) from a Phone, FS send the RTP to the
> private address (192.168.0.x) instead to the public NAT IP.
> The registration is ok:
>
> freeswitch at internal> sofia status profile tenant1.bs.dev.voip.clio.it reg
> 1
> Registrations:
>
> =================================================================================================
> Call-ID:        488014850 at 192.168.0.100
> User:           202 at tenant1.test.com
> Contact:        "Test 202" <
> sip:202 at 192.168.0.100:5062;fs_nat=yes;fs_path=sip%3A202%40
> <NAT_IP>%3A37710>
> Agent:          Yealink SIP-T20P 9.61.0.70
> Status:         Registered(UDP-NAT)(unknown) EXP(2012-06-06 19:01:55)
> EXPSECS(3232)
> Host:           localhost.localdomain
> IP:             <NAT_IP>
> Port:           37710
> Auth-User:      202
> Auth-Realm:     tenant1.test.com
> MWI-Account:    202 at tenant1.test.com
>
>
> How I can tell FS to send the RTP Packets to the right address? I think is
> needed a "comedia mode" like in Asterisk (or RTPProxy in openser)...
>
>
> Best regards,
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Wed Jun  6 20:38:08 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 6 Jun 2012 12:38:08 -0400
Subject: [Freeswitch-users] RTP NAT issue
In-Reply-To: <CAKV-jFxUFOtY6EzmKMv-3pZyvBf5WuyvFJOafh7=rWnWg=iW-w@mail.gmail.com>
References: <4FCF81C6.4010004@gmail.com>
	<CAKV-jFxUFOtY6EzmKMv-3pZyvBf5WuyvFJOafh7=rWnWg=iW-w@mail.gmail.com>
Message-ID: <CAKV-jFyj9NR6fmWF+rfS6WyUyc6M9uTuxLbtkJAXRBo+X98--A@mail.gmail.com>

NDLB-connectile-disfunction is probably what you're looking for, see:
http://wiki.freeswitch.org/wiki/NDLB

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 6, 2012 12:28 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> The issue is more likely the phone, as the phone is responsible for
> handing FS the correct IP. There is however a way to force this on the FS
> side but may break other devices. Please take a look at NDLB (No Device
> Left Behind) parameters for Sofia on the wiki.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 6, 2012 12:15 PM, "Carlo Dimaggio" <jaasmailing at gmail.com> wrote:
>
>>  Hi all,
>>
>> I have a problem with RTP and NAT.
>> The scenario is Hosted PBX and Natted phones (yealink):
>>
>> Phones (192.168.0.x) - NAT -> FS (public IP)
>>
>> When I call FS (for example an IVR) from a Phone, FS send the RTP to the
>> private address (192.168.0.x) instead to the public NAT IP.
>> The registration is ok:
>>
>> freeswitch at internal> sofia status profile tenant1.bs.dev.voip.clio.it reg
>> 1
>> Registrations:
>>
>> =================================================================================================
>> Call-ID:        488014850 at 192.168.0.100
>> User:           202 at tenant1.test.com
>> Contact:        "Test 202" <
>> sip:202 at 192.168.0.100:5062;fs_nat=yes;fs_path=sip%3A202%40
>> <NAT_IP>%3A37710>
>> Agent:          Yealink SIP-T20P 9.61.0.70
>> Status:         Registered(UDP-NAT)(unknown) EXP(2012-06-06 19:01:55)
>> EXPSECS(3232)
>> Host:           localhost.localdomain
>> IP:             <NAT_IP>
>> Port:           37710
>> Auth-User:      202
>> Auth-Realm:     tenant1.test.com
>> MWI-Account:    202 at tenant1.test.com
>>
>>
>> How I can tell FS to send the RTP Packets to the right address? I think
>> is needed a "comedia mode" like in Asterisk (or RTPProxy in openser)...
>>
>>
>> Best regards,
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
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From torstein.knutsen at gmail.com  Wed Jun  6 20:54:56 2012
From: torstein.knutsen at gmail.com (Torstein Knutsen)
Date: Wed, 6 Jun 2012 18:54:56 +0200
Subject: [Freeswitch-users] Controlling caller_id_name
In-Reply-To: <CAKV-jFwssLbwcFHOchkkwzFLckAzt6cmDsy3vY_wpAES4GehyA@mail.gmail.com>
References: <CAMnjH4QKpTbDw+pqwV0ZH3G4zS82AVBU_zs=rLFNDTRD+gh-0g@mail.gmail.com>
	<CAKV-jFwssLbwcFHOchkkwzFLckAzt6cmDsy3vY_wpAES4GehyA@mail.gmail.com>
Message-ID: <CAMnjH4SskCTc641+bMbMQRAP0X9qk42O6M4eELHYmkVrTMDGAw@mail.gmail.com>

Hi

Sure?  I did try that, but it does not work .. Im not doing bridge, but
conference .. ..

-T
On Jun 6, 2012 5:06 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> You need to be setting effective_caller_id_name before the bridge.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 6, 2012 6:19 AM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
> wrote:
>
>> Hi
>>
>> I'm trying to control the "caller_id_name" variable used in conferencing.
>> My dialplan looks like this :
>>
>> <extension name="public_extensions">
>> <condition field="destination_number" expression="^(0meet)$">
>> <action application="conference" data="72013 at webconf"/>
>>
>> When I call into the conference, I do an API "conference
>> xml_list" which gives details of the conferencees ..
>> I want to be able to change the caller_id_name in the listing, but all my
>> efforts for doing this via dialplan fails.
>>
>> Is this even possible somehow ?
>>
>> I tried to add this to the dialplan before executing the conference :
>> <action application="export" data="caller_id_name=Hobbes" />
>> <action application="set" data="caller_id_name=Hobbes" />
>> <action application="export" data="effective_caller_id_name=Hobbes" />
>> <action application="export" data="effective_caller_id_name=Hobbes" />
>> <action application="set" data="origination_caller_id_name=Hobbes" />
>> <action application="export" data="origination_caller_id_name=Hobbes" />
>>
>> But to no avail...
>>
>> Any pointers would be appreciated.
>>
>> Thank you
>> Torstein
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Wed Jun  6 21:01:35 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 6 Jun 2012 13:01:35 -0400
Subject: [Freeswitch-users] Controlling caller_id_name
In-Reply-To: <CAMnjH4SskCTc641+bMbMQRAP0X9qk42O6M4eELHYmkVrTMDGAw@mail.gmail.com>
References: <CAMnjH4QKpTbDw+pqwV0ZH3G4zS82AVBU_zs=rLFNDTRD+gh-0g@mail.gmail.com>
	<CAKV-jFwssLbwcFHOchkkwzFLckAzt6cmDsy3vY_wpAES4GehyA@mail.gmail.com>
	<CAMnjH4SskCTc641+bMbMQRAP0X9qk42O6M4eELHYmkVrTMDGAw@mail.gmail.com>
Message-ID: <CAKV-jFwFPDoYqbpUwA=rfAg6qN8=81dHYiPgx+c6sa63rYneZQ@mail.gmail.com>

Not sure if this will work but try executing the set inline

<action application="conference" data="1234 at default" inline="true"/>

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 6, 2012 12:56 PM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
wrote:

> Hi
>
> Sure?  I did try that, but it does not work .. Im not doing bridge, but
> conference .. ..
>
> -T
> On Jun 6, 2012 5:06 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>
>> You need to be setting effective_caller_id_name before the bridge.
>>
>> Brian Foster
>> Endigo Computer LLC
>>
>> Sent from a mobile device.
>> On Jun 6, 2012 6:19 AM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
>> wrote:
>>
>>> Hi
>>>
>>> I'm trying to control the "caller_id_name" variable used in conferencing.
>>> My dialplan looks like this :
>>>
>>> <extension name="public_extensions">
>>> <condition field="destination_number" expression="^(0meet)$">
>>> <action application="conference" data="72013 at webconf"/>
>>>
>>> When I call into the conference, I do an API "conference
>>> xml_list" which gives details of the conferencees ..
>>> I want to be able to change the caller_id_name in the listing, but all
>>> my efforts for doing this via dialplan fails.
>>>
>>> Is this even possible somehow ?
>>>
>>> I tried to add this to the dialplan before executing the conference :
>>> <action application="export" data="caller_id_name=Hobbes" />
>>> <action application="set" data="caller_id_name=Hobbes" />
>>> <action application="export" data="effective_caller_id_name=Hobbes" />
>>> <action application="export" data="effective_caller_id_name=Hobbes" />
>>> <action application="set" data="origination_caller_id_name=Hobbes" />
>>> <action application="export" data="origination_caller_id_name=Hobbes" />
>>>
>>> But to no avail...
>>>
>>> Any pointers would be appreciated.
>>>
>>> Thank you
>>> Torstein
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Wed Jun  6 21:02:35 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 6 Jun 2012 13:02:35 -0400
Subject: [Freeswitch-users] Controlling caller_id_name
In-Reply-To: <CAMnjH4SskCTc641+bMbMQRAP0X9qk42O6M4eELHYmkVrTMDGAw@mail.gmail.com>
References: <CAMnjH4QKpTbDw+pqwV0ZH3G4zS82AVBU_zs=rLFNDTRD+gh-0g@mail.gmail.com>
	<CAKV-jFwssLbwcFHOchkkwzFLckAzt6cmDsy3vY_wpAES4GehyA@mail.gmail.com>
	<CAMnjH4SskCTc641+bMbMQRAP0X9qk42O6M4eELHYmkVrTMDGAw@mail.gmail.com>
Message-ID: <CAKV-jFy2sN5fZs2prpdE0wD10CJXVfFgWQFQxg7iM_GJ8_SxXw@mail.gmail.com>

Crap... set the caller id inline not the conference line... sorry got a lot
going on today lol

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 6, 2012 12:56 PM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
wrote:

> Hi
>
> Sure?  I did try that, but it does not work .. Im not doing bridge, but
> conference .. ..
>
> -T
> On Jun 6, 2012 5:06 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>
>> You need to be setting effective_caller_id_name before the bridge.
>>
>> Brian Foster
>> Endigo Computer LLC
>>
>> Sent from a mobile device.
>> On Jun 6, 2012 6:19 AM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
>> wrote:
>>
>>> Hi
>>>
>>> I'm trying to control the "caller_id_name" variable used in conferencing.
>>> My dialplan looks like this :
>>>
>>> <extension name="public_extensions">
>>> <condition field="destination_number" expression="^(0meet)$">
>>> <action application="conference" data="72013 at webconf"/>
>>>
>>> When I call into the conference, I do an API "conference
>>> xml_list" which gives details of the conferencees ..
>>> I want to be able to change the caller_id_name in the listing, but all
>>> my efforts for doing this via dialplan fails.
>>>
>>> Is this even possible somehow ?
>>>
>>> I tried to add this to the dialplan before executing the conference :
>>> <action application="export" data="caller_id_name=Hobbes" />
>>> <action application="set" data="caller_id_name=Hobbes" />
>>> <action application="export" data="effective_caller_id_name=Hobbes" />
>>> <action application="export" data="effective_caller_id_name=Hobbes" />
>>> <action application="set" data="origination_caller_id_name=Hobbes" />
>>> <action application="export" data="origination_caller_id_name=Hobbes" />
>>>
>>> But to no avail...
>>>
>>> Any pointers would be appreciated.
>>>
>>> Thank you
>>> Torstein
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Jun  6 21:03:57 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Jun 2012 10:03:57 -0700
Subject: [Freeswitch-users] FreeSWITCH Conf Call Is On!
Message-ID: <CAKzWOxU5LLjQt7TrvfPF3WswVaz3XbKVuUuS59=ht+wRvX5nHQ@mail.gmail.com>

Come join us!

http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_06

We're going to talk about mod_httapi and a few other fun things.

Thanks!

-Michael
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From toddb at toddbailey.net  Wed Jun  6 19:36:07 2012
From: toddb at toddbailey.net (toddb at toddbailey.net)
Date: Wed, 06 Jun 2012 08:36:07 -0700
Subject: [Freeswitch-users] Skype on FreeSwitch install issues
Message-ID: <20120606083607.33e327b490679d2282e332758c73b55b.a8919b34fb.wbe@email14.secureserver.net>

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From jack at livecall.com  Wed Jun  6 21:10:39 2012
From: jack at livecall.com (Jack)
Date: Wed, 06 Jun 2012 10:10:39 -0700
Subject: [Freeswitch-users] FSClient UA error
Message-ID: <4FCF8F0F.8010004@livecall.com>

I am getting this error in my log when I try to run FSClient on windows xp:

2012-06-06 09:59:36.500000 [NOTICE] sofia.c:4892 Started Profile 
softphone [sofia_reg_softphone]
2012-06-06 09:59:36.500000 [DEBUG] mod_sofia.c:5471 Waiting for profiles 
to start
2012-06-06 09:59:36.546875 [ERR] sofia.c:2099 Error Creating SIP UA for 
profile: softphone

I also get a dialog box that says:

Warning the master sofia profile was not able to load and the phone will 
most likely _not_work,.....

I have checked the port and opened the firewall 5060 and tried reloading 
the sofia profile by editing the settings  and saving  and I still get 
the same.
Thanks,
  Jack


From gmaruzz at gmail.com  Wed Jun  6 21:17:15 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 6 Jun 2012 19:17:15 +0200
Subject: [Freeswitch-users] Skype on FreeSwitch install issues
In-Reply-To: <20120606083607.33e327b490679d2282e332758c73b55b.a8919b34fb.wbe@email14.secureserver.net>
References: <20120606083607.33e327b490679d2282e332758c73b55b.a8919b34fb.wbe@email14.secureserver.net>
Message-ID: <CALXCt0oygqw6oucQhLbLNq5pgJu9QCo_VgmSsfSc2osPAcKMDw@mail.gmail.com>

please use the automatic installer (install.pl) it will do it all for you

if not using automatic installer, do something similar to:

#Unload possible ALSA sound modules that would conflict with our OSS fake
module
rmmod snd_pcm_oss
rmmod snd_mixer_oss
rmmod snd_seq_oss
sleep 1
#Create the inode our fake sound driver will use
mknod /dev/dsp c 14 3
#Load our OSS fake module
insmod /usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko

this will work

Explanation is: your alsa installation creates automatically /dev/dsp for
compatibility with OSS, but you do not need it, if you are using ALSA. (You
would need it to use alsa from OSS application, but this is not what you
need).

-giovanni

On Wed, Jun 6, 2012 at 5:36 PM, <toddb at toddbailey.net> wrote:

> Hello,
>
> New to list & appologies if this is the wrong place to post questions.
>
> Assuming this is the place to be, here is a question.
>
> Whet I get to the build step for skypopen
> "make clean; make; insmod ./skypopen.ko; mknod /dev/dsp c 14 3"
>
> I get an error with the insmod cmd,  resource / device busy
>
>
> I think it's caused due to having several snd entries listed via the
> "lsmod |grep snd"
>
> The machine is question is a media, database and email server.
> And the media server component requires the use of ALSA components.
>
> Does this requirement prohibit the use of the skype component in
> Freeswitch or do I just need to prevent the snd modules from loading to
> complete the kernel mod? Then re enable snd modules at next boot.
>
>
> If this is the case any suggestions?   Maybe running FS in a Vbox to
> provide the functionality?
> Or do I need a dedicated box just running linux and FS components?
>
> current machine is fedora 14 x64 on a 4 core xeon @ 3ghz w/7 gig memory,
> w/ 35 % process load average
>
> thanks
>
>
> BTW: my desired configuration
>
> analog land line to Cisco SPA3102 router connected to the FS server with 4
> extensions registered.
> outbound local calls router to analog line, 1+ area code out bound calls
> routed to skype.
> Incoming calls routed to all extensions, unanswered calls set to a ivr to
> 1. leave voice mail or 2.forward to cell phone (via skype connection)
>
> any of this not do-able?
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From gmaruzz at gmail.com  Wed Jun  6 21:22:21 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 6 Jun 2012 19:22:21 +0200
Subject: [Freeswitch-users] Freeswitch + open vox G400P
In-Reply-To: <CAL8RuBPOK23EHNMobk2VT63UDMhY1FahHWn3M-nY96vgzcbDXw@mail.gmail.com>
References: <mailman.11174.1338905758.13443.freeswitch-users@lists.freeswitch.org>
	<CAL8RuBONbdxEWy9z5oTTuBZ_7B8hE4oNday=SBtDm6oaz2d8Zg@mail.gmail.com>
	<CAP2fbbf2VP452CgXRBbvXkqsA5grpQZGTcWApJyBfctzHoKomA@mail.gmail.com>
	<CAL8RuBN0VqGNHzvyW=NKKhe_oZVECfJ3GmqpbbG5r8-xGtaVAA@mail.gmail.com>
	<CAL8RuBN0ikW_mLW-LDVyoXH13K30KqyFUooWHVd4HCpgZTnNBg@mail.gmail.com>
	<CAL8RuBPOK23EHNMobk2VT63UDMhY1FahHWn3M-nY96vgzcbDXw@mail.gmail.com>
Message-ID: <CALXCt0r4u=0H7Zi9me-0SQ6big5YFVMco_+qR9j2pbs8QM3Asw@mail.gmail.com>

On Wed, Jun 6, 2012 at 4:47 PM, Kamminthang Nengzalam <
kamminthang.nengzalam at a-cti.com> wrote:

> Yep freeswitch sound quality is awesome n i am loving it. :-)
> Well regarding my gsmopen,
> i run it in my local desktop ubuntu.
> and login to the gsm using x-lite and a made sure the x-lite have gsm
> codec.
> after logging to the x-lite a dialout to my cell phone  and sometime a
> will not hear the ringtone on my x-lite and some times no audio. The same
> goes for when a call my gsm which i configure to ring in my x-lite.
>

Kam:
-do not use gsm codec, no need for it, and sounds bad. Use ulaw (g711) or
whatever
- do not use a desktop as a server. It will never work decently. X Windows,
mouse, dozen of other things will battle for IRQs, and this will trash all
the millisecond timers real time audio needs.
so:
-use a server installation
also, is kind of tricky to run both the sip client and the sip server on
the same machine (ports, ip addresses, etc). Better to use separate machines

-giovanni





> Hope this helps. Thank you guys.
>
> Kam
>
>
> On Tue, Jun 5, 2012 at 10:01 PM, Kamminthang Nengzalam <
> kamminthang.nengzalam at a-cti.com> wrote:
>
>> i followed the wiki correctly. And i'm getting no error. Can you lemme
>> kno how to get better sound quality......
>> Do i need to provide anything pliz lemme kno....
>>
>> Thanks,
>> Kam
>>
>>
>> On Tue, Jun 5, 2012 at 9:36 PM, Kamminthang Nengzalam <
>> kamminthang.nengzalam at a-cti.com> wrote:
>>
>>> i already have configure GSMopen for E1550 and it is working ok but the
>>> audio quality is not very good.
>>> Is there a way i can boost it or use openvox instead...
>>>
>>> Thanks,
>>> Kam
>>>
>>>
>>> On Tue, Jun 5, 2012 at 7:56 PM, Kamminthang Nengzalam <
>>> kamminthang.nengzalam at a-cti.com> wrote:
>>>
>>>>
>>>> Hi Guys,
>>>>
>>>> i want to configure openvox G400P in my freeswitch.
>>>> Can anyone lemme kno how i should do it?
>>>> will mod_gsmopen work and if so/not how a should make it work.
>>>> Pliz and thank you in advance for helping.....
>>>>
>>>> Thanks,
>>>> Kam
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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://staging.originateglobal.com/ClickToCall?id=%2244Rf0r6tLOxU8APXNT92IIxh93OzvT2clF0mEJCf8schNn%22>
>>>
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From torstein.knutsen at gmail.com  Wed Jun  6 21:41:41 2012
From: torstein.knutsen at gmail.com (Torstein Knutsen)
Date: Wed, 6 Jun 2012 19:41:41 +0200
Subject: [Freeswitch-users] Controlling caller_id_name
In-Reply-To: <CAKV-jFwFPDoYqbpUwA=rfAg6qN8=81dHYiPgx+c6sa63rYneZQ@mail.gmail.com>
References: <CAMnjH4QKpTbDw+pqwV0ZH3G4zS82AVBU_zs=rLFNDTRD+gh-0g@mail.gmail.com>
	<CAKV-jFwssLbwcFHOchkkwzFLckAzt6cmDsy3vY_wpAES4GehyA@mail.gmail.com>
	<CAMnjH4SskCTc641+bMbMQRAP0X9qk42O6M4eELHYmkVrTMDGAw@mail.gmail.com>
	<CAKV-jFwFPDoYqbpUwA=rfAg6qN8=81dHYiPgx+c6sa63rYneZQ@mail.gmail.com>
Message-ID: <CAMnjH4RkMcKPB2xHbCkO=2FdtdsZKk=pcsQJZf89rcUjGqqCmg@mail.gmail.com>

Thanks .. ill try, let you know monday .. :)

-Torstein
On Jun 6, 2012 7:02 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> Not sure if this will work but try executing the set inline
>
> <action application="conference" data="1234 at default" inline="true"/>
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 6, 2012 12:56 PM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
> wrote:
>
>> Hi
>>
>> Sure?  I did try that, but it does not work .. Im not doing bridge, but
>> conference .. ..
>>
>> -T
>> On Jun 6, 2012 5:06 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>>
>>> You need to be setting effective_caller_id_name before the bridge.
>>>
>>> Brian Foster
>>> Endigo Computer LLC
>>>
>>> Sent from a mobile device.
>>> On Jun 6, 2012 6:19 AM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
>>> wrote:
>>>
>>>> Hi
>>>>
>>>> I'm trying to control the "caller_id_name" variable used in
>>>> conferencing.
>>>> My dialplan looks like this :
>>>>
>>>> <extension name="public_extensions">
>>>> <condition field="destination_number" expression="^(0meet)$">
>>>> <action application="conference" data="72013 at webconf"/>
>>>>
>>>> When I call into the conference, I do an API "conference
>>>> xml_list" which gives details of the conferencees ..
>>>> I want to be able to change the caller_id_name in the listing, but all
>>>> my efforts for doing this via dialplan fails.
>>>>
>>>> Is this even possible somehow ?
>>>>
>>>> I tried to add this to the dialplan before executing the conference :
>>>> <action application="export" data="caller_id_name=Hobbes" />
>>>> <action application="set" data="caller_id_name=Hobbes" />
>>>> <action application="export" data="effective_caller_id_name=Hobbes" />
>>>> <action application="export" data="effective_caller_id_name=Hobbes" />
>>>> <action application="set" data="origination_caller_id_name=Hobbes" />
>>>> <action application="export" data="origination_caller_id_name=Hobbes" />
>>>>
>>>> But to no avail...
>>>>
>>>> Any pointers would be appreciated.
>>>>
>>>> Thank you
>>>> Torstein
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sdevoy at bizfocused.com  Wed Jun  6 21:45:08 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 6 Jun 2012 13:45:08 -0400
Subject: [Freeswitch-users] Maillist client software - NOOB
Message-ID: <0a2c01cd440c$1da39ad0$58ead070$@bizfocused.com>

HI All,

 

I have been getting more and more familiar with FS and can actually make
some contributions to the questions coming through in the maillist.
However, when I signed up I chose the option to bundle the messages before
sending them to me. - BIG MISTAKE -.  I use MS Outlook as my mail client.

 

When they all come as attachments I can access them and respond well enough,
but then I have no useful interface features like:

.         SEARCH

.         Next/Prev in Thread, etc.

 

So, I have 2 questions:

1.       How do I un-bundle the list mailings?

or better still



2.       Does anyone have a good client software package (or web site) that
provides a better user interface?

 

Thanks,

Sean

 

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From cliff at develix.com  Wed Jun  6 21:51:56 2012
From: cliff at develix.com (Cliff Wells)
Date: Wed, 06 Jun 2012 10:51:56 -0700
Subject: [Freeswitch-users] CDR recovery on crash
In-Reply-To: <BLU162-W4036FF37BC59F586695152880D0@phx.gbl>
References: <BLU162-W4036FF37BC59F586695152880D0@phx.gbl>
Message-ID: <1339005116.2103.10.camel@portable-evil>

On Wed, 2012-06-06 at 14:11 +0500, awais nazir wrote:
> Greetings freeswitchers,
> 
>  See if you can help in this.If a freeswitch crashes abnormally , can
> we somehow retrieve the CDR before its death.   A single abnormal
> crash in a month gives us some loss as I run around 1200 CC.

We're using cdr_pg_csv for a fairly high call volume and have dealt with
hardware issues that caused FS crashes without losing significant CDR
data.

Regards,
Cliff





From avi at avimarcus.net  Wed Jun  6 22:15:28 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 6 Jun 2012 21:15:28 +0300
Subject: [Freeswitch-users] CDR recovery on crash
In-Reply-To: <1339005116.2103.10.camel@portable-evil>
References: <BLU162-W4036FF37BC59F586695152880D0@phx.gbl>
	<1339005116.2103.10.camel@portable-evil>
Message-ID: <CALNZuwnxmTOqUcFRrYB0tPJ_CnyRUM7BCAVzqamLWurma9eAyA@mail.gmail.com>

Cliff, I believe he is referring to losing the CDRs for active calls. FS
only submits/writes the CDRs upon the end of the call, hence the issue. I
don't believe he's talking about data corruption.

-Avi


On Wed, Jun 6, 2012 at 8:51 PM, Cliff Wells <cliff at develix.com> wrote:

> On Wed, 2012-06-06 at 14:11 +0500, awais nazir wrote:
> > Greetings freeswitchers,
> >
> >  See if you can help in this.If a freeswitch crashes abnormally , can
> > we somehow retrieve the CDR before its death.   A single abnormal
> > crash in a month gives us some loss as I run around 1200 CC.
>
> We're using cdr_pg_csv for a fairly high call volume and have dealt with
> hardware issues that caused FS crashes without losing significant CDR
> data.
>
> Regards,
> Cliff
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun  6 22:24:47 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Jun 2012 13:24:47 -0500
Subject: [Freeswitch-users] CDR recovery on crash
In-Reply-To: <CALNZuwnxmTOqUcFRrYB0tPJ_CnyRUM7BCAVzqamLWurma9eAyA@mail.gmail.com>
References: <BLU162-W4036FF37BC59F586695152880D0@phx.gbl>
	<1339005116.2103.10.camel@portable-evil>
	<CALNZuwnxmTOqUcFRrYB0tPJ_CnyRUM7BCAVzqamLWurma9eAyA@mail.gmail.com>
Message-ID: <CAKbxfG_+Ju97cty5kL1=DEdUHSJbyrwPh8Ay85vqT2wwPP=wpQ@mail.gmail.com>

Track calls writes xml very into the db many times per call.
On Jun 6, 2012 1:17 PM, "Avi Marcus" <avi at avimarcus.net> wrote:

> Cliff, I believe he is referring to losing the CDRs for active calls. FS
> only submits/writes the CDRs upon the end of the call, hence the issue. I
> don't believe he's talking about data corruption.
>
> -Avi
>
>
> On Wed, Jun 6, 2012 at 8:51 PM, Cliff Wells <cliff at develix.com> wrote:
>
>> On Wed, 2012-06-06 at 14:11 +0500, awais nazir wrote:
>> > Greetings freeswitchers,
>> >
>> >  See if you can help in this.If a freeswitch crashes abnormally , can
>> > we somehow retrieve the CDR before its death.   A single abnormal
>> > crash in a month gives us some loss as I run around 1200 CC.
>>
>> We're using cdr_pg_csv for a fairly high call volume and have dealt with
>> hardware issues that caused FS crashes without losing significant CDR
>> data.
>>
>> Regards,
>> Cliff
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From toddb at toddbailey.net  Wed Jun  6 22:43:47 2012
From: toddb at toddbailey.net (toddb at toddbailey.net)
Date: Wed, 06 Jun 2012 11:43:47 -0700
Subject: [Freeswitch-users] Skype on FreeSwitch install issues
Message-ID: <20120606114347.33e327b490679d2282e332758c73b55b.9c3630c43f.wbe@email14.secureserver.net>

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From cliff at develix.com  Wed Jun  6 22:51:03 2012
From: cliff at develix.com (Cliff Wells)
Date: Wed, 06 Jun 2012 11:51:03 -0700
Subject: [Freeswitch-users] Maillist client software - NOOB
In-Reply-To: <0a2c01cd440c$1da39ad0$58ead070$@bizfocused.com>
References: <0a2c01cd440c$1da39ad0$58ead070$@bizfocused.com>
Message-ID: <1339008663.2103.12.camel@portable-evil>

On Wed, 2012-06-06 at 13:45 -0400, Sean Devoy wrote:
> HI All,
> 
>  
> 
> I have been getting more and more familiar with FS and can actually
> make some contributions to the questions coming through in the
> maillist.  However, when I signed up I chose the option to bundle the
> messages before sending them to me. ? BIG MISTAKE ?.  I use MS Outlook
> as my mail client.

Go to the mailman link at the bottom of every email.  Change your
preferences:

Would you like to receive list mail batched in a daily digest?  No.

Regards,
Cliff




From jchavanton at gmail.com  Wed Jun  6 23:30:23 2012
From: jchavanton at gmail.com (Julien Chavanton)
Date: Wed, 6 Jun 2012 15:30:23 -0400
Subject: [Freeswitch-users] scope of channel variables,
 when are they accessible ?
Message-ID: <CAKmcL2mEE4kpV6A507r+EWORq8561F1VRFBP+7_6ow2d8ruijA@mail.gmail.com>

I must have misunderstood something with (Bridge)

The same dialplan (context) was executed for the new outgoing leg.

I added this condition in the context / extension to stop the B Leg from
processing the dial plan.

<condition field="${direction}" expression="^outbound$"
break="on-true"></condition>

Strange I was not expecting Bridge to behave this way.



From: Julien Chavanton <jchavanton at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Date: Wed, 6 Jun 2012 11:10:03 -0400
> Subject: [Freeswitch-users] scope of channel variables, when are they
> accessible ?
> I do not understand the scope of channel variables, when are they
> accessible ?
>
> For example, I am looking for the sip username of the caller call leg.
>
> In this example ${sip_auth_username} is empty when logging or calling
> sf_update.pl but it is correct when calling sf_hangup.pl
>
> <include>
>         <extension name="outdialing">
>                 <condition field="destination_number" expression=".*)">
>                         <action application="answer"/>
>                         <action application="log" data="VARIABLES
> [${sip_auth_username}][${Caller-Username}][${sip_from_user}][${effective_caller_id_name}][${effective_caller_id_number}]"/>
>                         <action application="perl" data="/var/fs_hook/
> sf_update.pl ${sip_auth_username} ${sip_to_user}"/>
>                         <action application="set"
> data="api_hangup_hook=perl /var/fs_hook/sf_hangup.pl
> ${sip_auth_username} ${sip_to_user} "/>
>                         <action application="bridge"
>  data="{absolute_codec_string='PCMA,PCMU,G729',call_leg=B,session_in_hangup_hook=true'}sofia/gateway/XXXX/XXXX$1"/>
>
>                  </condition>
> </extension>
> </include>
>
>
>
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From jaasmailing at gmail.com  Thu Jun  7 00:13:08 2012
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Wed, 06 Jun 2012 22:13:08 +0200
Subject: [Freeswitch-users] RTP NAT issue
In-Reply-To: <CAKV-jFxUFOtY6EzmKMv-3pZyvBf5WuyvFJOafh7=rWnWg=iW-w@mail.gmail.com>
References: <4FCF81C6.4010004@gmail.com>
	<CAKV-jFxUFOtY6EzmKMv-3pZyvBf5WuyvFJOafh7=rWnWg=iW-w@mail.gmail.com>
Message-ID: <4FCFB9D4.1090504@gmail.com>

Ok, I'll try the parameter NDLB. Is it suitable in an hosted environment 
(thousand of extensions)?


Anyway, the problem is that I can't use STUN (phone side configuration) 
because I have two needs:

1) Phone A (192.168.0.100) calls Phone B (192.168.0.101), in this case 
the RTP flow should be sent directly between two endpoints. If I use 
STUN the SDP "A" contains the public IP and not the private IP (that 
must be used for device reachability).

2) Phone A (192.168.0.100) calls Freeswitch (Public IP), in this case 
the RTP flow shoud be sent to Freeswitch Public IP and Freeswitch shoud 
send the RTP to the Phone A NAT IP. Here the STUN is the best solution.

What is the best practice in this scenario?

Regards,


Il 06/06/12 18.28, Brian Foster ha scritto:
>
> The issue is more likely the phone, as the phone is responsible for 
> handing FS the correct IP. There is however a way to force this on the 
> FS side but may break other devices. Please take a look at NDLB (No 
> Device Left Behind) parameters for Sofia on the wiki.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
>
> On Jun 6, 2012 12:15 PM, "Carlo Dimaggio" <jaasmailing at gmail.com 
> <mailto:jaasmailing at gmail.com>> wrote:
>
>     Hi all,
>
>     I have a problem with RTP and NAT.
>     The scenario is Hosted PBX and Natted phones (yealink):
>
>     Phones (192.168.0.x) - NAT -> FS (public IP)
>
>     When I call FS (for example an IVR) from a Phone, FS send the RTP
>     to the private address (192.168.0.x) instead to the public NAT IP.
>     The registration is ok:
>

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From drk at drkngs.net  Thu Jun  7 00:59:19 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Wed, 06 Jun 2012 13:59:19 -0700
Subject: [Freeswitch-users] RTP NAT issue
In-Reply-To: <4FCFB9D4.1090504@gmail.com>
Message-ID: <20120606205919.8a01a11c@mail.tritonwest.net>

For a hosted environment, where you're not in control of the users devices/routers you should do the following:  
   
In the SIP profile, turn on agressive_nat_detection, on the client device have them turn off ALL nat mapping stuff, so the switch can detect it's nat, and if it don't work cause they have some broken router that's doing ALG, and only in that case then set the directory entry for that use to have the "NDLB-Connnectile-Dysfunction" be true.   
   
That should work every time.  
   
   
--Dave
      _____  

  From: Carlo Dimaggio [mailto:jaasmailing at gmail.com]
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, 06 Jun 2012 13:13:08 -0700
Subject: Re: [Freeswitch-users] RTP NAT issue

Ok, I'll try the parameter NDLB. Is it suitable in an hosted environment (thousand of extensions)?


Anyway, the problem is that I can't use STUN (phone side configuration) because I have two needs:

1) Phone A (192.168.0.100) calls Phone B (192.168.0.101), in this case the RTP flow should be sent directly between two endpoints. If I use STUN the SDP "A" contains the public IP and not the private IP (that must be used for device reachability).

2) Phone A (192.168.0.100) calls Freeswitch (Public IP), in this case the RTP flow shoud be sent to Freeswitch Public IP and Freeswitch shoud send the RTP to the Phone A NAT IP. Here the STUN is the best solution.

What is the best practice in this scenario?
 
Regards,


Il 06/06/12 18.28, Brian Foster ha scritto:     

The issue is more likely the phone, as the phone is responsible for handing FS the correct IP. There is however a way to force this on the FS side but may break other devices. Please take a look at NDLB (No Device Left Behind) parameters for Sofia on the wiki.  

Brian Foster
Endigo Computer LLC  

Sent from a mobile device.  
On Jun 6, 2012 12:15 PM, "Carlo Dimaggio" <jaasmailing at gmail.com> wrote:
    
Hi all,

I have a problem with RTP and NAT.
The scenario is Hosted PBX and Natted phones (yealink):

Phones (192.168.0.x) - NAT -> FS (public IP) 

When I call FS (for example an IVR) from a Phone, FS send the RTP to the private address (192.168.0.x) instead to the public NAT IP.
The registration is ok:

      
   
 
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From gmaruzz at gmail.com  Thu Jun  7 02:21:33 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 7 Jun 2012 00:21:33 +0200
Subject: [Freeswitch-users] Skype on FreeSwitch install issues
In-Reply-To: <20120606114347.33e327b490679d2282e332758c73b55b.9c3630c43f.wbe@email14.secureserver.net>
References: <20120606114347.33e327b490679d2282e332758c73b55b.9c3630c43f.wbe@email14.secureserver.net>
Message-ID: <CALXCt0oUA+e4KuxHKQ=csx6h=v-SmVgRT8KkP-qvoDjwOqQ2_A@mail.gmail.com>

On Wed, Jun 6, 2012 at 8:43 PM, <toddb at toddbailey.net> wrote:

>  Even though I had errors in the skypopen build process, I tried the
> install.pl script
> It resulted in more errors.
>

you need to follow strictly the wiki page, step by step:

1) build ALL the thingies (included skypopen.ko oss driver kernel module) -
without errors
2) run the install.pl script
3) run the startup script created by the install.pl script

the /boot directory has nothing to do with it, leave it alone.

your alsa install will probably create the /dev/dsp each reboot. So, you
need to run the startup script that will be created by the install script
after each reboot and before FreeSWITCH.

Please follow strictly step by step, or know exactly what you are doing ;) :

http://wiki.freeswitch.org/wiki/Skypopen#SHORT_BLUEPRINT:_STEPS_NEEDED_TO_USE_SKYPOPEN

http://wiki.freeswitch.org/wiki/Skypopen#Linux

at this last link, read the entire Linux subsection, at least until the
"windows" subsection begins, and you'll find peace and joy!

-giovanni





>
> Big concern is if I am able to sucessfully run the insmod (after disabling
> other snd devices) will doing so permanently disable them?  I'm thinking I
> need to make a backup of the /boot folder.   Any other folder that I need
> to address?
>
>
>
> -------- Original Message --------
> Subject: Re: [Freeswitch-users] Skype on FreeSwitch install issues
> From: Giovanni Maruzzelli <gmaruzz at gmail.com>
> Date: Wed, June 06, 2012 10:17 am
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
> please use the automatic installer (install.pl) it will do it all for you
>
> if not using automatic installer, do something similar to:
>
> #Unload possible ALSA sound modules that would conflict with our OSS fake
> module
> rmmod snd_pcm_oss
> rmmod snd_mixer_oss
> rmmod snd_seq_oss
> sleep 1
> #Create the inode our fake sound driver will use
> mknod /dev/dsp c 14 3
> #Load our OSS fake module
> insmod /usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko
>
> this will work
>
> Explanation is: your alsa installation creates automatically /dev/dsp for
> compatibility with OSS, but you do not need it, if you are using ALSA. (You
> would need it to use alsa from OSS application, but this is not what you
> need).
>
> -giovanni
>
> On Wed, Jun 6, 2012 at 5:36 PM, <toddb at toddbailey.net> wrote:
>
>>  Hello,
>>
>> New to list & appologies if this is the wrong place to post questions.
>>
>> Assuming this is the place to be, here is a question.
>>
>> Whet I get to the build step for skypopen
>> "make clean; make; insmod ./skypopen.ko; mknod /dev/dsp c 14 3"
>>
>> I get an error with the insmod cmd,  resource / device busy
>>
>>
>> I think it's caused due to having several snd entries listed via the
>> "lsmod |grep snd"
>>
>> The machine is question is a media, database and email server.
>> And the media server component requires the use of ALSA components.
>>
>> Does this requirement prohibit the use of the skype component in
>> Freeswitch or do I just need to prevent the snd modules from loading to
>> complete the kernel mod? Then re enable snd modules at next boot.
>>
>>
>> If this is the case any suggestions?   Maybe running FS in a Vbox to
>> provide the functionality?
>> Or do I need a dedicated box just running linux and FS components?
>>
>> current machine is fedora 14 x64 on a 4 core xeon @ 3ghz w/7 gig memory,
>> w/ 35 % process load average
>>
>> thanks
>>
>>
>> BTW: my desired configuration
>>
>> analog land line to Cisco SPA3102 router connected to the FS server with
>> 4 extensions registered.
>> outbound local calls router to analog line, 1+ area code out bound calls
>> routed to skype.
>> Incoming calls routed to all extensions, unanswered calls set to a ivr to
>> 1. leave voice mail or 2.forward to cell phone (via skype connection)
>>
>> any of this not do-able?
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From jeff at jefflenk.com  Thu Jun  7 02:50:06 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 6 Jun 2012 15:50:06 -0700 (PDT)
Subject: [Freeswitch-users] FSClient UA error
In-Reply-To: <4FCF8F0F.8010004@livecall.com>
References: <4FCF8F0F.8010004@livecall.com>
Message-ID: <1339023006371-7579490.post@n2.nabble.com>

Is something already using the ports that FSClient is trying to use? Are you
running fs on this box? Or another sip client?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-UA-error-tp7579476p7579490.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jack at livecall.com  Thu Jun  7 03:22:59 2012
From: jack at livecall.com (Jack)
Date: Wed, 06 Jun 2012 16:22:59 -0700
Subject: [Freeswitch-users] FSClient UA error
In-Reply-To: <1339023006371-7579490.post@n2.nabble.com>
References: <4FCF8F0F.8010004@livecall.com>
	<1339023006371-7579490.post@n2.nabble.com>
Message-ID: <4FCFE653.7020600@livecall.com>

5060 should be available, I do run other clients that use 5060  but make 
sure they are all the way closed out before trying FSClient.
FS is running on a different box.

On 6/6/2012 3:50 PM, Jeff Lenk wrote:
> Is something already using the ports that FSClient is trying to use? Are you
> running fs on this box? Or another sip client?
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-UA-error-tp7579476p7579490.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From sherifomran2000 at yahoo.com  Thu Jun  7 03:40:20 2012
From: sherifomran2000 at yahoo.com (Sherif Omran)
Date: Wed, 6 Jun 2012 16:40:20 -0700 (PDT)
Subject: [Freeswitch-users] debug xml curl
In-Reply-To: <4FCFE653.7020600@livecall.com>
Message-ID: <1339026020.26523.YahooMailClassic@web110811.mail.gq1.yahoo.com>

Hello guys,

any body has a good way to debug data sent by xml_curl rather than 

xml_curl debug on

I need to see it in the debug console ./fs_cli

Till now it produces xml files in the tmp

any ideas?

regards,
Sherif Omran



From mitch.capper at gmail.com  Thu Jun  7 03:54:14 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 6 Jun 2012 16:54:14 -0700
Subject: [Freeswitch-users] FSClient UA error
In-Reply-To: <4FCFE653.7020600@livecall.com>
References: <4FCF8F0F.8010004@livecall.com>
	<1339023006371-7579490.post@n2.nabble.com>
	<4FCFE653.7020600@livecall.com>
Message-ID: <CALn233qfV_onHuCJCXtm-Lp7V+DtLd-Fy=Ji3q3DV9ndfCjVtg@mail.gmail.com>

You can use fs_cli to connect up to see what error is happening but
make sure you have TLS disabled and change the ports in the sofia
settings and see if that fixes it.

~Mitch

On Wed, Jun 6, 2012 at 4:22 PM, Jack <jack at livecall.com> wrote:
> 5060 should be available, I do run other clients that use 5060 ?but make
> sure they are all the way closed out before trying FSClient.
> FS is running on a different box.
>
> On 6/6/2012 3:50 PM, Jeff Lenk wrote:
>> Is something already using the ports that FSClient is trying to use? Are you
>> running fs on this box? Or another sip client?
>>
>> --
>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-UA-error-tp7579476p7579490.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From dujinfang at gmail.com  Thu Jun  7 03:56:21 2012
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 7 Jun 2012 07:56:21 +0800
Subject: [Freeswitch-users] Sending arbitrary NOTIFY messages
In-Reply-To: <CAKzWOxVF_01Bf4=mKUjHeKtUict1uNO52nvuagZb5zQJcX6bXw@mail.gmail.com>
References: <CAKzWOxVF_01Bf4=mKUjHeKtUict1uNO52nvuagZb5zQJcX6bXw@mail.gmail.com>
Message-ID: <9442B870B78E4B18B72EDDFEB1E68547@gmail.com>

Hi michael, 

Doesn't Notify need a subscribe? If not the remote might return Call does not exists here or sth.

I had played some in the old days and eventually I made a patch

http://jira.freeswitch.org/browse/FS-3802


It looks that the current notify queries the sip_registrations table so non-regged doesn't work.

Attached some lua I played in the old days. they should work without the above patch, however, in n.lua I might had another patch which added an aor header(at least) so it's possible to use another remote ip but I cannot find the patch. Included a patch that similar to that.

Note: all the code was I played in the old days so I don't know if they are working. Let me know if it helps.


-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn

Sent with Sparrow (http://www.sparrowmailapp.com)


On Saturday, June 2, 2012 at 3:18 AM, Michael Collins wrote:

> Has anyone gotten the magic formula down for doing this? I'm trying to send a NOTIFY to a non-registered endpoint but I can't quite get the event headers correct. I found this old commit that Tony did about 3 years ago that suggests it's possible:
> http://fisheye.freeswitch.org/changelog/freeswitch.git/?showid=1fa1e961e4587475a51dcbadd31765b5a06d3115
> 
> If you know what the necessary headers are, or better yet, if you have a working example please let me know. And yes, I will gladly pay the wiki tax on this one. ;)
> 
> Thanks,
> MC
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> 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
> 
> 


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From sdevoy at bizfocused.com  Thu Jun  7 03:58:14 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 6 Jun 2012 19:58:14 -0400
Subject: [Freeswitch-users] Maillist client software - NOOB
In-Reply-To: <1339008663.2103.12.camel@portable-evil>
References: <0a2c01cd440c$1da39ad0$58ead070$@bizfocused.com>
	<1339008663.2103.12.camel@portable-evil>
Message-ID: <0c4801cd4440$3c748f80$b55dae80$@bizfocused.com>

Thanks Cliff.  I have tried that many, many times.  Every attempt gets me an email saying someone has tried to sign up my address again.  I assume I don't have the right password.

Thanks again



-----Original Message-----
From: Cliff Wells [mailto:cliff at develix.com] 
Sent: Wednesday, June 06, 2012 2:51 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Maillist client software - NOOB

On Wed, 2012-06-06 at 13:45 -0400, Sean Devoy wrote:
> HI All,
> 
>  
> 
> I have been getting more and more familiar with FS and can actually 
> make some contributions to the questions coming through in the 
> maillist.  However, when I signed up I chose the option to bundle the 
> messages before sending them to me. ? BIG MISTAKE ?.  I use MS Outlook 
> as my mail client.

Go to the mailman link at the bottom of every email.  Change your
preferences:

Would you like to receive list mail batched in a daily digest?  No.

Regards,
Cliff








From sdevoy at bizfocused.com  Thu Jun  7 04:00:47 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 6 Jun 2012 20:00:47 -0400
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
Message-ID: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>

The whole mailing list idea is so "1990's"

Is there interest in having a nice Message Forum instead of the mailing
list?  Viewing threaded items, Searching, seeing what's new and what's hot,
etc?

 

I have a hosting server with the capacity to host a forum site in ASP.NET or
PHP and MS Sql Server.  I am willing to purchase the software (if it is not
public domain) and maintain it on my server if people like the idea.  I just
don't want to throw down a couple hundred $ for nothing.  

 

Sean

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From brian at freeswitch.org  Thu Jun  7 04:07:34 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jun 2012 19:07:34 -0500
Subject: [Freeswitch-users] Sending arbitrary NOTIFY messages
In-Reply-To: <9442B870B78E4B18B72EDDFEB1E68547@gmail.com>
References: <CAKzWOxVF_01Bf4=mKUjHeKtUict1uNO52nvuagZb5zQJcX6bXw@mail.gmail.com>
	<9442B870B78E4B18B72EDDFEB1E68547@gmail.com>
Message-ID: <6853368876973493042@unknownmsgid>

it's the gratuitous notify on register for mwi

Sent from my iPad

On Jun 6, 2012, at 6:57 PM, Seven Du <dujinfang at gmail.com> wrote:

Hi michael,

Doesn't Notify need a subscribe? If not the remote might return Call does
not exists here or sth.

I had played some in the old days and eventually I made a patch

http://jira.freeswitch.org/browse/FS-3802


It looks that the current notify queries the sip_registrations table so
non-regged doesn't work.

Attached some lua I played in the old days. they should work without the
above patch, however, in n.lua I might had another patch which added an aor
header(at least) so it's possible to use another remote ip but I cannot
find the patch. Included a patch that similar to that.

Note: all the code was I played in the old days so I don't know if they are
working. Let me know if it helps.


-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn

Sent with Sparrow <http://www.sparrowmailapp.com>

On Saturday, June 2, 2012 at 3:18 AM, Michael Collins wrote:

Has anyone gotten the magic formula down for doing this? I'm trying to send
a NOTIFY to a non-registered endpoint but I can't quite get the event
headers correct. I found this old commit that Tony did about 3 years ago
that suggests it's possible:
http://fisheye.freeswitch.org/changelog/freeswitch.git/?showid=1fa1e961e4587475a51dcbadd31765b5a06d3115

If you know what the necessary headers are, or better yet, if you have a
working example please let me know. And yes, I will gladly pay the wiki tax
on this one. ;)

Thanks,
MC
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


 <notify.tar.gz>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From vvenkatar at gmail.com  Thu Jun  7 04:16:19 2012
From: vvenkatar at gmail.com (Venkatesh)
Date: Wed, 6 Jun 2012 17:16:19 -0700
Subject: [Freeswitch-users] Using CURL inside a JavaScript.
Message-ID: <CAP_69muchUh9-PWbO5GLGv_DBS=M5x1uTJh0T15UOo+rksqmow@mail.gmail.com>

Hi !

I am developing a simple IVR application using JS. One of the things I want
to do when my JS fires is to issue a WS API to fetch some data relevant to
the call; wait for the response and use information in the response to
process the incoming call. To meet that end, I wrote up a simple JS that
issues a CURL request providing appropriate call back function. However, I
dont see any callback being fired back to my application. The JS simply
proceeds to execute the next instruction. I know my WS returns a string
(verified by running the exact same command via a command line). I wanted
to know if there is anything I need to do to make the application "wait for
the call back".  Would appreciate any help.

Regards,
Venkatesh
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From freeswitch-list at puzzled.xs4all.nl  Thu Jun  7 05:45:52 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 07 Jun 2012 03:45:52 +0200
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
 list?
In-Reply-To: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
Message-ID: <4FD007D0.7010705@puzzled.xs4all.nl>

On 07-06-12 02:00, Sean Devoy wrote:
> The whole mailing list idea is so "1990's"
> 
> Is there interest in having a nice Message Forum instead of the mailing
> list?  Viewing threaded items, Searching, seeing what?s new and what?s
> hot, etc?

Since it has been a while I guess it was inevitable that someone would
come around with the forum idea again. Have you searched the mailing
list archives and read the previous discussion on this subject?
It did not work for the Asterisk community and I doubt it would work for
this community. If you want forum style, use nabble or gmane.

Regards,
Patrick


From brian at freeswitch.org  Thu Jun  7 05:48:06 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jun 2012 20:48:06 -0500
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <4FD007D0.7010705@puzzled.xs4all.nl>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
	<4FD007D0.7010705@puzzled.xs4all.nl>
Message-ID: <-3243025629320733747@unknownmsgid>

Hammer, meet Nail!

Sent from my iPad

On Jun 6, 2012, at 8:47 PM, Patrick Lists
<freeswitch-list at puzzled.xs4all.nl> wrote:

> Since it has been a while I guess it was inevitable that someone would
> come around with the forum idea again. Have you searched the mailing
> list archives and read the previous discussion on this subject?
> It did not work for the Asterisk community and I doubt it would work for
> this community. If you want forum style, use nabble or gmane.
>
> Regards,
> Patrick


From ifoundthetao at gmail.com  Thu Jun  7 05:50:21 2012
From: ifoundthetao at gmail.com (Timothy Bolton)
Date: Wed, 06 Jun 2012 20:50:21 -0500
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
 list?
In-Reply-To: <4FD007D0.7010705@puzzled.xs4all.nl>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
	<4FD007D0.7010705@puzzled.xs4all.nl>
Message-ID: <4FD008DD.4060206@gmail.com>

:)

Here's the link to the forum-ified mailing list: 
http://freeswitch-users.2379917.n2.nabble.com/

If you're unfamiliar with it, you can log in, and reply as well.  When 
you do that, you also reply to the list.

'We who cut mere stones must always be envisioning cathedrals.'
Quarry Worker's Creed


On 6/6/2012 8:45 PM, Patrick Lists wrote:
> On 07-06-12 02:00, Sean Devoy wrote:
>> The whole mailing list idea is so "1990's"
>>
>> Is there interest in having a nice Message Forum instead of the mailing
>> list?  Viewing threaded items, Searching, seeing what?s new and what?s
>> hot, etc?
> Since it has been a while I guess it was inevitable that someone would
> come around with the forum idea again. Have you searched the mailing
> list archives and read the previous discussion on this subject?
> It did not work for the Asterisk community and I doubt it would work for
> this community. If you want forum style, use nabble or gmane.
>
> Regards,
> Patrick
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Jun  7 05:54:37 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jun 2012 20:54:37 -0500
Subject: [Freeswitch-users] debug xml curl
In-Reply-To: <1339026020.26523.YahooMailClassic@web110811.mail.gq1.yahoo.com>
References: <1339026020.26523.YahooMailClassic@web110811.mail.gq1.yahoo.com>
Message-ID: <1852706752797320567@unknownmsgid>

Do not hijack threads please, start your own new message to the list.
clicking reply then changing the subject hijacks the thread.

/b

Sent from my iPad

On Jun 6, 2012, at 6:41 PM, Sherif Omran <sherifomran2000 at yahoo.com> wrote:

> Hello guys,
>
> any body has a good way to debug data sent by xml_curl rather than
>
> xml_curl debug on
>
> I need to see it in the debug console ./fs_cli
>
> Till now it produces xml files in the tmp
>
> any ideas?
>
> regards,
> Sherif Omran
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Jun  7 05:55:41 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jun 2012 20:55:41 -0500
Subject: [Freeswitch-users] RTP NAT issue
In-Reply-To: <20120606205919.8a01a11c@mail.tritonwest.net>
References: <20120606205919.8a01a11c@mail.tritonwest.net>
Message-ID: <5971058602431512241@unknownmsgid>

Suboptimal solution.  you'll break transfers among other things.

Sent from my iPad

On Jun 6, 2012, at 4:01 PM, "Dave R. Kompel" <drk at drkngs.net> wrote:

> For a hosted environment, where you're not in control of the users devices/routers you should do the following:
>
> In the SIP profile, turn on agressive_nat_detection, on the client device have them turn off ALL nat mapping stuff, so the switch can detect it's nat, and if it don't work cause they have some broken router that's doing ALG, and only in that case then set the directory entry for that use to have the "NDLB-Connnectile-Dysfunction" be true.
>
> That should work every time.
>
>
> --Dave


From bdfoster at endigotech.com  Thu Jun  7 06:11:41 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 6 Jun 2012 22:11:41 -0400
Subject: [Freeswitch-users] RTP NAT issue
In-Reply-To: <5971058602431512241@unknownmsgid>
References: <20120606205919.8a01a11c@mail.tritonwest.net>
	<5971058602431512241@unknownmsgid>
Message-ID: <CAKV-jFycaojiGoVRLvV9PHJbWML0rjQaQhC98n-XDz7GJt9+0w@mail.gmail.com>

Maybe doing a sip proxy on the nat'd side?

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 6, 2012 9:57 PM, "Brian West" <brian at freeswitch.org> wrote:

> Suboptimal solution.  you'll break transfers among other things.
>
> Sent from my iPad
>
> On Jun 6, 2012, at 4:01 PM, "Dave R. Kompel" <drk at drkngs.net> wrote:
>
> > For a hosted environment, where you're not in control of the users
> devices/routers you should do the following:
> >
> > In the SIP profile, turn on agressive_nat_detection, on the client
> device have them turn off ALL nat mapping stuff, so the switch can detect
> it's nat, and if it don't work cause they have some broken router that's
> doing ALG, and only in that case then set the directory entry for that use
> to have the "NDLB-Connnectile-Dysfunction" be true.
> >
> > That should work every time.
> >
> >
> > --Dave
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bdfoster at endigotech.com  Thu Jun  7 06:13:23 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 6 Jun 2012 22:13:23 -0400
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
Message-ID: <CAKV-jFwP0d8B_bZpXN+_mqO6ig9LEhMWhD4=g1oeiSGSYMTjwQ@mail.gmail.com>

Nabble works fine for that kind of thing ..

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 6, 2012 8:01 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> The whole mailing list idea is so "1990's"****
>
> Is there interest in having a nice Message Forum instead of the mailing
> list?  Viewing threaded items, Searching, seeing what?s new and what?s hot,
> etc?****
>
> ** **
>
> I have a hosting server with the capacity to host a forum site in ASP.NETor PHP and MS Sql Server.  I am willing to purchase the software (if it is
> not public domain) and maintain it on my server if people like the idea.  I
> just don't want to throw down a couple hundred $ for nothing.  ****
>
> ** **
>
> Sean****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ifoundthetao at gmail.com  Thu Jun  7 06:31:52 2012
From: ifoundthetao at gmail.com (Timothy Bolton)
Date: Wed, 06 Jun 2012 21:31:52 -0500
Subject: [Freeswitch-users] Using CURL inside a JavaScript.
In-Reply-To: <CAP_69muchUh9-PWbO5GLGv_DBS=M5x1uTJh0T15UOo+rksqmow@mail.gmail.com>
References: <CAP_69muchUh9-PWbO5GLGv_DBS=M5x1uTJh0T15UOo+rksqmow@mail.gmail.com>
Message-ID: <4FD01298.90402@gmail.com>

Do you have any code that you want to post (http://pastebin.freeswitch.org)?

Also, you may want to look into timers/intervals (setTimer 
setInterval).  I use those to keep tabs on responses from requests.  
Otherwise, the JavaScript will continue to execute without getting the 
response, which can be very frustrating.

'We who cut mere stones must always be envisioning cathedrals.'
Quarry Worker's Creed


On 6/6/2012 7:16 PM, Venkatesh wrote:
> Hi !
>
> I am developing a simple IVR application using JS. One of the things I 
> want to do when my JS fires is to issue a WS API to fetch some data 
> relevant to the call; wait for the response and use information in the 
> response to process the incoming call. To meet that end, I wrote up a 
> simple JS that issues a CURL request providing appropriate call back 
> function. However, I dont see any callback being fired back to my 
> application. The JS simply proceeds to execute the next instruction. I 
> know my WS returns a string (verified by running the exact same 
> command via a command line). I wanted to know if there is anything I 
> need to do to make the application "wait for the call back".  Would 
> appreciate any help.
>
> Regards,
> Venkatesh
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From ash at url.net.au  Thu Jun  7 06:32:12 2012
From: ash at url.net.au (Ashley Breeden)
Date: Thu, 7 Jun 2012 12:32:12 +1000
Subject: [Freeswitch-users] RTP NAT issue
In-Reply-To: <CAKV-jFxUFOtY6EzmKMv-3pZyvBf5WuyvFJOafh7=rWnWg=iW-w@mail.gmail.com>
References: <4FCF81C6.4010004@gmail.com>
	<CAKV-jFxUFOtY6EzmKMv-3pZyvBf5WuyvFJOafh7=rWnWg=iW-w@mail.gmail.com>
Message-ID: <DE54C6F5-9285-4454-B040-038FBE980AB1@url.net.au>

Try this:

- Enable agressive_nat_detection on the profile as per Dave's suggestion
- Make sure SIP ALG is not enabled on the NAT router
- On the Yealink set Rport enabled (Click Accounts, Advanced, set Rport drop down to enabled)

I find this works 99% of the time.  When I look at my registrations from a Yealink the contact line is like this:

"John" <sip:1001 at 192.168.15.6:5062;received=PUBLIC_IP:1032;fs_nat=yes;fs_path=sip%3A1001%40 PUBLIC_IP%3A1032>

Ash.


On 07/06/2012, at 2:28 AM, Brian Foster wrote:

> The issue is more likely the phone, as the phone is responsible for handing FS the correct IP. There is however a way to force this on the FS side but may break other devices. Please take a look at NDLB (No Device Left Behind) parameters for Sofia on the wiki.
> 
> Brian Foster
> Endigo Computer LLC
> 
> Sent from a mobile device.
> 
> On Jun 6, 2012 12:15 PM, "Carlo Dimaggio" <jaasmailing at gmail.com> wrote:
> Hi all,
> 
> I have a problem with RTP and NAT.
> The scenario is Hosted PBX and Natted phones (yealink):
> 
> Phones (192.168.0.x) - NAT -> FS (public IP) 
> 
> When I call FS (for example an IVR) from a Phone, FS send the RTP to the private address (192.168.0.x) instead to the public NAT IP.
> The registration is ok:
> 
> freeswitch at internal> sofia status profile tenant1.bs.dev.voip.clio.it reg
> 1
> Registrations:
> =================================================================================================
> Call-ID:        488014850 at 192.168.0.100
> User:           202 at tenant1.test.com
> Contact:        "Test 202" <sip:202 at 192.168.0.100:5062;fs_nat=yes;fs_path=sip%3A202%40<NAT_IP>%3A37710>
> Agent:          Yealink SIP-T20P 9.61.0.70
> Status:         Registered(UDP-NAT)(unknown) EXP(2012-06-06 19:01:55) EXPSECS(3232)
> Host:           localhost.localdomain
> IP:             <NAT_IP>
> Port:           37710
> Auth-User:      202
> Auth-Realm:     tenant1.test.com
> MWI-Account:    202 at tenant1.test.com
> 
> 
> How I can tell FS to send the RTP Packets to the right address? I think is needed a "comedia mode" like in Asterisk (or RTPProxy in openser)...
> 
> 
> Best regards,
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Regards,

Ashley Breeden
URL Networks Pty Ltd
Email: ash at url.net.au
WWW: http://www.url.net.au
Phone: 1300 33 11 78
Direct: 03 9008 5901
Mobile: 0411 112 056

HostedPBX - VoIP - Hosting - Wholesale

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From drk at drkngs.net  Thu Jun  7 07:23:55 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Wed, 06 Jun 2012 20:23:55 -0700
Subject: [Freeswitch-users] RTP NAT issue
In-Reply-To: <5971058602431512241@unknownmsgid>
Message-ID: <20120607032355.d0d9d846@mail.tritonwest.net>

Been using this configuration for service providers for years, and always works. Everything always works, transfers etc...  
   
--Dave
      _____  

  From: Brian West [mailto:brian at freeswitch.org]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Wed, 06 Jun 2012 18:55:41 -0700
Subject: Re: [Freeswitch-users] RTP NAT issue

Suboptimal solution. you'll break transfers among other things.

Sent from my iPad

On Jun 6, 2012, at 4:01 PM, "Dave R. Kompel" <drk at drkngs.net> wrote:

> For a hosted environment, where you're not in control of the users devices/routers you should do the following:
>
> In the SIP profile, turn on agressive_nat_detection, on the client device have them turn off ALL nat mapping stuff, so the switch can detect it's nat, and if it don't work cause they have some broken router that's doing ALG, and only in that case then set the directory entry for that use to have the "NDLB-Connnectile-Dysfunction" be true.
>
> That should work every time.
>
>
> --Dave

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
      
   
 
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From vvenkatar at gmail.com  Thu Jun  7 07:36:23 2012
From: vvenkatar at gmail.com (Venkatesh)
Date: Wed, 6 Jun 2012 20:36:23 -0700
Subject: [Freeswitch-users] Using CURL inside a JavaScript.
In-Reply-To: <4FD01298.90402@gmail.com>
References: <CAP_69muchUh9-PWbO5GLGv_DBS=M5x1uTJh0T15UOo+rksqmow@mail.gmail.com>
	<4FD01298.90402@gmail.com>
Message-ID: <CAP_69ms2t-BrwgkjmGjwh4U9+=O706zg9vOPC-6cFdsJvS=kdQ@mail.gmail.com>

Hi !

I have uploaded the relevant JS code to pastebin:
http://pastebin.freeswitch.org/19238

Would really appreciate any help!

Venkatesh

On Wed, Jun 6, 2012 at 7:31 PM, Timothy Bolton <ifoundthetao at gmail.com>wrote:

>  Do you have any code that you want to post (
> http://pastebin.freeswitch.org)?
>
> Also, you may want to look into timers/intervals (setTimer setInterval).
> I use those to keep tabs on responses from requests.  Otherwise, the
> JavaScript will continue to execute without getting the response, which can
> be very frustrating.
>
> 'We who cut mere stones must always be envisioning cathedrals.'
> Quarry Worker's Creed
>
>
> On 6/6/2012 7:16 PM, Venkatesh wrote:
>
> Hi !
>
>  I am developing a simple IVR application using JS. One of the things I
> want to do when my JS fires is to issue a WS API to fetch some data
> relevant to the call; wait for the response and use information in the
> response to process the incoming call. To meet that end, I wrote up a
> simple JS that issues a CURL request providing appropriate call back
> function. However, I dont see any callback being fired back to my
> application. The JS simply proceeds to execute the next instruction. I know
> my WS returns a string (verified by running the exact same command via a
> command line). I wanted to know if there is anything I need to do to make
> the application "wait for the call back".  Would appreciate any help.
>
>  Regards,
> Venkatesh
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Jun  7 09:48:00 2012
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Thu, 7 Jun 2012 10:18:00 +0430
Subject: [Freeswitch-users] Sending Event Back to RTMP Sessions
Message-ID: <CABK0fC6K6vMcTTbJTVgxjEr3XUts=NPAxAwK6fiZzP=zGktJdQ@mail.gmail.com>

Hi,
Looking for howto send back a event to rtmp session, after some inspection
in code it seems that needs to be a SWITCH_EVENT_CUSTOM event (subclassed
to RTMP_EVENT_CUSTOM) and have a "RTMP-Session-ID" header to be able to
deliver to far end of link. I'll appreciate if to know that is all
requirements.

BEST,
-- afshin
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From sherifomran2000 at yahoo.com  Thu Jun  7 10:05:33 2012
From: sherifomran2000 at yahoo.com (Sherif Omran)
Date: Wed, 6 Jun 2012 23:05:33 -0700 (PDT)
Subject: [Freeswitch-users] Debugging xml curl
Message-ID: <1339049133.55011.YahooMailClassic@web110811.mail.gq1.yahoo.com>

Hello guys,

any body has a good way to debug data sent by xml_curl rather than

xml_curl debug on

I need to see it in the debug console ./fs_cli

Till now it produces xml files in the tmp

any ideas?

regards,
Sherif Omran


From jaasmailing at gmail.com  Thu Jun  7 11:11:51 2012
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Thu, 07 Jun 2012 09:11:51 +0200
Subject: [Freeswitch-users] RTP NAT issue
In-Reply-To: <5971058602431512241@unknownmsgid>
References: <20120606205919.8a01a11c@mail.tritonwest.net>
	<5971058602431512241@unknownmsgid>
Message-ID: <4FD05437.8070603@gmail.com>

Hi Brian,

what is for you the optimal solution in an hosted environment?


Regards


Il 07/06/12 03.55, Brian West ha scritto:
> Suboptimal solution.  you'll break transfers among other things.
>
> Sent from my iPad
>
> On Jun 6, 2012, at 4:01 PM, "Dave R. Kompel"<drk at drkngs.net>  wrote:
>
>> For a hosted environment, where you're not in control of the users devices/routers you should do the following:
>>
>> In the SIP profile, turn on agressive_nat_detection, on the client device have them turn off ALL nat mapping stuff, so the switch can detect it's nat, and if it don't work cause they have some broken router that's doing ALG, and only in that case then set the directory entry for that use to have the "NDLB-Connnectile-Dysfunction" be true.
>>
>> That should work every time.
>>
>>
>> --Dave
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From belintinc at gmail.com  Thu Jun  7 08:01:49 2012
From: belintinc at gmail.com (BELint Inc)
Date: Thu, 7 Jun 2012 09:31:49 +0530
Subject: [Freeswitch-users] Freeswitch is responding to contact's port when
	client behind NAT
Message-ID: <CAEA+2KvO2KiV8jpDnbh5dqiOZMk4Nke6E3UFLD9rJbmpzpuNhQ@mail.gmail.com>

Hi All,

We`ve been using freeswitch since very long, It was good all over but it
got stuck to a simple yet complex situation. We have Yealink T28P series
phones behind NAT and freeswitch is unable to get them register. In the
scenario freeswitch is at public IP. The reason we think is that freeswitch
is using sip port in contact header to respond to.

Is there anyway we can avoid this or force it to always send the response
to destination IP instead of contact or via header ports because clients
are behind NAT and response can only reach them if it respond to
destination IP & PORT.

Please revert if someone know how to solve this problem. It is very urgent.

Regards
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From awais-nazeer at hotmail.com  Thu Jun  7 11:24:35 2012
From: awais-nazeer at hotmail.com (awais nazir)
Date: Thu, 7 Jun 2012 12:24:35 +0500
Subject: [Freeswitch-users] CDR recovery on crash
In-Reply-To: <BLU162-W4036FF37BC59F586695152880D0@phx.gbl>
References: <BLU162-W4036FF37BC59F586695152880D0@phx.gbl>
Message-ID: <BLU162-W515DFBEA14FE6EDDD14A188F30@phx.gbl>



Hi Guys 

Can I get some useful information for that please.
From: awais-nazeer at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: CDR recovery on crash
Date: Wed, 6 Jun 2012 14:11:50 +0500





Greetings freeswitchers,

 See if you can help in this.If a 
freeswitch crashes abnormally
 , can we somehow retrieve the CDR before its death.   A single abnormal
 crash in a month gives us some loss as I run around 1200 CC.

-- 
waisee 		 	   		   		 	   		  
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From jaasmailing at gmail.com  Thu Jun  7 11:28:33 2012
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Thu, 07 Jun 2012 09:28:33 +0200
Subject: [Freeswitch-users] RTP NAT issue
In-Reply-To: <20120607032355.d0d9d846@mail.tritonwest.net>
References: <20120607032355.d0d9d846@mail.tritonwest.net>
Message-ID: <4FD05821.30500@gmail.com>

Hi Dave,

I've tried the "aggressive_nat_detection" on my internal profile. 
Restarted the profile and I have this registration:

Contact:        "Test 202" 
<sip:202 at 192.168.2.100:5062;fs_nat=yes;fs_path=sip%3A202%40<FS IP>%3A37710>

but FS send the RTP to the private IP (192.168.2.100) and not to the 
public NAT IP.

Is there something wrong?


Best Regards



Il 07/06/12 05.23, Dave R. Kompel ha scritto:
> Been using this configuration for service providers for years, and 
> always works. Everything always works, transfers etc...
> --Dave
>
>     ------------------------------------------------------------------------
>     *From:* Brian West [mailto:brian at freeswitch.org]
>     *To:* FreeSWITCH Users Help
>     [mailto:freeswitch-users at lists.freeswitch.org]
>     *Sent:* Wed, 06 Jun 2012 18:55:41 -0700
>     *Subject:* Re: [Freeswitch-users] RTP NAT issue
>
>     Suboptimal solution. you'll break transfers among other things.
>
>     Sent from my iPad
>
>     On Jun 6, 2012, at 4:01 PM, "Dave R. Kompel" <drk at drkngs.net
>     <mailto:drk at drkngs.net>> wrote:
>
>     > For a hosted environment, where you're not in control of the
>     users devices/routers you should do the following:
>     >
>     > In the SIP profile, turn on agressive_nat_detection, on the
>     client device have them turn off ALL nat mapping stuff, so the
>     switch can detect it's nat, and if it don't work cause they have
>     some broken router that's doing ALG, and only in that case then
>     set the directory entry for that use to have the
>     "NDLB-Connnectile-Dysfunction" be true.
>     >
>     > That should work every time.
>     >
>     >
>     > --Dave
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com/>
>
>     
>      </>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org <http://www.freeswitch.org/>
>     http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>     http://www.cluecon.com <http://www.cluecon.com/>
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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/>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From avi at avimarcus.net  Thu Jun  7 11:43:46 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 7 Jun 2012 10:43:46 +0300
Subject: [Freeswitch-users] CDR recovery on crash
In-Reply-To: <BLU162-W515DFBEA14FE6EDDD14A188F30@phx.gbl>
References: <BLU162-W4036FF37BC59F586695152880D0@phx.gbl>
	<BLU162-W515DFBEA14FE6EDDD14A188F30@phx.gbl>
Message-ID: <CALNZuwn2ohNNUrt-ORsVWSZfEZpM=H51sbOC-pwzSyi-q-hORg@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Freeswitch_HA#Track_Calls

It will save XML representation of the current calls to your current DB.

-Avi


From steveayre at gmail.com  Thu Jun  7 13:21:03 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 7 Jun 2012 10:21:03 +0100
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
Message-ID: <CAFiqYu=GN4pKLP48OMPkWMjP84KtjtKzsGaxiONi_CDdCwJi-A@mail.gmail.com>

>
> Viewing threaded items, Searching, seeing what?s new and what?s hot, etc?


Strange, my mail client gives me all of that as well as being readable
offline and easier to read on the move.



On 7 June 2012 01:00, Sean Devoy <sdevoy at bizfocused.com> wrote:
> The whole mailing list idea is so "1990's"
>
> Is there interest in having a nice Message Forum instead of the mailing
> list?  Viewing threaded items, Searching, seeing what?s new and what?s
hot,
> etc?
>
>
>
> I have a hosting server with the capacity to host a forum site in ASP.NETor
> PHP and MS Sql Server.  I am willing to purchase the software (if it is
not
> public domain) and maintain it on my server if people like the idea.  I
just
> don't want to throw down a couple hundred $ for nothing.
>
>
>
> Sean
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Thu Jun  7 13:24:59 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 7 Jun 2012 10:24:59 +0100
Subject: [Freeswitch-users] Debugging xml curl
In-Reply-To: <1339049133.55011.YahooMailClassic@web110811.mail.gq1.yahoo.com>
References: <1339049133.55011.YahooMailClassic@web110811.mail.gq1.yahoo.com>
Message-ID: <CAFiqYukRfCS7bPT3WkkkJAqwyG1-Z2YHT7H5-PoZ6dwE3WMw7g@mail.gmail.com>

I don't think you can at the moment. I imagine it would be quite a simple
patch though.

-Steve


On 7 June 2012 07:05, Sherif Omran <sherifomran2000 at yahoo.com> wrote:

> Hello guys,
>
> any body has a good way to debug data sent by xml_curl rather than
>
> xml_curl debug on
>
> I need to see it in the debug console ./fs_cli
>
> Till now it produces xml files in the tmp
>
> any ideas?
>
> regards,
> Sherif Omran
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From cristian.re.work at gmail.com  Thu Jun  7 14:02:07 2012
From: cristian.re.work at gmail.com (cristian re)
Date: Thu, 7 Jun 2012 12:02:07 +0200
Subject: [Freeswitch-users] how to: get session variable after
	api_hangup_hook
In-Reply-To: <CAKzWOxUuNAjtuGwoTeud+_=ZHAtVshmqyZTxKfeNL961N6cm8w@mail.gmail.com>
References: <CABx_eV3-KAYSTaDC09uXpH-wWc1adJhgiAiDxg7yaRCwos73fg@mail.gmail.com>
	<CAKzWOxUuNAjtuGwoTeud+_=ZHAtVshmqyZTxKfeNL961N6cm8w@mail.gmail.com>
Message-ID: <CABx_eV0Xp=K08JeY8PovNKx4Me6h9GH6uajDY-r7fTD8-5ocVg@mail.gmail.com>

Yes of course,
I put in my dialplan:
<action application="set" data="session_in_hangup_hook=true"/>

But I read somewhere that it's possible to manage session after
"api_hangup_hook" only by Lua, Javascript or Perl scripts.
If my API is exposed in a module can I use it?
I try in my dialplan to call api with uuid parameter:

<action application="set" data="api_hangup_hook=my_hangup_api ${uuid}"/>

and in my c++ module I try to get session using uuid without success:

 switch_core_session_t * s = switch_core_session_locate(uuid);
if (s) {
   switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "SESSION\n");
   switch_core_session_rwunlock(s);
}






2012/6/5 Michael Collins <msc at freeswitch.org>

> Are you setting session_in_hangup_hook?
> http://wiki.freeswitch.org/wiki/Channel_Variables#session_in_hangup_hook
> -MC
>
> On Tue, Jun 5, 2012 at 8:33 AM, cristian re <cristian.re.work at gmail.com>wrote:
>
>> Hello,
>>
>> I write a custom cpp module for freeswitch that exposes an API:
>>
>> SWITCH_ADD_API(commands_api_interface, "my_hangup", "my_hangup",
>>  my_hangup_api , MY_HANGUP_USAGE);
>>
>> I want to call this api from dialplan (after bridge hangup) for reading
>> the variable "billmsec".
>> I put this into my dialplan:
>>
>> <action application="set" data="session_in_hangup_hook=true"/>
>> <action application="set" data="api_hangup_hook=my_hangup_api"/>
>>
>> After hangup freeswitch call correctly my API but I have not figured out
>> how to get the variable.
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Thu Jun  7 14:16:56 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 7 Jun 2012 13:16:56 +0300
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <CAFiqYu=GN4pKLP48OMPkWMjP84KtjtKzsGaxiONi_CDdCwJi-A@mail.gmail.com>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
	<CAFiqYu=GN4pKLP48OMPkWMjP84KtjtKzsGaxiONi_CDdCwJi-A@mail.gmail.com>
Message-ID: <CALNZuwn6+LnmiF8T2vPk7WgOBkKV4Pq21uP+rwhwQxcWZJLh=A@mail.gmail.com>

I make heavy use of gmail's mute (m) to "unsubscribe" from topics.

-Avi
(This message was painstakingly thumbed out on my mobile, so apologies for
brevity and errors.)

On Jun 7, 2012 12:23 PM, "Steven Ayre" <steveayre at gmail.com> wrote:

> Viewing threaded items, Searching, seeing what?s new and what?s hot, etc?

Strange, my mail client gives me all of that as well as being readable
offline and easier to read on the move.



On 7 June 2012 01:00, Sean Devoy <sdevoy at bizfocused.com> wrote:
> The whole mailing list idea is s...

> _________________________________________________________________________
> Professional FreeSWITC...

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From avi at avimarcus.net  Thu Jun  7 14:21:42 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 7 Jun 2012 13:21:42 +0300
Subject: [Freeswitch-users] Debugging xml curl
In-Reply-To: <CAFiqYukRfCS7bPT3WkkkJAqwyG1-Z2YHT7H5-PoZ6dwE3WMw7g@mail.gmail.com>
References: <1339049133.55011.YahooMailClassic@web110811.mail.gq1.yahoo.com>
	<CAFiqYukRfCS7bPT3WkkkJAqwyG1-Z2YHT7H5-PoZ6dwE3WMw7g@mail.gmail.com>
Message-ID: <CALNZuwmsCyOSFYXJi6FEHmYUVr2Q_ESHnV96nLjHDRUtWu7JFQ@mail.gmail.com>

I have a form that posts to the xml curl that allows me to set the CID,
destination, context, etc. Its not perfect but helps me debug almost
everything. Also, liberal LOG events generated by my code.

-Avi
(This message was painstakingly thumbed out on my mobile, so apologies for
brevity and errors.)

On Jun 7, 2012 12:26 PM, "Steven Ayre" <steveayre at gmail.com> wrote:

I don't think you can at the moment. I imagine it would be quite a simple
patch though.

-Steve




On 7 June 2012 07:05, Sherif Omran <sherifomran2000 at yahoo.com> wrote:
>
> Hello guys,
>
> any bod...

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From cristian.re.work at gmail.com  Thu Jun  7 14:39:42 2012
From: cristian.re.work at gmail.com (cristian re)
Date: Thu, 7 Jun 2012 12:39:42 +0200
Subject: [Freeswitch-users] how to: get session variable after
	api_hangup_hook
In-Reply-To: <CABx_eV0Xp=K08JeY8PovNKx4Me6h9GH6uajDY-r7fTD8-5ocVg@mail.gmail.com>
References: <CABx_eV3-KAYSTaDC09uXpH-wWc1adJhgiAiDxg7yaRCwos73fg@mail.gmail.com>
	<CAKzWOxUuNAjtuGwoTeud+_=ZHAtVshmqyZTxKfeNL961N6cm8w@mail.gmail.com>
	<CABx_eV0Xp=K08JeY8PovNKx4Me6h9GH6uajDY-r7fTD8-5ocVg@mail.gmail.com>
Message-ID: <CABx_eV3=f=8Chk56ssBYP_HzfGCuqAP_yC_c407sCdGbfZX7og@mail.gmail.com>

I understood I had to use session directly to obtain the channel...

thanks

2012/6/7 cristian re <cristian.re.work at gmail.com>

> Yes of course,
> I put in my dialplan:
> <action application="set" data="session_in_hangup_hook=true"/>
>
> But I read somewhere that it's possible to manage session after
> "api_hangup_hook" only by Lua, Javascript or Perl scripts.
> If my API is exposed in a module can I use it?
> I try in my dialplan to call api with uuid parameter:
>
> <action application="set" data="api_hangup_hook=my_hangup_api ${uuid}"/>
>
> and in my c++ module I try to get session using uuid without success:
>
>  switch_core_session_t * s = switch_core_session_locate(uuid);
> if (s) {
>    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "SESSION\n");
>    switch_core_session_rwunlock(s);
> }
>
>
>
>
>
>
> 2012/6/5 Michael Collins <msc at freeswitch.org>
>
>> Are you setting session_in_hangup_hook?
>> http://wiki.freeswitch.org/wiki/Channel_Variables#session_in_hangup_hook
>> -MC
>>
>> On Tue, Jun 5, 2012 at 8:33 AM, cristian re <cristian.re.work at gmail.com>wrote:
>>
>>>  Hello,
>>>
>>> I write a custom cpp module for freeswitch that exposes an API:
>>>
>>> SWITCH_ADD_API(commands_api_interface, "my_hangup", "my_hangup",
>>>  my_hangup_api , MY_HANGUP_USAGE);
>>>
>>> I want to call this api from dialplan (after bridge hangup) for reading
>>> the variable "billmsec".
>>> I put this into my dialplan:
>>>
>>> <action application="set" data="session_in_hangup_hook=true"/>
>>> <action application="set" data="api_hangup_hook=my_hangup_api"/>
>>>
>>> After hangup freeswitch call correctly my API but I have not figured out
>>> how to get the variable.
>>>
>>>
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From covici at ccs.covici.com  Thu Jun  7 16:32:05 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 07 Jun 2012 08:32:05 -0400
Subject: [Freeswitch-users] freeswitch and new version of sepstral
Message-ID: <12009.1339072325@ccs.covici.com>

Hi.  Does mod_cepstral work with the new version 6 cepstral voices -- I
don't want to install and try it before I have some idea.
Thanks.


-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From gmaruzz at gmail.com  Thu Jun  7 16:37:26 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 7 Jun 2012 14:37:26 +0200
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <CALNZuwn6+LnmiF8T2vPk7WgOBkKV4Pq21uP+rwhwQxcWZJLh=A@mail.gmail.com>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
	<CAFiqYu=GN4pKLP48OMPkWMjP84KtjtKzsGaxiONi_CDdCwJi-A@mail.gmail.com>
	<CALNZuwn6+LnmiF8T2vPk7WgOBkKV4Pq21uP+rwhwQxcWZJLh=A@mail.gmail.com>
Message-ID: <CALXCt0pQ9gOQc4JiBQF_a9HKigKuvvVSsG8DqyD_u_Rw=TR3Eg@mail.gmail.com>

Let's say loud:
"we're '90ish
we're proud"

On Thu, Jun 7, 2012 at 12:16 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I make heavy use of gmail's mute (m) to "unsubscribe" from topics.
>
> -Avi
> (This message was painstakingly thumbed out on my mobile, so apologies for
> brevity and errors.)
>
> On Jun 7, 2012 12:23 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>
> > Viewing threaded items, Searching, seeing what?s new and what?s hot, etc?
>
> Strange, my mail client gives me all of that as well as being readable
> offline and easier to read on the move.
>
>
>
> On 7 June 2012 01:00, Sean Devoy <sdevoy at bizfocused.com> wrote:
> > The whole mailing list idea is s...
>
> > _________________________________________________________________________
> > Professional FreeSWITC...
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From chris at gonumina.com  Thu Jun  7 16:55:29 2012
From: chris at gonumina.com (Chris Ferreira)
Date: Thu, 7 Jun 2012 08:55:29 -0400
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <CALXCt0pQ9gOQc4JiBQF_a9HKigKuvvVSsG8DqyD_u_Rw=TR3Eg@mail.gmail.com>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
	<CAFiqYu=GN4pKLP48OMPkWMjP84KtjtKzsGaxiONi_CDdCwJi-A@mail.gmail.com>
	<CALNZuwn6+LnmiF8T2vPk7WgOBkKV4Pq21uP+rwhwQxcWZJLh=A@mail.gmail.com>
	<CALXCt0pQ9gOQc4JiBQF_a9HKigKuvvVSsG8DqyD_u_Rw=TR3Eg@mail.gmail.com>
Message-ID: <-6651331130295024579@unknownmsgid>

Let's actually move to a BBS hosted on a FreeSWITCH Box.

___________________
Mobile Reply

On Jun 7, 2012, at 8:40 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

Let's say loud:
"we're '90ish
we're proud"

On Thu, Jun 7, 2012 at 12:16 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I make heavy use of gmail's mute (m) to "unsubscribe" from topics.
>
> -Avi
> (This message was painstakingly thumbed out on my mobile, so apologies for
> brevity and errors.)
>
> On Jun 7, 2012 12:23 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>
> > Viewing threaded items, Searching, seeing what?s new and what?s hot, etc?
>
> Strange, my mail client gives me all of that as well as being readable
> offline and easier to read on the move.
>
>
>
> On 7 June 2012 01:00, Sean Devoy <sdevoy at bizfocused.com> wrote:
> > The whole mailing list idea is s...
>
> > _________________________________________________________________________
> > Professional FreeSWITC...
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From vipkilla at gmail.com  Thu Jun  7 16:56:07 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Thu, 7 Jun 2012 08:56:07 -0400
Subject: [Freeswitch-users] XML_CURL and static XML (loaded in memory)
Message-ID: <CAC-LwPNO7JkC4-=VOh6ebbSEOdXtHub9jMUCA4XMHL6k=7fpVA@mail.gmail.com>

Is it possible to load freeswitch with static XML and then load mod_xml_curl?
It seems FreeSWITCH does not work correctly when this is done.
Here is an example:
In freeswitch.xml, I have:
  <section name="directory" description="User Directory">
    <X-PRE-PROCESS cmd="include" data="static/directory.xml"/>
  </section>

This loads my directory configuration statically in memory.
After I start FreeSWITCH, I load mod_xml_curl with 'fs_cli> load mod_xml_curl'
(mod_xml_curl is not loaded when freeswitch starts)

Here is my XML_CURL configuration:

    <configuration name="xml_curl.conf" description="cURL XML Gateway">
        <bindings>
            <binding name="default">
                <param name="gateway-url" value="192.168.0.158"
bindings="directory" />
            </binding>
        </bindings>
    </configuration>

FreeSWITCH does not seem to function properly after XML_CURL is loaded.
If I run the 'user_data' API, FS hangs there for a long time and
eventually returns the password from memory (it does not make an
XML_CURL request)
Here is example:
fs_cli> user_data 1000 at domain.com param password

I guess the ultimate question is this:
Is there any way to setup XML_CURL to failover to static XML (loaded in memory)?

Thanks!


From pkelly at gmail.com  Thu Jun  7 16:56:54 2012
From: pkelly at gmail.com (Pete Kelly)
Date: Thu, 7 Jun 2012 13:56:54 +0100
Subject: [Freeswitch-users] Problems compiling FS with mod_esl
Message-ID: <CAEWH9M8Wgf_eB7vcTofrZG+kKjjXYEP0wbWuf77HFP20ZjC3EQ@mail.gmail.com>

Hi

I am currently having some problems compiling freeswitch in lenny with
mod_esl.

I have done this:

- checked out FS
- ./bootstrap.sh, ./configure
- cd libs/esl && make && make luamod (as per wiki docs)
- uncomment mod_esl from modules.conf

I then run make all, and this error is produced when it gets to mod_esl.so:

Does anyone have any advice?

making all mod_esl
Creating mod_esl.so...
gcc: /usr/local/src/freeswitch_generic/libs/esl/libesl.so: No such file or
directory
gcc: /usr/local/src/freeswitch_generic/libs/esl/libesl.so: No such file or
directory
gcc -I/usr/local/src/freeswitch_generic/libs/esl/src/include
-I/usr/local/src/freeswitch_generic/libs/curl/include
-I/usr/local/src/freeswitch_generic/src/include
-I/usr/local/src/freeswitch_generic/src/include
-I/usr/local/src/freeswitch_generic/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic
-Wdeclaration-after-statement -D_GNU_SOURCE -shared -o .libs/mod_esl.so
-shared -Wl,-x .libs/mod_esl.o
/usr/local/src/freeswitch_generic/libs/esl/libesl.so
/usr/local/src/freeswitch_generic/libs/esl/libesl.so  -lm -lz
/usr/local/src/freeswitch_generic/.libs/libfreeswitch.so
-L/usr/local/src/freeswitch_generic/libs/esl -lesl   -Wl,--rpath
-Wl,/usr/local/freeswitch_generic/lib -Wl,--rpath
-Wl,/usr/local/freeswitch_generic/mod
make[5]: *** [mod_esl.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_esl-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
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From bdfoster at endigotech.com  Thu Jun  7 17:01:01 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 7 Jun 2012 09:01:01 -0400
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <CALXCt0pQ9gOQc4JiBQF_a9HKigKuvvVSsG8DqyD_u_Rw=TR3Eg@mail.gmail.com>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
	<CAFiqYu=GN4pKLP48OMPkWMjP84KtjtKzsGaxiONi_CDdCwJi-A@mail.gmail.com>
	<CALNZuwn6+LnmiF8T2vPk7WgOBkKV4Pq21uP+rwhwQxcWZJLh=A@mail.gmail.com>
	<CALXCt0pQ9gOQc4JiBQF_a9HKigKuvvVSsG8DqyD_u_Rw=TR3Eg@mail.gmail.com>
Message-ID: <CAKV-jFxPeN5uQSBDmn3uH-GqUc5v0HMbbSmbJN_V0g_VXQuhow@mail.gmail.com>

Id really just rather have a mailing list where it shows up in my inbox and
I can look at everything without having an internet connection and queue a
response if needed... makes life easy and since I check my email on a
constant basis, it just works better.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 7, 2012 8:39 AM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:

> Let's say loud:
> "we're '90ish
> we're proud"
>
> On Thu, Jun 7, 2012 at 12:16 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I make heavy use of gmail's mute (m) to "unsubscribe" from topics.
>>
>> -Avi
>> (This message was painstakingly thumbed out on my mobile, so apologies
>> for brevity and errors.)
>>
>> On Jun 7, 2012 12:23 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>>
>> > Viewing threaded items, Searching, seeing what?s new and what?s hot,
>> etc?
>>
>> Strange, my mail client gives me all of that as well as being readable
>> offline and easier to read on the move.
>>
>>
>>
>> On 7 June 2012 01:00, Sean Devoy <sdevoy at bizfocused.com> wrote:
>> > The whole mailing list idea is s...
>>
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITC...
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Thu Jun  7 17:22:52 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 7 Jun 2012 16:22:52 +0300
Subject: [Freeswitch-users] Problems compiling FS with mod_esl
In-Reply-To: <CAEWH9M8Wgf_eB7vcTofrZG+kKjjXYEP0wbWuf77HFP20ZjC3EQ@mail.gmail.com>
References: <CAEWH9M8Wgf_eB7vcTofrZG+kKjjXYEP0wbWuf77HFP20ZjC3EQ@mail.gmail.com>
Message-ID: <CALNZuw=KWgkHcXef9us=bC7EMXhhaRg94G=e-qUsoXgP3tAuAQ@mail.gmail.com>

All compilation and crash issues should be posted on
http://jira.freeswitch.org to ensure they get followed up on.

-Avi

On Thu, Jun 7, 2012 at 3:56 PM, Pete Kelly <pkelly at gmail.com> wrote:
> Hi
>
> I am currently having some problems compiling freeswitch in lenny with
> mod_esl.
>
> I have done this:
>
> - checked out FS
> - ./bootstrap.sh, ./configure
> - cd libs/esl && make && make luamod (as per wiki docs)
> - uncomment mod_esl from modules.conf
>
> I then run make all, and this error is produced when it gets to
mod_esl.so:
>
> Does anyone have any advice?
>
> making all mod_esl
> Creating mod_esl.so...
> gcc: /usr/local/src/freeswitch_generic/libs/esl/libesl.so: No such file or
> directory
> gcc: /usr/local/src/freeswitch_generic/libs/esl/libesl.so: No such file or
> directory
> gcc -I/usr/local/src/freeswitch_generic/libs/esl/src/include
> -I/usr/local/src/freeswitch_generic/libs/curl/include
> -I/usr/local/src/freeswitch_generic/src/include
> -I/usr/local/src/freeswitch_generic/src/include
> -I/usr/local/src/freeswitch_generic/libs/libteletone/src -fPIC -Werror
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
> -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic
-Wdeclaration-after-statement
> -D_GNU_SOURCE -shared -o .libs/mod_esl.so -shared -Wl,-x .libs/mod_esl.o
> /usr/local/src/freeswitch_generic/libs/esl/libesl.so
> /usr/local/src/freeswitch_generic/libs/esl/libesl.so  -lm -lz
> /usr/local/src/freeswitch_generic/.libs/libfreeswitch.so
> -L/usr/local/src/freeswitch_generic/libs/esl -lesl   -Wl,--rpath
> -Wl,/usr/local/freeswitch_generic/lib -Wl,--rpath
> -Wl,/usr/local/freeswitch_generic/mod
> make[5]: *** [mod_esl.so] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_esl-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun  7 17:34:06 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 7 Jun 2012 15:34:06 +0200
Subject: [Freeswitch-users] XML_CURL and static XML (loaded in memory)
In-Reply-To: <CAC-LwPNO7JkC4-=VOh6ebbSEOdXtHub9jMUCA4XMHL6k=7fpVA@mail.gmail.com>
References: <CAC-LwPNO7JkC4-=VOh6ebbSEOdXtHub9jMUCA4XMHL6k=7fpVA@mail.gmail.com>
Message-ID: <CAFGRPVrjsdgF3Jg9Wy78gj7T4VFQkaQ0+9hx0i9b7qO9UdM4JA@mail.gmail.com>

Hello,

I don't think you can use both at the same time...
One option is to write the xml dinamically from the xml-curl server before
reloading it...

cheers

On Thu, Jun 7, 2012 at 2:56 PM, Vik Killa <vipkilla at gmail.com> wrote:

> Is it possible to load freeswitch with static XML and then load
> mod_xml_curl?
> It seems FreeSWITCH does not work correctly when this is done.
> Here is an example:
> In freeswitch.xml, I have:
>  <section name="directory" description="User Directory">
>    <X-PRE-PROCESS cmd="include" data="static/directory.xml"/>
>  </section>
>
> This loads my directory configuration statically in memory.
> After I start FreeSWITCH, I load mod_xml_curl with 'fs_cli> load
> mod_xml_curl'
> (mod_xml_curl is not loaded when freeswitch starts)
>
> Here is my XML_CURL configuration:
>
>    <configuration name="xml_curl.conf" description="cURL XML Gateway">
>        <bindings>
>            <binding name="default">
>                <param name="gateway-url" value="192.168.0.158"
> bindings="directory" />
>            </binding>
>        </bindings>
>    </configuration>
>
> FreeSWITCH does not seem to function properly after XML_CURL is loaded.
> If I run the 'user_data' API, FS hangs there for a long time and
> eventually returns the password from memory (it does not make an
> XML_CURL request)
> Here is example:
> fs_cli> user_data 1000 at domain.com param password
>
> I guess the ultimate question is this:
> Is there any way to setup XML_CURL to failover to static XML (loaded in
> memory)?
>
> Thanks!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From alex at digitalmail.com  Thu Jun  7 17:56:15 2012
From: alex at digitalmail.com (Alex Lake)
Date: Thu, 07 Jun 2012 14:56:15 +0100
Subject: [Freeswitch-users]  att_xfer and loopback
Message-ID: <4FD0B2FF.70405@digitalmail.com>

(Reconnecting to Mailing List)

Hi guys - so I go for a break to celebrate 60 years of the queen and 
look what's here now!

Some really useful bits here. Thanks to Avi for answering in my place. 
The reasons he cites for desiring to use loopback are correct. However, 
because our scripts are auto-generated (from a base spec in our own XML 
format), it is viable (if somwhat undesirable) to spew out a 
superficially long and complex lua script as an alternative.

One of the (seemingly radical) things we're trying to do is to offer 
SIP-grade functionality to mixed SIP & PSTN endpoints. This is best 
illustrated with an example:

User Fred is a mobile kind of guy - although he will generally take 
calls on the SIP handset on his desk, he is often out and about, and so 
will also take calls on his mobile. His company has one of our hosted 
pbx services and he's extension 305. He's set things up so that the 
calls ring first on his SIP handset and, after a delay of 5s, on his 
mobile. Because mobiles have voicemail systems that can get in the way, 
he has the "voicemail defeat" option, whereby on answering the call (and 
hearing the whisper) he has to press 1 to show it's really him accepting 
the call.

A colleague, Bill, on extension 301 is in a very similar situation.

When Bill and Fred are both in the office, things are easy: If Bill gets 
a call that he wishes to handover to Fred (with a briefing), he can do a 
simple attended transfer SIP to SIP.

When they're both out of the office, things are a little more complex:

Firstly, Bill takes the call on his mobile (which doesn't have a 
transfer button) - he uses the * * key sequence to access a mid-call 
menu, selects the "attended transfer" facility and keys in Fred's 
extension number (305). The attended transfer script must do just what a 
normal call to Fred's DDI would do - i.e. Call the SIP handset then, 5s 
later, start trying to find him on his mobile (with the "push 1 to 
accept" feature).

Hopefully that gives an idea of how what we're doing is a bit complicated.

I will definitely try Antony's runes and see how that works.

I'm also up for trying to "improve" the wiki from the point of view of 
FreeSwitch beginners. The main thing I would suggest is much more use of 
examples (and fully contextualised examples, not just free-floating 
parameters!)

Anyway, thanks all (and hope this email gets through to the list)


From jeff at jefflenk.com  Thu Jun  7 18:10:58 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 7 Jun 2012 07:10:58 -0700 (PDT)
Subject: [Freeswitch-users] Problems compiling FS with mod_esl
In-Reply-To: <CALNZuw=KWgkHcXef9us=bC7EMXhhaRg94G=e-qUsoXgP3tAuAQ@mail.gmail.com>
References: <CAEWH9M8Wgf_eB7vcTofrZG+kKjjXYEP0wbWuf77HFP20ZjC3EQ@mail.gmail.com>
	<CALNZuw=KWgkHcXef9us=bC7EMXhhaRg94G=e-qUsoXgP3tAuAQ@mail.gmail.com>
Message-ID: <1339078257954-7579532.post@n2.nabble.com>

Also are you sure you intend to use mod_esl? That is not what fs uses for
normal esl connections, mod_event_socket does that.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problems-compiling-FS-with-mod-esl-tp7579527p7579532.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Hector.Geraldino at ipsoft.com  Thu Jun  7 18:45:31 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Thu, 7 Jun 2012 10:45:31 -0400
Subject: [Freeswitch-users] freeswitch and new version of sepstral
In-Reply-To: <12009.1339072325@ccs.covici.com>
References: <12009.1339072325@ccs.covici.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD022EE80FE4@NY1-EXMB-01.ip-soft.net>

Positive.

You just need to recompile the mod_cepstral module with the new libraries. I've it running in prod with the latest Cepstral version (6.0.1)

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
Sent: Thursday, June 07, 2012 8:32 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] freeswitch and new version of sepstral

Hi.  Does mod_cepstral work with the new version 6 cepstral voices -- I don't want to install and try it before I have some idea.
Thanks.


--
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


From covici at ccs.covici.com  Thu Jun  7 18:49:14 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 07 Jun 2012 10:49:14 -0400
Subject: [Freeswitch-users] freeswitch and new version of sepstral
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD022EE80FE4@NY1-EXMB-01.ip-soft.net>
References: <12009.1339072325@ccs.covici.com>
	<6A6B4C284AD15042B429EB9D904544AD022EE80FE4@NY1-EXMB-01.ip-soft.net>
Message-ID: <1270.1339080554@ccs.covici.com>

OK, thanks.

Hector Geraldino <Hector.Geraldino at ipsoft.com> wrote:

> Positive.
> 
> You just need to recompile the mod_cepstral module with the new libraries. I've it running in prod with the latest Cepstral version (6.0.1)
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> Sent: Thursday, June 07, 2012 8:32 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] freeswitch and new version of sepstral
> 
> Hi.  Does mod_cepstral work with the new version 6 cepstral voices -- I don't want to install and try it before I have some idea.
> Thanks.
> 
> 
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
>  
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From toddb at toddbailey.net  Thu Jun  7 19:06:03 2012
From: toddb at toddbailey.net (toddb at toddbailey.net)
Date: Thu, 07 Jun 2012 08:06:03 -0700
Subject: [Freeswitch-users]
 =?utf-8?q?How_about_a_USER_FORUM_and_kill_off_?=
 =?utf-8?q?the_mail_list=3F?=
Message-ID: <20120607080603.33e327b490679d2282e332758c73b55b.7f27a511f2.wbe@email14.secureserver.net>

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From ifoundthetao at gmail.com  Thu Jun  7 19:16:00 2012
From: ifoundthetao at gmail.com (Timothy Bolton)
Date: Thu, 07 Jun 2012 10:16:00 -0500
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
 list?
In-Reply-To: <20120607080603.33e327b490679d2282e332758c73b55b.7f27a511f2.wbe@email14.secureserver.net>
References: <20120607080603.33e327b490679d2282e332758c73b55b.7f27a511f2.wbe@email14.secureserver.net>
Message-ID: <4FD0C5B0.2070602@gmail.com>

You should really check out the link that I sent from Nabble.  It has 
everything that you're looking for.  It literally archives the list, and 
turns it into a forum.  It might have a 15 minute delay for the 
archiving, but that's not too bad.  Using the "mailing list" is just one 
way to get access to the information.  There's no reason why you can't 
use nabble as the main way of communicating with the users.  Plus, 
having a forum, separate from the mailing list, fractures the 
communication of the community, this way it's more orthogonal (remember: 
DRY).  :)

Maybe having a link to the nabble list on Freeswitch.org would be a good 
idea?  It may be there, but I didn't find it.

'We who cut mere stones must always be envisioning cathedrals.'
Quarry Worker's Creed


On 6/7/2012 10:06 AM, toddb at toddbailey.net wrote:
> I'm all for a forum where one can search for topics of interest 
> instead on having to post a question that has already been answered 
> several times in the past..
>
>     -------- Original Message --------
>     Subject: Re: [Freeswitch-users] How about a USER FORUM and kill
>     off the
>     mail list?
>     From: Giovanni Maruzzelli <gmaruzz at gmail.com
>     <mailto:gmaruzz at gmail.com>>
>     Date: Thu, June 07, 2012 5:37 am
>     To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>
>     Let's say loud:
>     "we're '90ish
>     we're proud"
>
>     On Thu, Jun 7, 2012 at 12:16 PM, Avi Marcus <avi at avimarcus.net
>     <mailto:avi at avimarcus.net>> wrote:
>
>         I make heavy use of gmail's mute (m) to "unsubscribe" from topics.
>         -Avi
>         (This message was painstakingly thumbed out on my mobile, so
>         apologies for brevity and errors.)
>>         On Jun 7, 2012 12:23 PM, "Steven Ayre" <steveayre at gmail.com
>>         <mailto:steveayre at gmail.com>> wrote:
>>
>>         > Viewing threaded items, Searching, seeing what's new and
>>         what's hot, etc?
>>
>>         Strange, my mail client gives me all of that as well as being
>>         readable offline and easier to read on the move.
>>
>>
>>
>>         On 7 June 2012 01:00, Sean Devoy <sdevoy at bizfocused.com
>>         <mailto:sdevoy at bizfocused.com>> wrote:
>>         > The whole mailing list idea is s...
>>         > _________________________________________________________________________
>>         > Professional FreeSWITC...
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>         <http://www.freeswitchsolutions.com/>
>>
>>         
>>          </>
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org <http://www.freeswitch.org/>
>>         http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>>         http://www.cluecon.com <http://www.cluecon.com/>
>>
>>         Join Us At ClueCon - Aug 7-9, 2012
>>
>>         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/>
>>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>         <http://www.freeswitchsolutions.com/>
>
>         
>          </>
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org <http://www.freeswitch.org/>
>         http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>         http://www.cluecon.com <http://www.cluecon.com/>
>
>         Join Us At ClueCon - Aug 7-9, 2012
>
>         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/>
>
>
>
>
>     -- 
>     Sincerely,
>
>     Giovanni Maruzzelli
>     Cell : +39-347-2665618
>
>     ------------------------------------------------------------------------
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From toddb at toddbailey.net  Thu Jun  7 19:16:45 2012
From: toddb at toddbailey.net (toddb at toddbailey.net)
Date: Thu, 07 Jun 2012 08:16:45 -0700
Subject: [Freeswitch-users] Skype on FreeSwitch install issues
Message-ID: <20120607081645.33e327b490679d2282e332758c73b55b.1d489bb130.wbe@email14.secureserver.net>

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From philippe at ppmt.org  Thu Jun  7 19:32:52 2012
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Thu, 7 Jun 2012 11:32:52 -0400
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <4FD0C5B0.2070602@gmail.com>
References: <20120607080603.33e327b490679d2282e332758c73b55b.7f27a511f2.wbe@email14.secureserver.net>
	<4FD0C5B0.2070602@gmail.com>
Message-ID: <CACUErJeiVSjGbJ0cZZQ4dSBJhDX0oX26Dng2aNNOuhBp_fPvig@mail.gmail.com>

I would agree for the link to nabble

I was not aware of it and it really is good when you are more into forum
format than mailing list.

Like that we have the best of both world. A mailing list and a forum.



On 7 June 2012 11:16, Timothy Bolton <ifoundthetao at gmail.com> wrote:

>  You should really check out the link that I sent from Nabble.  It has
> everything that you're looking for.  It literally archives the list, and
> turns it into a forum.  It might have a 15 minute delay for the archiving,
> but that's not too bad.  Using the "mailing list" is just one way to get
> access to the information.  There's no reason why you can't use nabble as
> the main way of communicating with the users.  Plus, having a forum,
> separate from the mailing list, fractures the communication of the
> community, this way it's more orthogonal (remember: DRY).  :)
>
> Maybe having a link to the nabble list on Freeswitch.org would be a good
> idea?  It may be there, but I didn't find it.
>
> 'We who cut mere stones must always be envisioning cathedrals.'
> Quarry Worker's Creed
>
>
> On 6/7/2012 10:06 AM, toddb at toddbailey.net wrote:
>
> I'm all for a forum where one can search for topics of interest instead on
> having to post a question that has already been answered several times in
> the past..
>
>
> -------- Original Message --------
> Subject: Re: [Freeswitch-users] How about a USER FORUM and kill off the
> mail list?
> From: Giovanni Maruzzelli <gmaruzz at gmail.com>
> Date: Thu, June 07, 2012 5:37 am
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
> Let's say loud:
> "we're '90ish
> we're proud"
>
> On Thu, Jun 7, 2012 at 12:16 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I make heavy use of gmail's mute (m) to "unsubscribe" from topics.
>> -Avi
>> (This message was painstakingly thumbed out on my mobile, so apologies
>> for brevity and errors.)
>>
>> On Jun 7, 2012 12:23 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>>
>> > Viewing threaded items, Searching, seeing what?s new and what?s hot,
>> etc?
>>
>>  Strange, my mail client gives me all of that as well as being readable
>> offline and easier to read on the move.
>>
>>
>>
>> On 7 June 2012 01:00, Sean Devoy <sdevoy at bizfocused.com> wrote:
>>  > The whole mailing list idea is s...
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITC...
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From alex at digitalmail.com  Thu Jun  7 19:36:12 2012
From: alex at digitalmail.com (Alex Lake)
Date: Thu, 07 Jun 2012 16:36:12 +0100
Subject: [Freeswitch-users] Problems with multiple-registrations
In-Reply-To: <4FD0B774.50809@digitalmail.com>
References: <4FD0B774.50809@digitalmail.com>
Message-ID: <4FD0CA6C.1000209@digitalmail.com>

OK - beginning to get to the bottom of this...

The sofia_contact command WAS showing multiple registrations, it just 
that I (for some unknown reason) expected to see them on separate lines, 
whereas they're shown on the same line, comma delimited. My bad....

However, there is (or was!) another problem in that one of our FS boxes 
is running a slightly old version (FreeSWITCH Version 1.0.head 
(git-139bd07 2011-11-21 08-27-25 -0600)) in which it would appear that 
multiple-registrations really is/was broken. Might that be possible?

Either way, I am intending to upgrade that box overnight.


On 07/06/2012 15:15, Alex Lake wrote:
> Trying to set up our Freeswitch boxes to support "a few" SIP 
> registrations against individual accounts.
>
> I've modified "internal.xml" to have <param 
> name="multiple-registrations" value="true"/> but it doesn't seem to 
> have had the desired effect.
>
> If I 2 SIP phones registering against the same user, I have never seen 
> more than one simultaneous registration (as evidenced by the 
> "sofia_contact" command). However (and this really spooks me!) if I 
> call the DDI associated with that account, sometimes both phones ring 
> (and accepting the call on either will cause the other to stop ringing).
>
>

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From sdevoy at bizfocused.com  Thu Jun  7 19:38:32 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 7 Jun 2012 11:38:32 -0400
Subject: [Freeswitch-users] How about a USER FORUM and kill off the
	mail	list?
In-Reply-To: <-3243025629320733747@unknownmsgid>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>	<4FD007D0.7010705@puzzled.xs4all.nl>
	<-3243025629320733747@unknownmsgid>
Message-ID: <0eef01cd44c3$98a82400$c9f86c00$@bizfocused.com>

WTF was that supposed to mean?

Nail meet Screw!
Hammer meet Screw Gun.

Better things come along as we progress.  Most of us  are in fact no longer
living in caves even though they provide great protection from tornados.

I was simply asking for a consensus on an idea.  I have an earlier post
asking if there was software or a website that allowed this type of
interface.  No responses to that question.

Thank you Timothy Bolton for the link to nabble, I had never heard of it.
It's not a great interface, but far more usable than dozens of unsearchable
email attachments from the "batched in daily digest".

I see peoples point about email accessibility off line.  I guess I have
lived where we have broadband, smart phone and abundant free wifi for so
long I did not consider being off the net.  That is certainly a valid point.
I must say it is counterintuitive for a "forum" on Internet Telephony to be
concerned about users without Internet Access.

I withdraw my offer an apologize for bringing up the idea.

Back to the caves!  :-)

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Wednesday, June 06, 2012 9:48 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How about a USER FORUM and kill off the mail
list?

Hammer, meet Nail!

Sent from my iPad

On Jun 6, 2012, at 8:47 PM, Patrick Lists
<freeswitch-list at puzzled.xs4all.nl> wrote:

> Since it has been a while I guess it was inevitable that someone would 
> come around with the forum idea again. Have you searched the mailing 
> list archives and read the previous discussion on this subject?
> It did not work for the Asterisk community and I doubt it would work 
> for this community. If you want forum style, use nabble or gmane.
>
> Regards,
> Patrick






From brian at freeswitch.org  Thu Jun  7 19:43:39 2012
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jun 2012 10:43:39 -0500
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <0eef01cd44c3$98a82400$c9f86c00$@bizfocused.com>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
	<4FD007D0.7010705@puzzled.xs4all.nl>
	<-3243025629320733747@unknownmsgid>
	<0eef01cd44c3$98a82400$c9f86c00$@bizfocused.com>
Message-ID: <-4913025585740026458@unknownmsgid>

nabble will be exactly what you're looking for.

Sent from my Cave

On Jun 7, 2012, at 10:39 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> I withdraw my offer an apologize for bringing up the idea.
>
> Back to the caves!  :-)


From bdfoster at endigotech.com  Thu Jun  7 19:52:45 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 7 Jun 2012 11:52:45 -0400
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <-4913025585740026458@unknownmsgid>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
	<4FD007D0.7010705@puzzled.xs4all.nl>
	<-3243025629320733747@unknownmsgid>
	<0eef01cd44c3$98a82400$c9f86c00$@bizfocused.com>
	<-4913025585740026458@unknownmsgid>
Message-ID: <CAKV-jFyxxJD=70noNN8EKR2tzeYcWNSVPr282QTAZ-pHcVNDCg@mail.gmail.com>

I've got all of those things too, but there's still the issue of "I have to
go to a website and waste time to help people." Nabble isn't perfect and
there are a few other names out there that do the same thing. The mailing
list "just works" and even though it isn't perfect it does what's needed.
This mailing list IS searchable from Google, so that's a non issue as well.

(other) Brian, you make me giggle :-)

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 7, 2012 11:44 AM, "Brian West" <brian at freeswitch.org> wrote:

> nabble will be exactly what you're looking for.
>
> Sent from my Cave
>
> On Jun 7, 2012, at 10:39 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>
> > I withdraw my offer an apologize for bringing up the idea.
> >
> > Back to the caves!  :-)
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 gmail.com  Thu Jun  7 19:58:31 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 7 Jun 2012 17:58:31 +0200
Subject: [Freeswitch-users] Skype on FreeSwitch install issues
In-Reply-To: <20120607081645.33e327b490679d2282e332758c73b55b.1d489bb130.wbe@email14.secureserver.net>
References: <20120607081645.33e327b490679d2282e332758c73b55b.1d489bb130.wbe@email14.secureserver.net>
Message-ID: <CALXCt0rXTfJKkqwSVBxCKQaEkEWreKB83Qito7==2YF2vgZv7w@mail.gmail.com>

On Thu, Jun 7, 2012 at 5:16 PM, <toddb at toddbailey.net> wrote:

> I followed the wiki to the letter and all goes well up to the step where I
> execute the ismod cmd.
> BUt again I suspect it's because there are several snd entries listed,
> nothing with 'oss' in the list but there is a /dev/dsp
>

1) if there is a /dev/dsp and is not created by skypopen.ko, then is
probably created by alsa
2) you have to find which kernel module is creating the /dev/dsp and rmmod
it before to insmod skypopen.ko

please do a "lsmod|grep snd" and paste here the results, probably I can
tell you which module to rmmod

-giovanni

-------- Original Message --------
Subject: Re: [Freeswitch-users] Skype on FreeSwitch install issues
From: Giovanni Maruzzelli <gmaruzz at gmail.com>
Date: Wed, June 06, 2012 3:21 pm
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>

On Wed, Jun 6, 2012 at 8:43 PM, <toddb at toddbailey.net> wrote:

>   Even though I had errors in the skypopen build process, I tried the
> install.pl script
> It resulted in more errors.
>

you need to follow strictly the wiki page, step by step:

1) build ALL the thingies (included skypopen.ko oss driver kernel module) -
without errors
2) run the install.pl script
3) run the startup script created by the install.pl script

the /boot directory has nothing to do with it, leave it alone.

your alsa install will probably create the /dev/dsp each reboot. So, you
need to run the startup script that will be created by the install script
after each reboot and before FreeSWITCH.

Please follow strictly step by step, or know exactly what you are doing ;) :

http://wiki.freeswitch.org/wiki/Skypopen#SHORT_BLUEPRINT:_STEPS_NEEDED_TO_USE_SKYPOPEN

http://wiki.freeswitch.org/wiki/Skypopen#Linux

at this last link, read the entire Linux subsection, at least until the
"windows" subsection begins, and you'll find peace and joy!

-giovanni





>
> Big concern is if I am able to sucessfully run the insmod (after disabling
> other snd devices) will doing so permanently disable them?  I'm thinking I
> need to make a backup of the /boot folder.   Any other folder that I need
> to address?
>
>
>
>  -------- Original Message --------
> Subject: Re: [Freeswitch-users] Skype on FreeSwitch install issues
> From: Giovanni Maruzzelli <gmaruzz at gmail.com>
> Date: Wed, June 06, 2012 10:17 am
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
> please use the automatic installer (install.pl) it will do it all for you
>
> if not using automatic installer, do something similar to:
>
> #Unload possible ALSA sound modules that would conflict with our OSS fake
> module
> rmmod snd_pcm_oss
> rmmod snd_mixer_oss
> rmmod snd_seq_oss
> sleep 1
> #Create the inode our fake sound driver will use
> mknod /dev/dsp c 14 3
> #Load our OSS fake module
> insmod /usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko
>
> this will work
>
> Explanation is: your alsa installation creates automatically /dev/dsp for
> compatibility with OSS, but you do not need it, if you are using ALSA. (You
> would need it to use alsa from OSS application, but this is not what you
> need).
>
> -giovanni
>
> On Wed, Jun 6, 2012 at 5:36 PM, <toddb at toddbailey.net> wrote:
>
>>  Hello,
>>
>> New to list & appologies if this is the wrong place to post questions.
>>
>> Assuming this is the place to be, here is a question.
>>
>> Whet I get to the build step for skypopen
>> "make clean; make; insmod ./skypopen.ko; mknod /dev/dsp c 14 3"
>>
>> I get an error with the insmod cmd,  resource / device busy
>>
>>
>> I think it's caused due to having several snd entries listed via the
>> "lsmod |grep snd"
>>
>> The machine is question is a media, database and email server.
>> And the media server component requires the use of ALSA components.
>>
>> Does this requirement prohibit the use of the skype component in
>> Freeswitch or do I just need to prevent the snd modules from loading to
>> complete the kernel mod? Then re enable snd modules at next boot.
>>
>>
>> If this is the case any suggestions?   Maybe running FS in a Vbox to
>> provide the functionality?
>> Or do I need a dedicated box just running linux and FS components?
>>
>> current machine is fedora 14 x64 on a 4 core xeon @ 3ghz w/7 gig memory,
>> w/ 35 % process load average
>>
>> thanks
>>
>>
>> BTW: my desired configuration
>>
>> analog land line to Cisco SPA3102 router connected to the FS server with
>> 4 extensions registered.
>> outbound local calls router to analog line, 1+ area code out bound calls
>> routed to skype.
>> Incoming calls routed to all extensions, unanswered calls set to a ivr to
>> 1. leave voice mail or 2.forward to cell phone (via skype connection)
>>
>> any of this not do-able?
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

------------------------------
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From abaci64 at gmail.com  Thu Jun  7 20:16:04 2012
From: abaci64 at gmail.com (Abaci)
Date: Thu, 07 Jun 2012 12:16:04 -0400
Subject: [Freeswitch-users] Per-channel fsctl loglevel?
In-Reply-To: <CALNZuwkgYayNDfAtn94-e+oJsgmxS0SWABXzsHxHqwPYpYYaXw@mail.gmail.com>
References: <CALNZuwn0oAG_bOXXD=XJJF25PihPh+EQHOMGP=Gf1cLasB=oOw@mail.gmail.com>
	<CAKzWOxU3R0U0xViHuW-G=3BKm+HQGnRy03KBJ8iQ54v2t++v9g@mail.gmail.com>
	<4FC64614.5040205@puzzled.xs4all.nl>
	<CAKzWOxU+T2bqp5+VDKU4seANggzxEK=cAqSvegESxscF584yWw@mail.gmail.com>
	<CALNZuwmwvN2UxsBoVJsa1LiXCcK97k8uNuu3cEJm9oMKHPN6kg@mail.gmail.com>
	<CAKzWOxWy9_E7DSPw2JmwSuPtuipSTYzFX+8rjNU5o0wKh0GgAQ@mail.gmail.com>
	<CALNZuwkLZ6Xagh=qqdWHpA7VQVQFdr4=MoT699JgZshkmydxHg@mail.gmail.com>
	<CAKzWOxWoRzq3OEcwMe7W=yPsJx28Kz-6pMpdsnFS4SSBkaVHoQ@mail.gmail.com>
	<CALNZuwkgYayNDfAtn94-e+oJsgmxS0SWABXzsHxHqwPYpYYaXw@mail.gmail.com>
Message-ID: <4FD0D3C4.5090200@gmail.com>

Just curious what other companies using FreeSWITCH and taking credit 
card over the phone are doing? there is no way you can be PCI compliant 
if you store the logs or CDRs, encrypted or not if it contains the CVV2. 
same goes for call recording. and iirc you can't use a regular voip line 
for credit cards (you have to use encryped linnes if it's voip)

On 5/30/2012 5:25 PM, Avi Marcus wrote:
> Mostly credit cards.. and anything you do with it, e.g. submit it via 
> https to authorize.net <http://authorize.net>, stripe, etc where then 
> you don't need the actual number anymore.
> So both the DTMF entry and the curl debug line. I can't think of 
> anything else in particular.
>
> -Avi
>
> On Wed, May 30, 2012 at 11:35 PM, Michael Collins <msc at freeswitch.org 
> <mailto:msc at freeswitch.org>> wrote:
>
>     Avi,
>
>     Can you think of any other places where the FS logging in general
>     might contain sensitive data? Reason I ask is that maybe we could
>     create something like "pcidss=true" and then use that as a flag to
>     disable logging anything that might be considered sensitive. Just
>     a thought.
>
>     -MC
>
>
>     On Wed, May 30, 2012 at 1:29 PM, Avi Marcus <avi at avimarcus.net
>     <mailto:avi at avimarcus.net>> wrote:
>
>         On Wed, May 30, 2012 at 11:18 PM, Michael Collins
>         <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>
>             How are you protecting everything else? If the XML CDR is
>             sent over HTTP instead of HTTPS then everything about the
>             call is plain text.
>
>         As far as I know, the only thing sensitive in the xml_cdr is
>         digits_dialed.
>
>             And what about the FS logs? Are you encrypting those
>             somehow? It seems to me that you need a more comprehensive
>             solution than just scrubbing a single channel variable.
>
>         No, I'm not encrypting them.. because t here wouldn't be
>         anything sensitive. As far as I can tell, the only issue is
>         the DTMF in DEBUG and the curl post message, again in DEBUG.
>         Since this is a lua IVR it seems nearly nothing else makes it
>         into the log. Only api:execute("curl",...) is in the log
>         because it's not a native direct curl command (like
>         session:playandgetdigits())
>
>             However, if you need an interim solution I would suggest
>             commenting out the line that sets digits_dialed:
>             http://fisheye.freeswitch.org/browse/freeswitch.git/src/switch_channel.c?r=HEAD#to3912
>
>             A more permanent solution might be to create a channel
>             variable that controls whether stuff like this gets
>             logged. Something like "no_dtmf_logging=true" or whatever.
>             That's a bit more involved because you have to decide if
>             there are other places where DTMF info gets logged and if
>             so, decide whether or not you want not to log them.
>
>         That's an interesting idea... it might be more encompassing to
>         have a loglevel=X channel variable instead that affects the
>         logging for that channel. But this is probably overkill...
>
>
>             What would be the ideal solution for your scenario? That
>             answer might yield the best course of action.
>             -MC
>
>
>             On Wed, May 30, 2012 at 11:20 AM, Avi Marcus
>             <avi at avimarcus.net <mailto:avi at avimarcus.net>> wrote:
>
>                 The PCI-DSS (Payment Card Industry Data Security
>                 Standard) requires encryption, not merely permission
>                 restriction, for sensitive data. Hence I'm looking at
>                 the DTMF logging which can probably be easily
>                 re-patterned back into the digits, the curl POST which
>                 also shows everything in the log, the dialed_digits in
>                 a standard xml_cdr..
>                 Otherwise, afaik, lua won't log things unless you
>                 explicitly tell it to.
>
>                 Any suggestions other than setting the entire switch
>                 to fsctl loglevel 6 and not storing the xml_cdrs in
>                 their raw form?
>
>                 -Avi
>
>                 On Wed, May 30, 2012 at 8:11 PM, Michael Collins
>                 <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>
>                     If it's a compliance issue then I'd triple-check
>                     to make sure that no one unauthorized can get to
>                     any of your FS logs or CDR data. I suspect that
>                     logging vs. not logging dialed_digits is not a
>                     make-or-break proposition. If you're doing
>                     xml_cdrs then you've probably got that same data
>                     in other log lines.
>
>                     -MC
>
>
>                     On Wed, May 30, 2012 at 9:08 AM, Patrick Lists
>                     <freeswitch-list at puzzled.xs4all.nl
>                     <mailto:freeswitch-list at puzzled.xs4all.nl>> wrote:
>
>                         On 30-05-12 17:48, Michael Collins wrote:
>                         >     And.. similarly is there a way to blank
>                         out the var digits_dialed in
>                         >     the xml_cdr, from within FS, before the
>                         end of the call?
>                         >
>                         > Why do you need to clear it out? What
>                         information does it collect that
>                         > you don't need?
>
>                         Since it's credit card data I can imagine Avi
>                         does not want it logged
>                         for security purposes.
>
>                         Regards,
>                         Patrick
>
>
>
>                     _________________________________________________________________________
>                     Professional FreeSWITCH Consulting Services:
>                     consulting at freeswitch.org
>                     <mailto:consulting at freeswitch.org>
>                     http://www.freeswitchsolutions.com
>
>                     FreeSWITCH-powered IP PBX: The CudaTel
>                     Communication Server
>                     
>
>                     Official FreeSWITCH Sites
>                     http://www.freeswitch.org
>                     http://wiki.freeswitch.org
>                     http://www.cluecon.com
>
>                     Join Us At ClueCon - Aug 7-9, 2012
>
>                     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
>
>
>
>                 _________________________________________________________________________
>                 Professional FreeSWITCH Consulting Services:
>                 consulting at freeswitch.org
>                 <mailto:consulting at freeswitch.org>
>                 http://www.freeswitchsolutions.com
>
>                 FreeSWITCH-powered IP PBX: The CudaTel Communication
>                 Server
>                 
>
>                 Official FreeSWITCH Sites
>                 http://www.freeswitch.org
>                 http://wiki.freeswitch.org
>                 http://www.cluecon.com
>
>                 Join Us At ClueCon - Aug 7-9, 2012
>
>                 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
>
>
>
>             _________________________________________________________________________
>             Professional FreeSWITCH Consulting Services:
>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>             http://www.freeswitchsolutions.com
>
>             
>             
>
>             Official FreeSWITCH Sites
>             http://www.freeswitch.org
>             http://wiki.freeswitch.org
>             http://www.cluecon.com
>
>             Join Us At ClueCon - Aug 7-9, 2012
>
>             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
>
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>
>         Join Us At ClueCon - Aug 7-9, 2012
>
>         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
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing 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 mgg at giagnocavo.net  Thu Jun  7 20:25:06 2012
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Thu, 7 Jun 2012 16:25:06 +0000
Subject: [Freeswitch-users] RTP NAT issue
In-Reply-To: <4FD05821.30500@gmail.com>
References: <20120607032355.d0d9d846@mail.tritonwest.net>
	<4FD05821.30500@gmail.com>
Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B6BA5BD0@BY2PRD0710MB390.namprd07.prod.outlook.com>

Well FreeSWITCH needs to get an RTP packet to know which port/IP to respond to, for the general case. Do a full UDP capture and open it in Wireshark. The VoIP calls flow view will show you RTP streams, too.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Carlo Dimaggio
Sent: Thursday, June 07, 2012 1:29 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] RTP NAT issue

Hi Dave,

I've tried the "aggressive_nat_detection" on my internal profile. Restarted the profile and I have this registration:

Contact:        "Test 202" <sip:202 at 192.168.2.100:5062;fs_nat=yes;fs_path=sip%3A202%40<FS IP>%3A37710>

but FS send the RTP to the private IP (192.168.2.100) and not to the public NAT IP.

Is there something wrong?


Best Regards



Il 07/06/12 05.23, Dave R. Kompel ha scritto:
Been using this configuration for service providers for years, and always works. Everything always works, transfers etc...

--Dave

________________________________
From: Brian West [mailto:brian at freeswitch.org]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Wed, 06 Jun 2012 18:55:41 -0700
Subject: Re: [Freeswitch-users] RTP NAT issue

Suboptimal solution. you'll break transfers among other things.

Sent from my iPad

On Jun 6, 2012, at 4:01 PM, "Dave R. Kompel" <drk at drkngs.net<mailto:drk at drkngs.net>> wrote:

> For a hosted environment, where you're not in control of the users devices/routers you should do the following:
>
> In the SIP profile, turn on agressive_nat_detection, on the client device have them turn off ALL nat mapping stuff, so the switch can detect it's nat, and if it don't work cause they have some broken router that's doing ALG, and only in that case then set the directory entry for that use to have the "NDLB-Connnectile-Dysfunction" be true.
>
> That should work every time.
>
>
> --Dave

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>


</>

Official FreeSWITCH Sites
http://www.freeswitch.org<http://www.freeswitch.org/>
http://wiki.freeswitch.org<http://wiki.freeswitch.org/>
http://www.cluecon.com<http://www.cluecon.com/>

Join Us At ClueCon - Aug 7-9, 2012

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/>






_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org<mailto:consulting at freeswitch.org>

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com



Join Us At ClueCon - Aug 7-9, 2012



FreeSWITCH-users mailing list

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From bdfoster at endigotech.com  Thu Jun  7 20:59:06 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 7 Jun 2012 12:59:06 -0400
Subject: [Freeswitch-users] Per-channel fsctl loglevel?
In-Reply-To: <4FD0D3C4.5090200@gmail.com>
References: <CALNZuwn0oAG_bOXXD=XJJF25PihPh+EQHOMGP=Gf1cLasB=oOw@mail.gmail.com>
	<CAKzWOxU3R0U0xViHuW-G=3BKm+HQGnRy03KBJ8iQ54v2t++v9g@mail.gmail.com>
	<4FC64614.5040205@puzzled.xs4all.nl>
	<CAKzWOxU+T2bqp5+VDKU4seANggzxEK=cAqSvegESxscF584yWw@mail.gmail.com>
	<CALNZuwmwvN2UxsBoVJsa1LiXCcK97k8uNuu3cEJm9oMKHPN6kg@mail.gmail.com>
	<CAKzWOxWy9_E7DSPw2JmwSuPtuipSTYzFX+8rjNU5o0wKh0GgAQ@mail.gmail.com>
	<CALNZuwkLZ6Xagh=qqdWHpA7VQVQFdr4=MoT699JgZshkmydxHg@mail.gmail.com>
	<CAKzWOxWoRzq3OEcwMe7W=yPsJx28Kz-6pMpdsnFS4SSBkaVHoQ@mail.gmail.com>
	<CALNZuwkgYayNDfAtn94-e+oJsgmxS0SWABXzsHxHqwPYpYYaXw@mail.gmail.com>
	<4FD0D3C4.5090200@gmail.com>
Message-ID: <CAKV-jFy43ZvTF1UHoM793YMP4AYfMSG+KmNcWecf555E1ZzTDw@mail.gmail.com>

More than likely those companies are somewhat oblivious to that. Its hard
to.find a VoIP provider who secures the calls end to end, and they may not
realize they are logging digits for the automated IVRs. I would think they
are smart enough to not record calls with the CVV. I think some companies
probably don't ask for the CVV2, there are some gateways where you can
avoid collecting it but pay higher rates for the risk.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 7, 2012 12:17 PM, "Abaci" <abaci64 at gmail.com> wrote:

>  Just curious what other companies using FreeSWITCH and taking credit card
> over the phone are doing? there is no way you can be PCI compliant if you
> store the logs or CDRs, encrypted or not if it contains the CVV2. same goes
> for call recording. and iirc you can't use a regular voip line for credit
> cards (you have to use encryped linnes if it's voip)
>
> On 5/30/2012 5:25 PM, Avi Marcus wrote:
>
> Mostly credit cards.. and anything you do with it, e.g. submit it via
> https to authorize.net, stripe, etc where then you don't need the actual
> number anymore.
> So both the DTMF entry and the curl debug line. I can't think of anything
> else in particular.
>
> -Avi
>
> On Wed, May 30, 2012 at 11:35 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Avi,
>>
>> Can you think of any other places where the FS logging in general might
>> contain sensitive data? Reason I ask is that maybe we could create
>> something like "pcidss=true" and then use that as a flag to disable logging
>> anything that might be considered sensitive. Just a thought.
>>
>> -MC
>>
>>
>> On Wed, May 30, 2012 at 1:29 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>>  On Wed, May 30, 2012 at 11:18 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> How are you protecting everything else? If the XML CDR is sent over
>>>> HTTP instead of HTTPS then everything about the call is plain text.
>>>
>>>  As far as I know, the only thing sensitive in the xml_cdr is
>>> digits_dialed.
>>>
>>>
>>>> And what about the FS logs? Are you encrypting those somehow? It seems
>>>> to me that you need a more comprehensive solution than just scrubbing a
>>>> single channel variable.
>>>>
>>>  No, I'm not encrypting them.. because t here wouldn't be anything
>>> sensitive. As far as I can tell, the only issue is the DTMF in DEBUG and
>>> the curl post message, again in DEBUG.
>>>  Since this is a lua IVR it seems nearly nothing else makes it into the
>>> log. Only api:execute("curl",...) is in the log because it's not a native
>>> direct curl command (like session:playandgetdigits())
>>>
>>>
>>>> However, if you need an interim solution I would suggest commenting out
>>>> the line that sets digits_dialed:
>>>>
>>>> http://fisheye.freeswitch.org/browse/freeswitch.git/src/switch_channel.c?r=HEAD#to3912
>>>>
>>>> A more permanent solution might be to create a channel variable that
>>>> controls whether stuff like this gets logged. Something like
>>>> "no_dtmf_logging=true" or whatever. That's a bit more involved because you
>>>> have to decide if there are other places where DTMF info gets logged and if
>>>> so, decide whether or not you want not to log them.
>>>>
>>>  That's an interesting idea... it might be more encompassing to have a
>>> loglevel=X channel variable instead that affects the logging for that
>>> channel. But this is probably overkill...
>>>
>>>>
>>>> What would be the ideal solution for your scenario? That answer might
>>>> yield the best course of action.
>>>> -MC
>>>>
>>>>
>>>>  On Wed, May 30, 2012 at 11:20 AM, Avi Marcus <avi at avimarcus.net>wrote:
>>>>
>>>>> The PCI-DSS (Payment Card Industry Data Security Standard) requires
>>>>> encryption, not merely permission restriction, for sensitive data. Hence
>>>>> I'm looking at the DTMF logging which can probably be easily re-patterned
>>>>> back into the digits, the curl POST which also shows everything in the log,
>>>>> the dialed_digits in a standard xml_cdr..
>>>>> Otherwise, afaik, lua won't log things unless you explicitly tell it
>>>>> to.
>>>>>
>>>>>  Any suggestions other than setting the entire switch to fsctl
>>>>> loglevel 6 and not storing the xml_cdrs in their raw form?
>>>>>
>>>>> -Avi
>>>>>
>>>>> On Wed, May 30, 2012 at 8:11 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> If it's a compliance issue then I'd triple-check to make sure that no
>>>>>> one unauthorized can get to any of your FS logs or CDR data. I suspect that
>>>>>> logging vs. not logging dialed_digits is not a make-or-break proposition.
>>>>>> If you're doing xml_cdrs then you've probably got that same data in other
>>>>>> log lines.
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> On Wed, May 30, 2012 at 9:08 AM, Patrick Lists <
>>>>>> freeswitch-list at puzzled.xs4all.nl> wrote:
>>>>>>
>>>>>>> On 30-05-12 17:48, Michael Collins wrote:
>>>>>>> >     And.. similarly is there a way to blank out the var
>>>>>>> digits_dialed in
>>>>>>> >     the xml_cdr, from within FS, before the end of the call?
>>>>>>> >
>>>>>>> > Why do you need to clear it out? What information does it collect
>>>>>>> that
>>>>>>> > you don't need?
>>>>>>>
>>>>>>> Since it's credit card data I can imagine Avi does not want it logged
>>>>>>> for security purposes.
>>>>>>>
>>>>>>> Regards,
>>>>>>> Patrick
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From toddb at toddbailey.net  Thu Jun  7 21:05:01 2012
From: toddb at toddbailey.net (toddbailey)
Date: Thu, 7 Jun 2012 10:05:01 -0700 (PDT)
Subject: [Freeswitch-users] FS server and SPA3102 connection
Message-ID: <1339088701300-7579547.post@n2.nabble.com>

Hi All,

My research leads be to believe I can use the Cisco SPA3102 as a device to
connect the phone line to the fs server (via a lan cable) and I won't need a
internal phone card installed.

Am I correct in this belief or do I need something like a x100P card, a
different phone card or compatible pci modem card to process
incoming/outgoing phone calls to an analog phone service?

thanks

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connection-tp7579547.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From bdfoster at endigotech.com  Thu Jun  7 21:07:04 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 7 Jun 2012 13:07:04 -0400
Subject: [Freeswitch-users] FS server and SPA3102 connection
In-Reply-To: <1339088701300-7579547.post@n2.nabble.com>
References: <1339088701300-7579547.post@n2.nabble.com>
Message-ID: <CAKV-jFxOBRaZ3aJmqqsBUODQrk_wuLad40vVuPf5fqrs9E2aSw@mail.gmail.com>

Spa3102 works just fine. No need for the card.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 7, 2012 1:05 PM, "toddbailey" <toddb at toddbailey.net> wrote:

> Hi All,
>
> My research leads be to believe I can use the Cisco SPA3102 as a device to
> connect the phone line to the fs server (via a lan cable) and I won't need
> a
> internal phone card installed.
>
> Am I correct in this belief or do I need something like a x100P card, a
> different phone card or compatible pci modem card to process
> incoming/outgoing phone calls to an analog phone service?
>
> thanks
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connection-tp7579547.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From itamar at ispbrasil.com.br  Thu Jun  7 21:12:14 2012
From: itamar at ispbrasil.com.br (Itamar Reis Peixoto)
Date: Thu, 7 Jun 2012 14:12:14 -0300
Subject: [Freeswitch-users] FS server and SPA3102 connection
In-Reply-To: <1339088701300-7579547.post@n2.nabble.com>
References: <1339088701300-7579547.post@n2.nabble.com>
Message-ID: <CACyzUwJUY-t3pYTJAM7uZyxuj2xSOgk060VRf++u95WBgz5HUg@mail.gmail.com>

On Thu, Jun 7, 2012 at 2:05 PM, toddbailey <toddb at toddbailey.net> wrote:
> Hi All,
>
> My research leads be to believe I can use the Cisco SPA3102 as a device to
> connect the phone line to the fs server (via a lan cable) and I won't need a
> internal phone card installed.

yes, thats right, works for me.


> Am I correct in this belief or do I need something like a x100P card, a
> different phone card or compatible pci modem card to process
> incoming/outgoing phone calls to an analog phone service?
>
> thanks



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

Itamar Reis Peixoto
msn, google talk: itamar at ispbrasil.com.br
+55 11 4063 5033 (FIXO SP)
+55 34 9158 9329 (TIM)
+55 34 8806 3989 (OI)
+55 34 3221 8599 (FIXO MG)


From covici at ccs.covici.com  Thu Jun  7 21:51:27 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 07 Jun 2012 13:51:27 -0400
Subject: [Freeswitch-users] FS server and SPA3102 connection
In-Reply-To: <CAKV-jFxOBRaZ3aJmqqsBUODQrk_wuLad40vVuPf5fqrs9E2aSw@mail.gmail.com>
References: <1339088701300-7579547.post@n2.nabble.com>
	<CAKV-jFxOBRaZ3aJmqqsBUODQrk_wuLad40vVuPf5fqrs9E2aSw@mail.gmail.com>
Message-ID: <31365.1339091487@ccs.covici.com>

It can be tricky to set up, there is a timing which has to be changed
from 30 milliseconds to 20 and flash is definitely a trip.

Brian Foster <bdfoster at endigotech.com> wrote:

> Spa3102 works just fine. No need for the card.
> 
> Brian Foster
> Endigo Computer LLC
> 
> Sent from a mobile device.
> On Jun 7, 2012 1:05 PM, "toddbailey" <toddb at toddbailey.net> wrote:
> 
> > Hi All,
> >
> > My research leads be to believe I can use the Cisco SPA3102 as a device to
> > connect the phone line to the fs server (via a lan cable) and I won't need
> > a
> > internal phone card installed.
> >
> > Am I correct in this belief or do I need something like a x100P card, a
> > different phone card or compatible pci modem card to process
> > incoming/outgoing phone calls to an analog phone service?
> >
> > thanks
> >
> > --
> > View this message in context:
> > http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connection-tp7579547.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From bdfoster at endigotech.com  Thu Jun  7 21:56:56 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 7 Jun 2012 13:56:56 -0400
Subject: [Freeswitch-users] FS server and SPA3102 connection
In-Reply-To: <31365.1339091487@ccs.covici.com>
References: <1339088701300-7579547.post@n2.nabble.com>
	<CAKV-jFxOBRaZ3aJmqqsBUODQrk_wuLad40vVuPf5fqrs9E2aSw@mail.gmail.com>
	<31365.1339091487@ccs.covici.com>
Message-ID: <CAKV-jFy-gaQ7Ni0VnNA4CEtOBjWcWGH2_-FBkAoxYvgFLWOgbA@mail.gmail.com>

If you have the correct settings in the device it will work just fine. I've
installed my SPA3102 all over the the world and although you have to change
the settings according to the Telecom and or country, it does work.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 7, 2012 1:52 PM, <covici at ccs.covici.com> wrote:

> It can be tricky to set up, there is a timing which has to be changed
> from 30 milliseconds to 20 and flash is definitely a trip.
>
> Brian Foster <bdfoster at endigotech.com> wrote:
>
> > Spa3102 works just fine. No need for the card.
> >
> > Brian Foster
> > Endigo Computer LLC
> >
> > Sent from a mobile device.
> > On Jun 7, 2012 1:05 PM, "toddbailey" <toddb at toddbailey.net> wrote:
> >
> > > Hi All,
> > >
> > > My research leads be to believe I can use the Cisco SPA3102 as a
> device to
> > > connect the phone line to the fs server (via a lan cable) and I won't
> need
> > > a
> > > internal phone card installed.
> > >
> > > Am I correct in this belief or do I need something like a x100P card, a
> > > different phone card or compatible pci modem card to process
> > > incoming/outgoing phone calls to an analog phone service?
> > >
> > > thanks
> > >
> > > --
> > > View this message in context:
> > >
> http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connection-tp7579547.html
> > > Sent from the freeswitch-users mailing list archive at Nabble.com.
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > Join Us At ClueCon - Aug 7-9, 2012
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> > ----------------------------------------------------
> > Alternatives:
> >
> > ----------------------------------------------------
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From syxtyn at hotmail.com  Thu Jun  7 20:14:26 2012
From: syxtyn at hotmail.com (Sixto Palacios)
Date: Thu, 7 Jun 2012 10:14:26 -0600
Subject: [Freeswitch-users] Help me please
In-Reply-To: <CALXCt0qZ34gXQi9diB-2YMm2zChrMLMFd05iAWuU4HPLmFJ1SA@mail.gmail.com>
References: <SNT103-W30DF3BB8366351BFEBD10DDEF30@phx.gbl>
	<SNT103-W407CBFF40DB185E2FE728DDEF30@phx.gbl>,
	<CALXCt0qZ34gXQi9diB-2YMm2zChrMLMFd05iAWuU4HPLmFJ1SA@mail.gmail.com>
Message-ID: <SNT103-W12AAA042684BA5B44FDF68DEF30@phx.gbl>




From: gmaruzz at gmail.com
Date: Thu, 7 Jun 2012 18:00:38 +0200
Subject: Re: FW: Help me please
To: syxtyn at hotmail.com

Please send help requests to the mailing list. You can find the mailing list at: http://lists.freeswitch.org/mailman/listinfo/freeswitch-users




On Thu, Jun 7, 2012 at 5:13 PM, Sixto Palacios <syxtyn at hotmail.com> wrote:








From: syxtyn at hotmail.com
To: gmaruzz at celliax.org


Subject: Help me please
Date: Thu, 7 Jun 2012 09:04:58 -0600





Hello friend, how're u?, need help with this error in freeswitch on Centos 6.2, please

sofia.c:5000 Ping succeeded asterisk with code 200 - count -1/1/1, state UP
2012-06-07 08:55:07.500404 [ERR] mod_skypopen.c:1818          [07bc7ba|07bc7ba] [ERRORA       1818 ][skype101       ][IDLE,IDLE] The Skype client to which we are connected FAILED to gave us CURRENTUSERHANDLE=voipufg, interface_id=1 FAILED to start. No Skype client logged in as 'voipufg' has been found. Please (re)launch a Skype client logged in as 'voipufg'. Skypopen exiting now


Violaci?n de segmento (`core' generado)

 		 	   		   		 	   		  


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

 		 	   		  
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From syxtyn at hotmail.com  Thu Jun  7 20:18:55 2012
From: syxtyn at hotmail.com (Sixto Palacios)
Date: Thu, 7 Jun 2012 10:18:55 -0600
Subject: [Freeswitch-users] Help me please with Skypopen module load error
Message-ID: <SNT103-W28AE7BC95B1E0DFA6B92A9DEF30@phx.gbl>


I have this problem when load the skypopen module in FS, I have my skype client running with the correct user name and password, but i dont know why this error persist, please help me

sofia.c:5000 Ping succeeded asterisk with code 200 - count -1/1/1, state UP
2012-06-07
 08:55:07.500404 [ERR] mod_skypopen.c:1818          [07bc7ba|07bc7ba] 
[ERRORA       1818 ][skype101       ][IDLE,IDLE] The Skype client to 
which we are connected FAILED to gave us CURRENTUSERHANDLE=voipufg, 
interface_id=1 FAILED to start. No Skype client logged in as 'voipufg' 
has been found. Please (re)launch a Skype client logged in as 'voipufg'.
 Skypopen exiting now


Violaci?n de segmento (`core' generado) 		 	   		  
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From belintinc at gmail.com  Thu Jun  7 21:56:07 2012
From: belintinc at gmail.com (BELint Inc)
Date: Thu, 7 Jun 2012 23:26:07 +0530
Subject: [Freeswitch-users] Freeswitch is responding to contact's port
	when client behind NAT
In-Reply-To: <CAEA+2KsCKPCVvMZiWx9qR21-0OpFPjMTH==f-JtrMCvUfm8gjQ@mail.gmail.com>
References: <CAEA+2KsCKPCVvMZiWx9qR21-0OpFPjMTH==f-JtrMCvUfm8gjQ@mail.gmail.com>
Message-ID: <CAEA+2Kt2eOj0VGjbU9bNu1sDjcp4t=cSt_nExuswcWpeQp4KZQ@mail.gmail.com>

Does any one know its solutions,
Guys please help, its urgent. I am even unable to register my phones :-(

Regards,

On Thu, Jun 7, 2012 at 9:20 AM, BELint Inc <belintinc at gmail.com> wrote:

> Hi All,
>
> We`ve been using freeswitch since very long, It was good all over but it
> got stuck to a simple yet complex situation. We have Yealink T28P series
> phones behind NAT and freeswitch is unable to get them register. In the
> scenario freeswitch is at public IP. The reason we think is that freeswitch
> is using sip port in contact header to respond to.
>
> Is there anyway we can avoid this or force it to always send the response
> to destination IP instead of contact or via header ports because clients
> are behind NAT and response can only reach them if it respond to
> destination IP & PORT.
>
> Please revert if someone know how to solve this problem. It is very urgent.
>
> Regards
> BELint
>
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From ifoundthetao at gmail.com  Thu Jun  7 22:25:52 2012
From: ifoundthetao at gmail.com (Timothy Bolton)
Date: Thu, 07 Jun 2012 13:25:52 -0500
Subject: [Freeswitch-users] Freeswitch is responding to contact's port
 when client behind NAT
In-Reply-To: <CAEA+2Kt2eOj0VGjbU9bNu1sDjcp4t=cSt_nExuswcWpeQp4KZQ@mail.gmail.com>
References: <CAEA+2KsCKPCVvMZiWx9qR21-0OpFPjMTH==f-JtrMCvUfm8gjQ@mail.gmail.com>
	<CAEA+2Kt2eOj0VGjbU9bNu1sDjcp4t=cSt_nExuswcWpeQp4KZQ@mail.gmail.com>
Message-ID: <4FD0F230.80905@gmail.com>

If this is urgent, you may want to look into commercial help.

The authors of FreeSWITCH offer commercial support, and there are others 
on the list as well.

'We who cut mere stones must always be envisioning cathedrals.'
Quarry Worker's Creed


On 6/7/2012 12:56 PM, BELint Inc wrote:
> Does any one know its solutions,
> Guys please help, its urgent. I am even unable to register my phones :-(
>
> Regards,
>
> On Thu, Jun 7, 2012 at 9:20 AM, BELint Inc <belintinc at gmail.com 
> <mailto:belintinc at gmail.com>> wrote:
>
>     Hi All,
>
>     We`ve been using freeswitch since very long, It was good all over
>     but it got stuck to a simple yet complex situation. We have
>     Yealink T28P series phones behind NAT and freeswitch is unable to
>     get them register. In the scenario freeswitch is at public IP. The
>     reason we think is that freeswitch is using sip port in contact
>     header to respond to.
>
>     Is there anyway we can avoid this or force it to always send the
>     response to destination IP instead of contact or via header ports
>     because clients are behind NAT and response can only reach them if
>     it respond to destination IP & PORT.
>
>     Please revert if someone know how to solve this problem. It is
>     very urgent.
>
>     Regards
>     BELint
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From toddb at toddbailey.net  Thu Jun  7 22:29:47 2012
From: toddb at toddbailey.net (toddbailey)
Date: Thu, 7 Jun 2012 11:29:47 -0700 (PDT)
Subject: [Freeswitch-users] FS server and SPA3102 connection
In-Reply-To: <CAKV-jFy-gaQ7Ni0VnNA4CEtOBjWcWGH2_-FBkAoxYvgFLWOgbA@mail.gmail.com>
References: <1339088701300-7579547.post@n2.nabble.com>
	<CAKV-jFxOBRaZ3aJmqqsBUODQrk_wuLad40vVuPf5fqrs9E2aSw@mail.gmail.com>
	<31365.1339091487@ccs.covici.com>
	<CAKV-jFy-gaQ7Ni0VnNA4CEtOBjWcWGH2_-FBkAoxYvgFLWOgbA@mail.gmail.com>
Message-ID: <20120607112928.33e327b490679d2282e332758c73b55b.b12fa06b36.wbe@email14.secureserver.net>

OK,&nbsp;So no "modem" card is needed. 
Good to know since I order the router last week and initially it sounds like the best solution, but doubt was raised as&nbsp;I studied the project and the various components. 
&nbsp; 
thanks 
&nbsp; 
&nbsp; 

-------- Original Message -------- Subject: Re: FS server and SPA3102 connection From: "Brian Foster [via freeswitch-users]" &lt; ml-node+s2379917n7579551h20 at n2.nabble.com &gt; Date: Thu, June 07, 2012 11:00 am To: toddbailey &lt; toddb at toddbailey.net &gt; 
If you have the correct settings in the device it will work just fine. I've installed my SPA3102 all over the the world and although you have to change the settings according to the Telecom and or country, it does work. 
Brian Foster Endigo Computer LLC 
Sent from a mobile device. 
On Jun 7, 2012 1:52 PM, &lt; [hidden email] &gt; wrote: 
It can be tricky to set up, there is a timing which has to be changed from 30 milliseconds to 20 and flash is definitely a trip. Brian Foster &lt; [hidden email] &gt; wrote: &gt; Spa3102 works just fine. No need for the card. &gt; &gt; Brian Foster &gt; Endigo Computer LLC &gt; &gt; Sent from a mobile device. &gt; On Jun 7, 2012 1:05 PM, "toddbailey" &lt; [hidden email] &gt; wrote: &gt; &gt; &gt; Hi All, &gt; &gt; &gt; &gt; My research leads be to believe I can use the Cisco SPA3102 as a device to &gt; &gt; connect the phone line to the fs server (via a lan cable) and I won't need &gt; &gt; a &gt; &gt; internal phone card installed. &gt; &gt; &gt; &gt; Am I correct in this belief or do I need something like a x100P card, a &gt; &gt; different phone card or compatible pci modem card to process &gt; &gt; incoming/outgoing phone calls to an analog phone service? &gt; &gt; &gt; &gt; thanks &gt; &gt; &gt; &gt; -- &gt; &gt; View this message in context: &gt; &gt; http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connection-tp7579547.html &gt; &gt; Sent from the freeswitch-users mailing list archive at Nabble.com . &gt; &gt; &gt; &gt; _________________________________________________________________________ &gt; &gt; Professional FreeSWITCH Consulting Services: &gt; &gt; [hidden email] &gt; &gt; http://www.freeswitchsolutions.com &gt; &gt; &gt; &gt;  &gt; &gt;  &gt; &gt; &gt; &gt; Official FreeSWITCH Sites &gt; &gt; http://www.freeswitch.org &gt; &gt; http://wiki.freeswitch.org &gt; &gt; http://www.cluecon.com &gt; &gt; &gt; &gt; Join Us At ClueCon - Aug 7-9, 2012 &gt; &gt; &gt; &gt; FreeSWITCH-users mailing list &gt; &gt; [hidden email] &gt; &gt; http://lists.freeswitch.org/mailman/listinfo/freeswitch-users &gt; &gt; UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users &gt; &gt; http://www.freeswitch.org &gt; &gt; &gt; &gt; ---------------------------------------------------- &gt; Alternatives: &gt; &gt; ---------------------------------------------------- &gt; _________________________________________________________________________ &gt; Professional FreeSWITCH Consulting Services: &gt; [hidden email] &gt; http://www.freeswitchsolutions.com &gt; &gt;  &gt;  &gt; &gt; Official FreeSWITCH Sites &gt; http://www.freeswitch.org &gt; http://wiki.freeswitch.org &gt; http://www.cluecon.com &gt; &gt; Join Us At ClueCon - Aug 7-9, 2012 &gt; &gt; FreeSWITCH-users mailing list &gt; [hidden email] &gt; http://lists.freeswitch.org/mailman/listinfo/freeswitch-users &gt; UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users &gt; http://www.freeswitch.org -- Your life is like a penny. &nbsp;You're going to lose it. &nbsp;The question is: How do you spend it? &nbsp; &nbsp; &nbsp; &nbsp; John Covici &nbsp; &nbsp; &nbsp; &nbsp; [hidden email] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: [hidden email] http://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: [hidden email]  http://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 



If you reply to this email, your message will be added to the discussion below: http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connection-tp7579547p7579551.html  
To unsubscribe from FS server and SPA3102 connection, click here . NAML  


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connection-tp7579547p7579556.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From bdfoster at endigotech.com  Thu Jun  7 22:33:02 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 7 Jun 2012 14:33:02 -0400
Subject: [Freeswitch-users] FS server and SPA3102 connection
In-Reply-To: <20120607112928.33e327b490679d2282e332758c73b55b.b12fa06b36.wbe@email14.secureserver.net>
References: <1339088701300-7579547.post@n2.nabble.com>
	<CAKV-jFxOBRaZ3aJmqqsBUODQrk_wuLad40vVuPf5fqrs9E2aSw@mail.gmail.com>
	<31365.1339091487@ccs.covici.com>
	<CAKV-jFy-gaQ7Ni0VnNA4CEtOBjWcWGH2_-FBkAoxYvgFLWOgbA@mail.gmail.com>
	<20120607112928.33e327b490679d2282e332758c73b55b.b12fa06b36.wbe@email14.secureserver.net>
Message-ID: <CAKV-jFwBuxLPMiwNYLmnOePFcUy9y_P0VDi7PVbpfCKMSV-B+g@mail.gmail.com>

To FS, the FXO is a SIP gateway and the FXS is just another IP ATA.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 7, 2012 2:30 PM, "toddbailey" <toddb at toddbailey.net> wrote:

> OK, So no "modem" card is needed.
> Good to know since I order the router last week and initially it sounds
> like the best solution, but doubt was raised as I studied the project and
> the various components.
>
> thanks
>
>
>
> -------- Original Message --------
> Subject: Re: FS server and SPA3102 connection
> From: "Brian Foster [via freeswitch-users]"
> <[hidden email] <http://user/SendEmail.jtp?type=node&node=7579556&i=0>>
> Date: Thu, June 07, 2012 11:00 am
> To: toddbailey <[hidden email]<http://user/SendEmail.jtp?type=node&node=7579556&i=1>
> >
>
> If you have the correct settings in the device it will work just fine.
> I've installed my SPA3102 all over the the world and although you have to
> change the settings according to the Telecom and or country, it does work.
> Brian Foster
> Endigo Computer LLC
> Sent from a mobile device.
> On Jun 7, 2012 1:52 PM, <[hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=0>>
> wrote:
>
>> It can be tricky to set up, there is a timing which has to be changed
>> from 30 milliseconds to 20 and flash is definitely a trip.
>>
>> Brian Foster <[hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=1>>
>> wrote:
>>
>> > Spa3102 works just fine. No need for the card.
>> >
>> > Brian Foster
>> > Endigo Computer LLC
>> >
>> > Sent from a mobile device.
>> > On Jun 7, 2012 1:05 PM, "toddbailey" <[hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=2>>
>> wrote:
>> >
>> > > Hi All,
>> > >
>> > > My research leads be to believe I can use the Cisco SPA3102 as a
>> device to
>> > > connect the phone line to the fs server (via a lan cable) and I won't
>> need
>> > > a
>> > > internal phone card installed.
>> > >
>> > > Am I correct in this belief or do I need something like a x100P card,
>> a
>> > > different phone card or compatible pci modem card to process
>> > > incoming/outgoing phone calls to an analog phone service?
>> > >
>> > > thanks
>> > >
>> > > --
>> > > View this message in context:
>> > >
>> http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connection-tp7579547.html
>> > > Sent from the freeswitch-users mailing list archive at Nabble.com.
>> > >
>> > >
>> _________________________________________________________________________
>> > > Professional FreeSWITCH Consulting Services:
>> > > [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=3>
>> > > http://www.freeswitchsolutions.com
>> > >
>> > > 
>> > > 
>> > >
>> > > Official FreeSWITCH Sites
>> > > http://www.freeswitch.org
>> > > http://wiki.freeswitch.org
>> > > http://www.cluecon.com
>> > >
>> > > Join Us At ClueCon - Aug 7-9, 2012
>> > >
>> > > FreeSWITCH-users mailing list
>> > > [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=4>
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> >
>> > ----------------------------------------------------
>> > Alternatives:
>> >
>> > ----------------------------------------------------
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=5>
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > Join Us At ClueCon - Aug 7-9, 2012
>> >
>> > FreeSWITCH-users mailing list
>> > [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=6>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> --
>> Your life is like a penny.  You're going to lose it.  The question is:
>> How do
>> you spend it?
>>
>>         John Covici
>>         [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=7>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=8>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=9>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=10>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=11>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
>  If you reply to this email, your message will be added to the discussion
> below:
>
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> View this message in context: RE: FS server and SPA3102 connection<http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connection-tp7579547p7579556.html>
> Sent from the freeswitch-users mailing list archive<http://freeswitch-users.2379917.n2.nabble.com/>at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
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From bdfoster at endigotech.com  Thu Jun  7 22:40:27 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 7 Jun 2012 14:40:27 -0400
Subject: [Freeswitch-users] Freeswitch is responding to contact's port
 when client behind NAT
In-Reply-To: <CAEA+2KvO2KiV8jpDnbh5dqiOZMk4Nke6E3UFLD9rJbmpzpuNhQ@mail.gmail.com>
References: <CAEA+2KvO2KiV8jpDnbh5dqiOZMk4Nke6E3UFLD9rJbmpzpuNhQ@mail.gmail.com>
Message-ID: <CAKV-jFxWBdxKfEs1MCRX=Jf30dFR_Yz311Oei2SqRwkBp3xwzQ@mail.gmail.com>

Please take a look at NDLB-connectile-disfunction parameter, details on the
wiki.

Seems to be a common question this week...

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 7, 2012 3:22 AM, "BELint Inc" <belintinc at gmail.com> wrote:

> Hi All,
>
> We`ve been using freeswitch since very long, It was good all over but it
> got stuck to a simple yet complex situation. We have Yealink T28P series
> phones behind NAT and freeswitch is unable to get them register. In the
> scenario freeswitch is at public IP. The reason we think is that freeswitch
> is using sip port in contact header to respond to.
>
> Is there anyway we can avoid this or force it to always send the response
> to destination IP instead of contact or via header ports because clients
> are behind NAT and response can only reach them if it respond to
> destination IP & PORT.
>
> Please revert if someone know how to solve this problem. It is very urgent.
>
> Regards
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From toddb at toddbailey.net  Thu Jun  7 22:49:45 2012
From: toddb at toddbailey.net (toddbailey)
Date: Thu, 7 Jun 2012 11:49:45 -0700 (PDT)
Subject: [Freeswitch-users] FS server and SPA3102 connection
In-Reply-To: <CAKV-jFwBuxLPMiwNYLmnOePFcUy9y_P0VDi7PVbpfCKMSV-B+g@mail.gmail.com>
References: <1339088701300-7579547.post@n2.nabble.com>
	<CAKV-jFxOBRaZ3aJmqqsBUODQrk_wuLad40vVuPf5fqrs9E2aSw@mail.gmail.com>
	<31365.1339091487@ccs.covici.com>
	<CAKV-jFy-gaQ7Ni0VnNA4CEtOBjWcWGH2_-FBkAoxYvgFLWOgbA@mail.gmail.com>
	<20120607112928.33e327b490679d2282e332758c73b55b.b12fa06b36.wbe@email14.secureserver.net>
	<CAKV-jFwBuxLPMiwNYLmnOePFcUy9y_P0VDi7PVbpfCKMSV-B+g@mail.gmail.com>
Message-ID: <20120607114930.33e327b490679d2282e332758c73b55b.6e7a70e9c0.wbe@email14.secureserver.net>

thanks for the update 
&nbsp; 
Joking stated all these tla's,&nbsp;new terms, features and technology&nbsp;is a real pita. 
&nbsp; 
lets see if I remember FXS is the line coming from the Telco and FXO is the phone,&nbsp; or maybe it's the reverse&nbsp; :) 
&nbsp; 
I'll have to research finding a glossary of terms for all this... 
&nbsp; 
The router specs for example,&nbsp; I read from top to bottom,&nbsp;and understood every word,&nbsp;but wasn't able to make much sense out of it.&nbsp; For each line item,&nbsp; I have to google what it means.&nbsp; 
&nbsp; 
My switch to linux over 8 years ago was a similar technoshock,&nbsp; all in good time...&nbsp;I guess 
&nbsp; 

-------- Original Message -------- Subject: Re: FS server and SPA3102 connection From: "Brian Foster [via freeswitch-users]" &lt; ml-node+s2379917n7579557h6 at n2.nabble.com &gt; Date: Thu, June 07, 2012 11:37 am To: toddbailey &lt; toddb at toddbailey.net &gt; 
To FS, the FXO is a SIP gateway and the FXS is just another IP ATA. 
Brian Foster Endigo Computer LLC 
Sent from a mobile device. 
On Jun 7, 2012 2:30 PM, "toddbailey" &lt; [hidden email] &gt; wrote: 

OK,&nbsp;So no "modem" card is needed. 
Good to know since I order the router last week and initially it sounds like the best solution, but doubt was raised as&nbsp;I studied the project and the various components. 
&nbsp; 
thanks 
&nbsp; 
&nbsp; 

-------- Original Message -------- Subject: Re: FS server and SPA3102 connection From: "Brian Foster [via freeswitch-users]" &lt; [hidden email] &gt; Date: Thu, June 07, 2012 11:00 am To: toddbailey &lt; [hidden email] &gt; 
If you have the correct settings in the device it will work just fine. I've installed my SPA3102 all over the the world and although you have to change the settings according to the Telecom and or country, it does work. 
Brian Foster Endigo Computer LLC 
Sent from a mobile device. 
On Jun 7, 2012 1:52 PM, &lt; [hidden email] &gt; wrote: 
It can be tricky to set up, there is a timing which has to be changed from 30 milliseconds to 20 and flash is definitely a trip. Brian Foster &lt; [hidden email] &gt; wrote: 
&gt; Spa3102 works just fine. No need for the card. &gt; &gt; Brian Foster &gt; Endigo Computer LLC &gt; &gt; Sent from a mobile device. &gt; On Jun 7, 2012 1:05 PM, "toddbailey" &lt; [hidden email] &gt; wrote: &gt; &gt; &gt; Hi All, &gt; &gt; &gt; &gt; My research leads be to believe I can use the Cisco SPA3102 as a device to &gt; &gt; connect the phone line to the fs server (via a lan cable) and I won't need &gt; &gt; a &gt; &gt; internal phone card installed. &gt; &gt; &gt; &gt; Am I correct in this belief or do I need something like a x100P card, a &gt; &gt; different phone card or compatible pci modem card to process &gt; &gt; incoming/outgoing phone calls to an analog phone service? &gt; &gt; &gt; &gt; thanks &gt; &gt; &gt; &gt; -- &gt; &gt; View this message in context: &gt; &gt; http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connection-tp7579547.html &gt; &gt; Sent from the freeswitch-users mailing list archive at Nabble.com . &gt; &gt; &gt; &gt; _________________________________________________________________________ &gt; &gt; Professional FreeSWITCH Consulting Services: &gt; &gt; [hidden email] &gt; &gt; http://www.freeswitchsolutions.com &gt; &gt; &gt; &gt;  &gt; &gt;  &gt; &gt; &gt; &gt; Official FreeSWITCH Sites &gt; &gt; http://www.freeswitch.org &gt; &gt; http://wiki.freeswitch.org &gt; &gt; http://www.cluecon.com &gt; &gt; &gt; &gt; Join Us At ClueCon - Aug 7-9, 2012 &gt; &gt; &gt; &gt; FreeSWITCH-users mailing list &gt; &gt; [hidden email] &gt; &gt; http://lists.freeswitch.org/mailman/listinfo/freeswitch-users &gt; &gt; UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users &gt; &gt; http://www.freeswitch.org &gt; &gt; &gt; &gt; ---------------------------------------------------- &gt; Alternatives: &gt; &gt; ---------------------------------------------------- &gt; _________________________________________________________________________ &gt; Professional FreeSWITCH Consulting Services: &gt; [hidden email] &gt; http://www.freeswitchsolutions.com &gt; &gt;  &gt;  &gt; &gt; Official FreeSWITCH Sites &gt; http://www.freeswitch.org &gt; http://wiki.freeswitch.org &gt; http://www.cluecon.com &gt; &gt; Join Us At ClueCon - Aug 7-9, 2012 &gt; &gt; FreeSWITCH-users mailing list &gt; [hidden email] &gt; http://lists.freeswitch.org/mailman/listinfo/freeswitch-users &gt; UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users &gt; http://www.freeswitch.org -- Your life is like a penny. &nbsp;You're going to lose it. &nbsp;The question is: How do you spend it? &nbsp; &nbsp; &nbsp; &nbsp; John Covici &nbsp; &nbsp; &nbsp; &nbsp; [hidden email] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: [hidden email] http://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: [hidden email]  http://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 



If you reply to this email, your message will be added to the discussion below: http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connection-tp7579547p7579551.html  
To unsubscribe from FS server and SPA3102 connection, click here . NAML  

View this message in context: RE: FS server and SPA3102 connection Sent from the freeswitch-users mailing list archive  at Nabble.com . _________________________________________________________________________ Professional FreeSWITCH Consulting Services: [hidden email] http://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: [hidden email]  http://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 



If you reply to this email, your message will be added to the discussion below: http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connection-tp7579547p7579557.html  
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View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connection-tp7579547p7579559.html
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From bdfoster at endigotech.com  Thu Jun  7 23:10:38 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 7 Jun 2012 15:10:38 -0400
Subject: [Freeswitch-users] FS server and SPA3102 connection
In-Reply-To: <20120607114930.33e327b490679d2282e332758c73b55b.6e7a70e9c0.wbe@email14.secureserver.net>
References: <1339088701300-7579547.post@n2.nabble.com>
	<CAKV-jFxOBRaZ3aJmqqsBUODQrk_wuLad40vVuPf5fqrs9E2aSw@mail.gmail.com>
	<31365.1339091487@ccs.covici.com>
	<CAKV-jFy-gaQ7Ni0VnNA4CEtOBjWcWGH2_-FBkAoxYvgFLWOgbA@mail.gmail.com>
	<20120607112928.33e327b490679d2282e332758c73b55b.b12fa06b36.wbe@email14.secureserver.net>
	<CAKV-jFwBuxLPMiwNYLmnOePFcUy9y_P0VDi7PVbpfCKMSV-B+g@mail.gmail.com>
	<20120607114930.33e327b490679d2282e332758c73b55b.6e7a70e9c0.wbe@email14.secureserver.net>
Message-ID: <CAKV-jFzQNgdycvTNL3OqWp40xoZTjJXGiUYvP+YkRa=Yz378Ng@mail.gmail.com>

Hey we're all learning. Its part of the process.

FXS is the handset side and FXO is the Telco side. don't mix them up :-)

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 7, 2012 2:50 PM, "toddbailey" <toddb at toddbailey.net> wrote:

> thanks for the update
>
> Joking stated all these tla's, new terms, features and technology is a
> real pita.
>
> lets see if I remember FXS is the line coming from the Telco and FXO is
> the phone,  or maybe it's the reverse  :)
>
> I'll have to research finding a glossary of terms for all this...
>
> The router specs for example,  I read from top to bottom, and understood
> every word, but wasn't able to make much sense out of it.  For each line
> item,  I have to google what it means.
>
> My switch to linux over 8 years ago was a similar technoshock,  all in
> good time... I guess
>
>
> -------- Original Message --------
> Subject: Re: FS server and SPA3102 connection
> From: "Brian Foster [via freeswitch-users]"
> <[hidden email] <http://user/SendEmail.jtp?type=node&node=7579559&i=0>>
> Date: Thu, June 07, 2012 11:37 am
> To: toddbailey <[hidden email]<http://user/SendEmail.jtp?type=node&node=7579559&i=1>
> >
>
> To FS, the FXO is a SIP gateway and the FXS is just another IP ATA.
> Brian Foster
> Endigo Computer LLC
> Sent from a mobile device.
> On Jun 7, 2012 2:30 PM, "toddbailey" <[hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579557&i=0>>
> wrote:
>
>> OK, So no "modem" card is needed.
>> Good to know since I order the router last week and initially it sounds
>> like the best solution, but doubt was raised as I studied the project and
>> the various components.
>>
>> thanks
>>
>>
>>
>> -------- Original Message --------
>> Subject: Re: FS server and SPA3102 connection
>> From: "Brian Foster [via freeswitch-users]"
>> <[hidden email] <http://user/SendEmail.jtp?type=node&node=7579556&i=0>>
>> Date: Thu, June 07, 2012 11:00 am
>> To: toddbailey <[hidden email]<http://user/SendEmail.jtp?type=node&node=7579556&i=1>
>> >
>>
>> If you have the correct settings in the device it will work just fine.
>> I've installed my SPA3102 all over the the world and although you have to
>> change the settings according to the Telecom and or country, it does work.
>> Brian Foster
>> Endigo Computer LLC
>> Sent from a mobile device.
>> On Jun 7, 2012 1:52 PM, <[hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=0>>
>> wrote:
>>
>>> It can be tricky to set up, there is a timing which has to be changed
>>> from 30 milliseconds to 20 and flash is definitely a trip.
>>>
>>> Brian Foster <[hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=1>>
>>> wrote:
>>>
>>> > Spa3102 works just fine. No need for the card.
>>> >
>>> > Brian Foster
>>> > Endigo Computer LLC
>>> >
>>> > Sent from a mobile device.
>>> > On Jun 7, 2012 1:05 PM, "toddbailey" <[hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=2>>
>>> wrote:
>>> >
>>> > > Hi All,
>>> > >
>>> > > My research leads be to believe I can use the Cisco SPA3102 as a
>>> device to
>>> > > connect the phone line to the fs server (via a lan cable) and I
>>> won't need
>>> > > a
>>> > > internal phone card installed.
>>> > >
>>> > > Am I correct in this belief or do I need something like a x100P
>>> card, a
>>> > > different phone card or compatible pci modem card to process
>>> > > incoming/outgoing phone calls to an analog phone service?
>>> > >
>>> > > thanks
>>> > >
>>> > > --
>>> > > View this message in context:
>>> > >
>>> http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connection-tp7579547.html
>>> > > Sent from the freeswitch-users mailing list archive at Nabble.com<http://nabble.com/>
>>> .
>>> > >
>>> > >
>>> _________________________________________________________________________
>>> > > Professional FreeSWITCH Consulting Services:
>>> > > [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=3>
>>> > > http://www.freeswitchsolutions.com
>>> > >
>>> > > 
>>> > > 
>>> > >
>>> > > Official FreeSWITCH Sites
>>> > > http://www.freeswitch.org
>>> > > http://wiki.freeswitch.org
>>> > > http://www.cluecon.com
>>> > >
>>> > > Join Us At ClueCon - Aug 7-9, 2012
>>> > >
>>> > > FreeSWITCH-users mailing list
>>> > > [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=4>
>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > > http://www.freeswitch.org
>>> > >
>>> >
>>> > ----------------------------------------------------
>>> > Alternatives:
>>> >
>>> > ----------------------------------------------------
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=5>
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > Join Us At ClueCon - Aug 7-9, 2012
>>> >
>>> > FreeSWITCH-users mailing list
>>> > [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=6>
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> --
>>> Your life is like a penny.  You're going to lose it.  The question is:
>>> How do
>>> you spend it?
>>>
>>>         John Covici
>>>         [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=7>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=8>
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> [hidden email]<http://email14.secureserver.net/user/SendEmail.jtp?type=node&node=7579551&i=9>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From avi at avimarcus.net  Fri Jun  8 01:36:51 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 8 Jun 2012 00:36:51 +0300
Subject: [Freeswitch-users] Per-channel fsctl loglevel?
In-Reply-To: <CAKV-jFy43ZvTF1UHoM793YMP4AYfMSG+KmNcWecf555E1ZzTDw@mail.gmail.com>
References: <CALNZuwn0oAG_bOXXD=XJJF25PihPh+EQHOMGP=Gf1cLasB=oOw@mail.gmail.com>
	<CAKzWOxU3R0U0xViHuW-G=3BKm+HQGnRy03KBJ8iQ54v2t++v9g@mail.gmail.com>
	<4FC64614.5040205@puzzled.xs4all.nl>
	<CAKzWOxU+T2bqp5+VDKU4seANggzxEK=cAqSvegESxscF584yWw@mail.gmail.com>
	<CALNZuwmwvN2UxsBoVJsa1LiXCcK97k8uNuu3cEJm9oMKHPN6kg@mail.gmail.com>
	<CAKzWOxWy9_E7DSPw2JmwSuPtuipSTYzFX+8rjNU5o0wKh0GgAQ@mail.gmail.com>
	<CALNZuwkLZ6Xagh=qqdWHpA7VQVQFdr4=MoT699JgZshkmydxHg@mail.gmail.com>
	<CAKzWOxWoRzq3OEcwMe7W=yPsJx28Kz-6pMpdsnFS4SSBkaVHoQ@mail.gmail.com>
	<CALNZuwkgYayNDfAtn94-e+oJsgmxS0SWABXzsHxHqwPYpYYaXw@mail.gmail.com>
	<4FD0D3C4.5090200@gmail.com>
	<CAKV-jFy43ZvTF1UHoM793YMP4AYfMSG+KmNcWecf555E1ZzTDw@mail.gmail.com>
Message-ID: <CALNZuwm0skph0SceYhvANDzVMpnM-nd59ixtL6QvGuXe_ypY1A@mail.gmail.com>

I had a big thread on Asterisk-biz (after asking here) and basically I
didn't hear anyone say they have a secured VoIP channel. Even if they did,
there's no way you can guarantee it's encrypted end to end. And some made
the point that PSTN is interceptable/hackable too directly at the house so
encryption from the Telco may already be too late.

Are you saying the full credit card number without CVV2 isn't considered
sensitive data at all, and can be saved unencrypted....?
-Avi


On Thu, Jun 7, 2012 at 7:59 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> More than likely those companies are somewhat oblivious to that. Its hard
> to.find a VoIP provider who secures the calls end to end, and they may not
> realize they are logging digits for the automated IVRs. I would think they
> are smart enough to not record calls with the CVV. I think some companies
> probably don't ask for the CVV2, there are some gateways where you can
> avoid collecting it but pay higher rates for the risk.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 7, 2012 12:17 PM, "Abaci" <abaci64 at gmail.com> wrote:
>
>>  Just curious what other companies using FreeSWITCH and taking credit
>> card over the phone are doing? there is no way you can be PCI compliant if
>> you store the logs or CDRs, encrypted or not if it contains the CVV2. same
>> goes for call recording. and iirc you can't use a regular voip line for
>> credit cards (you have to use encryped linnes if it's voip)
>>
>> On 5/30/2012 5:25 PM, Avi Marcus wrote:
>>
>> Mostly credit cards.. and anything you do with it, e.g. submit it via
>> https to authorize.net, stripe, etc where then you don't need the actual
>> number anymore.
>> So both the DTMF entry and the curl debug line. I can't think of anything
>> else in particular.
>>
>> -Avi
>>
>> On Wed, May 30, 2012 at 11:35 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Avi,
>>>
>>> Can you think of any other places where the FS logging in general might
>>> contain sensitive data? Reason I ask is that maybe we could create
>>> something like "pcidss=true" and then use that as a flag to disable logging
>>> anything that might be considered sensitive. Just a thought.
>>>
>>> -MC
>>>
>>>
>>> On Wed, May 30, 2012 at 1:29 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>>  On Wed, May 30, 2012 at 11:18 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> How are you protecting everything else? If the XML CDR is sent over
>>>>> HTTP instead of HTTPS then everything about the call is plain text.
>>>>
>>>>  As far as I know, the only thing sensitive in the xml_cdr is
>>>> digits_dialed.
>>>>
>>>>
>>>>> And what about the FS logs? Are you encrypting those somehow? It seems
>>>>> to me that you need a more comprehensive solution than just scrubbing a
>>>>> single channel variable.
>>>>>
>>>>  No, I'm not encrypting them.. because t here wouldn't be anything
>>>> sensitive. As far as I can tell, the only issue is the DTMF in DEBUG and
>>>> the curl post message, again in DEBUG.
>>>>  Since this is a lua IVR it seems nearly nothing else makes it into the
>>>> log. Only api:execute("curl",...) is in the log because it's not a native
>>>> direct curl command (like session:playandgetdigits())
>>>>
>>>>
>>>>> However, if you need an interim solution I would suggest commenting
>>>>> out the line that sets digits_dialed:
>>>>>
>>>>> http://fisheye.freeswitch.org/browse/freeswitch.git/src/switch_channel.c?r=HEAD#to3912
>>>>>
>>>>> A more permanent solution might be to create a channel variable that
>>>>> controls whether stuff like this gets logged. Something like
>>>>> "no_dtmf_logging=true" or whatever. That's a bit more involved because you
>>>>> have to decide if there are other places where DTMF info gets logged and if
>>>>> so, decide whether or not you want not to log them.
>>>>>
>>>>  That's an interesting idea... it might be more encompassing to have a
>>>> loglevel=X channel variable instead that affects the logging for that
>>>> channel. But this is probably overkill...
>>>>
>>>>>
>>>>> What would be the ideal solution for your scenario? That answer might
>>>>> yield the best course of action.
>>>>> -MC
>>>>>
>>>>>
>>>>>  On Wed, May 30, 2012 at 11:20 AM, Avi Marcus <avi at avimarcus.net>wrote:
>>>>>
>>>>>> The PCI-DSS (Payment Card Industry Data Security Standard) requires
>>>>>> encryption, not merely permission restriction, for sensitive data. Hence
>>>>>> I'm looking at the DTMF logging which can probably be easily re-patterned
>>>>>> back into the digits, the curl POST which also shows everything in the log,
>>>>>> the dialed_digits in a standard xml_cdr..
>>>>>> Otherwise, afaik, lua won't log things unless you explicitly tell it
>>>>>> to.
>>>>>>
>>>>>>  Any suggestions other than setting the entire switch to fsctl
>>>>>> loglevel 6 and not storing the xml_cdrs in their raw form?
>>>>>>
>>>>>> -Avi
>>>>>>
>>>>>> On Wed, May 30, 2012 at 8:11 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>
>>>>>>> If it's a compliance issue then I'd triple-check to make sure that
>>>>>>> no one unauthorized can get to any of your FS logs or CDR data. I suspect
>>>>>>> that logging vs. not logging dialed_digits is not a make-or-break
>>>>>>> proposition. If you're doing xml_cdrs then you've probably got that same
>>>>>>> data in other log lines.
>>>>>>>
>>>>>>> -MC
>>>>>>>
>>>>>>>
>>>>>>> On Wed, May 30, 2012 at 9:08 AM, Patrick Lists <
>>>>>>> freeswitch-list at puzzled.xs4all.nl> wrote:
>>>>>>>
>>>>>>>> On 30-05-12 17:48, Michael Collins wrote:
>>>>>>>> >     And.. similarly is there a way to blank out the var
>>>>>>>> digits_dialed in
>>>>>>>> >     the xml_cdr, from within FS, before the end of the call?
>>>>>>>> >
>>>>>>>> > Why do you need to clear it out? What information does it collect
>>>>>>>> that
>>>>>>>> > you don't need?
>>>>>>>>
>>>>>>>> Since it's credit card data I can imagine Avi does not want it
>>>>>>>> logged
>>>>>>>> for security purposes.
>>>>>>>>
>>>>>>>> Regards,
>>>>>>>> Patrick
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
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>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
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>>>>>>>
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>
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>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
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>>>>>> Join Us At ClueCon - Aug 7-9, 2012
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>>>>>
>>>>>
>>>>> _________________________________________________________________________
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>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
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>>>>
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>>>> consulting at freeswitch.org
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>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
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>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
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>>> Join Us At ClueCon - Aug 7-9, 2012
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>>
>> _________________________________________________________________________
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>>
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>>
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>>
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>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
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>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
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>>
>> Join Us At ClueCon - Aug 7-9, 2012
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>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
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>
> Join Us At ClueCon - Aug 7-9, 2012
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From mike.kallies at gmail.com  Fri Jun  8 01:47:48 2012
From: mike.kallies at gmail.com (Mike Kallies)
Date: Thu, 07 Jun 2012 17:47:48 -0400
Subject: [Freeswitch-users] Audiocodes MP-118 VoIP Gateway can't un-hold
Message-ID: <4FD12184.4000402@gmail.com>

Hey Everyone,

I'm using an Audiocodes MP-118 VoIP Gateway.  When making a call through
the device to a Freeswitch gateway on the other side, it sends the call
to my desk phone just fine.

When I mute the call, we get the hold music as per the dial plan,
everything is good.

Then when I unmute... the hold music stops, and... nothing.  The call
doesn't drop, it just sits there.

In the Audiocodes log, we see:


5d:1h:25m:32s (   lgr_psbrdex)(28879     )  recv <--
acEV_BROKEN_CONNECTION, Ch:0
5d:1h:25m:32s (      lgr_flow)(28880     )  #0:RTP_BROKEN_CONNECTION_EV
5d:1h:25m:32s (      lgr_flow)(28881     )  |
#0:RTP_BROKEN_CONNECTION_EV


I know this isn't entirely a Freeswitch question, but I figure the crowd
here might know the answer or where I should look for one.

Calls through our Grandstream device work fine, so I think this can be
narrowed down to the Audiocodes device.  But maybe there's a different
codec or option or something I could try to see if I can make it happy?

Thanks,

-Mike Kallies


From Hector.Geraldino at ipsoft.com  Fri Jun  8 02:02:22 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Thu, 7 Jun 2012 18:02:22 -0400
Subject: [Freeswitch-users] Per-channel fsctl loglevel?
In-Reply-To: <CALNZuwm0skph0SceYhvANDzVMpnM-nd59ixtL6QvGuXe_ypY1A@mail.gmail.com>
References: <CALNZuwn0oAG_bOXXD=XJJF25PihPh+EQHOMGP=Gf1cLasB=oOw@mail.gmail.com>
	<CAKzWOxU3R0U0xViHuW-G=3BKm+HQGnRy03KBJ8iQ54v2t++v9g@mail.gmail.com>
	<4FC64614.5040205@puzzled.xs4all.nl>
	<CAKzWOxU+T2bqp5+VDKU4seANggzxEK=cAqSvegESxscF584yWw@mail.gmail.com>
	<CALNZuwmwvN2UxsBoVJsa1LiXCcK97k8uNuu3cEJm9oMKHPN6kg@mail.gmail.com>
	<CAKzWOxWy9_E7DSPw2JmwSuPtuipSTYzFX+8rjNU5o0wKh0GgAQ@mail.gmail.com>
	<CALNZuwkLZ6Xagh=qqdWHpA7VQVQFdr4=MoT699JgZshkmydxHg@mail.gmail.com>
	<CAKzWOxWoRzq3OEcwMe7W=yPsJx28Kz-6pMpdsnFS4SSBkaVHoQ@mail.gmail.com>
	<CALNZuwkgYayNDfAtn94-e+oJsgmxS0SWABXzsHxHqwPYpYYaXw@mail.gmail.com>
	<4FD0D3C4.5090200@gmail.com>
	<CAKV-jFy43ZvTF1UHoM793YMP4AYfMSG+KmNcWecf555E1ZzTDw@mail.gmail.com>
	<CALNZuwm0skph0SceYhvANDzVMpnM-nd59ixtL6QvGuXe_ypY1A@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD022EE81043@NY1-EXMB-01.ip-soft.net>

According to the PCI DSS normative, you can store the PAN if it is encrypted. You must have a separate process in place governing the encryption (key storage, key rotation, key access, etc.)

However you can't, under any circumstances, store the CVV/CVV2 digits. Never.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus
Sent: Thursday, June 07, 2012 5:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Per-channel fsctl loglevel?

I had a big thread on Asterisk-biz (after asking here) and basically I didn't hear anyone say they have a secured VoIP channel. Even if they did, there's no way you can guarantee it's encrypted end to end. And some made the point that PSTN is interceptable/hackable too directly at the house so encryption from the Telco may already be too late.

Are you saying the full credit card number without CVV2 isn't considered sensitive data at all, and can be saved unencrypted....?
-Avi

On Thu, Jun 7, 2012 at 7:59 PM, Brian Foster <bdfoster at endigotech.com<mailto:bdfoster at endigotech.com>> wrote:

More than likely those companies are somewhat oblivious to that. Its hard to.find a VoIP provider who secures the calls end to end, and they may not realize they are logging digits for the automated IVRs. I would think they are smart enough to not record calls with the CVV. I think some companies probably don't ask for the CVV2, there are some gateways where you can avoid collecting it but pay higher rates for the risk.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 7, 2012 12:17 PM, "Abaci" <abaci64 at gmail.com<mailto:abaci64 at gmail.com>> wrote:
Just curious what other companies using FreeSWITCH and taking credit card over the phone are doing? there is no way you can be PCI compliant if you store the logs or CDRs, encrypted or not if it contains the CVV2. same goes for call recording. and iirc you can't use a regular voip line for credit cards (you have to use encryped linnes if it's voip)

On 5/30/2012 5:25 PM, Avi Marcus wrote:
Mostly credit cards.. and anything you do with it, e.g. submit it via https to authorize.net<http://authorize.net>, stripe, etc where then you don't need the actual number anymore.
So both the DTMF entry and the curl debug line. I can't think of anything else in particular.

-Avi

On Wed, May 30, 2012 at 11:35 PM, Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>> wrote:
Avi,

Can you think of any other places where the FS logging in general might contain sensitive data? Reason I ask is that maybe we could create something like "pcidss=true" and then use that as a flag to disable logging anything that might be considered sensitive. Just a thought.

-MC

On Wed, May 30, 2012 at 1:29 PM, Avi Marcus <avi at avimarcus.net<mailto:avi at avimarcus.net>> wrote:
On Wed, May 30, 2012 at 11:18 PM, Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>> wrote:
How are you protecting everything else? If the XML CDR is sent over HTTP instead of HTTPS then everything about the call is plain text.
As far as I know, the only thing sensitive in the xml_cdr is digits_dialed.

And what about the FS logs? Are you encrypting those somehow? It seems to me that you need a more comprehensive solution than just scrubbing a single channel variable.
No, I'm not encrypting them.. because t here wouldn't be anything sensitive. As far as I can tell, the only issue is the DTMF in DEBUG and the curl post message, again in DEBUG.
Since this is a lua IVR it seems nearly nothing else makes it into the log. Only api:execute("curl",...) is in the log because it's not a native direct curl command (like session:playandgetdigits())

However, if you need an interim solution I would suggest commenting out the line that sets digits_dialed:
http://fisheye.freeswitch.org/browse/freeswitch.git/src/switch_channel.c?r=HEAD#to3912

A more permanent solution might be to create a channel variable that controls whether stuff like this gets logged. Something like "no_dtmf_logging=true" or whatever. That's a bit more involved because you have to decide if there are other places where DTMF info gets logged and if so, decide whether or not you want not to log them.
That's an interesting idea... it might be more encompassing to have a loglevel=X channel variable instead that affects the logging for that channel. But this is probably overkill...

What would be the ideal solution for your scenario? That answer might yield the best course of action.
-MC

On Wed, May 30, 2012 at 11:20 AM, Avi Marcus <avi at avimarcus.net<mailto:avi at avimarcus.net>> wrote:
The PCI-DSS (Payment Card Industry Data Security Standard) requires encryption, not merely permission restriction, for sensitive data. Hence I'm looking at the DTMF logging which can probably be easily re-patterned back into the digits, the curl POST which also shows everything in the log, the dialed_digits in a standard xml_cdr..
Otherwise, afaik, lua won't log things unless you explicitly tell it to.

Any suggestions other than setting the entire switch to fsctl loglevel 6 and not storing the xml_cdrs in their raw form?

-Avi

On Wed, May 30, 2012 at 8:11 PM, Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>> wrote:
If it's a compliance issue then I'd triple-check to make sure that no one unauthorized can get to any of your FS logs or CDR data. I suspect that logging vs. not logging dialed_digits is not a make-or-break proposition. If you're doing xml_cdrs then you've probably got that same data in other log lines.

-MC

On Wed, May 30, 2012 at 9:08 AM, Patrick Lists <freeswitch-list at puzzled.xs4all.nl<mailto:freeswitch-list at puzzled.xs4all.nl>> wrote:
On 30-05-12 17:48, Michael Collins wrote:
>     And.. similarly is there a way to blank out the var digits_dialed in
>     the xml_cdr, from within FS, before the end of the call?
>
> Why do you need to clear it out? What information does it collect that
> you don't need?

Since it's credit card data I can imagine Avi does not want it logged
for security purposes.

Regards,
Patrick


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From gmaruzz at gmail.com  Fri Jun  8 02:33:23 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 8 Jun 2012 00:33:23 +0200
Subject: [Freeswitch-users] Help me please with Skypopen module load
	error
In-Reply-To: <SNT103-W28AE7BC95B1E0DFA6B92A9DEF30@phx.gbl>
References: <SNT103-W28AE7BC95B1E0DFA6B92A9DEF30@phx.gbl>
Message-ID: <CALXCt0q9MPbStcGtSdT8LGb9AO2jFt59wgpbeQ5L2pqYYTEk-A@mail.gmail.com>

On Thu, Jun 7, 2012 at 6:18 PM, Sixto Palacios <syxtyn at hotmail.com> wrote:

>  I have this problem when load the skypopen module in FS, I have my skype
> client running with the correct user name and password, but i dont know
> why this error persist, please help me
>
> sofia.c:5000 Ping succeeded asterisk with code 200 - count -1/1/1, state UP
> 2012-06-07 08:55:07.500404 [ERR] mod_skypopen.c:1818
> [07bc7ba|07bc7ba] [ERRORA       1818 ][skype101       ][IDLE,IDLE] The
> Skype client to which we are connected FAILED to gave us
> CURRENTUSERHANDLE=voipufg, interface_id=1 FAILED to start. No Skype client
> logged in as 'voipufg' has been found. Please (re)launch a Skype client
> logged in as 'voipufg'. Skypopen exiting now
> Violaci?n de segmento (`core' generado)
>
>
have you used install.pl ?

there are no skype client logged in the network with the right
credientials...

please use install.pl, will build a script to launch skype clients in the
right way


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From vvenkatar at gmail.com  Fri Jun  8 03:36:40 2012
From: vvenkatar at gmail.com (Venkatesh)
Date: Thu, 7 Jun 2012 16:36:40 -0700
Subject: [Freeswitch-users] Using CURL inside a JavaScript.
In-Reply-To: <CAP_69ms2t-BrwgkjmGjwh4U9+=O706zg9vOPC-6cFdsJvS=kdQ@mail.gmail.com>
References: <CAP_69muchUh9-PWbO5GLGv_DBS=M5x1uTJh0T15UOo+rksqmow@mail.gmail.com>
	<4FD01298.90402@gmail.com>
	<CAP_69ms2t-BrwgkjmGjwh4U9+=O706zg9vOPC-6cFdsJvS=kdQ@mail.gmail.com>
Message-ID: <CAP_69msPgGUUCAFxN3G3wyc1ndN4FDt7JrTRCH2RjnNL28nKrw@mail.gmail.com>

Sorry for the repost, but wanted to see if anybody had any ideas.... Would
really appreciate any help.

Venkatesh

On Wed, Jun 6, 2012 at 8:36 PM, Venkatesh <vvenkatar at gmail.com> wrote:

> Hi !
>
> I have uploaded the relevant JS code to pastebin:
> http://pastebin.freeswitch.org/19238
>
> Would really appreciate any help!
>
> Venkatesh
>
>
> On Wed, Jun 6, 2012 at 7:31 PM, Timothy Bolton <ifoundthetao at gmail.com>wrote:
>
>>  Do you have any code that you want to post (
>> http://pastebin.freeswitch.org)?
>>
>> Also, you may want to look into timers/intervals (setTimer setInterval).
>> I use those to keep tabs on responses from requests.  Otherwise, the
>> JavaScript will continue to execute without getting the response, which can
>> be very frustrating.
>>
>> 'We who cut mere stones must always be envisioning cathedrals.'
>> Quarry Worker's Creed
>>
>>
>> On 6/6/2012 7:16 PM, Venkatesh wrote:
>>
>> Hi !
>>
>>  I am developing a simple IVR application using JS. One of the things I
>> want to do when my JS fires is to issue a WS API to fetch some data
>> relevant to the call; wait for the response and use information in the
>> response to process the incoming call. To meet that end, I wrote up a
>> simple JS that issues a CURL request providing appropriate call back
>> function. However, I dont see any callback being fired back to my
>> application. The JS simply proceeds to execute the next instruction. I know
>> my WS returns a string (verified by running the exact same command via a
>> command line). I wanted to know if there is anything I need to do to make
>> the application "wait for the call back".  Would appreciate any help.
>>
>>  Regards,
>> Venkatesh
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
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>> http://www.freeswitch.org
>>
>>
>
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From th982a at googlemail.com  Fri Jun  8 03:49:22 2012
From: th982a at googlemail.com (Tamer Higazi)
Date: Fri, 08 Jun 2012 01:49:22 +0200
Subject: [Freeswitch-users] freeswitch and mISDN
Message-ID: <4FD13E02.4040202@googlemail.com>

Hi people!
I figured out how to activate freeswitch with mISDN (through freetdm).

How do I configure mISDN in freetdm, and what are the channel drivers,
with it's dial-plan respectively.


Tamer


From vvenkatar at gmail.com  Fri Jun  8 04:57:32 2012
From: vvenkatar at gmail.com (Venkatesh)
Date: Thu, 7 Jun 2012 17:57:32 -0700
Subject: [Freeswitch-users] Unable to use mod_java.
Message-ID: <CAP_69mtZuWq-22j-hiFBBuYzNKzqd46fR0MPjXgJYYd1kfjxnw@mail.gmail.com>

Hi !

I have been trying to enable mod_java with FS but am having problems. I was
able to compile mod_java fine. However FS is having trouble loading the
module. It errors out with the following error message:

2012-06-07 20:51:24.058126 [ERR] modjava.c:326 Error creating Java VM!,

I believe I have the CLASSPATH/PATH variables set up correctly in my
configuration files. In fact I put in some debug statements in modjava.c.
Below is the debug output showing the various variables:

2012-06-07 20:51:24.058126 [NOTICE] modjava.c:349 Java Framework Loading...
2012-06-07 20:51:24.058126 [ERR] modjava.c:299 VENKY optionCount: 7
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option:
-Djava.class.path=/usr/local/freeswitch/scripts
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option:
-Djava.library.path=/usr/java/jdk1.6.0_32/jre/lib:/usr/java/jdk1.6.0_32/lib
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -verbose:jni
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Xms=64
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Xmx=256
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option:
-agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=127.0.0.1:8000
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option:
-Djava.library.path=/usr/local/freeswitch/mod
2012-06-07 20:51:24.058126 [ERR] modjava.c:326 Error creating Java VM!,
error: -1
2012-06-07 20:51:24.058126 [INFO] switch_time.c:1123 Timezone reloaded 530
definitions
2012-06-07 20:51:24.058126 [CRIT] switch_loadable_module.c:1300 Error
Loading module /usr/local/freeswitch/mod/mod_java.so
**Module load routine returned an error**


Would really appreciate any ideas from others that may have tried enabling
the same?

Venkatesh
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From jaugenstine at gmail.com  Fri Jun  8 05:37:59 2012
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Thu, 7 Jun 2012 18:37:59 -0700
Subject: [Freeswitch-users] Unable to use mod_java.
In-Reply-To: <CAP_69mtZuWq-22j-hiFBBuYzNKzqd46fR0MPjXgJYYd1kfjxnw@mail.gmail.com>
References: <CAP_69mtZuWq-22j-hiFBBuYzNKzqd46fR0MPjXgJYYd1kfjxnw@mail.gmail.com>
Message-ID: <CAAq2U1By5VrsM=w0+Nbb=sxhh376-Gh0Gpwgs99FW-6CF5P=hQ@mail.gmail.com>

Venkatesh,

I can tell you from personal experience that mod_java does have issues.
Unless you really need Java I would recommend that you take a look at Lua.
If you really need or want to use Java, then I would recommend
investigating the event socket and XML-RPC interface.

Jonathan

On Thu, Jun 7, 2012 at 5:57 PM, Venkatesh <vvenkatar at gmail.com> wrote:

> Hi !
>
> I have been trying to enable mod_java with FS but am having problems. I
> was able to compile mod_java fine. However FS is having trouble loading the
> module. It errors out with the following error message:
>
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:326 Error creating Java VM!,
>
> I believe I have the CLASSPATH/PATH variables set up correctly in my
> configuration files. In fact I put in some debug statements in modjava.c.
> Below is the debug output showing the various variables:
>
> 2012-06-07 20:51:24.058126 [NOTICE] modjava.c:349 Java Framework Loading...
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:299 VENKY optionCount: 7
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option:
> -Djava.class.path=/usr/local/freeswitch/scripts
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option:
> -Djava.library.path=/usr/java/jdk1.6.0_32/jre/lib:/usr/java/jdk1.6.0_32/lib
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -verbose:jni
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Xms=64
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Xmx=256
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option:
> -agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=
> 127.0.0.1:8000
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option:
> -Djava.library.path=/usr/local/freeswitch/mod
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:326 Error creating Java VM!,
> error: -1
> 2012-06-07 20:51:24.058126 [INFO] switch_time.c:1123 Timezone reloaded 530
> definitions
> 2012-06-07 20:51:24.058126 [CRIT] switch_loadable_module.c:1300 Error
> Loading module /usr/local/freeswitch/mod/mod_java.so
> **Module load routine returned an error**
>
>
> Would really appreciate any ideas from others that may have tried enabling
> the same?
>
> Venkatesh
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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 yehavi.bourvine at gmail.com  Fri Jun  8 07:39:18 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 8 Jun 2012 06:39:18 +0300
Subject: [Freeswitch-users] Audiocodes MP-118 VoIP Gateway can't un-hold
In-Reply-To: <4FD12184.4000402@gmail.com>
References: <4FD12184.4000402@gmail.com>
Message-ID: <CAAtWKeFcB1=XiiL7D0PvHEX1WM+PDdwkOESzTQwVn8ZquapO3w@mail.gmail.com>

Search for the parameter DisconnectOnBrokenConnection and set it to zero. I
don't remember on which screen it is.

                        __Yehavi:

2012/6/8 Mike Kallies <mike.kallies at gmail.com>

> Hey Everyone,
>
> I'm using an Audiocodes MP-118 VoIP Gateway.  When making a call through
> the device to a Freeswitch gateway on the other side, it sends the call
> to my desk phone just fine.
>
> When I mute the call, we get the hold music as per the dial plan,
> everything is good.
>
> Then when I unmute... the hold music stops, and... nothing.  The call
> doesn't drop, it just sits there.
>
> In the Audiocodes log, we see:
>
>
> 5d:1h:25m:32s (   lgr_psbrdex)(28879     )  recv <--
> acEV_BROKEN_CONNECTION, Ch:0
> 5d:1h:25m:32s (      lgr_flow)(28880     )  #0:RTP_BROKEN_CONNECTION_EV
> 5d:1h:25m:32s (      lgr_flow)(28881     )  |
> #0:RTP_BROKEN_CONNECTION_EV
>
>
> I know this isn't entirely a Freeswitch question, but I figure the crowd
> here might know the answer or where I should look for one.
>
> Calls through our Grandstream device work fine, so I think this can be
> narrowed down to the Audiocodes device.  But maybe there's a different
> codec or option or something I could try to see if I can make it happy?
>
> Thanks,
>
> -Mike Kallies
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ahe.sanath at gmail.com  Fri Jun  8 08:17:59 2012
From: ahe.sanath at gmail.com (Sanath Prasanna)
Date: Fri, 8 Jun 2012 09:47:59 +0530
Subject: [Freeswitch-users] opensource project voicechat for freeswitch
In-Reply-To: <Pine.LNX.4.64.1206061742390.4294@tomback.antagus.de>
References: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
	<2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com>
	<1338558318.79806.YahooMailNeo@web120405.mail.ne1.yahoo.com>
	<Pine.LNX.4.64.1206041700001.22234@tomback.antagus.de>
	<CALQZz0zLpRBnFu3jce07Ycq=mn_OnAGrVFZgm-ibY_3eGg93Xw@mail.gmail.com>
	<Pine.LNX.4.64.1206061742390.4294@tomback.antagus.de>
Message-ID: <CALQZz0xy8da2jbtJOmphAKGSkY-qzxi90tBEx=Tu6q8p=QMbQg@mail.gmail.com>

Hi Thomas,
I wrote complete voice mail system using LUA. (store messages, recording
messages, retrieve messages, play messages with out bound dialing, send
message as email attachment to end user, etc)  if u can send complete
scope, I can change it & send.
Br,
Sanath

On Wed, Jun 6, 2012 at 9:22 PM, Thomas Hoellriegel <admin at blindi.net> wrote:

> Hi Sanath
>
> Thanks for your helpfull message
>
>
> I suggest: we do everything step by step.
>
> First: the voice mail creation and login.
>
> for example:
> press 1 for login.
> Press 2 to create your onw mailbox.
>  It is important that the mailbox already exists.
> If the Mailbox exists, play a error wavfile for example:
> Sorry this mailbox is allready exists, and go back to the creation option
> again.
>
> Thanks.
>
>
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/**listinfo/blinde-misc<http://www.blindi.net/mailman/listinfo/blinde-misc>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From drk at drkngs.net  Fri Jun  8 08:35:56 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Thu, 07 Jun 2012 21:35:56 -0700
Subject: [Freeswitch-users]
	=?iso-8859-1?q?Freeswitch_is_responding_to_con?=
	=?iso-8859-1?q?tact=27s_port=09when_client_behind_NAT?=
In-Reply-To: <CAEA+2Kt2eOj0VGjbU9bNu1sDjcp4t=cSt_nExuswcWpeQp4KZQ@mail.gmail.com>
Message-ID: <20120608043556.9b5b6644@mail.tritonwest.net>

Sorry for the long delay, but here's your answer:  
   
It's going to go to the address in the un-altered VIA header, unless RPORT gets sent it the VIA. If you have agressive_nat_detection turned on, and the client puts RPORT in the via, it will work fine. If you have a broken device that won't do it, then set the variable "NDLB-connectile-dysfunction=true" in the users directory enter (XML) or other, what ever you are using. That will cause the contact to be re-written to the transport address from where it came (both ip address and port) and the response will also be sent there.  
   
--Dave
      _____  

  From: BELint Inc [mailto:belintinc at gmail.com]
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, 07 Jun 2012 10:56:07 -0700
Subject: Re: [Freeswitch-users] Freeswitch is responding to contact's port when client behind NAT

Does any one know its solutions,  
Guys please help, its urgent. I am even unable to register my phones :-(  

  
Regards,

  
On Thu, Jun 7, 2012 at 9:20 AM, BELint Inc <belintinc at gmail.com> wrote:
  Hi All,

  
We`ve been using freeswitch since very long, It was good all over but it got stuck to a simple yet complex situation. We have Yealink T28P series phones behind NAT and freeswitch is unable to get them register. In the scenario freeswitch is at public IP. The reason we think is that freeswitch is using sip port in contact header to respond to.

Is there anyway we can avoid this or force it to always send the response to destination IP instead of contact or via header ports because clients are behind NAT and response can only reach them if it respond to destination IP & PORT.

Please revert if someone know how to solve this problem. It is very urgent.

Regards
BELint

      
   
 
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From ahe.sanath at gmail.com  Fri Jun  8 08:48:06 2012
From: ahe.sanath at gmail.com (Sanath Prasanna)
Date: Fri, 8 Jun 2012 10:18:06 +0530
Subject: [Freeswitch-users] "originate" Command not found in fs_cli
In-Reply-To: <CAKzWOxXiuP-s8ewW4gH5kEuuNYWE0tp7vbt7tVS+m_5FOMP4zg@mail.gmail.com>
References: <CALQZz0xQA0mP31=kbszjYB55it5gMU1mnUj5bNWuTR4FmSFuVQ@mail.gmail.com>
	<CAKV-jFztkkESpAXZbrB-htebmzNnf2JAormnSChPrn0-_kHiAQ@mail.gmail.com>
	<CALQZz0xacCGTQ7sijfCakPYWc44AP+UDpYNrip53VK6sOEefjA@mail.gmail.com>
	<CAKzWOxXiuP-s8ewW4gH5kEuuNYWE0tp7vbt7tVS+m_5FOMP4zg@mail.gmail.com>
Message-ID: <CALQZz0wu5GrGhfRXK=zrUmZN_Bk2TPNCAF1avckb_bY-aF8-=Q@mail.gmail.com>

Dear MC,
Sorry for inconvenience causes. That is happened due to comment of
mod_cmmands line in the modules.conf.xml. Now out bound dialing is also
work. Tx for great help.
Br,
Sanath


On Wed, Jun 6, 2012 at 10:38 AM, Michael Collins <msc at freeswitch.org> wrote:

> This is a bigger problem than originate. It looks like none of your
> commands are working. I would shut down FS and restart it in the
> foreground, i.e. don't use the "-nc" flag. Watch the console and see what
> errors pop up. I'm curious to know if there's an issue with mod_commands or
> something like that.
>
> -MC
>
>
> On Tue, Jun 5, 2012 at 9:31 PM, Sanath Prasanna <ahe.sanath at gmail.com>wrote:
>
>> Hi
>> Tx for advice. Here is the my profile
>> vi sip_profiles/external/ivr.xml
>> <include>
>>   <gateway name="ivr">
>>     <!-- Replace these value with your iCall Carrier Services username
>> and password. -->
>>     <!-- Even if you use an IP-based sub-account, FreeSWITCH needs these
>> values -->
>>     <param name="password" value="1234" />
>>     <param name="proxy" value="$${operator_sip_ip}" />
>>     <param name="realm" value="$${operator_realm}" />
>>     <!-- Set to "false" for IP-based accounts, or "true" for registration
>> based -->
>>    <!-- Here I need  IP-based account.Because operator is not provided
>> registration based sip account.-->
>>     <param name="register" value="false" />
>>     <!-- <param name="ping" value="30" /> -->
>>   </gateway>
>> </include>
>>
>>  vi dialplan/outbound.xml
>>
>> <include>
>>   <context name="outbound">
>>    <extension name="outbound">
>>       <condition field="destination_number" expression="^(54\d{9})$">
>>         <action application="bridge" data="sofia/gateway/ivr/$1"/>
>>       </condition>
>>     </extension>
>>   </context>
>> </include>
>> ~
>>
>> still same problem come
>> ===================
>> freeswitch at internal> originate sofia/gateway/ivr/14722796816 54722796817
>> -ERR originate sofia/gateway/dtl/ 14722796816   54722796817 Command not
>> found!
>>
>> also when run reloadxml
>> freeswitch at internal> reloadxml
>> -ERR reloadxml Command not found!
>> So when change conf, need to restart service every time.
>>
>> Pls advice to resolve this problems.
>> Br,
>> Sanath
>>
>>
>> On Tue, Jun 5, 2012 at 10:46 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>>
>>> Your not understanding the dialstring aspect...
>>>
>>> Example:
>>>
>>> originate sofia/gateway/flowroute/13175551212 user/1000 at 192.168.1.79
>>>
>>> Where sofia is the channel type, gateway is a keyword, flowroute is what
>>> I named my gateway, 13175551212 is the number I'm calling through my
>>> gateway flowroute, and user/1000 is the extension to send it to and
>>> 192.168.1.79 is the domain that user is attached to.
>>>
>>> Brian Foster
>>> Endigo Computer LLC
>>>
>>> Sent from a mobile device.
>>> On Jun 5, 2012 8:49 AM, "Sanath Prasanna" <ahe.sanath at gmail.com> wrote:
>>>
>>>> Hi,
>>>> when I run originate command in fs_cli following error coming. Pls
>>>> advice to resolve that.
>>>>
>>>> freeswitch at internal> originate sofia/external/ivr/9179123456 9179123458
>>>> -ERR originate  sofia/external/ivr/9179123456 9179123458 Command not
>>>> found!
>>>>
>>>> Br,
>>>> Sanath
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From abid_freeswitch at live.com  Fri Jun  8 11:32:32 2012
From: abid_freeswitch at live.com (Abid Saleem)
Date: Fri, 8 Jun 2012 12:32:32 +0500
Subject: [Freeswitch-users] Multilingual IVR in English and Thai
Message-ID: <BAY162-W57F9F3F7949FEC7FF9AAD1F4F20@phx.gbl>


Hi,
Hope to find you well!
Does Freeswitch support Multilingual IVR for example English and Thai at the same time. The customer should be given the option to chose either English or Thai. If yes, can you please specify how can it be configured?
Thanks in advance

 		 	   		  
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From andrew at cassidywebservices.co.uk  Fri Jun  8 11:35:23 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Fri, 8 Jun 2012 08:35:23 +0100
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <CAKV-jFyxxJD=70noNN8EKR2tzeYcWNSVPr282QTAZ-pHcVNDCg@mail.gmail.com>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
	<4FD007D0.7010705@puzzled.xs4all.nl>
	<-3243025629320733747@unknownmsgid>
	<0eef01cd44c3$98a82400$c9f86c00$@bizfocused.com>
	<-4913025585740026458@unknownmsgid>
	<CAKV-jFyxxJD=70noNN8EKR2tzeYcWNSVPr282QTAZ-pHcVNDCg@mail.gmail.com>
Message-ID: <CAMbW9oJAOhA3OrcG85AHer1gWwFLYGtAa0us+W_vS4zy9Ld91A@mail.gmail.com>

Gmail's search is good enough for me :)

On 7 June 2012 16:52, Brian Foster <bdfoster at endigotech.com> wrote:

> I've got all of those things too, but there's still the issue of "I have
> to go to a website and waste time to help people." Nabble isn't perfect and
> there are a few other names out there that do the same thing. The mailing
> list "just works" and even though it isn't perfect it does what's needed.
> This mailing list IS searchable from Google, so that's a non issue as well.
>
> (other) Brian, you make me giggle :-)
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 7, 2012 11:44 AM, "Brian West" <brian at freeswitch.org> wrote:
>
>> nabble will be exactly what you're looking for.
>>
>> Sent from my Cave
>>
>> On Jun 7, 2012, at 10:39 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>>
>> > I withdraw my offer an apologize for bringing up the idea.
>> >
>> > Back to the caves!  :-)
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From john.gathm at gmail.com  Fri Jun  8 12:31:10 2012
From: john.gathm at gmail.com (John Gathm)
Date: Fri, 8 Jun 2012 10:31:10 +0200
Subject: [Freeswitch-users] using APNS with diaplan : throttle a call until
	a certain condition changes
Message-ID: <CACU5aF6ma-W+VM98FzhS6i=kWHQhJPehpvPt1oS9E8662CaC2A@mail.gmail.com>

Hi

I am planing to use freeswitch with SIP clients on iPhone and I would like
to use APNS to notify clients of incoming calls (so that clients can launch)
When I receive an incoming call, I should look up in a DB the
extension/phone device id, send a notification to it, waits until client
logs on or a timeout is reached, then proceed to bridge the call to the
extension.

I could put this logic in a python script in the diaplan, that would block
the execution of the dialplan.

<action application="python" data="send_notification_and_wait.py"/>
<action application="bridge"
data="user/${user_data(${destination_number}@${domain_name} attr
id)}@${domain_name}" />

I am wondering if this implementation would be acceptable regarding
freeswitch design, or if diaplan/freeswitch offer cleaner ways to implement
it.

Regards.

John
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From vivekm at packtpub.com  Fri Jun  8 12:34:30 2012
From: vivekm at packtpub.com (Vivek Menon)
Date: Fri, 08 Jun 2012 14:04:30 +0530
Subject: [Freeswitch-users] FreeSWITCH Cookbook (Non-Technical Question)
Message-ID: <4FD1B916.3020103@packtpub.com>

Hi,

As the subject line says, this is not a technical question, but would 
anyone be interested in organizing a Giveaway on their/any referential 
website for our newly published title on FreeSWITCH? The book can be a 
useful resource for anyone who wish to enhance their understanding of 
FreeSWITCH while answering some of the common problems arising with the 
platform.

Written by members of the FreeSWITCH development team, this is a 
problem-solution approach to take your FreeSWITCH skills to the next 
level, where everything is explained in a practical way.

If you would be interested, you can read more about the book here: 
http://www.packtpub.com/freeswitch-telephony-advanced-cookbook/book. For 
those wishing to delve deeper,you may even download a Sample Chapter 
<http://www.packtpub.com/sites/default/files/5405OS-Chapter-3-Processing-Call-Detail-Records.pdf?utm_source=packtpub&utm_medium=free&utm_campaign=pdf> 
for free.

If the Giveaway manages to go live, you would be eligible to get a free 
print/e-copy of the book depending upon your current location of residency.

Looking forward to your comments.

Have a wonderful day ahead.

Kind Regards,

Vivek
--


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From vivekm at packtpub.com  Fri Jun  8 12:53:16 2012
From: vivekm at packtpub.com (Vivek Menon)
Date: Fri, 08 Jun 2012 14:23:16 +0530
Subject: [Freeswitch-users] FreeSWITCH Cookbook (Non-Technical Question)
In-Reply-To: <4FD1B916.3020103@packtpub.com>
References: <4FD1B916.3020103@packtpub.com>
Message-ID: <4FD1BD7C.8010901@packtpub.com>

Hi,

Apologies for not including my contact details earlier. If you wish to 
directly get in touch with me, kindly drop in a mail to vivekm at packtpub.com.

Looking forward to your response.

Kind Regards,

Vivek
--

On 08-06-2012 14:04, Vivek Menon wrote:
> Hi,
>
> As the subject line says, this is not a technical question, but would 
> anyone be interested in organizing a Giveaway on their/any referential 
> website for our newly published title on FreeSWITCH? The book can be a 
> useful resource for anyone who wish to enhance their understanding of 
> FreeSWITCH while answering some of the common problems arising with 
> the platform.
>
> Written by members of the FreeSWITCH development team, this is a 
> problem-solution approach to take your FreeSWITCH skills to the next 
> level, where everything is explained in a practical way.
>
> If you would be interested, you can read more about the book here: 
> http://www.packtpub.com/freeswitch-telephony-advanced-cookbook/book. 
> For those wishing to delve deeper,you may even download a Sample 
> Chapter 
> <http://www.packtpub.com/sites/default/files/5405OS-Chapter-3-Processing-Call-Detail-Records.pdf?utm_source=packtpub&utm_medium=free&utm_campaign=pdf> 
> for free.
>
> If the Giveaway manages to go live, you would be eligible to get a 
> free print/e-copy of the book depending upon your current location of 
> residency.
>
> Looking forward to your comments.
>
> Have a wonderful day ahead.
>
> Kind Regards,
>
> Vivek
> --
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From nathandownes at hotmail.com  Fri Jun  8 13:25:58 2012
From: nathandownes at hotmail.com (Mr Nathan Downes)
Date: Fri, 8 Jun 2012 19:25:58 +1000
Subject: [Freeswitch-users] Attenuation?
Message-ID: <BLU134-DS1CF5EB0468A94442DBC07D7F20@phx.gbl>

Hi!

 

Yet again another issue with fibre company, media level provided by them is
too high and causes clipping in the captures, which causes a static/crackle
sound to be heard on the call, I need confirmation that no
volume/gain/attenuation of any kind is done when FS handles the media. No
transcoding or anything is done on FS.  

 

Thanks again,

 

Nathan

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From saami_mh at ymail.com  Fri Jun  8 15:43:42 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Fri, 8 Jun 2012 04:43:42 -0700 (PDT)
Subject: [Freeswitch-users] how to Block inbound(internal) sofia profile
	calls ?(don't allow for internal calls)
Message-ID: <1339155822.9200.YahooMailNeo@web120104.mail.ne1.yahoo.com>

hi guys,
i have defined two extensions named:1000 & 1001
so i am going to allow the above extension make outbound calls and disable inbount calls
in other hand i dont want the ext:1000 able to make call to ext:1001 ,
and want to disable calling internal registerations sip?
how can i do that?
thanks so much
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From babak.freeswitch at gmail.com  Fri Jun  8 16:53:48 2012
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Fri, 8 Jun 2012 17:23:48 +0430
Subject: [Freeswitch-users] how to Block inbound(internal) sofia profile
 calls ?(don't allow for internal calls)
In-Reply-To: <1339155822.9200.YahooMailNeo@web120104.mail.ne1.yahoo.com>
References: <1339155822.9200.YahooMailNeo@web120104.mail.ne1.yahoo.com>
Message-ID: <CAEVfcab6Zv8cci7NB7hqtw24u4Z83AHeV=p5H6t-+Zq5-0he8Q@mail.gmail.com>

if you are using xml dialplan, to block calls between and to 1000 and 1001
you can just remove any extension that routes calls between and to 1000 and
1001 in your dialplan (such as "Local_Extension" in sample configuration in
default.xml)  and to have outbound calls just add an extension that has a
condition matching your destination.
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML

On Fri, Jun 8, 2012 at 4:13 PM, Samira Mh <saami_mh at ymail.com> wrote:

> hi guys,
> i have defined two extensions named:1000 & 1001
> so i am going to allow the above extension make outbound calls and disable
> inbount calls
> in other hand i dont want the ext:1000 able to make call to ext:1001 ,
> and want to disable calling internal registerations sip
> how can i do that?
> thanks so much
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From saami_mh at ymail.com  Fri Jun  8 17:19:35 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Fri, 8 Jun 2012 06:19:35 -0700 (PDT)
Subject: [Freeswitch-users] how to Block inbound(internal) sofia profile
	calls ?(don't allow for internal calls)
In-Reply-To: <CAEVfcab6Zv8cci7NB7hqtw24u4Z83AHeV=p5H6t-+Zq5-0he8Q@mail.gmail.com>
References: <1339155822.9200.YahooMailNeo@web120104.mail.ne1.yahoo.com>
	<CAEVfcab6Zv8cci7NB7hqtw24u4Z83AHeV=p5H6t-+Zq5-0he8Q@mail.gmail.com>
Message-ID: <1339161575.6898.YahooMailNeo@web120106.mail.ne1.yahoo.com>



hi,
thanks for your reply;
but default.xml is consist ad follow:

<?xml version="1.0" encoding="utf-8"?>
<include>
<context name="default">
? ? ? ? <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
</context>
</include>
but how can i limited internal calls?

________________________________
 From: babak yakhchali <babak.freeswitch at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Friday, June 8, 2012 5:23 PM
Subject: Re: [Freeswitch-users] how to Block inbound(internal) sofia profile calls ?(don't allow for internal calls)
 

if you are using xml dialplan, to block calls between and to 1000 and 1001 you can just remove any extension that routes calls between and to 1000 and 1001 in your dialplan (such as "Local_Extension" in sample configuration in default.xml) ?and to have outbound calls just add an extension that has a condition matching your destination.?
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML?


On Fri, Jun 8, 2012 at 4:13 PM, Samira Mh <saami_mh at ymail.com> wrote:

hi guys,
>i have defined two extensions named:1000 & 1001
>so i am going to allow the above extension make outbound calls and disable inbount calls
>in other hand i dont want the ext:1000 able to make call to ext:1001 ,
>and want to disable calling internal registerations sip?
>how can i do that?
>thanks so much
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From saami_mh at ymail.com  Fri Jun  8 17:41:22 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Fri, 8 Jun 2012 06:41:22 -0700 (PDT)
Subject: [Freeswitch-users] (solved) how to Block inbound(internal)
	sofia profile calls ?(don't allow for internal calls)
In-Reply-To: <1339161575.6898.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1339155822.9200.YahooMailNeo@web120104.mail.ne1.yahoo.com>
	<CAEVfcab6Zv8cci7NB7hqtw24u4Z83AHeV=p5H6t-+Zq5-0he8Q@mail.gmail.com>
	<1339161575.6898.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <1339162882.86991.YahooMailNeo@web120101.mail.ne1.yahoo.com>

hi,
i have renamed ?/usr/local/freeswitch/conf/dialplan/default/999_v_local_extension.xml ? ?to?999_v_local_extension.xml.noload
and reloadxml,and everythings worked correctly;;;
thanks...

?cat ?999_v_local_extension.xml
<extension name="local_extension" >
? ?<condition field="destination_number" expression="(^\d{2,7}$)" >
? ? ? ?<action application="set" data="dialed_extension=$1" />
? ? ? ?<action application="export" data="dialed_extension=$1" />
? ? ? ?<action application="limit" data="hash ${domain_name} $1 ${limit_max} ${limit_destination}" />
? ? ? ?<action application="bind_meta_app" data="1 b s execute_extension::dx XML features" />
? ? ? ?<action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.wav" />
? ? ? ?<action application="bind_meta_app" data="3 b s execute_extension::cf XML features" />
? ? ? ?<action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features" />
? ? ? ?<action application="set" data="ringback=${us-ring}" />
? ? ? ?<action application="set" data="transfer_ringback=$${hold_music}" />
? ? ? ?<action application="set" data="call_timeout=30" />
? ? ? ?<action application="set" data="hangup_after_bridge=true" />
? ? ? ?<action application="set" data="continue_on_fail=true" />
? ? ? ?<action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}" />
? ? ? ?<action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}" />
? ? ? ?<action application="set" data="called_party_call_group=${user_data(${dialed_extension}@${domain_name} var call_group)}" />
? ? ? ?<action application="hash" data="insert/${domain_name}-last_dial/${called_party_call_group}/${uuid}" />
? ? ? ?<action application="bridge" data="user/${user_data(${destination_number}@${domain_name} attr id)}@${domain_name}" />
? ? ? ?<action application="answer" />
? ? ? ?<action application="sleep" data="1000" />
? ? ? ?<action application="voicemail" data="default ${domain_name} ${dialed_extension}" />
? ?</condition>
</extension>


________________________________
 From: Samira Mh <saami_mh at ymail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Friday, June 8, 2012 5:49 PM
Subject: Re: [Freeswitch-users] how to Block inbound(internal) sofia profile calls ?(don't allow for internal calls)
 



hi,
thanks for your reply;
but default.xml is consist ad follow:


<?xml version="1.0" encoding="utf-8"?>
<include>
<context name="default">
? ? ? ? <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
</context>
</include>
but how can i limited internal calls?

________________________________
 From: babak yakhchali <babak.freeswitch at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Friday, June 8, 2012 5:23 PM
Subject: Re: [Freeswitch-users] how to Block inbound(internal) sofia profile calls ?(don't allow for internal calls)
 

if you are using xml dialplan, to block calls between and to 1000 and 1001 you can just remove any extension that routes calls between and to 1000 and 1001 in your dialplan (such as "Local_Extension" in sample configuration in default.xml) ?and to have outbound calls just add an extension that has a condition matching your destination.?
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML?


On Fri, Jun 8, 2012 at 4:13 PM, Samira Mh <saami_mh at ymail.com> wrote:

hi guys,
>i have defined two extensions named:1000 & 1001
>so i am going to allow the above extension make outbound calls and disable inbount calls
>in other hand i dont want the ext:1000 able to make call to ext:1001 ,
>and want to disable calling internal registerations sip?
>how can i do that?
>thanks so much
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From steveu at coppice.org  Fri Jun  8 18:40:39 2012
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 08 Jun 2012 22:40:39 +0800
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
 list?
In-Reply-To: <0eef01cd44c3$98a82400$c9f86c00$@bizfocused.com>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>	<4FD007D0.7010705@puzzled.xs4all.nl>
	<-3243025629320733747@unknownmsgid>
	<0eef01cd44c3$98a82400$c9f86c00$@bizfocused.com>
Message-ID: <4FD20EE7.3050203@coppice.org>

Mailing lists are very searchable in the real world. Try Googling 
something list "freeswitch mailing list web forum", and you can quickly 
go through some of the occasions this issue was raised before.

Mailing lists seldom turn into a war zone. News groups often do. Web 
forums only avoid being a war zone with *aggressive* policing. They are 
usually slow and clumsy to navigate through, too. Why waste the effort? 
Mailing lists sucks, but the alternatives available today are still worse.

Steve

On 06/07/2012 11:38 PM, Sean Devoy wrote:
> WTF was that supposed to mean?
>
> Nail meet Screw!
> Hammer meet Screw Gun.
>
> Better things come along as we progress.  Most of us  are in fact no longer
> living in caves even though they provide great protection from tornados.
>
> I was simply asking for a consensus on an idea.  I have an earlier post
> asking if there was software or a website that allowed this type of
> interface.  No responses to that question.
>
> Thank you Timothy Bolton for the link to nabble, I had never heard of it.
> It's not a great interface, but far more usable than dozens of unsearchable
> email attachments from the "batched in daily digest".
>
> I see peoples point about email accessibility off line.  I guess I have
> lived where we have broadband, smart phone and abundant free wifi for so
> long I did not consider being off the net.  That is certainly a valid point.
> I must say it is counterintuitive for a "forum" on Internet Telephony to be
> concerned about users without Internet Access.
>
> I withdraw my offer an apologize for bringing up the idea.
>
> Back to the caves!  :-)
>
> -----Original Message-----
> From: Brian West [mailto:brian at freeswitch.org]
> Sent: Wednesday, June 06, 2012 9:48 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] How about a USER FORUM and kill off the mail
> list?
>
> Hammer, meet Nail!
>
> Sent from my iPad
>
> On Jun 6, 2012, at 8:47 PM, Patrick Lists
> <freeswitch-list at puzzled.xs4all.nl>  wrote:
>
>> Since it has been a while I guess it was inevitable that someone would
>> come around with the forum idea again. Have you searched the mailing
>> list archives and read the previous discussion on this subject?
>> It did not work for the Asterisk community and I doubt it would work
>> for this community. If you want forum style, use nabble or gmane.
>>
>> Regards,
>> Patrick
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From toddb at toddbailey.net  Fri Jun  8 19:24:23 2012
From: toddb at toddbailey.net (toddb at toddbailey.net)
Date: Fri, 08 Jun 2012 08:24:23 -0700
Subject: [Freeswitch-users]
 =?utf-8?q?How_about_a_USER_FORUM_and_kill_off_?=
 =?utf-8?q?the_mail_list=3F?=
Message-ID: <20120608082423.33e327b490679d2282e332758c73b55b.7161020683.wbe@email14.secureserver.net>

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From evgeniy at bestnet.kharkov.ua  Fri Jun  8 11:21:30 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Fri, 08 Jun 2012 10:21:30 +0300
Subject: [Freeswitch-users] mod_nibblebill problem
In-Reply-To: <4FD199DA.7080805@bestnet.kharkov.ua>
References: <4FD199DA.7080805@bestnet.kharkov.ua>
Message-ID: <4FD1A7FA.3090009@bestnet.kharkov.ua>

Hi All,
when i'm calling from one user to another freeswitch user mod works
fine, but when i'm calling to external number (through gateway) billing
doesn't work:

2012-06-07 16:16:07.294840 [DEBUG] mod_nibblebill.c:612 Received request
via SESSION_HEARTBEAT!
2012-06-07 16:16:07.294840 [DEBUG] mod_nibblebill.c:453 Attempting to
bill at $1 per minute to account 7604504
2012-06-07 16:16:07.294840 [DEBUG] mod_nibblebill.c:465 Not billing
7604504 - call is not in answered state

I don't understand what this message means.

Sorry for my english:)
-- 
Evgeniy Movlyan,
BestNet Ltd.


From colins at succinct.co.za  Fri Jun  8 12:29:16 2012
From: colins at succinct.co.za (Colin Sindle)
Date: Fri, 8 Jun 2012 10:29:16 +0200
Subject: [Freeswitch-users] Garbled Leg Recording G729 with Asynchronous
	ptime
Message-ID: <CAFV278N8ShLqxz5x3ZJhc6OE5kxme1Vqaa8NTbscom_E5QCGjQ@mail.gmail.com>

Hi,

Please, I was wondering if there is anything obvious that we are missing
when trying to record a G729 call with asynchronous pimes (in: 20ms, out:
60ms).

The problem case call flow is:  inbound leg (ptime 20 ms), start recording,
playback a short message, bridge to outbound (destination can only handle a
ptime of 60 ms).   We are using mod_com_g729's codec.   The audio while on
the call in normally fine (rarely there is one way speech depending on
inbound/outbound provider), but *the recording has a garbled outbound
(Callee) leg* (inbound/Caller voice is fine.).

We have tried all the combinations of  G729 at 20i, G729 at 60i,
rtp-autofix-timing, and default-ptimes to no avail.

We are using git HEAD.

I do see this on fs_cli:

2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
0xa8894898 (nil)
2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
0xa8894898 (nil)
2012-06-07 16:56:45.583270 [WARNING] mod_sofia.c:1158 *Asynchronous PTIME
not supported, changing our end from 20 to 60*
2012-06-07 16:56:45.583270 [DEBUG] sofia_glue.c:2931 Changing Codec from
G729 at 20ms@8000hz to G729 at 60ms@8000hz
2012-06-07 16:56:45.623271 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
0xa86fe340 0xa86f1e78

Should this be working, are we missing something obvious?


Thank you,


Kind regards,


Colin
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From lazyvirus at gmx.com  Fri Jun  8 19:40:04 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Fri, 8 Jun 2012 17:40:04 +0200
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
 list?
In-Reply-To: <CACUErJeiVSjGbJ0cZZQ4dSBJhDX0oX26Dng2aNNOuhBp_fPvig@mail.gmail.com>
References: <20120607080603.33e327b490679d2282e332758c73b55b.7f27a511f2.wbe@email14.secureserver.net>
	<4FD0C5B0.2070602@gmail.com>
	<CACUErJeiVSjGbJ0cZZQ4dSBJhDX0oX26Dng2aNNOuhBp_fPvig@mail.gmail.com>
Message-ID: <20120608174004.585df222@anubis.defcon1>

On Thu, 7 Jun 2012 11:32:52 -0400
Philippe Le Toquin <philippe at ppmt.org> wrote:

> 
> Like that we have the best of both world. A mailing list and a
> forum.

I recently dig this question and found fudforum (http://fudforum.org)
that is a composite solution: it merges forums and MLs together, so
you can access what you want the way you want:)

My 0.0160151 ??
-- 
Being frustrated is disagreeable, but the real disasters in life
begin when you get what you want.


From Hector.Geraldino at ipsoft.com  Fri Jun  8 19:43:07 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Fri, 8 Jun 2012 11:43:07 -0400
Subject: [Freeswitch-users] Unable to use mod_java.
In-Reply-To: <CAP_69mtZuWq-22j-hiFBBuYzNKzqd46fR0MPjXgJYYd1kfjxnw@mail.gmail.com>
References: <CAP_69mtZuWq-22j-hiFBBuYzNKzqd46fR0MPjXgJYYd1kfjxnw@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD022EE810B8@NY1-EXMB-01.ip-soft.net>

Are you on the latest git?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Venkatesh
Sent: Thursday, June 07, 2012 8:58 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Unable to use mod_java.

Hi !

I have been trying to enable mod_java with FS but am having problems. I was able to compile mod_java fine. However FS is having trouble loading the module. It errors out with the following error message:

2012-06-07 20:51:24.058126 [ERR] modjava.c:326 Error creating Java VM!,

I believe I have the CLASSPATH/PATH variables set up correctly in my configuration files. In fact I put in some debug statements in modjava.c. Below is the debug output showing the various variables:

2012-06-07 20:51:24.058126 [NOTICE] modjava.c:349 Java Framework Loading...
2012-06-07 20:51:24.058126 [ERR] modjava.c:299 VENKY optionCount: 7
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Djava.class.path=/usr/local/freeswitch/scripts
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Djava.library.path=/usr/java/jdk1.6.0_32/jre/lib:/usr/java/jdk1.6.0_32/lib
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -verbose:jni
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Xms=64
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Xmx=256
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=127.0.0.1:8000<http://127.0.0.1:8000>
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Djava.library.path=/usr/local/freeswitch/mod
2012-06-07 20:51:24.058126 [ERR] modjava.c:326 Error creating Java VM!, error: -1
2012-06-07 20:51:24.058126 [INFO] switch_time.c:1123 Timezone reloaded 530 definitions
2012-06-07 20:51:24.058126 [CRIT] switch_loadable_module.c:1300 Error Loading module /usr/local/freeswitch/mod/mod_java.so
**Module load routine returned an error**


Would really appreciate any ideas from others that may have tried enabling the same?

Venkatesh
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From mike.kallies at gmail.com  Fri Jun  8 19:55:35 2012
From: mike.kallies at gmail.com (Mike Kallies)
Date: Fri, 08 Jun 2012 11:55:35 -0400
Subject: [Freeswitch-users] Audiocodes MP-118 VoIP Gateway can't un-hold
In-Reply-To: <CAAtWKeFcB1=XiiL7D0PvHEX1WM+PDdwkOESzTQwVn8ZquapO3w@mail.gmail.com>
References: <4FD12184.4000402@gmail.com>
	<CAAtWKeFcB1=XiiL7D0PvHEX1WM+PDdwkOESzTQwVn8ZquapO3w@mail.gmail.com>
Message-ID: <4FD22077.90004@gmail.com>

Hello Yehavi,

It was already set to not disconnect.  It doesn't drop the call, it just
doesn't unhold.  I can hold again and it will play music, then when I
unhold, the connection doesn't come back and the error message reappears
in the logs.

I'm not sure how 'hold' works in this system.  The phone sends a signal
to Freeswitch, then Freeswitch uses the dial plan to go to the hold
music?  Does that require negotiation of a new audio protocol or
something?  Then when we un-hold, it tries to negotiate back?

I'm a n00b with Freeswitch.  Is there a document I'm missing which
describes how hold works?

Thanks,

-Mike


On 07/06/2012 11:39 PM, Yehavi Bourvine wrote:
> Search for the parameter DisconnectOnBrokenConnection and set it to
> zero. I don't remember on which screen it is.
>  
>                         __Yehavi:
> 
> 2012/6/8 Mike Kallies <mike.kallies at gmail.com
> <mailto:mike.kallies at gmail.com>>
> 
>     Hey Everyone,
> 
>     I'm using an Audiocodes MP-118 VoIP Gateway.  When making a call through
>     the device to a Freeswitch gateway on the other side, it sends the call
>     to my desk phone just fine.
> 
>     When I mute the call, we get the hold music as per the dial plan,
>     everything is good.
> 
>     Then when I unmute... the hold music stops, and... nothing.  The call
>     doesn't drop, it just sits there.
> 
>     In the Audiocodes log, we see:
> 
> 
>     5d:1h:25m:32s (   lgr_psbrdex)(28879     )  recv <--
>     acEV_BROKEN_CONNECTION, Ch:0
>     5d:1h:25m:32s (      lgr_flow)(28880     )  #0:RTP_BROKEN_CONNECTION_EV
>     5d:1h:25m:32s (      lgr_flow)(28881     )  |
>     #0:RTP_BROKEN_CONNECTION_EV
> 
> 
>     I know this isn't entirely a Freeswitch question, but I figure the crowd
>     here might know the answer or where I should look for one.
> 
>     Calls through our Grandstream device work fine, so I think this can be
>     narrowed down to the Audiocodes device.  But maybe there's a different
>     codec or option or something I could try to see if I can make it happy?
> 
>     Thanks,
> 
>     -Mike Kallies
> 
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
> 
>     
>     
> 
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
> 
>     Join Us At ClueCon - Aug 7-9, 2012
> 
>     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
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From tnelson at rockbochs.com  Fri Jun  8 19:57:30 2012
From: tnelson at rockbochs.com (Tim Nelson)
Date: Fri, 8 Jun 2012 10:57:30 -0500 (CDT)
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
 list?
In-Reply-To: <20120608174004.585df222@anubis.defcon1>
Message-ID: <17731604.448460.1339171050549.JavaMail.root@rockbochs.com>

----- Original Message -----
> On Thu, 7 Jun 2012 11:32:52 -0400
> Philippe Le Toquin <philippe at ppmt.org> wrote:
> 
> > 
> > Like that we have the best of both world. A mailing list and a
> > forum.
> 
> I recently dig this question and found fudforum (http://fudforum.org)
> that is a composite solution: it merges forums and MLs together, so
> you can access what you want the way you want:)
> 

I was just about to suggest this approach as well. phpBB has a module to combine forums/mailing lists and it works swimmingly well. For a taste of operation, head over to the Vyatta forums/list [1] .

--Tim

[1] http://www.vyatta.org/forum/


From vvenkatar at gmail.com  Fri Jun  8 20:06:44 2012
From: vvenkatar at gmail.com (Venkatesh)
Date: Fri, 8 Jun 2012 09:06:44 -0700
Subject: [Freeswitch-users] Unable to use mod_java.
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD022EE810B8@NY1-EXMB-01.ip-soft.net>
References: <CAP_69mtZuWq-22j-hiFBBuYzNKzqd46fR0MPjXgJYYd1kfjxnw@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD022EE810B8@NY1-EXMB-01.ip-soft.net>
Message-ID: <CAP_69msEUaNHVk8OSQsy41-Shg=t0F5MjSSZYPwvFiLdaq+yMg@mail.gmail.com>

I believe I am on the latest version.

Venkatesh

On Fri, Jun 8, 2012 at 8:43 AM, Hector Geraldino <
Hector.Geraldino at ipsoft.com> wrote:

> Are you on the latest git?****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Venkatesh
> *Sent:* Thursday, June 07, 2012 8:58 PM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Unable to use mod_java.****
>
> ** **
>
> Hi !****
>
> ** **
>
> I have been trying to enable mod_java with FS but am having problems. I
> was able to compile mod_java fine. However FS is having trouble loading the
> module. It errors out with the following error message:****
>
> ** **
>
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:326 Error creating Java VM!,***
> *
>
> ** **
>
> I believe I have the CLASSPATH/PATH variables set up correctly in my
> configuration files. In fact I put in some debug statements in modjava.c.
> Below is the debug output showing the various variables:****
>
> ** **
>
> 2012-06-07 20:51:24.058126 [NOTICE] modjava.c:349 Java Framework Loading...
> ****
>
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:299 VENKY optionCount: 7****
>
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option:
> -Djava.class.path=/usr/local/freeswitch/scripts****
>
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option:
> -Djava.library.path=/usr/java/jdk1.6.0_32/jre/lib:/usr/java/jdk1.6.0_32/lib
> ****
>
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -verbose:jni*
> ***
>
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Xms=64****
>
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Xmx=256****
>
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option:
> -agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=
> 127.0.0.1:8000****
>
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option:
> -Djava.library.path=/usr/local/freeswitch/mod****
>
> 2012-06-07 20:51:24.058126 [ERR] modjava.c:326 Error creating Java VM!,
> error: -1****
>
> 2012-06-07 20:51:24.058126 [INFO] switch_time.c:1123 Timezone reloaded 530
> definitions****
>
> 2012-06-07 20:51:24.058126 [CRIT] switch_loadable_module.c:1300 Error
> Loading module /usr/local/freeswitch/mod/mod_java.so****
>
> **Module load routine returned an error******
>
> ** **
>
> ** **
>
> Would really appreciate any ideas from others that may have tried enabling
> the same? ****
>
> ** **
>
> Venkatesh****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From admin at blindi.net  Fri Jun  8 20:39:55 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Fri, 8 Jun 2012 18:39:55 +0200 (CEST)
Subject: [Freeswitch-users] opensource project voicechat for freeswitch
In-Reply-To: <CALQZz0xy8da2jbtJOmphAKGSkY-qzxi90tBEx=Tu6q8p=QMbQg@mail.gmail.com>
References: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
	<2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com>
	<1338558318.79806.YahooMailNeo@web120405.mail.ne1.yahoo.com>
	<Pine.LNX.4.64.1206041700001.22234@tomback.antagus.de>
	<CALQZz0zLpRBnFu3jce07Ycq=mn_OnAGrVFZgm-ibY_3eGg93Xw@mail.gmail.com>
	<Pine.LNX.4.64.1206061742390.4294@tomback.antagus.de>
	<CALQZz0xy8da2jbtJOmphAKGSkY-qzxi90tBEx=Tu6q8p=QMbQg@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1206081829500.19137@tomback.antagus.de>

Hi Sanath,

> I wrote complete voice mail system using LUA. (store messages, recording
> messages, retrieve messages, play messages with out bound dialing, send
> message as email attachment to end user, etc)

Thanks!! Very cool!
I think a global Coffig file we will need.
For example for the databaseaccess.

Can you send the Lua script please?
  I am inputting it into the tar.gz at:
http://www.blindi.net/downloads/open_vocechat.tar.gz
I create a changelog..
And i like to publish your name, and your nice work.
Can i do this?

---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From bdfoster at endigotech.com  Fri Jun  8 20:58:14 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 8 Jun 2012 12:58:14 -0400
Subject: [Freeswitch-users] mod_nibblebill problem
In-Reply-To: <4FD1A7FA.3090009@bestnet.kharkov.ua>
References: <4FD199DA.7080805@bestnet.kharkov.ua>
	<4FD1A7FA.3090009@bestnet.kharkov.ua>
Message-ID: <CAKV-jFxT1UdG1xea8CEsKodHZ3+zn+TczgDOg9BvK5FKtY5P-w@mail.gmail.com>

Calls aren't billed when the phone is ringing, only id/when call is
answered.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 8, 2012 11:26 AM, "Evgeniy Movlyan" <evgeniy at bestnet.kharkov.ua>
wrote:

> Hi All,
> when i'm calling from one user to another freeswitch user mod works
> fine, but when i'm calling to external number (through gateway) billing
> doesn't work:
>
> 2012-06-07 16:16:07.294840 [DEBUG] mod_nibblebill.c:612 Received request
> via SESSION_HEARTBEAT!
> 2012-06-07 16:16:07.294840 [DEBUG] mod_nibblebill.c:453 Attempting to
> bill at $1 per minute to account 7604504
> 2012-06-07 16:16:07.294840 [DEBUG] mod_nibblebill.c:465 Not billing
> 7604504 - call is not in answered state
>
> I don't understand what this message means.
>
> Sorry for my english:)
> --
> Evgeniy Movlyan,
> BestNet Ltd.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sdevoy at bizfocused.com  Fri Jun  8 21:04:29 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 8 Jun 2012 13:04:29 -0400
Subject: [Freeswitch-users] List Admin help
Message-ID: <016201cd4598$c47219d0$4d564d70$@bizfocused.com>

OK, I see we are stuck with a mailing list and I am very sorry to have
brought it up.  Fine.

 

Will someone please tell me how to change my "digest" setting?

DO NOT TELL ME TO GO TO:
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

It does not work.  Every effort I make sends me a stupid message saying I am
already enrolled.

 

I even assumed I don't know the password.  I wrote to:
freeswitch-users-owner at lists.freeswitch.org  that was equally ineffective,
but instead of a meaningless reply, I got no reply.

Is there some way to get my password or even reset it?

 

 

Thanks,

A very frustrated Sean.

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From cliff at develix.com  Fri Jun  8 21:12:53 2012
From: cliff at develix.com (Cliff Wells)
Date: Fri, 08 Jun 2012 10:12:53 -0700
Subject: [Freeswitch-users] Maillist client software - NOOB
In-Reply-To: <0c4801cd4440$3c748f80$b55dae80$@bizfocused.com>
References: <0a2c01cd440c$1da39ad0$58ead070$@bizfocused.com>
	<1339008663.2103.12.camel@portable-evil>
	<0c4801cd4440$3c748f80$b55dae80$@bizfocused.com>
Message-ID: <1339175573.7901.5.camel@portable-evil>

Can you unsubscribe and re-subscribe with the correct options set?

Mailman does suck quite a bit.

Cliff

On Wed, 2012-06-06 at 19:58 -0400, Sean Devoy wrote:
> Thanks Cliff.  I have tried that many, many times.  Every attempt gets me an email saying someone has tried to sign up my address again.  I assume I don't have the right password.
> 
> Thanks again
> 
> 
> 
> -----Original Message-----
> From: Cliff Wells [mailto:cliff at develix.com] 
> Sent: Wednesday, June 06, 2012 2:51 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Maillist client software - NOOB
> 
> On Wed, 2012-06-06 at 13:45 -0400, Sean Devoy wrote:
> > HI All,
> > 
> >  
> > 
> > I have been getting more and more familiar with FS and can actually 
> > make some contributions to the questions coming through in the 
> > maillist.  However, when I signed up I chose the option to bundle the 
> > messages before sending them to me. ? BIG MISTAKE ?.  I use MS Outlook 
> > as my mail client.
> 
> Go to the mailman link at the bottom of every email.  Change your
> preferences:
> 
> Would you like to receive list mail batched in a daily digest?  No.
> 
> Regards,
> Cliff
> 
> 
> 
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From sdevoy at bizfocused.com  Fri Jun  8 21:12:58 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 8 Jun 2012 13:12:58 -0400
Subject: [Freeswitch-users] Audio Connection Delay with SCA
Message-ID: <018101cd4599$f3b2b320$db181960$@bizfocused.com>

Hi,

 

I finally got Shared Call Appearance working to 4 NAT connected phones!!!!!

 

Now the users say when they answer that line, they must wait a second or so
to start talking (no big deal), but they all claim it takes nearly 5 seconds
to be able to hear the caller.  They answer the standard "Thank You for
calling XYZ, how can I help you today" which the caller DOES HEAR, but they
cannot hear his response for the first 5 seconds and ALWAYS have to ask them
to repeat it.

 

In the logs, I see what I believe is them picking up extension 299:

    2012-06-08 11:44:15.115249 [NOTICE] switch_channel.c:926 New Channel
sofia/external/sip:299 at 69.251.170.6:5063 .

And then the RPT finally connecting:

   2012-06-08 11:44:20.665094 [DEBUG] sofia_glue.c:3248 AUDIO RTP
[sofia/external/sip:299 at 69.251.170.6:5064] 10.10.40.185 port 23500 ->
69.251.170.6 port 16498 codec: 0 ms: 20

 

Note the 5.5 second delay!  Granted there are 100+ lines between those if
you want to see them, I will post them.

After that audio is fine.

 

Any thoughts would be appreciated.

Sean

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From jaasmailing at gmail.com  Fri Jun  8 21:12:59 2012
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Fri, 08 Jun 2012 19:12:59 +0200
Subject: [Freeswitch-users] FS sends Hold signal to remote gateway
Message-ID: <4FD2329B.3010509@gmail.com>

Hi all,

I'm experiencing a strange behaviour when calls are on hold.

Situation:

Call 1. Phone A (internal profile) calls a number (PSTN endpoint) 
through a gateway on the external profile. If the phone sets on hold all 
is ok and the remote PSTN endpoint hears the FreeSWITCH music on hold.

Call 2. PSTN endpoint calls through the same gateway the Phone A. If the 
Phone A sets on hold the call, FreeSWITCH sends an hold signal to the 
remote gateway, so there is no Music on hold.

Is a normal behaviour? How I can have the same behaviour of call 1?


Regards,
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From lazyvirus at gmx.com  Fri Jun  8 21:15:52 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Fri, 8 Jun 2012 19:15:52 +0200
Subject: [Freeswitch-users] List Admin help
In-Reply-To: <016201cd4598$c47219d0$4d564d70$@bizfocused.com>
References: <016201cd4598$c47219d0$4d564d70$@bizfocused.com>
Message-ID: <20120608191552.05a8d973@anubis.defcon1>

On Fri, 8 Jun 2012 13:04:29 -0400
"Sean Devoy" <sdevoy at bizfocused.com> wrote:

> 
> DO NOT TELL ME TO GO TO:
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

I WON'T, this is part of your subscriber page, the one the welcome
e-mail is referencing.

JY
-- 
At the end of your life there'll be a good rest, and no further
activities are scheduled.


From cliff at develix.com  Fri Jun  8 21:17:54 2012
From: cliff at develix.com (Cliff Wells)
Date: Fri, 08 Jun 2012 10:17:54 -0700
Subject: [Freeswitch-users] List Admin help
In-Reply-To: <016201cd4598$c47219d0$4d564d70$@bizfocused.com>
References: <016201cd4598$c47219d0$4d564d70$@bizfocused.com>
Message-ID: <1339175874.7901.8.camel@portable-evil>

On Fri, 2012-06-08 at 13:04 -0400, Sean Devoy wrote:
> OK, I see we are stuck with a mailing list and I am very sorry to have
> brought it up.  Fine.
> 
>  
> 
> Will someone please tell me how to change my ?digest? setting?
> 
> DO NOT TELL ME TO GO TO:
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> It does not work.  Every effort I make sends me a stupid message
> saying I am already enrolled.
> 
>  
> 
> I even assumed I don?t know the password.  I wrote to:
> freeswitch-users-owner at lists.freeswitch.org  that was equally
> ineffective, but instead of a meaningless reply, I got no reply.
> 
> Is there some way to get my password or even reset it?

I just went to the link you mentioned above, scrolled down and clicked
the button that says "unsubscribe or edit options".  I left the email
field blank.

On the next page, I filled in my email address, scrolled down to the
"Password reminder" section and clicked the "remind" button. Within two
minutes I had my password in my inbox.

Regards,
Cliff







From brian at freeswitch.org  Fri Jun  8 22:18:47 2012
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jun 2012 13:18:47 -0500
Subject: [Freeswitch-users] List Admin help
In-Reply-To: <1339175874.7901.8.camel@portable-evil>
References: <016201cd4598$c47219d0$4d564d70$@bizfocused.com>
	<1339175874.7901.8.camel@portable-evil>
Message-ID: <3953782102430870665@unknownmsgid>

can someone add this to the FAQ on the wiki?

/b

Sent from my Cave

On Jun 8, 2012, at 12:18 PM, Cliff Wells <cliff at develix.com> wrote:

> On Fri, 2012-06-08 at 13:04 -0400, Sean Devoy wrote:
>> OK, I see we are stuck with a mailing list and I am very sorry to have
>> brought it up.  Fine.
>>
>>
>>
>> Will someone please tell me how to change my ?gdigest?h setting?
>>
>> DO NOT TELL ME TO GO TO:
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> It does not work.  Every effort I make sends me a stupid message
>> saying I am already enrolled.
>>
>>
>>
>> I even assumed I don?ft know the password.  I wrote to:
>> freeswitch-users-owner at lists.freeswitch.org  that was equally
>> ineffective, but instead of a meaningless reply, I got no reply.
>>
>> Is there some way to get my password or even reset it?
>
> I just went to the link you mentioned above, scrolled down and clicked
> the button that says "unsubscribe or edit options".  I left the email
> field blank.
>
> On the next page, I filled in my email address, scrolled down to the
> "Password reminder" section and clicked the "remind" button. Within two
> minutes I had my password in my inbox.
>
> Regards,
> Cliff
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Fri Jun  8 22:20:51 2012
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jun 2012 13:20:51 -0500
Subject: [Freeswitch-users] Maillist client software - NOOB
In-Reply-To: <1339175573.7901.5.camel@portable-evil>
References: <0a2c01cd440c$1da39ad0$58ead070$@bizfocused.com>
	<1339008663.2103.12.camel@portable-evil>
	<0c4801cd4440$3c748f80$b55dae80$@bizfocused.com>
	<1339175573.7901.5.camel@portable-evil>
Message-ID: <-8172420677588195851@unknownmsgid>

Why not just read the page, login and edit your options.  Mailman is
fine, its attention to details the users are tripping on.  It's a
common issue.

Sent from my Cave

On Jun 8, 2012, at 12:13 PM, Cliff Wells <cliff at develix.com> wrote:

> Can you unsubscribe and re-subscribe with the correct options set?
>
> Mailman does suck quite a bit.
>
> Cliff


From brian at freeswitch.org  Fri Jun  8 22:22:20 2012
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jun 2012 13:22:20 -0500
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <4FD20EE7.3050203@coppice.org>
References: <0c5201cd4440$97f03530$c7d09f90$@bizfocused.com>
	<4FD007D0.7010705@puzzled.xs4all.nl>
	<-3243025629320733747@unknownmsgid>
	<0eef01cd44c3$98a82400$c9f86c00$@bizfocused.com>
	<4FD20EE7.3050203@coppice.org>
Message-ID: <-6143061278004381769@unknownmsgid>

well put!

Sent from my beach hut (cave upgrade)

On Jun 8, 2012, at 9:42 AM, Steve Underwood <steveu at coppice.org> wrote:

> Mailing lists are very searchable in the real world. Try Googling
> something list "freeswitch mailing list web forum", and you can quickly
> go through some of the occasions this issue was raised before.
>
> Mailing lists seldom turn into a war zone. News groups often do. Web
> forums only avoid being a war zone with *aggressive* policing. They are
> usually slow and clumsy to navigate through, too. Why waste the effort?
> Mailing lists sucks, but the alternatives available today are still worse.
>
> Steve


From robert.longfield at klinsight.com  Fri Jun  8 22:25:41 2012
From: robert.longfield at klinsight.com (Robert Longfield)
Date: Fri, 8 Jun 2012 14:25:41 -0400
Subject: [Freeswitch-users] Time needed for ringing a call group
Message-ID: <D3BC5C093B2E4150AF352558D6E077BE@KITPC003>

I have support call group that works as it should. When a caller selects the support call group it rings all the support extensions as it should. The problem arises when a support member is logged in to their extension but is not at their computer. The call will continue to ring the call group forever. I would like to change this so it will only ring for a set amount of time.

Currently if the support group is busy or they manually decline the call the incoming call will be transferred to the general extension. 
I would like to mimic this behaviour when the call goes unanswered after 20 secs or so.

Here is the Call Group Settings:

    <extension name="group_dial_support">
      <condition field="destination_number" expression="^2001$">
        <action application="set" data="call_timeout=2000"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="group/support@$${domain}"/>
        <action application="answer"/>
        <action application="sleep" data="100"/>
        <action application="bridge" data="sofia/internal/1000@$${domain}"/>
      </condition>
    </extension>

I hope I?ve explained my problem clearly.

-Rob

Robert Longfield
SEO Specialist
KL Insight
http://klinsight.com/
Tel: 613-344-2116 | Fax: 613.634.7029
993 Princess Street, Suite 212
Kingston, ON K7L 1H3, Canada


Notice of Confidentiality:
The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review re-transmission dissemination or other use of or taking of any action in reliance upon this information by persons or entities other than the intended recipient is prohibited. If you received this in error please contact the sender immediately by return electronic transmission and then immediately delete this transmission including all attachments without copying distributing or disclosing same.
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From yehavi.bourvine at gmail.com  Fri Jun  8 22:33:38 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 8 Jun 2012 21:33:38 +0300
Subject: [Freeswitch-users] Audiocodes MP-118 VoIP Gateway can't un-hold
In-Reply-To: <4FD22077.90004@gmail.com>
References: <4FD12184.4000402@gmail.com>
	<CAAtWKeFcB1=XiiL7D0PvHEX1WM+PDdwkOESzTQwVn8ZquapO3w@mail.gmail.com>
	<4FD22077.90004@gmail.com>
Message-ID: <CAAtWKeFh+gMDqOAnV_3jeu6Bhtoxvg9RC4rd84BgNR9Lk09d+w@mail.gmail.com>

Hello Mike,

  As far as I know hold is handled natively by FreeSwitch and you do not
have to do anything in your dialplan.

There are two ways for the MP to ask for hold: One is to set the media
address to 0.0.0.0 and the other is to set it to send-only or receive-only.
As far as I recall (I am at home now) the MP supports both; try changing
the method and see whether it helps,

                           __Yehavi:

2012/6/8 Mike Kallies <mike.kallies at gmail.com>

> Hello Yehavi,
>
> It was already set to not disconnect.  It doesn't drop the call, it just
> doesn't unhold.  I can hold again and it will play music, then when I
> unhold, the connection doesn't come back and the error message reappears
> in the logs.
>
> I'm not sure how 'hold' works in this system.  The phone sends a signal
> to Freeswitch, then Freeswitch uses the dial plan to go to the hold
> music?  Does that require negotiation of a new audio protocol or
> something?  Then when we un-hold, it tries to negotiate back?
>
> I'm a n00b with Freeswitch.  Is there a document I'm missing which
> describes how hold works?
>
> Thanks,
>
> -Mike
>
>
> On 07/06/2012 11:39 PM, Yehavi Bourvine wrote:
> > Search for the parameter DisconnectOnBrokenConnection and set it to
> > zero. I don't remember on which screen it is.
> >
> >                         __Yehavi:
> >
> > 2012/6/8 Mike Kallies <mike.kallies at gmail.com
> > <mailto:mike.kallies at gmail.com>>
> >
> >     Hey Everyone,
> >
> >     I'm using an Audiocodes MP-118 VoIP Gateway.  When making a call
> through
> >     the device to a Freeswitch gateway on the other side, it sends the
> call
> >     to my desk phone just fine.
> >
> >     When I mute the call, we get the hold music as per the dial plan,
> >     everything is good.
> >
> >     Then when I unmute... the hold music stops, and... nothing.  The call
> >     doesn't drop, it just sits there.
> >
> >     In the Audiocodes log, we see:
> >
> >
> >     5d:1h:25m:32s (   lgr_psbrdex)(28879     )  recv <--
> >     acEV_BROKEN_CONNECTION, Ch:0
> >     5d:1h:25m:32s (      lgr_flow)(28880     )
>  #0:RTP_BROKEN_CONNECTION_EV
> >     5d:1h:25m:32s (      lgr_flow)(28881     )  |
> >     #0:RTP_BROKEN_CONNECTION_EV
> >
> >
> >     I know this isn't entirely a Freeswitch question, but I figure the
> crowd
> >     here might know the answer or where I should look for one.
> >
> >     Calls through our Grandstream device work fine, so I think this can
> be
> >     narrowed down to the Audiocodes device.  But maybe there's a
> different
> >     codec or option or something I could try to see if I can make it
> happy?
> >
> >     Thanks,
> >
> >     -Mike Kallies
> >
> >
> _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com
> >
> >     
> >     
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org
> >     http://wiki.freeswitch.org
> >     http://www.cluecon.com
> >
> >     Join Us At ClueCon - Aug 7-9, 2012
> >
> >     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
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ifoundthetao at gmail.com  Fri Jun  8 22:34:29 2012
From: ifoundthetao at gmail.com (Timothy Bolton)
Date: Fri, 08 Jun 2012 13:34:29 -0500
Subject: [Freeswitch-users] Hardware build recommendations
Message-ID: <4FD245B5.9010708@gmail.com>

Does anyone have any ideas for a good hardware build for a SOHO server?

Here's what I'd like running on it: OpenVPN, FreeSWITCH, and a firewall 
(probably pfSense).

My home network doesn't have too much traffic, and I don't get that many 
calls.

-- 
'We who cut mere stones must always be envisioning cathedrals.'
Quarry Worker's Creed



From anthony.minessale at gmail.com  Fri Jun  8 23:03:42 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jun 2012 14:03:42 -0500
Subject: [Freeswitch-users] Garbled Leg Recording G729 with Asynchronous
	ptime
In-Reply-To: <CAFV278N8ShLqxz5x3ZJhc6OE5kxme1Vqaa8NTbscom_E5QCGjQ@mail.gmail.com>
References: <CAFV278N8ShLqxz5x3ZJhc6OE5kxme1Vqaa8NTbscom_E5QCGjQ@mail.gmail.com>
Message-ID: <CAKbxfG-g5fqC+ZQA5ZJzXwwzoy-zS71JB0MjuWkEaYcvN+bUbw@mail.gmail.com>

have you tried updating lately?
also try global_setvar passthru_ptime_mismatch=true


On Fri, Jun 8, 2012 at 3:29 AM, Colin Sindle <colins at succinct.co.za> wrote:
> Hi,
>
> Please, I was wondering if there is anything obvious that we are missing
> when trying to record a?G729 call with asynchronous pimes (in: 20ms, out:
> 60ms).
>
> The problem case call flow is: ?inbound leg (ptime 20 ms), start recording,
> playback a short message, bridge to outbound (destination can only handle a
> ptime of 60 ms). ? We are using mod_com_g729's codec. ? The audio while on
> the call in normally fine (rarely there is one way speech depending on
> inbound/outbound provider), but the recording has a garbled outbound
> (Callee) leg (inbound/Caller voice is fine.).
>
> We have tried all the?combinations?of ?G729 at 20i, G729 at 60i,
> rtp-autofix-timing, and?default-ptimes to no avail.
>
> We are using git HEAD.
>
> I do see this on fs_cli:
>
> 2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0xa8894898 (nil)
> 2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> 0xa8894898 (nil)
> 2012-06-07 16:56:45.583270 [WARNING] mod_sofia.c:1158 Asynchronous PTIME not
> supported, changing our end from 20 to 60
> 2012-06-07 16:56:45.583270 [DEBUG] sofia_glue.c:2931 Changing Codec from
> G729 at 20ms@8000hz to G729 at 60ms@8000hz
> 2012-06-07 16:56:45.623271 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> 0xa86fe340 0xa86f1e78
>
> Should this be working, are we missing something obvious?
>
>
> Thank you,
>
>
> Kind regards,
>
>
> Colin
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Fri Jun  8 23:28:02 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jun 2012 14:28:02 -0500
Subject: [Freeswitch-users] Audio Connection Delay with SCA
In-Reply-To: <018101cd4599$f3b2b320$db181960$@bizfocused.com>
References: <018101cd4599$f3b2b320$db181960$@bizfocused.com>
Message-ID: <CAKbxfG-RiMFttXuZQjZSbyG-Uk6gZLWT-XzFYZfqUGdhdPJL-Q@mail.gmail.com>

what type of phone?  If it has a jitter buffer, you can try reducing
or disabling it.
its probably a combo of jitter buffers charging up on the phone, and
the auto-ajust trying to fix the broken sdp for NAT.


On Fri, Jun 8, 2012 at 12:12 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> Hi,
>
>
>
> I finally got Shared Call Appearance working to 4 NAT connected phones!!!!!
>
>
>
> Now the users say when they answer that line, they must wait a second or so
> to start talking (no big deal), but they all claim it takes nearly 5 seconds
> to be able to hear the caller.? They answer the standard ?Thank You for
> calling XYZ, how can I help you today? which the caller DOES HEAR, but they
> cannot hear his response for the first 5 seconds and ALWAYS have to ask them
> to repeat it.
>
>
>
> In the logs, I see what I believe is them picking up extension 299:
>
> ??? 2012-06-08 11:44:15.115249 [NOTICE] switch_channel.c:926 New Channel
> sofia/external/sip:299 at 69.251.170.6:5063 ?
>
> And then the RPT finally connecting:
>
> ?? 2012-06-08 11:44:20.665094 [DEBUG] sofia_glue.c:3248 AUDIO RTP
> [sofia/external/sip:299 at 69.251.170.6:5064] 10.10.40.185 port 23500 ->
> 69.251.170.6 port 16498 codec: 0 ms: 20
>
>
>
> Note the 5.5 second delay!? Granted there are 100+ lines between those if
> you want to see them, I will post them.
>
> After that audio is fine.
>
>
>
> Any thoughts would be appreciated.
>
> Sean
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From sdevoy at bizfocused.com  Fri Jun  8 23:29:46 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 8 Jun 2012 15:29:46 -0400
Subject: [Freeswitch-users] List Admin help
In-Reply-To: <20120608191552.05a8d973@anubis.defcon1>
References: <016201cd4598$c47219d0$4d564d70$@bizfocused.com>
	<20120608191552.05a8d973@anubis.defcon1>
Message-ID: <026e01cd45ad$105a1410$310e3c30$@bizfocused.com>

I found the problem.  I was misguided when I was told to just resubmit my
information with the new preference at that link.

In fact, very clearly on that page in the section that says "Enter your
admin address and password to visit the subscribers list:"  is the
"unsubscribe and edit" button.  Where else would it be?  Up by the Subscribe
button? Nay Nay, people might find it there.

Silly me.

The stupid quips you send are as amusing as they are helpful.

Thanks.

-----Original Message-----
From: Bzzz [mailto:lazyvirus at gmx.com] 
Sent: Friday, June 08, 2012 1:16 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] List Admin help

On Fri, 8 Jun 2012 13:04:29 -0400
"Sean Devoy" <sdevoy at bizfocused.com> wrote:

> 
> DO NOT TELL ME TO GO TO:
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

I WON'T, this is part of your subscriber page, the one the welcome e-mail is
referencing.

JY
--
At the end of your life there'll be a good rest, and no further activities
are scheduled.






From brian at freeswitch.org  Fri Jun  8 23:37:34 2012
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jun 2012 14:37:34 -0500
Subject: [Freeswitch-users] FS sends Hold signal to remote gateway
In-Reply-To: <4FD2329B.3010509@gmail.com>
References: <4FD2329B.3010509@gmail.com>
Message-ID: <8709089764425887813@unknownmsgid>

sip traces and logs, please put them ?p on p??t?b??, Also have yo? tr??d
the ??t??t ??d??

Sent from my beach hut (cave 2.0 upgrade)

On Jun 8, 2012, at 12:13 PM, Carlo Dimaggio <jaasmailing at gmail.com> wrote:

 Hi all,

I'm experiencing a strange behaviour when calls are on hold.

Situation:

Call 1. Phone A (internal profile) calls a number (PSTN endpoint) through a
gateway on the external profile. If the phone sets on hold all is ok and the
remote PSTN endpoint hears the FreeSWITCH music on hold.

Call 2. PSTN endpoint calls through the same gateway the Phone A. If the
Phone A sets on hold the call, FreeSWITCH sends an hold signal to the
remote gateway, so there is no Music on hold.

Is a normal behaviour? How I can have the same behaviour of call 1?


Regards,

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 Jun  8 23:39:26 2012
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jun 2012 14:39:26 -0500
Subject: [Freeswitch-users] Time needed for ringing a call group
In-Reply-To: <D3BC5C093B2E4150AF352558D6E077BE@KITPC003>
References: <D3BC5C093B2E4150AF352558D6E077BE@KITPC003>
Message-ID: <-6559196381009927779@unknownmsgid>

do ?o? really mean 2000 second? h?r??

Sent from my biplane at 450ft (beach hut 2.0 upgrade)

On Jun 8, 2012, at 1:26 PM, Robert Longfield
<robert.longfield at klinsight.com> wrote:

>         <action application="set" data="call_timeout=2000"/>


From brian at freeswitch.org  Fri Jun  8 23:43:50 2012
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jun 2012 14:43:50 -0500
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <17731604.448460.1339171050549.JavaMail.root@rockbochs.com>
References: <17731604.448460.1339171050549.JavaMail.root@rockbochs.com>
Message-ID: <-3835502817330869863@unknownmsgid>

Why do topics like these get tons of interactions, but real issues
fall to the wayside.  I think the horse is dead Jim, beam me up
already.

Sent from my spaceship orbiting the moon (biplane upgrade)

On Jun 8, 2012, at 10:58 AM, Tim Nelson <tnelson at rockbochs.com> wrote:

> I was just about to suggest this approach as well. phpBB has a module to combine forums/mailing lists and it works swimmingly well. For a taste of operation, head over to the Vyatta forums/list [1] .
>
> --Tim
>
> [1] http://www.vyatta.org/forum/?


From ifoundthetao at gmail.com  Fri Jun  8 23:46:37 2012
From: ifoundthetao at gmail.com (Timothy Bolton)
Date: Fri, 08 Jun 2012 14:46:37 -0500
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
 list?
In-Reply-To: <-3835502817330869863@unknownmsgid>
References: <17731604.448460.1339171050549.JavaMail.root@rockbochs.com>
	<-3835502817330869863@unknownmsgid>
Message-ID: <4FD2569D.9080504@gmail.com>

It can be much more fun to put off work and b.s. by the virtual water 
cooler of an off-topic conversation. These topics lend themselves more 
toward opinion than the knowledge required to answer a question.

'We who cut mere stones must always be envisioning cathedrals.'
Quarry Worker's Creed

On 6/8/2012 2:43 PM, Brian West wrote:
> Why do topics like these get tons of interactions, but real issues
> fall to the wayside.  I think the horse is dead Jim, beam me up
> already.
>
> Sent from my spaceship orbiting the moon (biplane upgrade)
>
> On Jun 8, 2012, at 10:58 AM, Tim Nelson <tnelson at rockbochs.com> wrote:
>
>> I was just about to suggest this approach as well. phpBB has a module to combine forums/mailing lists and it works swimmingly well. For a taste of operation, head over to the Vyatta forums/list [1] .
>>
>> --Tim
>>
>> [1] http://www.vyatta.org/forum/?
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Fri Jun  8 23:52:05 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jun 2012 14:52:05 -0500
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <4FD2569D.9080504@gmail.com>
References: <17731604.448460.1339171050549.JavaMail.root@rockbochs.com>
	<-3835502817330869863@unknownmsgid> <4FD2569D.9080504@gmail.com>
Message-ID: <CAKbxfG8a+0vBbA=yO2fUNkeWwqOFMSVYnGmB2c3ZzG50S-RWGQ@mail.gmail.com>

we've looked at some alternate mailing lists that had some forum-like
features but we need time and energy to pursue things like that.
We need to get our branching and packaging done first.

I wish nabble and friends would actually go away, they horribly
pollute the google searches and their purpose is to draw banner ads
from our content.


On Fri, Jun 8, 2012 at 2:46 PM, Timothy Bolton <ifoundthetao at gmail.com> wrote:
> It can be much more fun to put off work and b.s. by the virtual water
> cooler of an off-topic conversation. These topics lend themselves more
> toward opinion than the knowledge required to answer a question.
>
> 'We who cut mere stones must always be envisioning cathedrals.'
> Quarry Worker's Creed
>
> On 6/8/2012 2:43 PM, Brian West wrote:
>> Why do topics like these get tons of interactions, but real issues
>> fall to the wayside. ?I think the horse is dead Jim, beam me up
>> already.
>>
>> Sent from my spaceship orbiting the moon (biplane upgrade)
>>
>> On Jun 8, 2012, at 10:58 AM, Tim Nelson <tnelson at rockbochs.com> wrote:
>>
>>> I was just about to suggest this approach as well. phpBB has a module to combine forums/mailing lists and it works swimmingly well. For a taste of operation, head over to the Vyatta forums/list [1] .
>>>
>>> --Tim
>>>
>>> [1] http://www.vyatta.org/forum/?
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From sdevoy at bizfocused.com  Fri Jun  8 23:52:22 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 8 Jun 2012 15:52:22 -0400
Subject: [Freeswitch-users] Audio Connection Delay with SCA
In-Reply-To: <CAKbxfG-RiMFttXuZQjZSbyG-Uk6gZLWT-XzFYZfqUGdhdPJL-Q@mail.gmail.com>
References: <018101cd4599$f3b2b320$db181960$@bizfocused.com>
	<CAKbxfG-RiMFttXuZQjZSbyG-Uk6gZLWT-XzFYZfqUGdhdPJL-Q@mail.gmail.com>
Message-ID: <029601cd45b0$38d29d60$aa77d820$@bizfocused.com>

Anthony,
Thanks for the reply.

We use all CSICO SPA504G phones.
I did have the jitter buffer set up "HIGH" (choices: Low, Med, High, Very
High, Extremely High).  I dropped them down to Low now for testing.  They do
have pretty good network speeds (business level cable modem).

There is also something labeled:  " Jitter Buffer Adjustment:  "  Choices
(Up, Down, Both, Disabled).  I have both.

However, the issue is present, but far less pronounced in non-shared lines.
They have always said there was a connection delay at this site, but with
SCA it seems to be compounded.

Maybe Group Call Pick will work better for them, they are all in the area.

Sean

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Friday, June 08, 2012 3:28 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Audio Connection Delay with SCA

what type of phone?  If it has a jitter buffer, you can try reducing or
disabling it.
its probably a combo of jitter buffers charging up on the phone, and the
auto-ajust trying to fix the broken sdp for NAT.


On Fri, Jun 8, 2012 at 12:12 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> Hi,
>
>
>
> I finally got Shared Call Appearance working to 4 NAT connected
phones!!!!!
>
>
>
> Now the users say when they answer that line, they must wait a second 
> or so to start talking (no big deal), but they all claim it takes 
> nearly 5 seconds to be able to hear the caller.? They answer the 
> standard ?Thank You for calling XYZ, how can I help you today? which 
> the caller DOES HEAR, but they cannot hear his response for the first 
> 5 seconds and ALWAYS have to ask them to repeat it.
>
>
>
> In the logs, I see what I believe is them picking up extension 299:
>
> ??? 2012-06-08 11:44:15.115249 [NOTICE] switch_channel.c:926 New 
> Channel
> sofia/external/sip:299 at 69.251.170.6:5063 

>
> And then the RPT finally connecting:
>
> ?? 2012-06-08 11:44:20.665094 [DEBUG] sofia_glue.c:3248 AUDIO RTP 
> [sofia/external/sip:299 at 69.251.170.6:5064] 10.10.40.185 port 23500 ->
> 69.251.170.6 port 16498 codec: 0 ms: 20
>
>
>
> Note the 5.5 second delay!? Granted there are 100+ lines between those 
> if you want to see them, I will post them.
>
> After that audio is fine.
>
>
>
> Any thoughts would be appreciated.
>
> Sean
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users 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
>



--
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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  Fri Jun  8 23:59:44 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jun 2012 14:59:44 -0500
Subject: [Freeswitch-users] Audio Connection Delay with SCA
In-Reply-To: <029601cd45b0$38d29d60$aa77d820$@bizfocused.com>
References: <018101cd4599$f3b2b320$db181960$@bizfocused.com>
	<CAKbxfG-RiMFttXuZQjZSbyG-Uk6gZLWT-XzFYZfqUGdhdPJL-Q@mail.gmail.com>
	<029601cd45b0$38d29d60$aa77d820$@bizfocused.com>
Message-ID: <CAKbxfG_a5QfK+FhqqMWmtRZhogw47rwyufnJRDOPk9unWyXxgQ@mail.gmail.com>

from FS point of view there should be little different from SCA and
not SCA besides the presence signaling, you should try some on-site
testing with the same phones.


On Fri, Jun 8, 2012 at 2:52 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> Anthony,
> Thanks for the reply.
>
> We use all CSICO SPA504G phones.
> I did have the jitter buffer set up "HIGH" (choices: Low, Med, High, Very
> High, Extremely High). ?I dropped them down to Low now for testing. ?They do
> have pretty good network speeds (business level cable modem).
>
> There is also something labeled: ?" Jitter Buffer Adjustment: ?" ?Choices
> (Up, Down, Both, Disabled). ?I have both.
>
> However, the issue is present, but far less pronounced in non-shared lines.
> They have always said there was a connection delay at this site, but with
> SCA it seems to be compounded.
>
> Maybe Group Call Pick will work better for them, they are all in the area.
>
> Sean
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
> Minessale
> Sent: Friday, June 08, 2012 3:28 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Audio Connection Delay with SCA
>
> what type of phone? ?If it has a jitter buffer, you can try reducing or
> disabling it.
> its probably a combo of jitter buffers charging up on the phone, and the
> auto-ajust trying to fix the broken sdp for NAT.
>
>
> On Fri, Jun 8, 2012 at 12:12 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>> Hi,
>>
>>
>>
>> I finally got Shared Call Appearance working to 4 NAT connected
> phones!!!!!
>>
>>
>>
>> Now the users say when they answer that line, they must wait a second
>> or so to start talking (no big deal), but they all claim it takes
>> nearly 5 seconds to be able to hear the caller.? They answer the
>> standard ?Thank You for calling XYZ, how can I help you today? which
>> the caller DOES HEAR, but they cannot hear his response for the first
>> 5 seconds and ALWAYS have to ask them to repeat it.
>>
>>
>>
>> In the logs, I see what I believe is them picking up extension 299:
>>
>> ??? 2012-06-08 11:44:15.115249 [NOTICE] switch_channel.c:926 New
>> Channel
>> sofia/external/sip:299 at 69.251.170.6:5063 ?
>>
>> And then the RPT finally connecting:
>>
>> ?? 2012-06-08 11:44:20.665094 [DEBUG] sofia_glue.c:3248 AUDIO RTP
>> [sofia/external/sip:299 at 69.251.170.6:5064] 10.10.40.185 port 23500 ->
>> 69.251.170.6 port 16498 codec: 0 ms: 20
>>
>>
>>
>> Note the 5.5 second delay!? Granted there are 100+ lines between those
>> if you want to see them, I will post them.
>>
>> After that audio is fine.
>>
>>
>>
>> Any thoughts would be appreciated.
>>
>> Sean
>>
>>
>> ______________________________________________________________________
>> ___ Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users 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
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From jmoran at secureachsystems.com  Sat Jun  9 00:16:02 2012
From: jmoran at secureachsystems.com (Jason Moran)
Date: Fri, 8 Jun 2012 16:16:02 -0400
Subject: [Freeswitch-users] JS File Operation Error with latest FS
Message-ID: <361E98F99D3CC3439EED59BC1924ED6966467E@SERVER2003.SecuReachSystems.local>

A spidermonkey File operation (often) causes the output of "ASS #####"
followed by a "File operation pclose failed" error now that I've done a
make current (last  update was 4 weeks ago).   The code puts a 0 or 1
into the variable fdFoundFile for later testing. This has worked for
months until the make current.  Now it fails with the pclose error 4/5
times.

 

CODE:

    console_log("info","-3-\n");

//Do a check to make sure we can read the file 

var fd = new File("|[ -f "+myFileName+" ] && echo 1 || echo 0");

    console_log("info","-4-\n");

fd.open("read");

    console_log("info","-5-\n");

var fdFoundFile = fd.read(1);

    console_log("info","-6-\n");

fd.close();

    console_log("info","-7-\n");

 

LOG:

2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -3-

2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -4-

2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -5-

ASS 30937

2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -6-

2012-06-08 15:52:14.717917 [ERR] Testcall.js:258  Error: File operation
pclose failed

 

The # following "ASS" always changes and keeps on increasing.

 

Thanks,
Jason M

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From bdfoster at endigotech.com  Sat Jun  9 00:23:50 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 8 Jun 2012 16:23:50 -0400
Subject: [Freeswitch-users] Hardware build recommendations
In-Reply-To: <4FD245B5.9010708@gmail.com>
References: <4FD245B5.9010708@gmail.com>
Message-ID: <CAKV-jFyD303PvFTXiVrzbS2w56b-1jKQwA8g05-EN0giJpUv3w@mail.gmail.com>

Raspberry Pi?

Shoot for 1.6 ghz dual core with 1gb memory, 2 NICs. That should be
overkill (engineering translation: bare minimum).

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 8, 2012 2:35 PM, "Timothy Bolton" <ifoundthetao at gmail.com> wrote:

> Does anyone have any ideas for a good hardware build for a SOHO server?
>
> Here's what I'd like running on it: OpenVPN, FreeSWITCH, and a firewall
> (probably pfSense).
>
> My home network doesn't have too much traffic, and I don't get that many
> calls.
>
> --
> 'We who cut mere stones must always be envisioning cathedrals.'
> Quarry Worker's Creed
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lazyvirus at gmx.com  Sat Jun  9 00:30:10 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Fri, 8 Jun 2012 22:30:10 +0200
Subject: [Freeswitch-users] Time needed for ringing a call group
In-Reply-To: <-6559196381009927779@unknownmsgid>
References: <D3BC5C093B2E4150AF352558D6E077BE@KITPC003>
	<-6559196381009927779@unknownmsgid>
Message-ID: <20120608223010.2d079108@anubis.defcon1>

On Fri, 8 Jun 2012 14:39:26 -0500
Brian West <brian at freeswitch.org> wrote:

> do ?o? really mean 2000 second? h?r??
     ^ ^                        ^  ^ ^ 
> Sent from my biplane at 450ft (beach hut 2.0 upgrade)

Hmm, either you're kidding or you have a keyboard|board problem
Brian; you're answer to Carlo also contains strange characters.

JY

Sent from my bitch phone (B-bra update)
-- 
You will be dead within a year.


From curriegrad2004 at gmail.com  Sat Jun  9 01:02:29 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 8 Jun 2012 14:02:29 -0700
Subject: [Freeswitch-users] Hardware build recommendations
In-Reply-To: <CAKV-jFyD303PvFTXiVrzbS2w56b-1jKQwA8g05-EN0giJpUv3w@mail.gmail.com>
References: <4FD245B5.9010708@gmail.com>
	<CAKV-jFyD303PvFTXiVrzbS2w56b-1jKQwA8g05-EN0giJpUv3w@mail.gmail.com>
Message-ID: <CAMWiMy84NibiUaFoGLZXhAKRY5HJXkLsJwunCmAHBvs_qETRLQ@mail.gmail.com>

Try aiming for a 64-bit OS if you could. For the OpenVPN part, if you
could shoot for a Sandy Bridge or later CPU you'd probably find the
AES-NI part of the CPU quite suitable. If you're on a budget you can
always certainly aim for an AMD E-350 or something along the lines of
that. Plenty of CPU power for SOHO boxes.

On Fri, Jun 8, 2012 at 1:23 PM, Brian Foster <bdfoster at endigotech.com> wrote:
> Raspberry Pi?
>
> Shoot for 1.6 ghz dual core with 1gb memory, 2 NICs. That should be overkill
> (engineering translation: bare minimum).
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
>
> On Jun 8, 2012 2:35 PM, "Timothy Bolton" <ifoundthetao at gmail.com> wrote:
>>
>> Does anyone have any ideas for a good hardware build for a SOHO server?
>>
>> Here's what I'd like running on it: OpenVPN, FreeSWITCH, and a firewall
>> (probably pfSense).
>>
>> My home network doesn't have too much traffic, and I don't get that many
>> calls.
>>
>> --
>> 'We who cut mere stones must always be envisioning cathedrals.'
>> Quarry Worker's Creed
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From bdfoster at endigotech.com  Sat Jun  9 01:09:08 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 8 Jun 2012 17:09:08 -0400
Subject: [Freeswitch-users] JS File Operation Error with latest FS
In-Reply-To: <361E98F99D3CC3439EED59BC1924ED6966467E@SERVER2003.SecuReachSystems.local>
References: <361E98F99D3CC3439EED59BC1924ED6966467E@SERVER2003.SecuReachSystems.local>
Message-ID: <CAKV-jFwN1ZTXR9bm6vHXhRVzPJ=t7RZbddVjwdG3opSC4TMPbQ@mail.gmail.com>

Too many #'s, hole is only 4 letters.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 8, 2012 4:16 PM, "Jason Moran" <jmoran at secureachsystems.com> wrote:

>  A spidermonkey File operation (often) causes the output of ?ASS #####?
> followed by a ?File operation pclose failed? error now that I?ve done a
> make current (last  update was 4 weeks ago).   The code puts a 0 or 1 into
> the variable fdFoundFile for later testing. This has worked for months
> until the make current.  Now it fails with the pclose error 4/5 times.****
>
> ** **
>
> CODE:****
>
>     console_log("info","-3-\n");****
>
> //Do a check to make sure we can read the file ****
>
> var fd = new File("|[ -f "+myFileName+" ] && echo 1 || echo 0");****
>
>     console_log("info","-4-\n");****
>
> fd.open("read");****
>
>     console_log("info","-5-\n");****
>
> var fdFoundFile = fd.read(1);****
>
>     console_log("info","-6-\n");****
>
> fd.close();****
>
>     console_log("info","-7-\n");****
>
> ** **
>
> LOG:****
>
> 2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -3-****
>
> 2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -4-****
>
> 2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -5-****
>
> ASS 30937****
>
> 2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -6-****
>
> 2012-06-08 15:52:14.717917 [ERR] Testcall.js:258  Error: File operation
> pclose failed****
>
> ** **
>
> The # following ?ASS? always changes and keeps on increasing.****
>
> ** **
>
> Thanks,
> Jason M****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmoran at secureachsystems.com  Sat Jun  9 01:19:33 2012
From: jmoran at secureachsystems.com (Jason Moran)
Date: Fri, 8 Jun 2012 17:19:33 -0400
Subject: [Freeswitch-users] JS File Operation Error with latest FS
References: <361E98F99D3CC3439EED59BC1924ED6966467E@SERVER2003.SecuReachSystems.local>
Message-ID: <361E98F99D3CC3439EED59BC1924ED69664684@SERVER2003.SecuReachSystems.local>

This code achieves the same functionality without creating an exception:

var fd = new File("/tmp/blah");

if (fd.exists) {

  console_log((fd.name) + "exists\n");

} else {

  console_log((fd.name) + "does not exist\n");

}

 

However, the question remains: why did this pclose error crop up
recently?

-Jason

 

From: Jason Moran 
Sent: Friday, June 08, 2012 4:16 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] JS File Operation Error with latest FS

 

A spidermonkey File operation (often) causes the output of "ASS #####"
followed by a "File operation pclose failed" error now that I've done a
make current (last  update was 4 weeks ago).   The code puts a 0 or 1
into the variable fdFoundFile for later testing. This has worked for
months until the make current.  Now it fails with the pclose error 4/5
times.

 

CODE:

    console_log("info","-3-\n");

//Do a check to make sure we can read the file 

var fd = new File("|[ -f "+myFileName+" ] && echo 1 || echo 0");

    console_log("info","-4-\n");

fd.open("read");

    console_log("info","-5-\n");

var fdFoundFile = fd.read(1);

    console_log("info","-6-\n");

fd.close();

    console_log("info","-7-\n");

 

LOG:

2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -3-

2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -4-

2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -5-

ASS 30937

2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -6-

2012-06-08 15:52:14.717917 [ERR] Testcall.js:258  Error: File operation
pclose failed

 

The # following "ASS" always changes and keeps on increasing.

 

Thanks,
Jason M

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From Hector.Geraldino at ipsoft.com  Sat Jun  9 01:30:47 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Fri, 8 Jun 2012 17:30:47 -0400
Subject: [Freeswitch-users] Unable to use mod_java.
In-Reply-To: <CAP_69mtZuWq-22j-hiFBBuYzNKzqd46fR0MPjXgJYYd1kfjxnw@mail.gmail.com>
References: <CAP_69mtZuWq-22j-hiFBBuYzNKzqd46fR0MPjXgJYYd1kfjxnw@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD022EE810EB@NY1-EXMB-01.ip-soft.net>

Check the settings in the java.conf.xml file (conf/autoload_config). Make sure it's referencing a valid libjvm.so  file/path.

Also, be sure to have the freeswitch.jar jar file in the freeswitch/scripts folder. Check also the port you're using for debugging (8000): is this port being used by another app?

Good luck!

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Venkatesh
Sent: Thursday, June 07, 2012 8:58 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Unable to use mod_java.

Hi !

I have been trying to enable mod_java with FS but am having problems. I was able to compile mod_java fine. However FS is having trouble loading the module. It errors out with the following error message:

2012-06-07 20:51:24.058126 [ERR] modjava.c:326 Error creating Java VM!,

I believe I have the CLASSPATH/PATH variables set up correctly in my configuration files. In fact I put in some debug statements in modjava.c. Below is the debug output showing the various variables:

2012-06-07 20:51:24.058126 [NOTICE] modjava.c:349 Java Framework Loading...
2012-06-07 20:51:24.058126 [ERR] modjava.c:299 VENKY optionCount: 7
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Djava.class.path=/usr/local/freeswitch/scripts
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Djava.library.path=/usr/java/jdk1.6.0_32/jre/lib:/usr/java/jdk1.6.0_32/lib
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -verbose:jni
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Xms=64
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Xmx=256
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=127.0.0.1:8000<http://127.0.0.1:8000>
2012-06-07 20:51:24.058126 [ERR] modjava.c:302 VENKY option: -Djava.library.path=/usr/local/freeswitch/mod
2012-06-07 20:51:24.058126 [ERR] modjava.c:326 Error creating Java VM!, error: -1
2012-06-07 20:51:24.058126 [INFO] switch_time.c:1123 Timezone reloaded 530 definitions
2012-06-07 20:51:24.058126 [CRIT] switch_loadable_module.c:1300 Error Loading module /usr/local/freeswitch/mod/mod_java.so
**Module load routine returned an error**


Would really appreciate any ideas from others that may have tried enabling the same?

Venkatesh
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From bdfoster at endigotech.com  Sat Jun  9 02:52:34 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 8 Jun 2012 18:52:34 -0400
Subject: [Freeswitch-users] Time needed for ringing a call group
In-Reply-To: <20120608223010.2d079108@anubis.defcon1>
References: <D3BC5C093B2E4150AF352558D6E077BE@KITPC003>
	<-6559196381009927779@unknownmsgid>
	<20120608223010.2d079108@anubis.defcon1>
Message-ID: <CAKV-jFywjjsMSwbVRLeDB5Uba19J-LnWxE3jmTjGHHyup4hoSg@mail.gmail.com>

It's worse, he has an iPad.

Brian Foster
Endigo Computer LLC

Sent from my Android phone (the mother of all upgrades)
On Jun 8, 2012 4:29 PM, "Bzzz" <lazyvirus at gmx.com> wrote:

> On Fri, 8 Jun 2012 14:39:26 -0500
> Brian West <brian at freeswitch.org> wrote:
>
> > do ?o? really mean 2000 second? h?r??
>     ^ ^                        ^  ^ ^
> > Sent from my biplane at 450ft (beach hut 2.0 upgrade)
>
> Hmm, either you're kidding or you have a keyboard|board problem
> Brian; you're answer to Carlo also contains strange characters.
>
> JY
>
> Sent from my bitch phone (B-bra update)
> --
> You will be dead within a year.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lazyvirus at gmx.com  Sat Jun  9 03:10:48 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Sat, 9 Jun 2012 01:10:48 +0200
Subject: [Freeswitch-users] Time needed for ringing a call group
In-Reply-To: <CAKV-jFywjjsMSwbVRLeDB5Uba19J-LnWxE3jmTjGHHyup4hoSg@mail.gmail.com>
References: <D3BC5C093B2E4150AF352558D6E077BE@KITPC003>
	<-6559196381009927779@unknownmsgid>
	<20120608223010.2d079108@anubis.defcon1>
	<CAKV-jFywjjsMSwbVRLeDB5Uba19J-LnWxE3jmTjGHHyup4hoSg@mail.gmail.com>
Message-ID: <20120609011048.39b24021@anubis.defcon1>

On Fri, 8 Jun 2012 18:52:34 -0400
Brian Foster <bdfoster at endigotech.com> wrote:

> It's worse, he has an iPad.
 
OMFG, does he know that he risks an 
eternal 406 damnation from the SIP god?

-- 
* Omnic looks at his 33.6k link and then looks at Joy
* Mercury cuddles his cable modem.. (=:]


From victortho at gmail.com  Sat Jun  9 03:11:47 2012
From: victortho at gmail.com (Victor)
Date: Fri, 8 Jun 2012 16:11:47 -0700
Subject: [Freeswitch-users] Time needed for ringing a call group
In-Reply-To: <D3BC5C093B2E4150AF352558D6E077BE@KITPC003>
References: <D3BC5C093B2E4150AF352558D6E077BE@KITPC003>
Message-ID: <CAG0Fp+8Tz7QKyLUhjH7KAa4QsjkVjg6FT-GeeE6q1N+6wp-H2w@mail.gmail.com>

Your call_timeout is set to 2000 seconds. Change it to
<action application="set" data="call_timeout=20"/>

--Victor


On Fri, Jun 8, 2012 at 11:25 AM, Robert Longfield
<robert.longfield at klinsight.com> wrote:
> I have support call group that works as it should. When a caller selects the
> support call group it rings all the support extensions as it should. The
> problem arises when a support member is logged in to their extension but is
> not at their computer. The call will continue to ring the call group
> forever. I would like to change this so it will only ring for a set amount
> of time.
>
> Currently if the support group is busy or they manually decline the call the
> incoming call will be transferred to the general extension.
> I would like to mimic this behaviour when the call goes unanswered after 20
> secs or so.
>
> Here is the Call Group Settings:
>
> ??? <extension name="group_dial_support">
> ????? <condition field="destination_number" expression="^2001$">
> ??????? <action application="set" data="call_timeout=2000"/>
> ??????? <action application="set" data="hangup_after_bridge=true"/>
> ??????? <action application="set" data="continue_on_fail=true"/>
> ??????? <action application="bridge" data="group/support@$${domain}"/>
> ??????? <action application="answer"/>
> ??????? <action application="sleep" data="100"/>
> ??????? <action application="bridge" data="sofia/internal/1000@$${domain}"/>
> ????? </condition>
> ??? </extension>
>
> I hope I?ve explained my problem clearly.
>
> -Rob
>
> Robert Longfield
> SEO Specialist
> KL Insight
> http://klinsight.com/
> Tel: 613-344-2116 | Fax: 613.634.7029
> 993 Princess Street, Suite 212
> Kingston, ON K7L 1H3, Canada
>
>
> Notice of Confidentiality:
> The information transmitted is intended only for the person or entity to
> which it is addressed and may contain confidential and/or privileged
> material. Any review re-transmission dissemination or other use of or taking
> of any action in reliance upon this information by persons or entities other
> than the intended recipient is prohibited. If you received this in error
> please contact the sender immediately by return electronic transmission and
> then immediately delete this transmission including all attachments without
> copying distributing or disclosing same.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From stkn at freeswitch.org  Sat Jun  9 04:29:54 2012
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Sat, 09 Jun 2012 02:29:54 +0200
Subject: [Freeswitch-users] freeswitch and mISDN
In-Reply-To: <4FD13E02.4040202@googlemail.com>
References: <4FD13E02.4040202@googlemail.com>
Message-ID: <4FD29902.3080702@freeswitch.org>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 06/08/12 01:49, Tamer Higazi wrote:
> How do I configure mISDN in freetdm, and what are the channel drivers, with it's dial-plan respectively.


The initial commit of ftmod_misdn contains the freetdm.conf/mISDN-specific setup instructions:

   http://git.freeswitch.org/git/freeswitch/commit/?id=09a61f5025b1c7dee05259db0e169f39a565b62d

   (Complete history of the module: http://git.freeswitch.org/git/freeswitch/log/libs/freetdm/src/ftmod/ftmod_misdn/ftmod_misdn.c )

This covers only the I/O part (ftmod_misdn, like ftmod_zap or ftmod_wanpipe, is an I/O module),
for the signalling part you'll have to (build, install and) configure ftmod_libpri in freetdm.conf.xml:

   http://stkn.techmage.de/archives/191

is (very) a short summary (ignore the dahdi/zaptel specific configuration parts).


ftmod_libpri span parameters:

      context  -  FreeSWITCH dialplan context to send incoming calls to

      dialplan -  FreeSWITCH dialplan (e.g. "XML")


      mode    -  "user" or "net", the latter being unsupported with ftmod_misdn,
                 due to restrictions caused by the way FreeTDM handles configuration of I/O modules.

      dialect -  Q.931 protocol dialect used for signalling: "euroisdn"(/"q931"),
                 (other possible values: "ni1", "ni2", "dms100", "5ess", "4ess", "gr303eoc", "gr303tmc")

      opts    -  Comma separated list of flags:

                     "suggest_channel"         : Not sure what this one does, ignore this one.

                     "omit_display"            : Disables sending callerid name, some switches don't like this.

                     "omit_redirecting_number" : The same for REDIRECTING NUMBER information elements.

                     "aoc"                     : Enables some Advice of Charge bits in ftmod_libpri, supposed to
                                                 handle incoming AOC messages and output some details at NOTICE loglevel
                                                 (experimental).

      ton     -  Default Type-of-Number for outgoing calls: "international", "national", "local", "private", "unknown" (default).
                 Better not touch this unless you really have to.

      layer1  -  B-Channel Layer1 protocol, "alaw" or "ulaw"   NOTE: ftmod_libpri will auto-select the right one
                 based on the the chosen dialect (a-law for EuroISDN/Q.931, u-law for everything else).

      overlapdial - "yes" (or "both"), "receive", "send", "no"

      debug   -  Comma separated list of debug flags for libpri:

                     "q921_state" + "q921_dump" + "q921_raw"     = "q921_all" : Enables Q.921 logging (State changes, decoded messages, raw messages)
                     "q931_state" + "q931_dump" + "q931_anomaly" = "q931_all" : Enables Q.931 logging (State changes, decoded messages, message anomalies)

                     "aoc"    : AOC debug messages
                     "apdu"   : Q.932 / ROSE debug messages(?)
                     "config" : libpri configuration debug messages

                     "all"  : Enable all of the above
                     "none" : Disables all


      service_message_support   - Enables support for maintenance / restart / service messages  (not EuroISDN/Q,931)


      And there are a couple of new features (MSN filter, Numer prefix/ToN autoselection) for ftmod_libpri sitting in my post-1.2.0 queue.


Dialplan:  Incoming calls should be sent to their own ISDN-specific dialplan context, makes handling them easier
           (like the public dialplan for the external SIP profile).


I'll stop here now, it's past 2 AM and i can't concentrate anymore.

stkn
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From gabe at gundy.org  Sat Jun  9 05:16:22 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 8 Jun 2012 19:16:22 -0600
Subject: [Freeswitch-users] using APNS with diaplan : throttle a call
 until a certain condition changes
In-Reply-To: <CACU5aF6ma-W+VM98FzhS6i=kWHQhJPehpvPt1oS9E8662CaC2A@mail.gmail.com>
References: <CACU5aF6ma-W+VM98FzhS6i=kWHQhJPehpvPt1oS9E8662CaC2A@mail.gmail.com>
Message-ID: <CAMwZ-tLtXYEWbxo+R9ooudh+PZ-F_bnpPrtawZLMFjK13Z2ceQ@mail.gmail.com>

On Fri, Jun 8, 2012 at 2:31 AM, John Gathm <john.gathm at gmail.com> wrote:
> I am wondering if this implementation would be acceptable regarding
> freeswitch design, or if diaplan/freeswitch offer cleaner ways to implement
> it.

What if you used your dialplan to try to call the phone and if you do
NOT find it, park the call. Then you might use the event socket to
watch for the parking of the call so you can send your notification.
Next, use the event socket to watch for sofia registrations to know
when the app has received the notification, launched and is ready to
take calls. Once you get that event, unpark the call and send it on
its way.


Best,
Gabe

P.S. go python!


From gabe at gundy.org  Sat Jun  9 05:24:16 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 8 Jun 2012 19:24:16 -0600
Subject: [Freeswitch-users] Multilingual IVR in English and Thai
In-Reply-To: <BAY162-W57F9F3F7949FEC7FF9AAD1F4F20@phx.gbl>
References: <BAY162-W57F9F3F7949FEC7FF9AAD1F4F20@phx.gbl>
Message-ID: <CAMwZ-t+46E8jcswhPRgNa8WLd8vfMmT7jDhp3qoM-8khmPsNXQ@mail.gmail.com>

On Fri, Jun 8, 2012 at 1:32 AM, Abid Saleem <abid_freeswitch at live.com> wrote:
> Does Freeswitch support Multilingual IVR for example English and Thai at the
> same time. The customer should be given the option to chose either English
> or Thai. If yes, can you please specify how can it be configured?

FreeSWITCH supports IVRs and Speech Phrase Management. It's your job
to put them together :)

http://wiki.freeswitch.org/wiki/Speech_Phrase_Management#Features
(Multilingual Support)

http://wiki.freeswitch.org/wiki/IVR_Menu#Options (see the part where
you specify "phrase:my_phrase")

Good luck and happy hacking!


Gabe

P.S. I hope this is an app that let's me order yummy Thai food :)


From gabe at gundy.org  Sat Jun  9 06:08:28 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 8 Jun 2012 20:08:28 -0600
Subject: [Freeswitch-users]
	=?windows-1252?q?Freeswitch_=93real=94_respons?=
	=?windows-1252?q?e_to_api_chat=3F_Having_a_problem_determining_res?=
	=?windows-1252?q?ult_via_esl=2E?=
In-Reply-To: <5ABA6F8E7A50AB4B84651BF8EF5B557457DA0F@MAILBOX.millicorp.com>
References: <5ABA6F8E7A50AB4B84651BF8EF5B557457DA0F@MAILBOX.millicorp.com>
Message-ID: <CAMwZ-tLMRUqQ9CpJoArsgorT+CB2EOzWi2Dkw8=4J2S6e76q1Q@mail.gmail.com>

On Wed, Jun 6, 2012 at 8:26 AM, Bill Ryder <Bill.Ryder at millicorp.com> wrote:
> ...however over the esl socket I get the same response event no matter what,
> just "sent". Also, when I bind to the esl port and attempt to spit out every
> message possible (no filters), we see most of what comes through a sip
> trace, but again this bad result sip packet seems not to generate an esl
> event.
>
> Is there a way over esl to probe raw sip messages so that we can test for
> such a packet manually, or some other way of seeking the final async result?

This sounds like a bug/feature request. I'd recommend opening a ticket
over at http://jira.freeswitch.org

Gabe


From gabe at gundy.org  Sat Jun  9 06:14:22 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 8 Jun 2012 20:14:22 -0600
Subject: [Freeswitch-users] Freeswitch send 423 (Interval Too Brief)
In-Reply-To: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57E7E@CGNMSG01.oc.qsc.de>
References: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57E7E@CGNMSG01.oc.qsc.de>
Message-ID: <CAMwZ-t+eaXUktbM1bsh1gO_r0qNCj-xfaG1Yu3eLO=hR-d5e8A@mail.gmail.com>

On Tue, Jun 5, 2012 at 8:51 AM, Lappe, Adam <Adam.Lappe at qsc.de> wrote:
> is there a way to make freeswitch send a 423 (Interval Too Brief) during a
> SIP REGISTER request?
>
> I want to prevent clients to (re)register with a low expire-time and thus
> spam our network.

Have you greped the code for it? Seems like it's be easy to spot. If
not, open a ticket. Or, better yet, submit a patch :)


> The ?sip-force-expires?-variable would work but don?t seem to be
> SIP-compliant to me.

Have you followed the SIP traces to see exactly how this works? I
haven't, but perhaps it really does work how you'd hope (I don't know,
I've never used it).


Best,
Gabe


From gabe at gundy.org  Sat Jun  9 06:18:51 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 8 Jun 2012 20:18:51 -0600
Subject: [Freeswitch-users] Missing event for "180 Ringing"?
In-Reply-To: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
Message-ID: <CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>

On Tue, Jun 5, 2012 at 7:17 AM, Dennis <odermann at googlemail.com> wrote:
> for us it seems as if there is an event missing for 180 Ringing. Is
> there a way to receive/activate this event?
>
> We are using ESL and are getting nearly all events we need. For "183
> Session Progress SDP" we get "channel_progress_media" and for "200 OK
> SDP" we get "channel_answer". So we only need one more event...

How are you registering for events? I'm assuming you're registering
for all of them.

There *might* be a reason that it doesn't fire an event for ringing. I
can't think of one, but that doesn't mean anything :)

I'd check out the code and make sure it's not there and then open a
ticket. http://jira.freeswitch.org/


Good luck,
Gabe


From gabe at gundy.org  Sat Jun  9 06:22:48 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 8 Jun 2012 20:22:48 -0600
Subject: [Freeswitch-users] Capture media codes?
In-Reply-To: <4FCC4942.3000605@the800group.com>
References: <4FCC4942.3000605@the800group.com>
Message-ID: <CAMwZ-tLdAqwVhEwtnEB3nZhTuedwtF7=Vb+9-n2vGjekpXJu1A@mail.gmail.com>

On Sun, Jun 3, 2012 at 11:36 PM, ocset <ocset at the800group.com> wrote:
> Is there a way to capture and display the media codes being sent from a
> provider.

Anyone of these should work for you:

http://wiki.freeswitch.org/wiki/Packet_Capture

Best,
Gabe


From gabe at gundy.org  Sat Jun  9 06:31:03 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 8 Jun 2012 20:31:03 -0600
Subject: [Freeswitch-users] mod_directory and LUA for database users
In-Reply-To: <96004A14-ACA2-430E-AD9F-702ACEFFD554@thinksimplicity.net>
References: <B2C68DE4-E421-4900-BAB7-1767E5D0DE88@thinksimplicity.net>
	<96004A14-ACA2-430E-AD9F-702ACEFFD554@thinksimplicity.net>
Message-ID: <CAMwZ-tLLiSY==wTJAatDMPX+mvHMfVO9-XVNU+QA2poDh1E_UQ@mail.gmail.com>

On Fri, Jun 1, 2012 at 12:42 PM, Perovich Alexander
<alexanderjp at thinksimplicity.net> wrote:
> I have been trying to get the mod_directory to query the users that are
> stored in a database and accessed using a lua script, however every attempt
> comes back match cannot be found.

When you say users are stored in a DB, do you mean that you're
supplying the XML in the FreeSWITCH user directory section? And are
you're using Lua to generate that XML?

If so, maybe you're seeing something similar to this:
http://wiki.freeswitch.org/wiki/Mod_directory#Interaction_with_xml_curl

If not, we'll need more info to help you out.

Best,
Gabe


From gabe at gundy.org  Sat Jun  9 07:27:16 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 8 Jun 2012 21:27:16 -0600
Subject: [Freeswitch-users] State of GUIs
In-Reply-To: <CAMbW9o+eOmWmNbz0rtsvQCu3TiQ+R38_SHxeoBjZWA7j3q8hhA@mail.gmail.com>
References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net>
	<CAMbW9oJ6qBUkWrB=W9O+wGbLERMEWX-ujnXnpw6cVjP9-qxVMQ@mail.gmail.com>
	<03f301ccfd03$d7777080$86665180$@launch3.net>
	<CAKbxfG835-ROigp-7RtCeVH-7UtVLtTeQgo7GAF5Zo+gDnx+Eg@mail.gmail.com>
	<CAAYf2LvoFzeBrbgA=_WaOCSWxqGOXceBx8S_vxMnx7zbpCCVrQ@mail.gmail.com>
	<CAC-LwPPi4EmN0YxRnOvQAE9akpr6=4rnmH0apM82JFw-nDyUfg@mail.gmail.com>
	<CAMbW9o+eOmWmNbz0rtsvQCu3TiQ+R38_SHxeoBjZWA7j3q8hhA@mail.gmail.com>
Message-ID: <CAMwZ-tK884W-t16ChmLawFA61jiKbg3NDkKT5-u-V+k1WuKg5g@mail.gmail.com>

On Fri, Mar 9, 2012 at 9:04 AM, Andrew Cassidy
<andrew at cassidywebservices.co.uk> wrote:
> As with a number of other people, I'm writing one to suit my needs too. Mine
> is being written in Python/Django and is aimed at being multi tenant in a
> cloud setting or single user on a manged device setting sharing as much of
> the code from both scenarios as possible.

Hey Andrew, how did this project go?


Gabe


From gabe at gundy.org  Sat Jun  9 08:08:40 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 8 Jun 2012 22:08:40 -0600
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <50037FB5-DE1A-485C-980D-8F6ECAC3928E@freeswitch.org>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
	<4BFC4A64.5090208@gmail.com>
	<AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>
	<D311BDCB06324D0DB77898FC6FF4B0EB@dell9400>
	<AANLkTilAMe1lRKK4R6Nmr29FzfI-pP3wRa7Foabmd8Dp@mail.gmail.com>
	<4BFD4562.5030804@gmail.com>
	<AANLkTikGQlZ4N3zfYv18COvbubEi-8LSOZxnAy1KC5sP@mail.gmail.com>
	<4BFEE96A.2010003@gmail.com>
	<AANLkTikSgh4xN9kb3Lha5dNlacgTA9DqrkHqU3qCWZUT@mail.gmail.com>
	<D110995190DF45D78114CB61426F9946@dell9400>
	<AANLkTillSAX6L4gum_g9SNjP6O8hGEtXTfcIakZT96_D@mail.gmail.com>
	<4C057EB7.3090005@gmail.com>
	<AANLkTima1WBHX_2n0fN9qSXlYNFmbMckIDBlO_Ga7RTo@mail.gmail.com>
	<50037FB5-DE1A-485C-980D-8F6ECAC3928E@freeswitch.org>
Message-ID: <CAMwZ-t+t0Z1pus4oiDU0hs5jNF0YPK-VjWcFAwhm8_GSbMu9vg@mail.gmail.com>

On Tue, Jun 1, 2010 at 4:48 PM, Brian West <brian at freeswitch.org> wrote:
> Moving forward can we please highlight the area of the email we wish to reply to and then click reply, ?include the relevant portions of the email we are replying to... I see no need to have the past 40 emails in the reply on the thread. ?It wastes time, bandwidth and disk space and is harder to follow when trying to see if its something I need to pay attention to.

This is old, but I have to ask, can I get an AMEN?!


Gabe


From saami_mh at ymail.com  Sat Jun  9 11:48:36 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Sat, 9 Jun 2012 00:48:36 -0700 (PDT)
Subject: [Freeswitch-users] how to play file using lua script within
	dialplan without need to answer session?
Message-ID: <1339228116.47309.YahooMailNeo@web120102.mail.ne1.yahoo.com>

hi guys;
i am going to playback .wav file using lua scripts without neet to answer seeeion;
how can i do that?
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From saami_mh at ymail.com  Sat Jun  9 12:41:36 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Sat, 9 Jun 2012 01:41:36 -0700 (PDT)
Subject: [Freeswitch-users] how to disable sending early media ?
Message-ID: <1339231296.22293.YahooMailNeo@web120102.mail.ne1.yahoo.com>

<action application="set" data="ignore_early_media=true"/>i have set the above channel variable but?when i am going to playback .wav file using lua scripts without issue the  seeeion:answer ; i got error :sending early media 
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From peter.olsson at visionutveckling.se  Sat Jun  9 12:58:52 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 9 Jun 2012 08:58:52 +0000
Subject: [Freeswitch-users] how to play file using lua script
	within	dialplan without need to answer session?
In-Reply-To: <1339228116.47309.YahooMailNeo@web120102.mail.ne1.yahoo.com>
References: <1339228116.47309.YahooMailNeo@web120102.mail.ne1.yahoo.com>
Message-ID: <FD021270-99F9-4AF3-A8F0-E5E258F491E3@visionutveckling.se>

It should be enough to just start playing, FS will try to enable early media if you don't have media already.

/Peter

9 jun 2012 kl. 09:56 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com>>:

hi guys;
i am going to playback .wav file using lua scripts without neet to answer seeeion;
how can i do that?
!DSPAM:4fd2fe6b32761835411859!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
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!DSPAM:4fd2fe6b32761835411859!


From peter.olsson at visionutveckling.se  Sat Jun  9 13:00:14 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 9 Jun 2012 09:00:14 +0000
Subject: [Freeswitch-users] how to disable sending early media ?
In-Reply-To: <1339231296.22293.YahooMailNeo@web120102.mail.ne1.yahoo.com>
References: <1339231296.22293.YahooMailNeo@web120102.mail.ne1.yahoo.com>
Message-ID: <1A866A23-F1AF-49A0-8DFB-7F9C8D5C9B2D@visionutveckling.se>

I guess you need to decide if you want early media or not... Playback before answer needs early media.


/Peter

9 jun 2012 kl. 10:47 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com>>:


<action application="set" data="ignore_early_media=true"/>

i have set the above channel variable but when i am going to playback .wav file using lua scripts without issue the seeeion:answer ; i got error :sending early media

!DSPAM:4fd30a8732761348221455!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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!DSPAM:4fd30a8732761348221455!


From saami_mh at ymail.com  Sat Jun  9 13:13:40 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Sat, 9 Jun 2012 02:13:40 -0700 (PDT)
Subject: [Freeswitch-users] how to disable sending early media ?
In-Reply-To: <1A866A23-F1AF-49A0-8DFB-7F9C8D5C9B2D@visionutveckling.se>
References: <1339231296.22293.YahooMailNeo@web120102.mail.ne1.yahoo.com>
	<1A866A23-F1AF-49A0-8DFB-7F9C8D5C9B2D@visionutveckling.se>
Message-ID: <1339233220.88823.YahooMailNeo@web120101.mail.ne1.yahoo.com>

hi thanks alot for your reply;
i don't want to use command "session:answer"?
and want to only play .wav files in lua?
i don't want get sending early media ?


________________________________
 From: Peter Olsson <peter.olsson at visionutveckling.se>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Saturday, June 9, 2012 1:30 PM
Subject: Re: [Freeswitch-users] how to disable sending early media ?
 
I guess you need to decide if you want early media or not... Playback before answer needs early media.


/Peter

9 jun 2012 kl. 10:47 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com>>:


<action application="set" data="ignore_early_media=true"/>

i have set the above channel variable but when i am going to playback .wav file using lua scripts without issue the seeeion:answer ; i got error :sending early media

!DSPAM:4fd30a8732761348221455!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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!DSPAM:4fd30a8732761348221455!

_________________________________________________________________________
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consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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Join Us At ClueCon - Aug 7-9, 2012

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From monemran at gmail.com  Sat Jun  9 13:40:39 2012
From: monemran at gmail.com (Mohammad Emran)
Date: Sat, 9 Jun 2012 15:40:39 +0600
Subject: [Freeswitch-users] how to disable sending early media ?
In-Reply-To: <1339233220.88823.YahooMailNeo@web120101.mail.ne1.yahoo.com>
References: <1339231296.22293.YahooMailNeo@web120102.mail.ne1.yahoo.com>
	<1A866A23-F1AF-49A0-8DFB-7F9C8D5C9B2D@visionutveckling.se>
	<1339233220.88823.YahooMailNeo@web120101.mail.ne1.yahoo.com>
Message-ID: <6DC1D7D0-2D51-41E1-84CD-B32C445C81F5@gmail.com>

Use pre_answer instead of answer.


Sent from my iPhone

On Jun 9, 2012, at 3:13 PM, Samira Mh <saami_mh at ymail.com> wrote:

> hi thanks alot for your reply;
> i don't want to use command "session:answer" 
> and want to only play .wav files in lua 
> i don't want get sending early media  
> 
> From: Peter Olsson <peter.olsson at visionutveckling.se>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
> Sent: Saturday, June 9, 2012 1:30 PM
> Subject: Re: [Freeswitch-users] how to disable sending early media ?
> 
> I guess you need to decide if you want early media or not... Playback before answer needs early media.
> 
> 
> /Peter
> 
> 9 jun 2012 kl. 10:47 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com>>:
> 
> 
> <action application="set" data="ignore_early_media=true"/>
> 
> i have set the above channel variable but when i am going to playback .wav file using lua scripts without issue the seeeion:answer ; i got error :sending early media
> 
> !DSPAM:4fd30a8732761348221455!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> 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
> 
> 
> !DSPAM:4fd30a8732761348221455!
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From saami_mh at ymail.com  Sat Jun  9 13:49:18 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Sat, 9 Jun 2012 02:49:18 -0700 (PDT)
Subject: [Freeswitch-users] how to disable sending early media ?
In-Reply-To: <6DC1D7D0-2D51-41E1-84CD-B32C445C81F5@gmail.com>
References: <1339231296.22293.YahooMailNeo@web120102.mail.ne1.yahoo.com>
	<1A866A23-F1AF-49A0-8DFB-7F9C8D5C9B2D@visionutveckling.se>
	<1339233220.88823.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<6DC1D7D0-2D51-41E1-84CD-B32C445C81F5@gmail.com>
Message-ID: <1339235358.94030.YahooMailNeo@web120106.mail.ne1.yahoo.com>

hi,?
thanks alot for your reply;
but i suue the following command in the lua script
session:preAnswer();?

?and again got the below error and .wav file coudn't play?
switch_cpp.cpp:617 Sending early media




________________________________
 From: Mohammad Emran <monemran at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Saturday, June 9, 2012 2:10 PM
Subject: Re: [Freeswitch-users] how to disable sending early media ?
 

Use pre_answer instead of answer.


Sent from my iPhone

On Jun 9, 2012, at 3:13 PM, Samira Mh <saami_mh at ymail.com> wrote:


hi thanks alot for your reply;
>i don't want to use command "session:answer"?
>and want to only play .wav files in lua?
>i don't want get sending early media ?
>
>
>
>________________________________
> From: Peter Olsson <peter.olsson at visionutveckling.se>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Saturday, June 9, 2012 1:30 PM
>Subject: Re: [Freeswitch-users] how to disable sending early media ?
> 
>I guess you need to decide if you want early media or not... Playback before answer needs early media.
>
>
>/Peter
>
>9 jun 2012 kl. 10:47 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com>>:
>
>
><action application="set" data="ignore_early_media=true"/>
>
>i have set the above channel variable but when i am going to playback .wav file using lua scripts without issue the seeeion:answer ; i got error :sending early media
>
>!DSPAM:4fd30a8732761348221455!
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>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
>
>
>!DSPAM:4fd30a8732761348221455!
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users
 mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing 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 monemran at gmail.com  Sat Jun  9 13:59:57 2012
From: monemran at gmail.com (M.Emran)
Date: Sat, 9 Jun 2012 15:59:57 +0600
Subject: [Freeswitch-users] how to disable sending early media ?
In-Reply-To: <1339235358.94030.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1339231296.22293.YahooMailNeo@web120102.mail.ne1.yahoo.com>
	<1A866A23-F1AF-49A0-8DFB-7F9C8D5C9B2D@visionutveckling.se>
	<1339233220.88823.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<6DC1D7D0-2D51-41E1-84CD-B32C445C81F5@gmail.com>
	<1339235358.94030.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <CAMbKPxYuD8Uyt_9Y4pTG2i7Wg0R-1JkEyWWfs5rxX5Uj8h67gg@mail.gmail.com>

it seems ur wav file has an issue.it must be played.


On Sat, Jun 9, 2012 at 3:49 PM, Samira Mh <saami_mh at ymail.com> wrote:

> hi,
> thanks alot for your reply;
> but i suue the following command in the lua script
> session:preAnswer();
>  and again got the below error and .wav file coudn't play
> switch_cpp.cpp:617 Sending early media
>
>   ------------------------------
> *From:* Mohammad Emran <monemran at gmail.com>
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Cc:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Saturday, June 9, 2012 2:10 PM
>
> *Subject:* Re: [Freeswitch-users] how to disable sending early media ?
>
> Use pre_answer instead of answer.
>
>
> Sent from my iPhone
>
> On Jun 9, 2012, at 3:13 PM, Samira Mh <saami_mh at ymail.com> wrote:
>
> hi thanks alot for your reply;
> i don't want to use command "session:answer"
> and want to only play .wav files in lua
> i don't want get sending early media
>
>   ------------------------------
> *From:* Peter Olsson <peter.olsson at visionutveckling.se>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Saturday, June 9, 2012 1:30 PM
> *Subject:* Re: [Freeswitch-users] how to disable sending early media ?
>
> I guess you need to decide if you want early media or not... Playback
> before answer needs early media.
>
>
> /Peter
>
> 9 jun 2012 kl. 10:47 skrev "Samira Mh" <saami_mh at ymail.com<mailto:
> saami_mh at ymail.com>>:
>
>
> <action application="set" data="ignore_early_media=true"/>
>
> i have set the above channel variable but when i am going to playback .wav
> file using lua scripts without issue the seeeion:answer ; i got error
> :sending early media
>
> !DSPAM:4fd30a8732761348221455!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>
> !DSPAM:4fd30a8732761348221455!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards
----------
M Emran
Chief Executive Officer
E-SOFT BILLING (pvt). LTD.

HB Tower(3rd Floor)
House No # 1/A
Road No # 23
Gulshan # 1
Dhaka-1212, Bangladesh.
*Phone:* +880-2-8822312,+880-2-8822384
Fax : +880-2-8822254
E-Mail: info at e-softbilling.com
Web: www.e-softbilling.com
         www.isoftswitch.com
         www.howtonix.com
         www.sipmobiledialer.com
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From saami_mh at ymail.com  Sat Jun  9 14:17:54 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Sat, 9 Jun 2012 03:17:54 -0700 (PDT)
Subject: [Freeswitch-users] how to disable sending early media ?
In-Reply-To: <CAMbKPxYuD8Uyt_9Y4pTG2i7Wg0R-1JkEyWWfs5rxX5Uj8h67gg@mail.gmail.com>
References: <1339231296.22293.YahooMailNeo@web120102.mail.ne1.yahoo.com>
	<1A866A23-F1AF-49A0-8DFB-7F9C8D5C9B2D@visionutveckling.se>
	<1339233220.88823.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<6DC1D7D0-2D51-41E1-84CD-B32C445C81F5@gmail.com>
	<1339235358.94030.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<CAMbKPxYuD8Uyt_9Y4pTG2i7Wg0R-1JkEyWWfs5rxX5Uj8h67gg@mail.gmail.com>
Message-ID: <1339237074.24777.YahooMailNeo@web120106.mail.ne1.yahoo.com>

whaaaaat?
i have checked ?with alot of files but it coudn't work


________________________________
 From: M.Emran <monemran at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Saturday, June 9, 2012 2:29 PM
Subject: Re: [Freeswitch-users] how to disable sending early media ?
 

it seems ur wav file has an issue.it must be played.



On Sat, Jun 9, 2012 at 3:49 PM, Samira Mh <saami_mh at ymail.com> wrote:

hi,?
>thanks alot for your reply;
>but i suue the following command in the lua script
>session:preAnswer();?
>
>?and again got the below error and .wav file coudn't play?
>switch_cpp.cpp:617 Sending early media
>
>
>
>
>________________________________
> From: Mohammad Emran <monemran at gmail.com>
>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Saturday, June 9, 2012 2:10 PM
>
>Subject: Re: [Freeswitch-users] how to disable sending early media ?
> 
>
>
>Use pre_answer instead of answer.
>
>
>Sent from my iPhone
>
>On Jun 9, 2012, at 3:13 PM, Samira Mh <saami_mh at ymail.com> wrote:
>
>
>hi thanks alot for your reply;
>>i don't want to use command "session:answer"?
>>and want to only play .wav files in lua?
>>i don't want get sending early media ?
>>
>>
>>
>>________________________________
>> From: Peter Olsson <peter.olsson at visionutveckling.se>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>>Sent: Saturday, June 9, 2012 1:30 PM
>>Subject: Re: [Freeswitch-users] how to disable sending early media ?
>> 
>>I guess you need to decide if you want early media or not... Playback before answer needs early media.
>>
>>
>>/Peter
>>
>>9 jun 2012 kl. 10:47 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com>>:
>>
>>
>><action application="set" data="ignore_early_media=true"/>
>>
>>i have set the above channel variable but when i am going to playback .wav file using lua scripts without issue the seeeion:answer ; i got error :sending early media
>>
>>!DSPAM:4fd30a8732761348221455!
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
>>consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
>>http://www.freeswitch.org
>>http://wiki.freeswitch.org
>>http://www.cluecon.com
>>
>>Join Us At ClueCon - Aug 7-9, 2012
>>
>>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
>>
>>
>>!DSPAM:4fd30a8732761348221455!
>>
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
>>consulting at freeswitch.org
>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
>>http://www.freeswitch.org
>>http://wiki.freeswitch.org
>>http://www.cluecon.com
>>
>>Join Us At ClueCon - Aug 7-9, 2012
>>
>>FreeSWITCH-users
 mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
>>consulting at freeswitch.org
>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
>>http://www.freeswitch.org
>>http://wiki.freeswitch.org
>>http://www.cluecon.com
>>
>>Join Us At ClueCon - Aug 7-9, 2012
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>FreeSWITCH-powered
 IP PBX: The CudaTel Communication Server
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
Regards 
---------- 
M Emran
Chief Executive Officer
E-SOFT BILLING (pvt). LTD.

HB Tower(3rd Floor)
House No # 1/A
Road No # 23
Gulshan # 1
Dhaka-1212, Bangladesh.
Phone: +880-2-8822312,+880-2-8822384
Fax :
+880-2-8822254
E-Mail: info at e-softbilling.com
Web: www.e-softbilling.com
???????? www.isoftswitch.com
???????? www.howtonix.com
???????? www.sipmobiledialer.com

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From andrew at cassidywebservices.co.uk  Sat Jun  9 14:43:06 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sat, 9 Jun 2012 11:43:06 +0100
Subject: [Freeswitch-users] State of GUIs
In-Reply-To: <CAMwZ-tK884W-t16ChmLawFA61jiKbg3NDkKT5-u-V+k1WuKg5g@mail.gmail.com>
References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net>
	<CAMbW9oJ6qBUkWrB=W9O+wGbLERMEWX-ujnXnpw6cVjP9-qxVMQ@mail.gmail.com>
	<03f301ccfd03$d7777080$86665180$@launch3.net>
	<CAKbxfG835-ROigp-7RtCeVH-7UtVLtTeQgo7GAF5Zo+gDnx+Eg@mail.gmail.com>
	<CAAYf2LvoFzeBrbgA=_WaOCSWxqGOXceBx8S_vxMnx7zbpCCVrQ@mail.gmail.com>
	<CAC-LwPPi4EmN0YxRnOvQAE9akpr6=4rnmH0apM82JFw-nDyUfg@mail.gmail.com>
	<CAMbW9o+eOmWmNbz0rtsvQCu3TiQ+R38_SHxeoBjZWA7j3q8hhA@mail.gmail.com>
	<CAMwZ-tK884W-t16ChmLawFA61jiKbg3NDkKT5-u-V+k1WuKg5g@mail.gmail.com>
Message-ID: <CAMbW9o+WVbQgzBdKYGDOw2fFj0mgoDWEmF_oqPJFRSyivb49yQ@mail.gmail.com>

Hi Gabe, It's still in the works but is going well. Since my last post I've
added Queues and Conference call support, although queues are going to need
a little work doing to force the freeswitch instances to load the
configuration of new ones.

Hoping to add faxing in shortly (time permitting, of course) and we'll see
where we go from there.

On 9 June 2012 04:27, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Fri, Mar 9, 2012 at 9:04 AM, Andrew Cassidy
> <andrew at cassidywebservices.co.uk> wrote:
> > As with a number of other people, I'm writing one to suit my needs too.
> Mine
> > is being written in Python/Django and is aimed at being multi tenant in a
> > cloud setting or single user on a manged device setting sharing as much
> of
> > the code from both scenarios as possible.
>
> Hey Andrew, how did this project go?
>
>
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
-------------- next part --------------
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From peter.olsson at visionutveckling.se  Sat Jun  9 15:10:07 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 9 Jun 2012 11:10:07 +0000
Subject: [Freeswitch-users] how to disable sending early media ?
In-Reply-To: <1339237074.24777.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1339231296.22293.YahooMailNeo@web120102.mail.ne1.yahoo.com>
	<1A866A23-F1AF-49A0-8DFB-7F9C8D5C9B2D@visionutveckling.se>
	<1339233220.88823.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<6DC1D7D0-2D51-41E1-84CD-B32C445C81F5@gmail.com>
	<1339235358.94030.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<CAMbKPxYuD8Uyt_9Y4pTG2i7Wg0R-1JkEyWWfs5rxX5Uj8h67gg@mail.gmail.com>,
	<1339237074.24777.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <1FFF97C269757C458224B7C895F35F15100283@cantor.std.visionutv.se>

Playing wav before answer IS early media.

So - no early media, no playback before answer.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Samira Mh [saami_mh at ymail.com]
Skickat: den 9 juni 2012 12:17
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] how to disable sending early media ?

whaaaaat?
i have checked  with alot of files but it coudn't work

________________________________
From: M.Emran <monemran at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, June 9, 2012 2:29 PM
Subject: Re: [Freeswitch-users] how to disable sending early media ?

it seems ur wav file has an issue.it<http://issue.it> must be played.


On Sat, Jun 9, 2012 at 3:49 PM, Samira Mh <saami_mh at ymail.com<mailto:saami_mh at ymail.com>> wrote:
hi,
thanks alot for your reply;
but i suue the following command in the lua script
session:preAnswer();
 and again got the below error and .wav file coudn't play
switch_cpp.cpp:617 Sending early media

________________________________
From: Mohammad Emran <monemran at gmail.com<mailto:monemran at gmail.com>>

To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Saturday, June 9, 2012 2:10 PM

Subject: Re: [Freeswitch-users] how to disable sending early media ?

Use pre_answer instead of answer.


Sent from my iPhone

On Jun 9, 2012, at 3:13 PM, Samira Mh <saami_mh at ymail.com<mailto:saami_mh at ymail.com>> wrote:

hi thanks alot for your reply;
i don't want to use command "session:answer"
and want to only play .wav files in lua
i don't want get sending early media

________________________________
From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Saturday, June 9, 2012 1:30 PM
Subject: Re: [Freeswitch-users] how to disable sending early media ?

I guess you need to decide if you want early media or not... Playback before answer needs early media.


/Peter

9 jun 2012 kl. 10:47 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com><mailto:saami_mh at ymail.com<mailto:saami_mh at ymail.com>>>:


<action application="set" data="ignore_early_media=true"/>

i have set the above channel variable but when i am going to playback .wav file using lua scripts without issue the seeeion:answer ; i got error :sending early media


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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>>
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


!DSPAM:4fd30a8732761348221455!

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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




--
Regards
----------
M Emran
Chief Executive Officer
E-SOFT BILLING (pvt). LTD.

HB Tower(3rd Floor)
House No # 1/A
Road No # 23
Gulshan # 1
Dhaka-1212, Bangladesh.
Phone: +880-2-8822312,+880-2-8822384
Fax : +880-2-8822254
E-Mail: info at e-softbilling.com<mailto:info at e-softbilling.com>
Web: www.e-softbilling.com<http://www.e-softbilling.com/>
         www.isoftswitch.com<http://www.isoftswitch.com/>
         www.howtonix.com<http://www.howtonix.com/>
         www.sipmobiledialer.com<http://www.sipmobiledialer.com/>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


!DSPAM:4fd321b532761506015947!


From saami_mh at ymail.com  Sat Jun  9 15:20:50 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Sat, 9 Jun 2012 04:20:50 -0700 (PDT)
Subject: [Freeswitch-users] how to disable sending early media ?
In-Reply-To: <1FFF97C269757C458224B7C895F35F15100283@cantor.std.visionutv.se>
References: <1339231296.22293.YahooMailNeo@web120102.mail.ne1.yahoo.com>
	<1A866A23-F1AF-49A0-8DFB-7F9C8D5C9B2D@visionutveckling.se>
	<1339233220.88823.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<6DC1D7D0-2D51-41E1-84CD-B32C445C81F5@gmail.com>
	<1339235358.94030.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<CAMbKPxYuD8Uyt_9Y4pTG2i7Wg0R-1JkEyWWfs5rxX5Uj8h67gg@mail.gmail.com>,
	<1339237074.24777.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<1FFF97C269757C458224B7C895F35F15100283@cantor.std.visionutv.se>
Message-ID: <1339240850.28527.YahooMailNeo@web120101.mail.ne1.yahoo.com>

but is there ?any way to "no early media " but??playback before answer?


________________________________
 From: Peter Olsson <peter.olsson at visionutveckling.se>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Saturday, June 9, 2012 3:40 PM
Subject: Re: [Freeswitch-users] how to disable sending early media ?
 
Playing wav before answer IS early media.

So - no early media, no playback before answer.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Samira Mh [saami_mh at ymail.com]
Skickat: den 9 juni 2012 12:17
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] how to disable sending early media ?

whaaaaat?
i have checked? with alot of files but it coudn't work

________________________________
From: M.Emran <monemran at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, June 9, 2012 2:29 PM
Subject: Re: [Freeswitch-users] how to disable sending early media ?

it seems ur wav file has an issue.it<http://issue.it> must be played.


On Sat, Jun 9, 2012 at 3:49 PM, Samira Mh <saami_mh at ymail.com<mailto:saami_mh at ymail.com>> wrote:
hi,
thanks alot for your reply;
but i suue the following command in the lua script
session:preAnswer();
and again got the below error and .wav file coudn't play
switch_cpp.cpp:617 Sending early media

________________________________
From: Mohammad Emran <monemran at gmail.com<mailto:monemran at gmail.com>>

To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Saturday, June 9, 2012 2:10 PM

Subject: Re: [Freeswitch-users] how to disable sending early media ?

Use pre_answer instead of answer.


Sent from my iPhone

On Jun 9, 2012, at 3:13 PM, Samira Mh <saami_mh at ymail.com<mailto:saami_mh at ymail.com>> wrote:

hi thanks alot for your reply;
i don't want to use command "session:answer"
and want to only play .wav files in lua
i don't want get sending early media

________________________________
From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Saturday, June 9, 2012 1:30 PM
Subject: Re: [Freeswitch-users] how to disable sending early media ?

I guess you need to decide if you want early media or not... Playback before answer needs early media.


/Peter

9 jun 2012 kl. 10:47 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com><mailto:saami_mh at ymail.com<mailto:saami_mh at ymail.com>>>:


<action application="set" data="ignore_early_media=true"/>

i have set the above channel variable but when i am going to playback .wav file using lua scripts without issue the seeeion:answer ; i got error :sending early media


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


!DSPAM:4fd30a8732761348221455!

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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




--
Regards
----------
M Emran
Chief Executive Officer
E-SOFT BILLING (pvt). LTD.

HB Tower(3rd Floor)
House No # 1/A
Road No # 23
Gulshan # 1
Dhaka-1212, Bangladesh.
Phone: +880-2-8822312,+880-2-8822384
Fax : +880-2-8822254
E-Mail: info at e-softbilling.com<mailto:info at e-softbilling.com>
Web: www.e-softbilling.com<http://www.e-softbilling.com/>
? ? ? ?  www.isoftswitch.com<http://www.isoftswitch.com/>
? ? ? ?  www.howtonix.com<http://www.howtonix.com/>
? ? ? ?  www.sipmobiledialer.com<http://www.sipmobiledialer.com/>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


!DSPAM:4fd321b532761506015947!

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at 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 --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120609/53a30fe9/attachment-0001.html 

From peter.olsson at visionutveckling.se  Sat Jun  9 15:35:27 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 9 Jun 2012 11:35:27 +0000
Subject: [Freeswitch-users] how to disable sending early media ?
In-Reply-To: <1339240850.28527.YahooMailNeo@web120101.mail.ne1.yahoo.com>
References: <1339231296.22293.YahooMailNeo@web120102.mail.ne1.yahoo.com>
	<1A866A23-F1AF-49A0-8DFB-7F9C8D5C9B2D@visionutveckling.se>
	<1339233220.88823.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<6DC1D7D0-2D51-41E1-84CD-B32C445C81F5@gmail.com>
	<1339235358.94030.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<CAMbKPxYuD8Uyt_9Y4pTG2i7Wg0R-1JkEyWWfs5rxX5Uj8h67gg@mail.gmail.com>,
	<1339237074.24777.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<1FFF97C269757C458224B7C895F35F15100283@cantor.std.visionutv.se>,
	<1339240850.28527.YahooMailNeo@web120101.mail.ne1.yahoo.com>
Message-ID: <1FFF97C269757C458224B7C895F35F151002CC@cantor.std.visionutv.se>

Nope. Early media means "bring up media as soon as possible (before the call is answered)", so that's the whole idea. If you don't have early media, no media will be setup until the call has been answered, so it's not possible to play a file.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Samira Mh [saami_mh at ymail.com]
Skickat: den 9 juni 2012 13:20
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] how to disable sending early media ?

but is there  any way to "no early media " but  playback before answer?

________________________________
From: Peter Olsson <peter.olsson at visionutveckling.se>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, June 9, 2012 3:40 PM
Subject: Re: [Freeswitch-users] how to disable sending early media ?

Playing wav before answer IS early media.

So - no early media, no playback before answer.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] f?r Samira Mh [saami_mh at ymail.com<mailto:saami_mh at ymail.com>]
Skickat: den 9 juni 2012 12:17
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] how to disable sending early media ?

whaaaaat?
i have checked  with alot of files but it coudn't work

________________________________
From: M.Emran <monemran at gmail.com<mailto:monemran at gmail.com>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Saturday, June 9, 2012 2:29 PM
Subject: Re: [Freeswitch-users] how to disable sending early media ?

it seems ur wav file has an issue.it<http://issue.it> must be played.


On Sat, Jun 9, 2012 at 3:49 PM, Samira Mh <saami_mh at ymail.com<mailto:saami_mh at ymail.com><mailto:saami_mh at ymail.com<mailto:saami_mh at ymail.com>>> wrote:
hi,
thanks alot for your reply;
but i suue the following command in the lua script
session:preAnswer();
and again got the below error and .wav file coudn't play
switch_cpp.cpp:617 Sending early media

________________________________
From: Mohammad Emran <monemran at gmail.com<mailto:monemran at gmail.com><mailto:monemran at gmail.com<mailto:monemran at gmail.com>>>

To: FreeSWITCH Users Help <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>>>
Cc: FreeSWITCH Users Help <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>>>
Sent: Saturday, June 9, 2012 2:10 PM

Subject: Re: [Freeswitch-users] how to disable sending early media ?

Use pre_answer instead of answer.


Sent from my iPhone

On Jun 9, 2012, at 3:13 PM, Samira Mh <saami_mh at ymail.com<mailto:saami_mh at ymail.com><mailto:saami_mh at ymail.com<mailto:saami_mh at ymail.com>>> wrote:

hi thanks alot for your reply;
i don't want to use command "session:answer"
and want to only play .wav files in lua
i don't want get sending early media

________________________________
From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se><mailto:peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>>
To: FreeSWITCH Users Help <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>>>
Sent: Saturday, June 9, 2012 1:30 PM
Subject: Re: [Freeswitch-users] how to disable sending early media ?

I guess you need to decide if you want early media or not... Playback before answer needs early media.


/Peter

9 jun 2012 kl. 10:47 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com><mailto:saami_mh at ymail.com<mailto:saami_mh at ymail.com>><mailto:saami_mh at ymail.com<mailto:saami_mh at ymail.com><mailto:saami_mh at ymail.com<mailto:saami_mh at ymail.com>>>>:


<action application="set" data="ignore_early_media=true"/>

i have set the above channel variable but when i am going to playback .wav file using lua scripts without issue the seeeion:answer ; i got error :sending early media


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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--
Regards
----------
M Emran
Chief Executive Officer
E-SOFT BILLING (pvt). LTD.

HB Tower(3rd Floor)
House No # 1/A
Road No # 23
Gulshan # 1
Dhaka-1212, Bangladesh.
Phone: +880-2-8822312,+880-2-8822384
Fax : +880-2-8822254
E-Mail: info at e-softbilling.com<mailto:info at e-softbilling.com><mailto:info at e-softbilling.com<mailto:info at e-softbilling.com>>
Web: www.e-softbilling.com<http://www.e-softbilling.com/>
        www.isoftswitch.com<http://www.isoftswitch.com/>
        www.howtonix.com<http://www.howtonix.com/>
        www.sipmobiledialer.com<http://www.sipmobiledialer.com/>

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!DSPAM:4fd321b532761506015947!

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From chris at gonumina.com  Sat Jun  9 17:46:56 2012
From: chris at gonumina.com (Chris Ferreira)
Date: Sat, 9 Jun 2012 09:46:56 -0400
Subject: [Freeswitch-users] State of GUIs
In-Reply-To: <CAMbW9o+WVbQgzBdKYGDOw2fFj0mgoDWEmF_oqPJFRSyivb49yQ@mail.gmail.com>
References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net>
	<CAMbW9oJ6qBUkWrB=W9O+wGbLERMEWX-ujnXnpw6cVjP9-qxVMQ@mail.gmail.com>
	<03f301ccfd03$d7777080$86665180$@launch3.net>
	<CAKbxfG835-ROigp-7RtCeVH-7UtVLtTeQgo7GAF5Zo+gDnx+Eg@mail.gmail.com>
	<CAAYf2LvoFzeBrbgA=_WaOCSWxqGOXceBx8S_vxMnx7zbpCCVrQ@mail.gmail.com>
	<CAC-LwPPi4EmN0YxRnOvQAE9akpr6=4rnmH0apM82JFw-nDyUfg@mail.gmail.com>
	<CAMbW9o+eOmWmNbz0rtsvQCu3TiQ+R38_SHxeoBjZWA7j3q8hhA@mail.gmail.com>
	<CAMwZ-tK884W-t16ChmLawFA61jiKbg3NDkKT5-u-V+k1WuKg5g@mail.gmail.com>
	<CAMbW9o+WVbQgzBdKYGDOw2fFj0mgoDWEmF_oqPJFRSyivb49yQ@mail.gmail.com>
Message-ID: <-614380732171954257@unknownmsgid>

Hi Andrew,

Do you plan to only use this internally, sell it, or open source it? I ask
because it sounds a lot like what we intend on doing.


Thanks,

-Chris

___________________
Mobile Reply

On Jun 9, 2012, at 6:45 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk>
wrote:

Hi Gabe, It's still in the works but is going well. Since my last post I've
added Queues and Conference call support, although queues are going to need
a little work doing to force the freeswitch instances to load the
configuration of new ones.

Hoping to add faxing in shortly (time permitting, of course) and we'll see
where we go from there.

On 9 June 2012 04:27, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Fri, Mar 9, 2012 at 9:04 AM, Andrew Cassidy
> <andrew at cassidywebservices.co.uk> wrote:
> > As with a number of other people, I'm writing one to suit my needs too.
> Mine
> > is being written in Python/Django and is aimed at being multi tenant in a
> > cloud setting or single user on a manged device setting sharing as much
> of
> > the code from both scenarios as possible.
>
> Hey Andrew, how did this project go?
>
>
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
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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
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk

 _________________________________________________________________________
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From Nabble_01394 at slickdeals.endjunk.com  Sat Jun  9 18:35:05 2012
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Sat, 9 Jun 2012 07:35:05 -0700 (PDT)
Subject: [Freeswitch-users] State of GUIs
In-Reply-To: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net>
References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net>
Message-ID: <1339252505739-7579650.post@n2.nabble.com>


Brett Wilson wrote
> It seems that most of the gui projects (blue.box, fusionpbx) have been
> mostly abandoned in terms of development.
I don't know blue.box; however, developers @FusionPBX seems to be very
aggressive in developing FusionPBX. Take a look at the output from *svn
info* and mine is as shown below:



 The problem I have with fusionpbx is that config files are overwritten by
whatever is in the database. If you hand-edit a config file, fusion will not
parse the file and load those settings  into the interface.
For FusionPBX, if you want to retain your manually (hand) edited
configuration file(s), you will need to use the FusionPBX built-in editor to
edit your configuration file(s). Or, you can use its GUI to edit (add/del)
items in your configuration file(s). Once that is done, you will need to
save your configuration file(s). You can confirm the changes by manually
editing your configuration files using your favorite text editor, i.e. vi,
etc., from a shell.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/State-of-GUIs-tp7351199p7579650.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From belintinc at gmail.com  Sat Jun  9 18:57:51 2012
From: belintinc at gmail.com (BELint Inc)
Date: Sat, 9 Jun 2012 20:27:51 +0530
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 72, Issue 66
In-Reply-To: <mailman.11403.1339130921.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.11403.1339130921.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAEA+2KtNM-q9OUmBYL3hvQ88y01i26tyqK+xnneMq5m-hEyrwg@mail.gmail.com>

 Thanks Dave,

We already tried "NDLB-connectile-dysfunction=true" as the client (Yealink
T28P) is not using rport in its VIA header. But still no luck freeswitch
still sending the response to the port in Contact header (5062) instead of
the recieving port (14335). 5062 port is actually not valid as the client
is behind NAT and only can get the response if freeswitch will send it to
14335 port.

Do you have any other idea about this?
:-( .........

Regards,
Belint


> ---------- Forwarded message ----------
> From: "Dave R. Kompel" <drk at drkngs.net>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Thu, 07 Jun 2012 21:35:56 -0700
> Subject: Re: [Freeswitch-users] Freeswitch is responding to contact's port
> when client behind NAT
> **
> Sorry for the long delay, but here's your answer:
>
> It's going to go to the address in the un-altered VIA header, unless RPORT
> gets sent it the VIA. If you have agressive_nat_detection turned on, and
> the client puts RPORT in the via, it will work fine. If you have a broken
> device that won't do it, then set the variable
> "NDLB-connectile-dysfunction=true" in the users directory enter (XML) or
> other, what ever you are using. That will cause the contact to be
> re-written to the transport address from where it came (both ip address and
> port) and the response will also be sent there.
>
> --Dave
>
>  ------------------------------
> *From:* BELint Inc [mailto:belintinc at gmail.com]
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thu, 07 Jun 2012 10:56:07 -0700
> *Subject:* Re: [Freeswitch-users] Freeswitch is responding to contact's
> port when client behind NAT
>
> Does any one know its solutions,
> Guys please help, its urgent. I am even unable to register my phones :-(
>
> Regards,
>
> On Thu, Jun 7, 2012 at 9:20 AM, BELint Inc <belintinc at gmail.com> wrote:
>
>> Hi All,
>>
>> We`ve been using freeswitch since very long, It was good all over but it
>> got stuck to a simple yet complex situation. We have Yealink T28P series
>> phones behind NAT and freeswitch is unable to get them register. In the
>> scenario freeswitch is at public IP. The reason we think is that freeswitch
>> is using sip port in contact header to respond to.
>>
>> Is there anyway we can avoid this or force it to always send the response
>> to destination IP instead of contact or via header ports because clients
>> are behind NAT and response can only reach them if it respond to
>> destination IP & PORT.
>>
>> Please revert if someone know how to solve this problem. It is very
>> urgent.
>>
>> Regards
>> BELint
>>
>
>
>
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From andrew at cassidywebservices.co.uk  Sat Jun  9 19:21:17 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sat, 9 Jun 2012 16:21:17 +0100
Subject: [Freeswitch-users] State of GUIs
In-Reply-To: <1339252505739-7579650.post@n2.nabble.com>
References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net>
	<1339252505739-7579650.post@n2.nabble.com>
Message-ID: <CAMbW9o+88S5WudFGHWL+JeJ9Yb1=Tzc9_Lq18TysQKi3rGtxoQ@mail.gmail.com>

Hi Chris, it's something that's intended to be sold and is being funded by
a company that wants to use it as the basis for a hosted telephony platform.

There are already many front-ends available for freeswitch, such as
bluebox, freepybx, etc but I didn't feel any of them fit in with my
intended model.

On 9 June 2012 15:35, mazilo <Nabble_01394 at slickdeals.endjunk.com> wrote:

>
> Brett Wilson wrote
> > It seems that most of the gui projects (blue.box, fusionpbx) have been
> > mostly abandoned in terms of development.
> I don't know blue.box; however, developers @FusionPBX seems to be very
> aggressive in developing FusionPBX. Take a look at the output from *svn
> info* and mine is as shown below:
>
>
>
>  The problem I have with fusionpbx is that config files are overwritten by
> whatever is in the database. If you hand-edit a config file, fusion will
> not
> parse the file and load those settings  into the interface.
> For FusionPBX, if you want to retain your manually (hand) edited
> configuration file(s), you will need to use the FusionPBX built-in editor
> to
> edit your configuration file(s). Or, you can use its GUI to edit (add/del)
> items in your configuration file(s). Once that is done, you will need to
> save your configuration file(s). You can confirm the changes by manually
> editing your configuration files using your favorite text editor, i.e. vi,
> etc., from a shell.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
> Watts of electricity.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/State-of-GUIs-tp7351199p7579650.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From toddb at toddbailey.net  Sat Jun  9 21:58:18 2012
From: toddb at toddbailey.net (toddb at toddbailey.net)
Date: Sat, 09 Jun 2012 10:58:18 -0700
Subject: [Freeswitch-users] Hardware build recommendations
Message-ID: <20120609105818.33e327b490679d2282e332758c73b55b.80fee05aed.wbe@email14.secureserver.net>

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From brian at freeswitch.org  Sat Jun  9 22:39:05 2012
From: brian at freeswitch.org (Brian West)
Date: Sat, 9 Jun 2012 13:39:05 -0500
Subject: [Freeswitch-users] Freeswitch send 423 (Interval Too Brief)
In-Reply-To: <CAMwZ-t+eaXUktbM1bsh1gO_r0qNCj-xfaG1Yu3eLO=hR-d5e8A@mail.gmail.com>
References: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57E7E@CGNMSG01.oc.qsc.de>
	<CAMwZ-t+eaXUktbM1bsh1gO_r0qNCj-xfaG1Yu3eLO=hR-d5e8A@mail.gmail.com>
Message-ID: <9176366786866033429@unknownmsgid>

This means the invite contains a session timer that is lower than the
allow minimum per the RFC, it's asterisk? if so it just might be
sending -1 as the min se.

Sent from my iPad

On Jun 8, 2012, at 9:15 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Tue, Jun 5, 2012 at 8:51 AM, Lappe, Adam <Adam.Lappe at qsc.de> wrote:
>> is there a way to make freeswitch send a 423 (Interval Too Brief) during a
>> SIP REGISTER request?
>>
>> I want to prevent clients to (re)register with a low expire-time and thus
>> spam our network.
>
> Have you greped the code for it? Seems like it's be easy to spot. If
> not, open a ticket. Or, better yet, submit a patch :)
>
>
>> The ?gsip-force-expires?h-variable would work but don?ft seem to be
>> SIP-compliant to me.
>
> Have you followed the SIP traces to see exactly how this works? I
> haven't, but perhaps it really does work how you'd hope (I don't know,
> I've never used it).
>
>
> Best,
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Jun  9 22:40:24 2012
From: brian at freeswitch.org (Brian West)
Date: Sat, 9 Jun 2012 13:40:24 -0500
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <CAMwZ-t+t0Z1pus4oiDU0hs5jNF0YPK-VjWcFAwhm8_GSbMu9vg@mail.gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
	<4BFC4A64.5090208@gmail.com>
	<AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>
	<D311BDCB06324D0DB77898FC6FF4B0EB@dell9400>
	<AANLkTilAMe1lRKK4R6Nmr29FzfI-pP3wRa7Foabmd8Dp@mail.gmail.com>
	<4BFD4562.5030804@gmail.com>
	<AANLkTikGQlZ4N3zfYv18COvbubEi-8LSOZxnAy1KC5sP@mail.gmail.com>
	<4BFEE96A.2010003@gmail.com>
	<AANLkTikSgh4xN9kb3Lha5dNlacgTA9DqrkHqU3qCWZUT@mail.gmail.com>
	<D110995190DF45D78114CB61426F9946@dell9400>
	<AANLkTillSAX6L4gum_g9SNjP6O8hGEtXTfcIakZT96_D@mail.gmail.com>
	<4C057EB7.3090005@gmail.com>
	<AANLkTima1WBHX_2n0fN9qSXlYNFmbMckIDBlO_Ga7RTo@mail.gmail.com>
	<50037FB5-DE1A-485C-980D-8F6ECAC3928E@freeswitch.org>
	<CAMwZ-t+t0Z1pus4oiDU0hs5jNF0YPK-VjWcFAwhm8_GSbMu9vg@mail.gmail.com>
Message-ID: <-2289143762373440304@unknownmsgid>

I don't even recall sending this email.  I'm losing it!

Sent from my iPad

On Jun 8, 2012, at 11:10 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> This is old, but I have to ask, can I get an AMEN?!
>
>
> Gabe


From brian at freeswitch.org  Sat Jun  9 22:42:43 2012
From: brian at freeswitch.org (Brian West)
Date: Sat, 9 Jun 2012 13:42:43 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 72, Issue 66
In-Reply-To: <CAEA+2KtNM-q9OUmBYL3hvQ88y01i26tyqK+xnneMq5m-hEyrwg@mail.gmail.com>
References: <mailman.11403.1339130921.13443.freeswitch-users@lists.freeswitch.org>
	<CAEA+2KtNM-q9OUmBYL3hvQ88y01i26tyqK+xnneMq5m-hEyrwg@mail.gmail.com>
Message-ID: <8505475230999962314@unknownmsgid>

um this is the purpose of rport, enable it on the endpoint for pain relief.

Sent from my iPad

On Jun 9, 2012, at 9:59 AM, BELint Inc <belintinc at gmail.com> wrote:

Thanks Dave,

We already tried "NDLB-connectile-dysfunction=true" as the client (Yealink
T28P) is not using rport in its VIA header. But still no luck freeswitch
still sending the response to the port in Contact header (5062) instead of
the recieving port (14335). 5062 port is actually not valid as the client
is behind NAT and only can get the response if freeswitch will send it to
14335 port.

Do you have any other idea about this?
:-( .........

Regards,
Belint
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From bdfoster at endigotech.com  Sat Jun  9 22:44:39 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 9 Jun 2012 14:44:39 -0400
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <-2289143762373440304@unknownmsgid>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
	<4BFC4A64.5090208@gmail.com>
	<AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>
	<D311BDCB06324D0DB77898FC6FF4B0EB@dell9400>
	<AANLkTilAMe1lRKK4R6Nmr29FzfI-pP3wRa7Foabmd8Dp@mail.gmail.com>
	<4BFD4562.5030804@gmail.com>
	<AANLkTikGQlZ4N3zfYv18COvbubEi-8LSOZxnAy1KC5sP@mail.gmail.com>
	<4BFEE96A.2010003@gmail.com>
	<AANLkTikSgh4xN9kb3Lha5dNlacgTA9DqrkHqU3qCWZUT@mail.gmail.com>
	<D110995190DF45D78114CB61426F9946@dell9400>
	<AANLkTillSAX6L4gum_g9SNjP6O8hGEtXTfcIakZT96_D@mail.gmail.com>
	<4C057EB7.3090005@gmail.com>
	<AANLkTima1WBHX_2n0fN9qSXlYNFmbMckIDBlO_Ga7RTo@mail.gmail.com>
	<50037FB5-DE1A-485C-980D-8F6ECAC3928E@freeswitch.org>
	<CAMwZ-t+t0Z1pus4oiDU0hs5jNF0YPK-VjWcFAwhm8_GSbMu9vg@mail.gmail.com>
	<-2289143762373440304@unknownmsgid>
Message-ID: <CAKV-jFxf1NAte2jEo3k3-9Yx-tDbavC8C515PrDJzADN4CE1XQ@mail.gmail.com>

Well for starters it was an email from two years ago lol

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 9, 2012 2:41 PM, "Brian West" <brian at freeswitch.org> wrote:

> I don't even recall sending this email.  I'm losing it!
>
> Sent from my iPad
>
> On Jun 8, 2012, at 11:10 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
>
> > This is old, but I have to ask, can I get an AMEN?!
> >
> >
> > Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Jun  9 22:49:25 2012
From: brian at freeswitch.org (Brian West)
Date: Sat, 9 Jun 2012 13:49:25 -0500
Subject: [Freeswitch-users] Time needed for ringing a call group
In-Reply-To: <20120608223010.2d079108@anubis.defcon1>
References: <D3BC5C093B2E4150AF352558D6E077BE@KITPC003>
	<-6559196381009927779@unknownmsgid>
	<20120608223010.2d079108@anubis.defcon1>
Message-ID: <-8988759419527339202@unknownmsgid>

I alway kid, but my question real.  2000 seconds was exce???v?!

(^-^)/

Sent from my iPad

On Jun 8, 2012, at 3:30 PM, Bzzz <lazyvirus at gmx.com> wrote:

> Hmm, either you're kidding or you have a keyboard|board problem


From brian at freeswitch.org  Sat Jun  9 22:50:35 2012
From: brian at freeswitch.org (Brian West)
Date: Sat, 9 Jun 2012 13:50:35 -0500
Subject: [Freeswitch-users] Unable to use mod_java.
In-Reply-To: <CAAq2U1By5VrsM=w0+Nbb=sxhh376-Gh0Gpwgs99FW-6CF5P=hQ@mail.gmail.com>
References: <CAP_69mtZuWq-22j-hiFBBuYzNKzqd46fR0MPjXgJYYd1kfjxnw@mail.gmail.com>
	<CAAq2U1By5VrsM=w0+Nbb=sxhh376-Gh0Gpwgs99FW-6CF5P=hQ@mail.gmail.com>
Message-ID: <-5197386564220120599@unknownmsgid>

or the libesl java bits in libs/esl

Sent from my iPad

On Jun 7, 2012, at 8:39 PM, jonathan augenstine <jaugenstine at gmail.com> wrote:

> Venkatesh,
>
> I can tell you from personal experience that mod_java does have issues.  Unless you really need Java I would recommend that you take a look at Lua.  If you really need or want to use Java, then I would recommend investigating the event socket and XML-RPC interface.
>
> Jonathan


From bdfoster at endigotech.com  Sat Jun  9 22:53:10 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 9 Jun 2012 14:53:10 -0400
Subject: [Freeswitch-users] Hardware build recommendations
In-Reply-To: <20120609105818.33e327b490679d2282e332758c73b55b.80fee05aed.wbe@email14.secureserver.net>
References: <20120609105818.33e327b490679d2282e332758c73b55b.80fee05aed.wbe@email14.secureserver.net>
Message-ID: <CAKV-jFy4kOo69V8sboRtxRWMtxTx+J_Rf7C0csZKOpFRWpvuvQ@mail.gmail.com>

Well his calls per hour are probably irrelevent. Probably much less than
one. Many of us (including myself) have production servers and use a home
server as a dev machine (which has a ton of other stuff running on it).
Freeswitch, OpenVPN, pfsense, etc are all pretty easy to deal with on a
home server. Its when you get things like mythtv-backend, Subsonic, etc.
that you can easily get in trouble with. I've got an old Dell Precision 490
(ok maybe 5 years old) w/ a Intel Xeon 5150 and 1GB of ram that runs gnome,
mythtv-backend, OpenVPN, FS, Subsonic, MediaTomb, 2 websites, a file
server, and ZNC and it runs pretty well. there's even a few MySQL
databases. Load avg probably hovers at around 0.25 for the most part.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 9, 2012 2:00 PM, <toddb at toddbailey.net> wrote:

> I'm not expert on this topic but I think to get a more accurate answer you
> you provide more detail on the the box is going to be use for.  Dedicated
> or shared usage.
>
> For example:   I'm running Fedora 14 x64 w/8gig memory and several
> terabytes disk storage
> It's a dual proc w/ 2x Intel 5160 (dual core)  3ghz  (HP x8400 workstation
> system board transplanted into a server chassis).  I'm using it as home
> based media server with 4 clients (DVR& Music), a database, email,web file
> server.  I just added Freeswitch and see little impact to resources.  My
> cost off ebay was about $400
>
> For Freeswitch it might be useful to state estimated call volume per hour
> and number of simultaneous calls and desired features,
>
>
>  -------- Original Message --------
> Subject: Re: [Freeswitch-users] Hardware build recommendations
> From: curriegrad2004 <curriegrad2004 at gmail.com>
> Date: Fri, June 08, 2012 2:02 pm
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
> Try aiming for a 64-bit OS if you could. For the OpenVPN part, if you
> could shoot for a Sandy Bridge or later CPU you'd probably find the
> AES-NI part of the CPU quite suitable. If you're on a budget you can
> always certainly aim for an AMD E-350 or something along the lines of
> that. Plenty of CPU power for SOHO boxes.
>
> On Fri, Jun 8, 2012 at 1:23 PM, Brian Foster <bdfoster at endigotech.com>
> wrote:
> > Raspberry Pi?
> >
> > Shoot for 1.6 ghz dual core with 1gb memory, 2 NICs. That should be
> overkill
> > (engineering translation: bare minimum).
> >
> > Brian Foster
> > Endigo Computer LLC
> >
> > Sent from a mobile device.
> >
> > On Jun 8, 2012 2:35 PM, "Timothy Bolton" <ifoundthetao at gmail.com> wrote:
> >>
> >> Does anyone have any ideas for a good hardware build for a SOHO server?
> >>
> >> Here's what I'd like running on it: OpenVPN, FreeSWITCH, and a firewall
> >> (probably pfSense).
> >>
> >> My home network doesn't have too much traffic, and I don't get that many
> >> calls.
> >>
> >> --
> >> 'We who cut mere stones must always be envisioning cathedrals.'
> >> Quarry Worker's Creed
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From drk at drkngs.net  Sat Jun  9 22:54:51 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Sat, 09 Jun 2012 11:54:51 -0700
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 72, Issue 66
In-Reply-To: <8505475230999962314@unknownmsgid>
Message-ID: <20120609185451.82827004@mail.tritonwest.net>

Brian,  
   
There are still some devices that are broken, and won't send RPORT. It sux...  
   
--Dave
      _____  

  From: Brian West [mailto:brian at freeswitch.org]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Sat, 09 Jun 2012 11:42:43 -0700
Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 72, Issue 66

  
um this is the purpose of rport, enable it on the endpoint for pain relief.

Sent from my iPad  

On Jun 9, 2012, at 9:59 AM, BELint Inc <belintinc at gmail.com> wrote:

    
Thanks Dave,    
We already tried "NDLB-connectile-dysfunction=true" as the client (Yealink T28P) is not using rport in its VIA header. But still no luck freeswitch still sending the response to the port in Contact header (5062) instead of the recieving port (14335). 5062 port is actually not valid as the client is behind NAT and only can get the response if freeswitch will send it to 14335 port.  
   
Do you have any other idea about this?  
:-( .........  
Regards,  
Belint  
      
   
 
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From bdfoster at endigotech.com  Sat Jun  9 22:59:31 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 9 Jun 2012 14:59:31 -0400
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 72, Issue 66
In-Reply-To: <20120609185451.82827004@mail.tritonwest.net>
References: <8505475230999962314@unknownmsgid>
	<20120609185451.82827004@mail.tritonwest.net>
Message-ID: <CAKV-jFxgjXVgyhbt=QOypmdRWiyGkj_E+b=dY2D2UuwhLp=DLQ@mail.gmail.com>

Set NDLB-force-rport on FS... see the wiki for this common issue (btw
please send a reply on the actual thread not the digest).

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 9, 2012 2:56 PM, "Dave R. Kompel" <drk at drkngs.net> wrote:

> **
> Brian,
>
> There are still some devices that are broken, and won't send RPORT. It
> sux...
>
> --Dave
>
>  ------------------------------
> *From:* Brian West [mailto:brian at freeswitch.org]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> *Sent:* Sat, 09 Jun 2012 11:42:43 -0700
> *Subject:* Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 72, Issue
> 66
>
> um this is the purpose of rport, enable it on the endpoint for pain relief.
>
> Sent from my iPad
>
> On Jun 9, 2012, at 9:59 AM, BELint Inc <belintinc at gmail.com> wrote:
>
>  Thanks Dave,
>
> We already tried "NDLB-connectile-dysfunction=true" as the client (Yealink
> T28P) is not using rport in its VIA header. But still no luck freeswitch
> still sending the response to the port in Contact header (5062) instead of
> the recieving port (14335). 5062 port is actually not valid as the client
> is behind NAT and only can get the response if freeswitch will send it to
> 14335 port.
>
> Do you have any other idea about this?
> :-( .........
>
> Regards,
> Belint
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From toddb at toddbailey.net  Sat Jun  9 23:20:51 2012
From: toddb at toddbailey.net (toddb at toddbailey.net)
Date: Sat, 09 Jun 2012 12:20:51 -0700
Subject: [Freeswitch-users] FS & SPA 3102 Setup
Message-ID: <20120609122051.33e327b490679d2282e332758c73b55b.857425d83e.wbe@email14.secureserver.net>

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From drk at drkngs.net  Sat Jun  9 23:22:07 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Sat, 09 Jun 2012 12:22:07 -0700
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 72, Issue 66
In-Reply-To: <CAKV-jFxgjXVgyhbt=QOypmdRWiyGkj_E+b=dY2D2UuwhLp=DLQ@mail.gmail.com>
Message-ID: <20120609192207.d43581e2@mail.tritonwest.net>

Yes, that can also work, but I wouldn't do that on the main profile you're using. I would set up one running on a different port, just for the devices that need it. That way it won't mess up your clients that work right.  
   
--Dave
      _____  

  From: Brian Foster [mailto:bdfoster at endigotech.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Sat, 09 Jun 2012 11:59:31 -0700
Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 72, Issue 66

  

Set NDLB-force-rport on FS... see the wiki for this common issue (btw please send a reply on the actual thread not the digest).  

Brian Foster
Endigo Computer LLC  

Sent from a mobile device.  
On Jun 9, 2012 2:56 PM, "Dave R. Kompel" <drk at drkngs.net> wrote:
    
  
Brian,  
   
There are still some devices that are broken, and won't send RPORT. It sux...  
   
--Dave
      _____  

  From: Brian West [mailto:brian at freeswitch.org]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Sat, 09 Jun 2012 11:42:43 -0700
Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 72, Issue 66

  
um this is the purpose of rport, enable it on the endpoint for pain relief.

Sent from my iPad  

On Jun 9, 2012, at 9:59 AM, BELint Inc <belintinc at gmail.com> wrote:

    
Thanks Dave,    
We already tried "NDLB-connectile-dysfunction=true" as the client (Yealink T28P) is not using rport in its VIA header. But still no luck freeswitch still sending the response to the port in Contact header (5062) instead of the recieving port (14335). 5062 port is actually not valid as the client is behind NAT and only can get the response if freeswitch will send it to 14335 port.  
   
Do you have any other idea about this?  
:-( .........  
Regards,  
Belint  
  
   
 
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

      
   
 
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From belintinc at gmail.com  Sat Jun  9 23:49:10 2012
From: belintinc at gmail.com (BELint Inc)
Date: Sun, 10 Jun 2012 01:19:10 +0530
Subject: [Freeswitch-users] (Offlist)  FreeSWITCH-users Digest, Vol 72,
 Issue 66
In-Reply-To: <20120609182428.790cd56f@mail.tritonwest.net>
References: <CAEA+2KtNM-q9OUmBYL3hvQ88y01i26tyqK+xnneMq5m-hEyrwg@mail.gmail.com>
	<20120609182428.790cd56f@mail.tritonwest.net>
Message-ID: <CAEA+2Ksa8nHpbgC+sA46s0JnK4PU4MHBO1dsjUF34Rxyo4wRvg@mail.gmail.com>

Hi Dave,

Sure we can try that, meanwhile we also are interested in , which way it
can help us solve the problem.

Regards
Belint



On Sat, Jun 9, 2012 at 11:54 PM, Dave R. Kompel <drk at drkngs.net> wrote:

> **
> Can we set up a test? I would like you to try to register your device in
> one of my lab switches. Since I make a comercial product which is a carrier
> grade softswitch, based on FS, my customers all are using FS as their main
> switch for ITSP/CLEC.
>
> Although I use custom logic to handle registrations, the options should
> work for you. I would like to see actually what is going on.
>
> --Dave
>
>  ------------------------------
> *From:* BELint Inc [mailto:belintinc at gmail.com]
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Sat, 09 Jun 2012 07:57:51 -0700
> *Subject:* Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 72, Issue
> 66
>
>  Thanks Dave,
>
> We already tried "NDLB-connectile-dysfunction=true" as the client (Yealink
> T28P) is not using rport in its VIA header. But still no luck freeswitch
> still sending the response to the port in Contact header (5062) instead of
> the recieving port (14335). 5062 port is actually not valid as the client
> is behind NAT and only can get the response if freeswitch will send it to
> 14335 port.
>
> Do you have any other idea about this?
> :-( .........
>
> Regards,
> Belint
>
>
>> ---------- Forwarded message ----------
>> From: "Dave R. Kompel" <drk at drkngs.net>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Date: Thu, 07 Jun 2012 21:35:56 -0700
>> Subject: Re: [Freeswitch-users] Freeswitch is responding to contact's
>> port when client behind NAT
>> **
>> Sorry for the long delay, but here's your answer:
>>
>> It's going to go to the address in the un-altered VIA header, unless
>> RPORT gets sent it the VIA. If you have agressive_nat_detection turned on,
>> and the client puts RPORT in the via, it will work fine. If you have a
>> broken device that won't do it, then set the variable
>> "NDLB-connectile-dysfunction=true" in the users directory enter (XML) or
>> other, what ever you are using. That will cause the contact to be
>> re-written to the transport address from where it came (both ip address and
>> port) and the response will also be sent there.
>>
>> --Dave
>>
>>  ------------------------------
>> *From:* BELint Inc [mailto:belintinc at gmail.com]
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Thu, 07 Jun 2012 10:56:07 -0700
>> *Subject:* Re: [Freeswitch-users] Freeswitch is responding to contact's
>> port when client behind NAT
>>
>> Does any one know its solutions,
>> Guys please help, its urgent. I am even unable to register my phones :-(
>>
>> Regards,
>>
>> On Thu, Jun 7, 2012 at 9:20 AM, BELint Inc <belintinc at gmail.com> wrote:
>>
>>> Hi All,
>>>
>>> We`ve been using freeswitch since very long, It was good all over but it
>>> got stuck to a simple yet complex situation. We have Yealink T28P series
>>> phones behind NAT and freeswitch is unable to get them register. In the
>>> scenario freeswitch is at public IP. The reason we think is that freeswitch
>>> is using sip port in contact header to respond to.
>>>
>>> Is there anyway we can avoid this or force it to always send the
>>> response to destination IP instead of contact or via header ports because
>>> clients are behind NAT and response can only reach them if it respond to
>>> destination IP & PORT.
>>>
>>> Please revert if someone know how to solve this problem. It is very
>>> urgent.
>>>
>>> Regards
>>> BELint
>>>
>>
>>
>>
>
>
>
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From ifoundthetao at gmail.com  Sun Jun 10 00:08:17 2012
From: ifoundthetao at gmail.com (Timothy Bolton)
Date: Sat, 09 Jun 2012 15:08:17 -0500
Subject: [Freeswitch-users] Hardware build recommendations
In-Reply-To: <20120609105818.33e327b490679d2282e332758c73b55b.80fee05aed.wbe@email14.secureserver.net>
References: <20120609105818.33e327b490679d2282e332758c73b55b.80fee05aed.wbe@email14.secureserver.net>
Message-ID: <4FD3AD31.4090506@gmail.com>

I thought I lined those out in the initial email:
 >> Here's what I'd like running on it: OpenVPN, FreeSWITCH, and a firewall
 >> (probably pfSense).
 >>
 >> My home network doesn't have too much traffic, and I don't get that many
 >> calls.

But I suppose my language may have been too generic.

I get less than a call an hour.  Most of this is just me playing around: 
initiating calls from websites, integrating that into ticketing systems 
/ emails, etc..  So not a high demand, very few simultaneous calls on 
the inbound, though I would like for many simultaneous calls to be possible.

As far as other things go: CentOS 6, no mail server, OpenVPN, pfSense, 
probably OpenSSL too.  Access to this network via VPN would be on 
occasion when I'm out of the office, or if other people need to get in, 
which would be rare.  That's about all this would be used for.  Other 
boxes are set up for other uses.

'We who cut mere stones must always be envisioning cathedrals.'
Quarry Worker's Creed

On 6/9/2012 12:58 PM, toddb at toddbailey.net wrote:
> I'm not expert on this topic but I think to get a more accurate answer 
> you you provide more detail on the the box is going to be use for.  
> Dedicated or shared usage.
>
> For example:   I'm running Fedora 14 x64 w/8gig memory and several 
> terabytes disk storage
> It's a dual proc w/ 2x Intel 5160 (dual core)  3ghz  (HP x8400 
> workstation system board transplanted into a server chassis).  I'm 
> using it as home based media server with 4 clients (DVR& Music), a 
> database, email,web file server. I just added Freeswitch and see 
> little impact to resources. My cost off ebay was about $400
>
> For Freeswitch it might be useful to state estimated call volume per 
> hour and number of simultaneous calls and desired features,
>
>
>     -------- Original Message --------
>     Subject: Re: [Freeswitch-users] Hardware build recommendations
>     From: curriegrad2004 <curriegrad2004 at gmail.com
>     <mailto:curriegrad2004 at gmail.com>>
>     Date: Fri, June 08, 2012 2:02 pm
>     To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>
>     Try aiming for a 64-bit OS if you could. For the OpenVPN part, if you
>     could shoot for a Sandy Bridge or later CPU you'd probably find the
>     AES-NI part of the CPU quite suitable. If you're on a budget you can
>     always certainly aim for an AMD E-350 or something along the lines of
>     that. Plenty of CPU power for SOHO boxes.
>
>     On Fri, Jun 8, 2012 at 1:23 PM, Brian Foster
>     <bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>> wrote:
>     > Raspberry Pi?
>     >
>     > Shoot for 1.6 ghz dual core with 1gb memory, 2 NICs. That should
>     be overkill
>     > (engineering translation: bare minimum).
>     >
>     > Brian Foster
>     > Endigo Computer LLC
>     >
>     > Sent from a mobile device.
>     >
>     > On Jun 8, 2012 2:35 PM, "Timothy Bolton" <ifoundthetao at gmail.com
>     <mailto:ifoundthetao at gmail.com>> wrote:
>     >>
>     >> Does anyone have any ideas for a good hardware build for a SOHO
>     server?
>     >>
>     >> Here's what I'd like running on it: OpenVPN, FreeSWITCH, and a
>     firewall
>     >> (probably pfSense).
>     >>
>     >> My home network doesn't have too much traffic, and I don't get
>     that many
>     >> calls.
>     >>
>     >> --
>     >> 'We who cut mere stones must always be envisioning cathedrals.'
>     >> Quarry Worker's Creed
>     >>
>     >>
>     >>
>     _________________________________________________________________________
>     >> Professional FreeSWITCH Consulting Services:
>     >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     >> http://www.freeswitchsolutions.com
>     >>
>     >> 
>     >> 
>     >>
>     >> Official FreeSWITCH Sites
>     >> http://www.freeswitch.org
>     >> http://wiki.freeswitch.org
>     >> http://www.cluecon.com
>     >>
>     >> Join Us At ClueCon - Aug 7-9, 2012
>     >>
>     >> 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
>     >
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     > http://www.freeswitchsolutions.com
>     >
>     > 
>     > 
>     >
>     > Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://wiki.freeswitch.org
>     > http://www.cluecon.com
>     >
>     > Join Us At ClueCon - Aug 7-9, 2012
>     >
>     > 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
>     >
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From gabe at gundy.org  Sun Jun 10 01:36:49 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 9 Jun 2012 15:36:49 -0600
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <-2289143762373440304@unknownmsgid>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
	<4BFC4A64.5090208@gmail.com>
	<AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>
	<D311BDCB06324D0DB77898FC6FF4B0EB@dell9400>
	<AANLkTilAMe1lRKK4R6Nmr29FzfI-pP3wRa7Foabmd8Dp@mail.gmail.com>
	<4BFD4562.5030804@gmail.com>
	<AANLkTikGQlZ4N3zfYv18COvbubEi-8LSOZxnAy1KC5sP@mail.gmail.com>
	<4BFEE96A.2010003@gmail.com>
	<AANLkTikSgh4xN9kb3Lha5dNlacgTA9DqrkHqU3qCWZUT@mail.gmail.com>
	<D110995190DF45D78114CB61426F9946@dell9400>
	<AANLkTillSAX6L4gum_g9SNjP6O8hGEtXTfcIakZT96_D@mail.gmail.com>
	<4C057EB7.3090005@gmail.com>
	<AANLkTima1WBHX_2n0fN9qSXlYNFmbMckIDBlO_Ga7RTo@mail.gmail.com>
	<50037FB5-DE1A-485C-980D-8F6ECAC3928E@freeswitch.org>
	<CAMwZ-t+t0Z1pus4oiDU0hs5jNF0YPK-VjWcFAwhm8_GSbMu9vg@mail.gmail.com>
	<-2289143762373440304@unknownmsgid>
Message-ID: <CAMwZ-tLtYG056pCAQKzHNAMOi5R83jBXYM+wkRyVaDSfDcE5Kg@mail.gmail.com>

On Sat, Jun 9, 2012 at 12:40 PM, Brian West <brian at freeswitch.org> wrote:
> I don't even recall sending this email. ?I'm losing it!

It *was* two years ago!

Gabe


From anthony.minessale at gmail.com  Sun Jun 10 08:01:18 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 9 Jun 2012 23:01:18 -0500
Subject: [Freeswitch-users] State of GUIs
In-Reply-To: <CAMbW9o+88S5WudFGHWL+JeJ9Yb1=Tzc9_Lq18TysQKi3rGtxoQ@mail.gmail.com>
References: <02d401ccfc3d$454cd6d0$cfe68470$@launch3.net>
	<1339252505739-7579650.post@n2.nabble.com>
	<CAMbW9o+88S5WudFGHWL+JeJ9Yb1=Tzc9_Lq18TysQKi3rGtxoQ@mail.gmail.com>
Message-ID: <CAKbxfG9yj421gK0kmChsaRkgjEOyhj=e8sgSmsHya5Soc6P10g@mail.gmail.com>

Cool can you ask them to call Brian and sponsor cluecon 2012?  The
conference that makes things like this possible? =p
On Jun 9, 2012 10:22 AM, "Andrew Cassidy" <andrew at cassidywebservices.co.uk>
wrote:

> Hi Chris, it's something that's intended to be sold and is being funded by
> a company that wants to use it as the basis for a hosted telephony platform.
>
> There are already many front-ends available for freeswitch, such as
> bluebox, freepybx, etc but I didn't feel any of them fit in with my
> intended model.
>
> On 9 June 2012 15:35, mazilo <Nabble_01394 at slickdeals.endjunk.com> wrote:
>
>>
>> Brett Wilson wrote
>> > It seems that most of the gui projects (blue.box, fusionpbx) have been
>> > mostly abandoned in terms of development.
>> I don't know blue.box; however, developers @FusionPBX seems to be very
>> aggressive in developing FusionPBX. Take a look at the output from *svn
>> info* and mine is as shown below:
>>
>>
>>
>>  The problem I have with fusionpbx is that config files are overwritten by
>> whatever is in the database. If you hand-edit a config file, fusion will
>> not
>> parse the file and load those settings  into the interface.
>> For FusionPBX, if you want to retain your manually (hand) edited
>> configuration file(s), you will need to use the FusionPBX built-in editor
>> to
>> edit your configuration file(s). Or, you can use its GUI to edit (add/del)
>> items in your configuration file(s). Once that is done, you will need to
>> save your configuration file(s). You can confirm the changes by manually
>> editing your configuration files using your favorite text editor, i.e. vi,
>> etc., from a shell.
>>
>> -----
>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
>> Watts of electricity.
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/State-of-GUIs-tp7351199p7579650.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at opencsta.org  Sun Jun 10 15:25:18 2012
From: chris at opencsta.org (Chris Mylonas)
Date: Sun, 10 Jun 2012 21:25:18 +1000
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <CAKbxfG8a+0vBbA=yO2fUNkeWwqOFMSVYnGmB2c3ZzG50S-RWGQ@mail.gmail.com>
References: <17731604.448460.1339171050549.JavaMail.root@rockbochs.com>
	<-3835502817330869863@unknownmsgid> <4FD2569D.9080504@gmail.com>
	<CAKbxfG8a+0vBbA=yO2fUNkeWwqOFMSVYnGmB2c3ZzG50S-RWGQ@mail.gmail.com>
Message-ID: <E75CB347-A4E2-4F1F-AC5E-168E55D347A7@opencsta.org>

i think the mailing list work well.
i think the mailing list works very well when used in conjunction with IRC.

this thread has started to get to IRC standards in it's casual nature, although some networking (and not software or voip forums do get this way) follow this trend.

the "time and energy" anthm relates to is one of those seemingly frustrating tasks of not making the information available (or that becomes available) or not available in the future in a manner that "semantically" links it to all the rest of "related" topics ... if ya get my gist.

i guess this is why we have publishers ;)

the paying attention to 100 ML responses is the PITA in regards to an email client that doesn't read your mind is the ML problem...

my 2c

cheers
chris

On 09/06/2012, at 5:52 AM, Anthony Minessale wrote:

> we've looked at some alternate mailing lists that had some forum-like
> features but we need time and energy to pursue things like that.
> We need to get our branching and packaging done first.
> 
> I wish nabble and friends would actually go away, they horribly
> pollute the google searches and their purpose is to draw banner ads
> from our content.
> 
> 
> On Fri, Jun 8, 2012 at 2:46 PM, Timothy Bolton <ifoundthetao at gmail.com> wrote:
>> It can be much more fun to put off work and b.s. by the virtual water
>> cooler of an off-topic conversation. These topics lend themselves more
>> toward opinion than the knowledge required to answer a question.
>> 
>> 'We who cut mere stones must always be envisioning cathedrals.'
>> Quarry Worker's Creed
>> 
>> On 6/8/2012 2:43 PM, Brian West wrote:
>>> Why do topics like these get tons of interactions, but real issues
>>> fall to the wayside.  I think the horse is dead Jim, beam me up
>>> already.
>>> 
>>> Sent from my spaceship orbiting the moon (biplane upgrade)
>>> 
>>> On Jun 8, 2012, at 10:58 AM, Tim Nelson <tnelson at rockbochs.com> wrote:
>>> 
>>>> I was just about to suggest this approach as well. phpBB has a module to combine forums/mailing lists and it works swimmingly well. For a taste of operation, head over to the Vyatta forums/list [1] .
>>>> 
>>>> --Tim
>>>> 
>>>> [1] http://www.vyatta.org/forum/?
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> Join Us At ClueCon - Aug 7-9, 2012
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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.gathm at gmail.com  Sun Jun 10 23:48:38 2012
From: john.gathm at gmail.com (John Gathm)
Date: Sun, 10 Jun 2012 21:48:38 +0200
Subject: [Freeswitch-users] using APNS with diaplan : throttle a call
 until a certain condition changes
In-Reply-To: <CAMwZ-tLtXYEWbxo+R9ooudh+PZ-F_bnpPrtawZLMFjK13Z2ceQ@mail.gmail.com>
References: <CACU5aF6ma-W+VM98FzhS6i=kWHQhJPehpvPt1oS9E8662CaC2A@mail.gmail.com>
	<CAMwZ-tLtXYEWbxo+R9ooudh+PZ-F_bnpPrtawZLMFjK13Z2ceQ@mail.gmail.com>
Message-ID: <CACU5aF4rP_03cmKXCp5LgJ5mU=Ynxkz1zqLbLwCKKD3b-TJR8Q@mail.gmail.com>

Hi Gabriel,
Thanks for your answer

so to summarize, I should (I did tome tries)

1) kind of park if failure - kind of replace the voicemail

       <action application="bridge"
data="user/${user_data(${destination_number}@${domain_name} attr
id)}@${domain_name}" />
       <action application="answer" />
       <action application="park"/>
       <!--<action application="sleep" data="1000" />
       <action application="voicemail" data="default ${domain_name}
${dialed_extension}" />-->

2) subscribe to "event xml CHANNEL_PARK"

Then I can get
<Core-UUID>7e4fd02a-b260-11e1-9687-c13084909128</Core-UUID>
<variable_dialed_user>1234</variable_dialed_user> (there the extension my
dialplan initially transfered the call to)
<Unique-ID>43cca2b4-b311-11e1-a4d1-c13084909128</Unique-ID>
<Channel-Call-UUID>43cca2b4-b311-11e1-a4d1-c13084909128</Channel-Call-UUID>

==> which UDID is the one I should watch to be able to unpark the call ?

==> which command from mod_command should I then use to "resume" the call ?
or transfer/bridge again to extension 1234

Regards,

John


On Sat, Jun 9, 2012 at 3:16 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Fri, Jun 8, 2012 at 2:31 AM, John Gathm <john.gathm at gmail.com> wrote:
> > I am wondering if this implementation would be acceptable regarding
> > freeswitch design, or if diaplan/freeswitch offer cleaner ways to
> implement
> > it.
>
> What if you used your dialplan to try to call the phone and if you do
> NOT find it, park the call. Then you might use the event socket to
> watch for the parking of the call so you can send your notification.
> Next, use the event socket to watch for sofia registrations to know
> when the app has received the notification, launched and is ready to
> take calls. Once you get that event, unpark the call and send it on
> its way.
>
>
> Best,
> Gabe
>
> P.S. go python!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From avi at avimarcus.net  Mon Jun 11 00:31:15 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 10 Jun 2012 23:31:15 +0300
Subject: [Freeswitch-users] Context features not found -- some explanation?
Message-ID: <CALNZuwkxejFY234xK8JJ=LZ1oNWiVuQFtenhkwPN3sB1HBfh2w@mail.gmail.com>

2012-06-10 23:28:21.280407 [DEBUG] switch_ivr.c:598
sofia/internal/sip:1102 at 79.182.136.133:5076 Command Execute
execute_extension(att_xfer XML features)
EXECUTE sofia/internal/sip:1102 at 79.182.136.133:5076
execute_extension(att_xfer XML features)
2012-06-10 23:28:21.280407 [INFO] mod_dialplan_xml.c:485 Processing
Avi Marcus <1000>->att_xfer in context features
2012-06-10 23:28:21.290403 [WARNING] mod_dialplan_xml.c:515 Context
features not found

I tried *4 to use the default binding for an att_xfer - I turned it
back on in the static xml (+reloadxml) and put it into my xml_curl ...
hence my extreme surprise to find "context features not found". My
xml_curl testing script perfectly returns att_xfer in context
features. Is there some sort of permission issue with contexts that
I've never heard of..?

Thanks,
-Avi


From patrick at sunsus.net  Sun Jun 10 17:27:43 2012
From: patrick at sunsus.net (sunsus)
Date: Sun, 10 Jun 2012 06:27:43 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch
Message-ID: <1339334863709-7579670.post@n2.nabble.com>

Hello 

I've tryed to use netvoip.ch as a gateway. But I have a lot of connection
problems. Because netvoip has a load balancing between two servers
sip.netvoip.ch (sip-1.netvoip.ch and sip-2.netvoip.ch). 

Now the problem is if you are getting with the INVITE to the first server
you're getting a 304 Unauthorized with the nonce of the first server. For
some reason FreeSWITCH is now sending a INVITE with the digist
Authentication to the second server and this server is now answering with
304 Unauthorized with an other nonce. FreeSWITCH now ends the call with 500
internal server error. 

Is this a FreeSWITCH related issue or a netvoip related issue? Or is there
any configuration in the gateway to say to use the same server for the
authentication process? 

Regards

Patrick 


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-tp7579670.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From engelster at gmail.com  Sun Jun 10 23:43:41 2012
From: engelster at gmail.com (Der Engel)
Date: Sun, 10 Jun 2012 14:43:41 -0500
Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail
	list?
In-Reply-To: <E75CB347-A4E2-4F1F-AC5E-168E55D347A7@opencsta.org>
References: <17731604.448460.1339171050549.JavaMail.root@rockbochs.com>
	<-3835502817330869863@unknownmsgid> <4FD2569D.9080504@gmail.com>
	<CAKbxfG8a+0vBbA=yO2fUNkeWwqOFMSVYnGmB2c3ZzG50S-RWGQ@mail.gmail.com>
	<E75CB347-A4E2-4F1F-AC5E-168E55D347A7@opencsta.org>
Message-ID: <CAFf7Ch3BrgKz61fGMnASAqoD40G5N5PjQxPR8Z0ZsPPj9j5DUw@mail.gmail.com>

How about using google groups?

Is like a mailing list but has all the features people are requesting here.

On Sun, Jun 10, 2012 at 6:25 AM, Chris Mylonas <chris at opencsta.org> wrote:
> i think the mailing list work well.
> i think the mailing list works very well when used in conjunction with IRC.
>
> this thread has started to get to IRC standards in it's casual nature, although some networking (and not software or voip forums do get this way) follow this trend.
>
> the "time and energy" anthm relates to is one of those seemingly frustrating tasks of not making the information available (or that becomes available) or not available in the future in a manner that "semantically" links it to all the rest of "related" topics ... if ya get my gist.
>
> i guess this is why we have publishers ;)
>
> the paying attention to 100 ML responses is the PITA in regards to an email client that doesn't read your mind is the ML problem...
>
> my 2c
>
> cheers
> chris
>
> On 09/06/2012, at 5:52 AM, Anthony Minessale wrote:
>
>> we've looked at some alternate mailing lists that had some forum-like
>> features but we need time and energy to pursue things like that.
>> We need to get our branching and packaging done first.
>>
>> I wish nabble and friends would actually go away, they horribly
>> pollute the google searches and their purpose is to draw banner ads
>> from our content.
>>
>>
>> On Fri, Jun 8, 2012 at 2:46 PM, Timothy Bolton <ifoundthetao at gmail.com> wrote:
>>> It can be much more fun to put off work and b.s. by the virtual water
>>> cooler of an off-topic conversation. These topics lend themselves more
>>> toward opinion than the knowledge required to answer a question.
>>>
>>> 'We who cut mere stones must always be envisioning cathedrals.'
>>> Quarry Worker's Creed
>>>
>>> On 6/8/2012 2:43 PM, Brian West wrote:
>>>> Why do topics like these get tons of interactions, but real issues
>>>> fall to the wayside. ?I think the horse is dead Jim, beam me up
>>>> already.
>>>>
>>>> Sent from my spaceship orbiting the moon (biplane upgrade)
>>>>
>>>> On Jun 8, 2012, at 10:58 AM, Tim Nelson <tnelson at rockbochs.com> wrote:
>>>>
>>>>> I was just about to suggest this approach as well. phpBB has a module to combine forums/mailing lists and it works swimmingly well. For a taste of operation, head over to the Vyatta forums/list [1] .
>>>>>
>>>>> --Tim
>>>>>
>>>>> [1] http://www.vyatta.org/forum/?
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mel0torme at gmail.com  Mon Jun 11 01:28:47 2012
From: mel0torme at gmail.com (Tom C)
Date: Sun, 10 Jun 2012 14:28:47 -0700
Subject: [Freeswitch-users] (no subject)
Message-ID: <CAO7OxTxeQAmMUxko3+95+Qq3+6VDfdvXeEjy5KYo3jH7hZVHDQ@mail.gmail.com>

On the "Performance testing and configurations" page of the wiki, under
Recommended SIP Settings, it says the following:

libsofia only handles 1 thread per profile, so if that is your bottle
neck use more profiles


Is this still true?  It appears to me that every call, and maybe every
channel, is getting its own thread.  Or is the above comment referring to
call management rather than actually processing audio?
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From bdfoster at endigotech.com  Mon Jun 11 01:33:53 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 10 Jun 2012 17:33:53 -0400
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAO7OxTxeQAmMUxko3+95+Qq3+6VDfdvXeEjy5KYo3jH7hZVHDQ@mail.gmail.com>
References: <CAO7OxTxeQAmMUxko3+95+Qq3+6VDfdvXeEjy5KYo3jH7hZVHDQ@mail.gmail.com>
Message-ID: <CAKV-jFye+5cvZD8MBqgKTgzQzQH4tfL3DSuZiC61Y4H66hBCtQ@mail.gmail.com>

It's refering to SIP messages, audio gets its own threads. If that's your
bottleneck Id be very surprised.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 10, 2012 5:29 PM, "Tom C" <mel0torme at gmail.com> wrote:

> On the "Performance testing and configurations" page of the wiki, under
> Recommended SIP Settings, it says the following:
>
>
> libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles
>
>
> Is this still true?  It appears to me that every call, and maybe every
> channel, is getting its own thread.  Or is the above comment referring to
> call management rather than actually processing audio?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From drk at drkngs.net  Mon Jun 11 01:38:15 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Sun, 10 Jun 2012 14:38:15 -0700
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAO7OxTxeQAmMUxko3+95+Qq3+6VDfdvXeEjy5KYo3jH7hZVHDQ@mail.gmail.com>
Message-ID: <20120610213815.90abc494@mail.tritonwest.net>

That's out of date... someone wanna pay the Wiki tax?
      _____  

  From: Tom C [mailto:mel0torme at gmail.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Sun, 10 Jun 2012 14:28:47 -0700
Subject: [Freeswitch-users] (no subject)

On the "Performance testing and configurations" page of the wiki, under Recommended SIP Settings, it says the following:  

  


libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles  

  
Is this still true?  It appears to me that every call, and maybe every channel, is getting its own thread.  Or is the above comment referring to call management rather than actually processing audio?      
   
 
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From bdfoster at endigotech.com  Mon Jun 11 01:47:32 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 10 Jun 2012 17:47:32 -0400
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <20120610213815.90abc494@mail.tritonwest.net>
References: <CAO7OxTxeQAmMUxko3+95+Qq3+6VDfdvXeEjy5KYo3jH7hZVHDQ@mail.gmail.com>
	<20120610213815.90abc494@mail.tritonwest.net>
Message-ID: <CAKV-jFyD_CFhAa721pJVKD3=MdmLVxejCovtS_oibJPVPBcEvg@mail.gmail.com>

It's not out of date...

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 10, 2012 5:39 PM, "Dave R. Kompel" <drk at drkngs.net> wrote:

> **
> That's out of date... someone wanna pay the Wiki tax?
>
>  ------------------------------
> *From:* Tom C [mailto:mel0torme at gmail.com]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> *Sent:* Sun, 10 Jun 2012 14:28:47 -0700
> *Subject:* [Freeswitch-users] (no subject)
>
> On the "Performance testing and configurations" page of the wiki, under
> Recommended SIP Settings, it says the following:
>
> libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles
>
>
> Is this still true?  It appears to me that every call, and maybe every
> channel, is getting its own thread.  Or is the above comment referring to
> call management rather than actually processing audio?
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From drk at drkngs.net  Mon Jun 11 02:18:02 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Sun, 10 Jun 2012 15:18:02 -0700
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAKV-jFyD_CFhAa721pJVKD3=MdmLVxejCovtS_oibJPVPBcEvg@mail.gmail.com>
Message-ID: <20120610221802.371d0f51@mail.tritonwest.net>

I remember like the end of last year Tony doing a lot of work to make parts of it not single threaded. 
      _____  

  From: Brian Foster [mailto:bdfoster at endigotech.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Sun, 10 Jun 2012 14:47:32 -0700
Subject: Re: [Freeswitch-users] (no subject)

  

It's not out of date...  

Brian Foster
Endigo Computer LLC  

Sent from a mobile device.  
On Jun 10, 2012 5:39 PM, "Dave R. Kompel" <drk at drkngs.net> wrote:
    
  
That's out of date... someone wanna pay the Wiki tax?
      _____  

  From: Tom C [mailto:mel0torme at gmail.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Sun, 10 Jun 2012 14:28:47 -0700
Subject: [Freeswitch-users] (no subject)

On the "Performance testing and configurations" page of the wiki, under Recommended SIP Settings, it says the following:   

  


libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles  

  
Is this still true?  It appears to me that every call, and maybe every channel, is getting its own thread.  Or is the above comment referring to call management rather than actually processing audio?  
   
 
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

      
   
 
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From mel0torme at gmail.com  Mon Jun 11 03:09:54 2012
From: mel0torme at gmail.com (Tom C)
Date: Sun, 10 Jun 2012 16:09:54 -0700
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch
In-Reply-To: <1339334863709-7579670.post@n2.nabble.com>
References: <1339334863709-7579670.post@n2.nabble.com>
Message-ID: <CAO7OxTwFdpNS9Yjpbg2Do3gN9QA3J9JRDyHn_us1bhyx4Kay_Q@mail.gmail.com>

Not sure if it's the same problem, but it sounds similar to the issues we
had a couple years ago getting FreeSwitch to work with Whistlephone.
 Whistlephone also used multiple servers for load balancing, and while
outgoing calls worked fine, incoming calls were unreliable.

For more technical details, you can search this mailing list for
Whistlephone, about a year and a half ago.

The solution was to add a separate gateway for each proxy server.

So, in conf/sip_profiles/external, I added whistlephone.xml, with the stuff
below.  Pay attention to proxy vs proxy1 in the server names, and the added
"from-domain" in the secondary gateways.  Using the "realm" parameter
seemed to help me, but it took experimentation to find what worked.

--------------------------------------------------------------------
<code>

<include>
<gateway name="WhistlePhone">
   <param name="username" value="xxxxxxxxxxx"/>
   <param name="from-user" value="xxxxxxxxxx"/>
   <param name="proxy" value="proxy.WhistlePhone.com"/>
   <param name="realm" value="WhistlePhone"/>
   <param name="password" value="xxxx"/>
   <param name="expire-seconds" value="600"/>
   <param name="register" value="true"/>
   <param name="retry-seconds" value="30"/>
</gateway>

<gateway name="WhistlePhoneProxy1">
   <param name="username" value="xxxxxxxxxxx"/>
   <param name="from-user" value="xxxxxxxxxx"/>
   <param name="proxy" value="proxy1.WhistlePhone.com"/>
   <param name="from-domain" value="proxy.Whistlephone.com"/>
   <param name="password" value="xxxx"/>
   <param name="expire-seconds" value="600"/>
   <param name="register" value="true"/>
   <param name="retry-seconds" value="30"/>
</gateway>

 repeat for each proxy server.....

</include>

</code>
------------------------------------------------------------------------
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From peter.olsson at visionutveckling.se  Mon Jun 11 09:41:40 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 11 Jun 2012 05:41:40 +0000
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <20120610221802.371d0f51@mail.tritonwest.net>
References: <CAKV-jFyD_CFhAa721pJVKD3=MdmLVxejCovtS_oibJPVPBcEvg@mail.gmail.com>,
	<20120610221802.371d0f51@mail.tritonwest.net>
Message-ID: <E80A41DB-AAD7-49AD-955D-E4FD0D4E5FBF@visionutveckling.se>

Yes, Tony added extra signalling threads within FS, however, libsofia is still limited to one thread per profile.

So, now you have one thread per sofia profile, a few signalling threads withing FS (to where events are queued from sofia), and of course, one thread per channel, for media handling etc.



/Peter

11 jun 2012 kl. 00:26 skrev "Dave R. Kompel" <drk at drkngs.net<mailto:drk at drkngs.net>>:

I remember like the end of last year Tony doing a lot of work to make parts of it not single threaded.

________________________________
From: Brian Foster [mailto:bdfoster at endigotech.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Sun, 10 Jun 2012 14:47:32 -0700
Subject: Re: [Freeswitch-users] (no subject)


It's not out of date...

Brian Foster
Endigo Computer LLC

Sent from a mobile device.

On Jun 10, 2012 5:39 PM, "Dave R. Kompel" <drk at drkngs.net<mailto:drk at drkngs.net>> wrote:
That's out of date... someone wanna pay the Wiki tax?

________________________________
From: Tom C [mailto:mel0torme at gmail.com<mailto:mel0torme at gmail.com>]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>]
Sent: Sun, 10 Jun 2012 14:28:47 -0700
Subject: [Freeswitch-users] (no subject)

On the "Performance testing and configurations" page of the wiki, under Recommended SIP Settings, it says the following:


libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles

Is this still true?  It appears to me that every call, and maybe every channel, is getting its own thread.  Or is the above comment referring to call management rather than actually processing audio?



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>


</>

Official FreeSWITCH Sites
http://www.freeswitch.org<http://www.freeswitch.org/>
http://wiki.freeswitch.org<http://wiki.freeswitch.org/>
http://www.cluecon.com<http://www.cluecon.com/>

Join Us At ClueCon - Aug 7-9, 2012

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/>



!DSPAM:4fd51bea32761541018091!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


!DSPAM:4fd51bea32761541018091!


From evgeniy at bestnet.kharkov.ua  Mon Jun 11 10:01:29 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Mon, 11 Jun 2012 09:01:29 +0300
Subject: [Freeswitch-users] mod_nibblebill problem
In-Reply-To: <CAKV-jFxT1UdG1xea8CEsKodHZ3+zn+TczgDOg9BvK5FKtY5P-w@mail.gmail.com>
References: <4FD199DA.7080805@bestnet.kharkov.ua>
	<4FD1A7FA.3090009@bestnet.kharkov.ua>
	<CAKV-jFxT1UdG1xea8CEsKodHZ3+zn+TczgDOg9BvK5FKtY5P-w@mail.gmail.com>
Message-ID: <4FD589B9.5010909@bestnet.kharkov.ua>

I'm answered the call, but still gets this message: "Not billing
7604504 - call is not in answered state".

08.06.2012 19:58, Brian Foster ???????:
> Calls aren't billed when the phone is ringing, only id/when call
> is answered.
> 
> Brian Foster Endigo Computer LLC
> 
> Sent from a mobile device. On Jun 8, 2012 11:26 AM, "Evgeniy
> Movlyan" <evgeniy at bestnet.kharkov.ua> wrote:
> 
>> Hi All, when i'm calling from one user to another freeswitch user
>> mod works fine, but when i'm calling to external number (through
>> gateway) billing doesn't work:
>> 
>> 2012-06-07 16:16:07.294840 [DEBUG] mod_nibblebill.c:612 Received
>> request via SESSION_HEARTBEAT! 2012-06-07 16:16:07.294840 [DEBUG]
>> mod_nibblebill.c:453 Attempting to bill at $1 per minute to
>> account 7604504 2012-06-07 16:16:07.294840 [DEBUG]
>> mod_nibblebill.c:465 Not billing 7604504 - call is not in
>> answered state
>> 
>> I don't understand what this message means.
>> 
>> Sorry for my english:) -- Evgeniy Movlyan, BestNet Ltd.
>> 
>> _________________________________________________________________________
>>
>> 
Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org http://www.freeswitchsolutions.com
>> 
>>  
>> 
>> 
>> Official FreeSWITCH Sites http://www.freeswitch.org 
>> http://wiki.freeswitch.org http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list 
>> FreeSWITCH-users at lists.freeswitch.org 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> 
http://www.freeswitch.org
>> 
> 
> 
> 
> _________________________________________________________________________
>
> 
Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org http://www.freeswitchsolutions.com
> 
>  
> 
> 
> Official FreeSWITCH Sites http://www.freeswitch.org 
> http://wiki.freeswitch.org http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list 
> FreeSWITCH-users at lists.freeswitch.org 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> 
http://www.freeswitch.org

-- 
Evgeniy Movlyan,
BestNet Ltd.


From patrick at sunsus.net  Mon Jun 11 10:08:28 2012
From: patrick at sunsus.net (=?ISO-8859-1?Q?Patrick_D=FCrsteler?=)
Date: Mon, 11 Jun 2012 08:08:28 +0200
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch
In-Reply-To: <CAO7OxTwFdpNS9Yjpbg2Do3gN9QA3J9JRDyHn_us1bhyx4Kay_Q@mail.gmail.com>
References: <1339334863709-7579670.post@n2.nabble.com>
	<CAO7OxTwFdpNS9Yjpbg2Do3gN9QA3J9JRDyHn_us1bhyx4Kay_Q@mail.gmail.com>
Message-ID: <CAFLdx6MqmuUHvKxwn2x+qszKVx_24ytHNp0Hr375hp2byikDcw@mail.gmail.com>

Hi Tom

Thanks for your answer, but we have the problem with the outgoing calls.
See attached sip trace.

regards

Patrick

On Mon, Jun 11, 2012 at 1:09 AM, Tom C <mel0torme at gmail.com> wrote:

> Not sure if it's the same problem, but it sounds similar to the issues we
> had a couple years ago getting FreeSwitch to work with Whistlephone.
>  Whistlephone also used multiple servers for load balancing, and while
> outgoing calls worked fine, incoming calls were unreliable.
>
> For more technical details, you can search this mailing list for
> Whistlephone, about a year and a half ago.
>
> The solution was to add a separate gateway for each proxy server.
>
> So, in conf/sip_profiles/external, I added whistlephone.xml, with the
> stuff below.  Pay attention to proxy vs proxy1 in the server names, and the
> added "from-domain" in the secondary gateways.  Using the "realm" parameter
> seemed to help me, but it took experimentation to find what worked.
>
> --------------------------------------------------------------------
> <code>
>
> <include>
> <gateway name="WhistlePhone">
>    <param name="username" value="xxxxxxxxxxx"/>
>    <param name="from-user" value="xxxxxxxxxx"/>
>    <param name="proxy" value="proxy.WhistlePhone.com"/>
>    <param name="realm" value="WhistlePhone"/>
>    <param name="password" value="xxxx"/>
>    <param name="expire-seconds" value="600"/>
>    <param name="register" value="true"/>
>    <param name="retry-seconds" value="30"/>
> </gateway>
>
> <gateway name="WhistlePhoneProxy1">
>    <param name="username" value="xxxxxxxxxxx"/>
>    <param name="from-user" value="xxxxxxxxxx"/>
>    <param name="proxy" value="proxy1.WhistlePhone.com"/>
>    <param name="from-domain" value="proxy.Whistlephone.com"/>
>    <param name="password" value="xxxx"/>
>    <param name="expire-seconds" value="600"/>
>    <param name="register" value="true"/>
>    <param name="retry-seconds" value="30"/>
> </gateway>
>
>  repeat for each proxy server.....
>
> </include>
>
> </code>
> ------------------------------------------------------------------------
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From saami_mh at ymail.com  Mon Jun 11 12:15:45 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Mon, 11 Jun 2012 01:15:45 -0700 (PDT)
Subject: [Freeswitch-users] how to use  codec g729 on freeswitch ?
Message-ID: <1339402545.76120.YahooMailNeo@web120102.mail.ne1.yahoo.com>

hi guys,
i am going to make call using codec g729,
because the carrier ?support only codec g729
is it possible to get free that?
in freeswitch implement only g729 passthrough .
but how can i get it ?
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From awais-nazeer at hotmail.com  Mon Jun 11 13:43:05 2012
From: awais-nazeer at hotmail.com (awais nazir)
Date: Mon, 11 Jun 2012 14:43:05 +0500
Subject: [Freeswitch-users] Gateway failover and ports limitation
Message-ID: <BLU162-W166D7524853458B5E54BEA88F70@phx.gbl>


Hi

Can u help me a bit here. I have a scenario like

GW_priority_1_30_ports
GW_priority_2_60_ports


So if first gatway is down or fails with giving cause code (480 
or 503 ) OR its 30 channels are occupied then 
next carrier should be selected. And so on.

I can see failover config in dial plan http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover . But blending it with limit and doing failover on Gateway down and on the basis of some cause code is confusing.

Your help and hints are appreciated.

--waisee
 		 	   		  
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From peter.olsson at visionutveckling.se  Mon Jun 11 13:51:46 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 11 Jun 2012 09:51:46 +0000
Subject: [Freeswitch-users] how to use  codec g729 on freeswitch ?
Message-ID: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>

The passthrough codec for G.729 is built by default, just make sure to load the module mod_g729. If you need transcoding, or use IVR's, voicemail etc, you must purchase commercial licenses for G729.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh
Skickat: den 11 juni 2012 10:16
Till: Free SWITCH Users Help
?mne: [Freeswitch-users] how to use codec g729 on freeswitch ?

hi guys,
i am going to make call using codec g729,
because the carrier  support only codec g729
is it possible to get free that?
in freeswitch implement only g729 passthrough .
but how can i get it ?
!DSPAM:4fd5a7d032762000267653!
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From peter.olsson at visionutveckling.se  Mon Jun 11 13:56:14 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 11 Jun 2012 09:56:14 +0000
Subject: [Freeswitch-users] Gateway failover and ports limitation
Message-ID: <1FFF97C269757C458224B7C895F35F15102499@cantor.std.visionutv.se>

If the provider only allows you to use 30 channels, I guess they will send a specific error to you if you try to do more calls, so just handle that error as you handle 480, 503 etc.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r awais nazir
Skickat: den 11 juni 2012 11:43
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Gateway failover and ports limitation

Hi

Can u help me a bit here. I have a scenario like

GW_priority_1_30_ports
GW_priority_2_60_ports


So if first gatway is down or fails with giving cause code (480 or 503 ) OR its 30 channels are occupied then next carrier should be selected. And so on.

I can see failover config in dial plan http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover . But blending it with limit and doing failover on Gateway down and on the basis of some cause code is confusing.

Your help and hints are appreciated.

--waisee
!DSPAM:4fd5bc0432761819510119!
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From torstein.knutsen at gmail.com  Mon Jun 11 14:01:08 2012
From: torstein.knutsen at gmail.com (Torstein Knutsen)
Date: Mon, 11 Jun 2012 12:01:08 +0200
Subject: [Freeswitch-users] Controlling caller_id_name
In-Reply-To: <CAKV-jFy2sN5fZs2prpdE0wD10CJXVfFgWQFQxg7iM_GJ8_SxXw@mail.gmail.com>
References: <CAMnjH4QKpTbDw+pqwV0ZH3G4zS82AVBU_zs=rLFNDTRD+gh-0g@mail.gmail.com>
	<CAKV-jFwssLbwcFHOchkkwzFLckAzt6cmDsy3vY_wpAES4GehyA@mail.gmail.com>
	<CAMnjH4SskCTc641+bMbMQRAP0X9qk42O6M4eELHYmkVrTMDGAw@mail.gmail.com>
	<CAKV-jFy2sN5fZs2prpdE0wD10CJXVfFgWQFQxg7iM_GJ8_SxXw@mail.gmail.com>
Message-ID: <CAMnjH4RfUyooOutm2i5bCEByAVU6Es9d1ELBijfMUN5C=gc3ow@mail.gmail.com>

Hi

I tried :
<action inline="true" application="set"
data="effective_caller_id_name=Hobbes"/>
I tried :
<action inline="true" application="set" data="caller_id_name=Hobbes"/>
I tried :
<action inline="true" application="set"
data="origination_caller_id_name=Hobbes"/>

But fs_cli "conference xml_list" still shows the "original" caller_id in
the conference detail listing on all attempts...

Thank you for your continued support :-) !
-Torstein

On 6 June 2012 19:02, Brian Foster <bdfoster at endigotech.com> wrote:

> Crap... set the caller id inline not the conference line... sorry got a
> lot going on today lol
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 6, 2012 12:56 PM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
> wrote:
>
>> Hi
>>
>> Sure?  I did try that, but it does not work .. Im not doing bridge, but
>> conference .. ..
>>
>> -T
>> On Jun 6, 2012 5:06 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>>
>>> You need to be setting effective_caller_id_name before the bridge.
>>>
>>> Brian Foster
>>> Endigo Computer LLC
>>>
>>> Sent from a mobile device.
>>> On Jun 6, 2012 6:19 AM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
>>> wrote:
>>>
>>>> Hi
>>>>
>>>> I'm trying to control the "caller_id_name" variable used in
>>>> conferencing.
>>>> My dialplan looks like this :
>>>>
>>>> <extension name="public_extensions">
>>>> <condition field="destination_number" expression="^(0meet)$">
>>>> <action application="conference" data="72013 at webconf"/>
>>>>
>>>> When I call into the conference, I do an API "conference
>>>> xml_list" which gives details of the conferencees ..
>>>> I want to be able to change the caller_id_name in the listing, but all
>>>> my efforts for doing this via dialplan fails.
>>>>
>>>> Is this even possible somehow ?
>>>>
>>>> I tried to add this to the dialplan before executing the conference :
>>>> <action application="export" data="caller_id_name=Hobbes" />
>>>> <action application="set" data="caller_id_name=Hobbes" />
>>>> <action application="export" data="effective_caller_id_name=Hobbes" />
>>>> <action application="export" data="effective_caller_id_name=Hobbes" />
>>>> <action application="set" data="origination_caller_id_name=Hobbes" />
>>>> <action application="export" data="origination_caller_id_name=Hobbes" />
>>>>
>>>> But to no avail...
>>>>
>>>> Any pointers would be appreciated.
>>>>
>>>> Thank you
>>>> Torstein
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From saami_mh at ymail.com  Mon Jun 11 14:13:31 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Mon, 11 Jun 2012 03:13:31 -0700 (PDT)
Subject: [Freeswitch-users] how to use  codec g729 on freeswitch ?
In-Reply-To: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
Message-ID: <1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>

but for making call from one country to another need to set codec to G729(no passthrough) because the carrier support only G729 ?and i want the free one not licensing ,
is it posiible to use media proxy so that use G729 without need to use??commercial license?


________________________________
 From: Peter Olsson <peter.olsson at visionutveckling.se>
To: Free SWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Monday, June 11, 2012 2:21 PM
Subject: Re: [Freeswitch-users] how to use  codec g729 on freeswitch ?
 

 
The passthrough codec for G.729 is built by default, just make sure to load the module mod_g729. If you need transcoding, or use IVR?s, voicemail etc, you must purchase commercial licenses for G729.
?
/Peter
?
?
Fr?n:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh
Skickat: den 11 juni 2012 10:16
Till: Free SWITCH Users Help
?mne: [Freeswitch-users] how to use codec g729 on freeswitch ?
?
hi guys,
i am going to make call using codec g729,
because the carrier ?support only codec g729
is it possible to get free that?
in freeswitch implement only g729 passthrough .
but how can i get it ?
!DSPAM:4fd5a7d032762000267653! 
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
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From awais-nazeer at hotmail.com  Mon Jun 11 14:47:28 2012
From: awais-nazeer at hotmail.com (awais nazir)
Date: Mon, 11 Jun 2012 15:47:28 +0500
Subject: [Freeswitch-users] Gateway failover and ports limitation
Message-ID: <BLU162-W226945D20696AEAD94430C88F70@phx.gbl>


Hi, 

Provider expects me to control limit at my end. I have to do the scenario with static dialplan not with database/odbc.

Thanks in advance for helping.



On Mon, Jun 11, 2012 at 2:56 PM, Peter Olsson <peter.olsson at visionutveckling.se> wrote:








If
 the provider only allows you to use 30 channels, I guess they will send
 a specific error to you if you try to do more calls, so just handle
 that error as you handle 480, 503 etc.
 
/Peter
 


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]
F?r awais nazir

Skickat: den 11 juni 2012 11:43

Till: freeswitch-users at lists.freeswitch.org

?mne: [Freeswitch-users] Gateway failover and ports limitation


 

Hi



Can u help me a bit here. I have a scenario like



GW_priority_1_30_ports

GW_priority_2_60_ports




So if first gatway is down or fails with giving cause code (480 or 503 )
 OR its 30 channels are occupied then next carrier should be selected. 
And so on.



I can see failover config in dial plan 
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover . But blending it with limit and doing failover on Gateway down and on the basis of some cause code is confusing.



Your help and hints are appreciated.



--waisee

!DSPAM:4fd5bc0432761819510119!




 		 	   		  
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From gcd at i.ph  Mon Jun 11 15:17:31 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Mon, 11 Jun 2012 19:17:31 +0800
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
Message-ID: <CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>

samira,

if your IP phone, ATA or gateway supports G729, then you can use FS free
G729 codec. just make sure the codec is set in the dialplan.

-nandy

On Mon, Jun 11, 2012 at 6:13 PM, Samira Mh <saami_mh at ymail.com> wrote:

> but for making call from one country to another need to set codec to
> G729(no passthrough) because the carrier support only G729  and i want the
> free one not licensing ,
> is it posiible to use media proxy so that use G729 without need to use  commercial
> license?
>
>   ------------------------------
> *From:* Peter Olsson <peter.olsson at visionutveckling.se>
> *To:* Free SWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, June 11, 2012 2:21 PM
> *Subject:* Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
>
>   The passthrough codec for G.729 is built by default, just make sure to
> load the module mod_g729. If you need transcoding, or use IVR?s, voicemail
> etc, you must purchase commercial licenses for G729.
>
> /Peter
>
>
>  *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Samira Mh
> *Skickat:* den 11 juni 2012 10:16
> *Till:* Free SWITCH Users Help
> *?mne:* [Freeswitch-users] how to use codec g729 on freeswitch ?
>
>  hi guys,
>  i am going to make call using codec g729,
>  because the carrier  support only codec g729
>  is it possible to get free that?
>  in freeswitch implement only g729 passthrough .
>  but how can i get it ?
>  !DSPAM:4fd5a7d032762000267653!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From saami_mh at ymail.com  Mon Jun 11 15:33:15 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Mon, 11 Jun 2012 04:33:15 -0700 (PDT)
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
Message-ID: <1339414395.62082.YahooMailNeo@web120105.mail.ne1.yahoo.com>

yes, my voipgateway support G729 and i set "G729" on freeswitch but when making call the error occure:
G729 only for passthrough !!


________________________________
 From: Nandy Dagondon <gcd at i.ph>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Monday, June 11, 2012 3:47 PM
Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
 

samira,

if your IP phone, ATA or gateway supports G729, then you can use FS free G729 codec. just make sure the codec is set in the dialplan.

-nandy


On Mon, Jun 11, 2012 at 6:13 PM, Samira Mh <saami_mh at ymail.com> wrote:

but for making call from one country to another need to set codec to G729(no passthrough) because the carrier support only G729 ?and i want the free one not licensing ,
>is it posiible to use media proxy so that use G729 without need to use??commercial license?
>
>
>
>________________________________
> From: Peter Olsson <peter.olsson at visionutveckling.se>
>To: Free SWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Monday, June 11, 2012 2:21 PM
>Subject: Re: [Freeswitch-users] how to use  codec g729 on freeswitch ?
> 
>
>
>The passthrough codec for G.729 is built by default, just make sure to load the module mod_g729. If you need transcoding, or use IVR?s, voicemail etc, you must purchase commercial licenses for G729.
>?
>/Peter
>?
>?
>Fr?n:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh
>Skickat: den 11 juni 2012 10:16
>Till: Free SWITCH Users Help
>?mne: [Freeswitch-users] how to use codec g729 on freeswitch ?
>?
>hi guys,
>i am going to make call using codec g729,
>because the carrier ?support only codec g729
>is it possible to get free that?
>in freeswitch implement only g729 passthrough .
>but how can i get it ?
>!DSPAM:4fd5a7d032762000267653! 
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing 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 chris at opencsta.org  Mon Jun 11 15:40:38 2012
From: chris at opencsta.org (Chris Mylonas)
Date: Mon, 11 Jun 2012 21:40:38 +1000
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <1339414395.62082.YahooMailNeo@web120105.mail.ne1.yahoo.com>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
	<1339414395.62082.YahooMailNeo@web120105.mail.ne1.yahoo.com>
Message-ID: <30DFAA60-ADB7-4B3A-A5FE-9085491FD83E@opencsta.org>

Are you bridging a call to this gateway or are you playing a soundfile to it?

"passthrough" means no file based stuff is to happen on the FS server - only the codec can pass through the switch.
in order to play "files" or record to a file, you will need a g729 license.

HTH
Chris



On 11/06/2012, at 9:33 PM, Samira Mh wrote:

> yes, my voipgateway support G729 and i set "G729" on freeswitch but when making call the error occure:
> G729 only for passthrough !!
> 
> From: Nandy Dagondon <gcd at i.ph>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
> Sent: Monday, June 11, 2012 3:47 PM
> Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
> 
> samira,
> 
> if your IP phone, ATA or gateway supports G729, then you can use FS free G729 codec. just make sure the codec is set in the dialplan.
> 
> -nandy
> 
> On Mon, Jun 11, 2012 at 6:13 PM, Samira Mh <saami_mh at ymail.com> wrote:
> but for making call from one country to another need to set codec to G729(no passthrough) because the carrier support only G729  and i want the free one not licensing ,
> is it posiible to use media proxy so that use G729 without need to use  commercial license?
> 
> From: Peter Olsson <peter.olsson at visionutveckling.se>
> To: Free SWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
> Sent: Monday, June 11, 2012 2:21 PM
> Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
> 
> The passthrough codec for G.729 is built by default, just make sure to load the module mod_g729. If you need transcoding, or use IVR?s, voicemail etc, you must purchase commercial licenses for G729.
>  
> /Peter
>  
>  
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh
> Skickat: den 11 juni 2012 10:16
> Till: Free SWITCH Users Help
> ?mne: [Freeswitch-users] how to use codec g729 on freeswitch ?
>  
> hi guys,
> i am going to make call using codec g729,
> because the carrier  support only codec g729
> is it possible to get free that?
> in freeswitch implement only g729 passthrough .
> but how can i get it ?
> !DSPAM:4fd5a7d032762000267653!
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From alex at digitalmail.com  Mon Jun 11 15:47:54 2012
From: alex at digitalmail.com (Alex Lake)
Date: Mon, 11 Jun 2012 12:47:54 +0100
Subject: [Freeswitch-users] Only calling the first element in the list in
	this mode
Message-ID: <4FD5DAEA.4000601@digitalmail.com>

Having upgraded to the latest build (we were previously on a build from 
2011), I'm finding that multiple SIP registrations are now working, but 
that multi-destination bridge commands are not.

Getting messages
2012-06-11 11:45:00.565775 [WARNING] switch_ivr_originate.c:2353 Only 
calling the first element in the list in this mode.

Is there some special "allow multi destination" setting that I need to add?

Rgds,
Alex


From bdfoster at endigotech.com  Mon Jun 11 16:03:00 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 11 Jun 2012 08:03:00 -0400
Subject: [Freeswitch-users] Controlling caller_id_name
In-Reply-To: <CAMnjH4RfUyooOutm2i5bCEByAVU6Es9d1ELBijfMUN5C=gc3ow@mail.gmail.com>
References: <CAMnjH4QKpTbDw+pqwV0ZH3G4zS82AVBU_zs=rLFNDTRD+gh-0g@mail.gmail.com>
	<CAKV-jFwssLbwcFHOchkkwzFLckAzt6cmDsy3vY_wpAES4GehyA@mail.gmail.com>
	<CAMnjH4SskCTc641+bMbMQRAP0X9qk42O6M4eELHYmkVrTMDGAw@mail.gmail.com>
	<CAKV-jFy2sN5fZs2prpdE0wD10CJXVfFgWQFQxg7iM_GJ8_SxXw@mail.gmail.com>
	<CAMnjH4RfUyooOutm2i5bCEByAVU6Es9d1ELBijfMUN5C=gc3ow@mail.gmail.com>
Message-ID: <CAKV-jFxSg7Rda0tYvzQT8ZxUNVHvCAPM7LOQVi9tfuiRS_H_fA@mail.gmail.com>

<action application="set" data="efffective_caller_id_name=Hobbes"
inline="true" />

This might be a case where you need to do a loopback.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 11, 2012 6:01 AM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
wrote:

> Hi
>
> I tried :
> <action inline="true" application="set"
> data="effective_caller_id_name=Hobbes"/>
> I tried :
> <action inline="true" application="set" data="caller_id_name=Hobbes"/>
> I tried :
> <action inline="true" application="set"
> data="origination_caller_id_name=Hobbes"/>
>
> But fs_cli "conference xml_list" still shows the "original" caller_id in
> the conference detail listing on all attempts...
>
> Thank you for your continued support :-) !
> -Torstein
>
> On 6 June 2012 19:02, Brian Foster <bdfoster at endigotech.com> wrote:
>
>> Crap... set the caller id inline not the conference line... sorry got a
>> lot going on today lol
>>
>> Brian Foster
>> Endigo Computer LLC
>>
>> Sent from a mobile device.
>> On Jun 6, 2012 12:56 PM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
>> wrote:
>>
>>> Hi
>>>
>>> Sure?  I did try that, but it does not work .. Im not doing bridge, but
>>> conference .. ..
>>>
>>> -T
>>> On Jun 6, 2012 5:06 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>>>
>>>> You need to be setting effective_caller_id_name before the bridge.
>>>>
>>>> Brian Foster
>>>> Endigo Computer LLC
>>>>
>>>> Sent from a mobile device.
>>>> On Jun 6, 2012 6:19 AM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
>>>> wrote:
>>>>
>>>>> Hi
>>>>>
>>>>> I'm trying to control the "caller_id_name" variable used in
>>>>> conferencing.
>>>>> My dialplan looks like this :
>>>>>
>>>>> <extension name="public_extensions">
>>>>> <condition field="destination_number" expression="^(0meet)$">
>>>>> <action application="conference" data="72013 at webconf"/>
>>>>>
>>>>> When I call into the conference, I do an API "conference
>>>>> xml_list" which gives details of the conferencees ..
>>>>> I want to be able to change the caller_id_name in the listing, but all
>>>>> my efforts for doing this via dialplan fails.
>>>>>
>>>>> Is this even possible somehow ?
>>>>>
>>>>> I tried to add this to the dialplan before executing the conference :
>>>>> <action application="export" data="caller_id_name=Hobbes" />
>>>>> <action application="set" data="caller_id_name=Hobbes" />
>>>>> <action application="export" data="effective_caller_id_name=Hobbes" />
>>>>> <action application="export" data="effective_caller_id_name=Hobbes" />
>>>>> <action application="set" data="origination_caller_id_name=Hobbes" />
>>>>> <action application="export" data="origination_caller_id_name=Hobbes"
>>>>> />
>>>>>
>>>>> But to no avail...
>>>>>
>>>>> Any pointers would be appreciated.
>>>>>
>>>>> Thank you
>>>>> Torstein
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Mon Jun 11 16:10:32 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 11 Jun 2012 08:10:32 -0400
Subject: [Freeswitch-users] Only calling the first element in the list
 in this mode
In-Reply-To: <4FD5DAEA.4000601@digitalmail.com>
References: <4FD5DAEA.4000601@digitalmail.com>
Message-ID: <CAKV-jFwX7qV73-VCNhzco0Jyb-zmCE8C4AFQjPnc9uKYCZvkZw@mail.gmail.com>

Set multiple-registrations to true in the affected Sofia profile if you
haven't already done so and report back.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 11, 2012 7:49 AM, "Alex Lake" <alex at digitalmail.com> wrote:

> Having upgraded to the latest build (we were previously on a build from
> 2011), I'm finding that multiple SIP registrations are now working, but
> that multi-destination bridge commands are not.
>
> Getting messages
> 2012-06-11 11:45:00.565775 [WARNING] switch_ivr_originate.c:2353 Only
> calling the first element in the list in this mode.
>
> Is there some special "allow multi destination" setting that I need to add?
>
> Rgds,
> Alex
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bdfoster at endigotech.com  Mon Jun 11 16:14:21 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 11 Jun 2012 08:14:21 -0400
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <30DFAA60-ADB7-4B3A-A5FE-9085491FD83E@opencsta.org>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
	<1339414395.62082.YahooMailNeo@web120105.mail.ne1.yahoo.com>
	<30DFAA60-ADB7-4B3A-A5FE-9085491FD83E@opencsta.org>
Message-ID: <CAKV-jFxHEYwM8cj7tNPmPytRf9_41N6z2gH4xuk2dCctnP-sDQ@mail.gmail.com>

Adding to comments by Chris, this includes voicemail, faxing, basically
anything media related that freeswitch would have to accomplish on it's
own. If your getting that message then you are in fact doing something
media relates inside freeswitch and its telling you that you've been a
naughty, naughty boy.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 11, 2012 7:41 AM, "Chris Mylonas" <chris at opencsta.org> wrote:

> Are you bridging a call to this gateway or are you playing a soundfile to
> it?
>
> "passthrough" means no file based stuff is to happen on the FS server -
> only the codec can pass through the switch.
> in order to play "files" or record to a file, you will need a g729 license.
>
> HTH
> Chris
>
>
>
> On 11/06/2012, at 9:33 PM, Samira Mh wrote:
>
> yes, my voipgateway support G729 and i set "G729" on freeswitch but when
> making call the error occure:
> G729 only for passthrough !!
>
>   ------------------------------
> *From:* Nandy Dagondon <gcd at i.ph>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, June 11, 2012 3:47 PM
> *Subject:* Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
>
> samira,
>
> if your IP phone, ATA or gateway supports G729, then you can use FS free
> G729 codec. just make sure the codec is set in the dialplan.
>
> -nandy
>
> On Mon, Jun 11, 2012 at 6:13 PM, Samira Mh <saami_mh at ymail.com> wrote:
>
> but for making call from one country to another need to set codec to
> G729(no passthrough) because the carrier support only G729  and i want the
> free one not licensing ,
> is it posiible to use media proxy so that use G729 without need to use  commercial
> license?
>
>   ------------------------------
> *From:* Peter Olsson <peter.olsson at visionutveckling.se>
> *To:* Free SWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, June 11, 2012 2:21 PM
> *Subject:* Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
>
>   The passthrough codec for G.729 is built by default, just make sure to
> load the module mod_g729. If you need transcoding, or use IVR?s, voicemail
> etc, you must purchase commercial licenses for G729.
>
> /Peter
>
>
>  *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Samira Mh
> *Skickat:* den 11 juni 2012 10:16
> *Till:* Free SWITCH Users Help
> *?mne:* [Freeswitch-users] how to use codec g729 on freeswitch ?
>
>  hi guys,
>  i am going to make call using codec g729,
>  because the carrier  support only codec g729
>  is it possible to get free that?
>  in freeswitch implement only g729 passthrough .
>  but how can i get it ?
>  !DSPAM:4fd5a7d032762000267653!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From torstein.knutsen at gmail.com  Mon Jun 11 16:58:00 2012
From: torstein.knutsen at gmail.com (Torstein Knutsen)
Date: Mon, 11 Jun 2012 14:58:00 +0200
Subject: [Freeswitch-users] Controlling caller_id_name
In-Reply-To: <CAKV-jFxSg7Rda0tYvzQT8ZxUNVHvCAPM7LOQVi9tfuiRS_H_fA@mail.gmail.com>
References: <CAMnjH4QKpTbDw+pqwV0ZH3G4zS82AVBU_zs=rLFNDTRD+gh-0g@mail.gmail.com>
	<CAKV-jFwssLbwcFHOchkkwzFLckAzt6cmDsy3vY_wpAES4GehyA@mail.gmail.com>
	<CAMnjH4SskCTc641+bMbMQRAP0X9qk42O6M4eELHYmkVrTMDGAw@mail.gmail.com>
	<CAKV-jFy2sN5fZs2prpdE0wD10CJXVfFgWQFQxg7iM_GJ8_SxXw@mail.gmail.com>
	<CAMnjH4RfUyooOutm2i5bCEByAVU6Es9d1ELBijfMUN5C=gc3ow@mail.gmail.com>
	<CAKV-jFxSg7Rda0tYvzQT8ZxUNVHvCAPM7LOQVi9tfuiRS_H_fA@mail.gmail.com>
Message-ID: <CAMnjH4QDqdRy0h5V_wDNobnxVH8hE0C_WGW1p2zaOscU1p0Erg@mail.gmail.com>

Hi

The loopback solution works, I'd guess I have to use that option.
I'm noticing that I end up with double CDR's, and triple "active calls"
("fs_cli status"). So I have to investigate furter on that...

If that's not normal behaviour ?

Thank you for pointing me here !

br
Torstein

On 11 June 2012 14:03, Brian Foster <bdfoster at endigotech.com> wrote:

> <action application="set" data="efffective_caller_id_name=Hobbes"
> inline="true" />
>
> This might be a case where you need to do a loopback.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 11, 2012 6:01 AM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
> wrote:
>
>> Hi
>>
>> I tried :
>> <action inline="true" application="set"
>> data="effective_caller_id_name=Hobbes"/>
>> I tried :
>> <action inline="true" application="set" data="caller_id_name=Hobbes"/>
>> I tried :
>> <action inline="true" application="set"
>> data="origination_caller_id_name=Hobbes"/>
>>
>> But fs_cli "conference xml_list" still shows the "original" caller_id in
>> the conference detail listing on all attempts...
>>
>> Thank you for your continued support :-) !
>> -Torstein
>>
>> On 6 June 2012 19:02, Brian Foster <bdfoster at endigotech.com> wrote:
>>
>>> Crap... set the caller id inline not the conference line... sorry got a
>>> lot going on today lol
>>>
>>> Brian Foster
>>> Endigo Computer LLC
>>>
>>> Sent from a mobile device.
>>> On Jun 6, 2012 12:56 PM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
>>> wrote:
>>>
>>>> Hi
>>>>
>>>> Sure?  I did try that, but it does not work .. Im not doing bridge, but
>>>> conference .. ..
>>>>
>>>> -T
>>>> On Jun 6, 2012 5:06 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>>>>
>>>>> You need to be setting effective_caller_id_name before the bridge.
>>>>>
>>>>> Brian Foster
>>>>> Endigo Computer LLC
>>>>>
>>>>> Sent from a mobile device.
>>>>> On Jun 6, 2012 6:19 AM, "Torstein Knutsen" <torstein.knutsen at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hi
>>>>>>
>>>>>> I'm trying to control the "caller_id_name" variable used in
>>>>>> conferencing.
>>>>>> My dialplan looks like this :
>>>>>>
>>>>>> <extension name="public_extensions">
>>>>>> <condition field="destination_number" expression="^(0meet)$">
>>>>>> <action application="conference" data="72013 at webconf"/>
>>>>>>
>>>>>> When I call into the conference, I do an API "conference
>>>>>> xml_list" which gives details of the conferencees ..
>>>>>> I want to be able to change the caller_id_name in the listing, but
>>>>>> all my efforts for doing this via dialplan fails.
>>>>>>
>>>>>> Is this even possible somehow ?
>>>>>>
>>>>>> I tried to add this to the dialplan before executing the conference :
>>>>>> <action application="export" data="caller_id_name=Hobbes" />
>>>>>> <action application="set" data="caller_id_name=Hobbes" />
>>>>>> <action application="export" data="effective_caller_id_name=Hobbes" />
>>>>>> <action application="export" data="effective_caller_id_name=Hobbes" />
>>>>>> <action application="set" data="origination_caller_id_name=Hobbes" />
>>>>>> <action application="export" data="origination_caller_id_name=Hobbes"
>>>>>> />
>>>>>>
>>>>>> But to no avail...
>>>>>>
>>>>>> Any pointers would be appreciated.
>>>>>>
>>>>>> Thank you
>>>>>> Torstein
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From patrick at sunsus.net  Mon Jun 11 13:09:28 2012
From: patrick at sunsus.net (=?ISO-8859-1?Q?Patrick_D=FCrsteler?=)
Date: Mon, 11 Jun 2012 11:09:28 +0200
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch
In-Reply-To: <CAFLdx6MqmuUHvKxwn2x+qszKVx_24ytHNp0Hr375hp2byikDcw@mail.gmail.com>
References: <1339334863709-7579670.post@n2.nabble.com>
	<CAO7OxTwFdpNS9Yjpbg2Do3gN9QA3J9JRDyHn_us1bhyx4Kay_Q@mail.gmail.com>
	<CAFLdx6MqmuUHvKxwn2x+qszKVx_24ytHNp0Hr375hp2byikDcw@mail.gmail.com>
Message-ID: <CAFLdx6OmybxeAxDpPrbCjeBTou5beLemzsZYCRhGnsPS-DgV0A@mail.gmail.com>

Hello

I just got some information form the Provider, they use DNS
for Loadbalancing, is there a way to tell Freeswitch to don't lookup agin.

regards

Patrick


On Mon, Jun 11, 2012 at 8:08 AM, Patrick D?rsteler <patrick at sunsus.net>wrote:

> Hi Tom
>
> Thanks for your answer, but we have the problem with the outgoing calls.
> See attached sip trace.
>
> regards
>
> Patrick
>
> On Mon, Jun 11, 2012 at 1:09 AM, Tom C <mel0torme at gmail.com> wrote:
>
>> Not sure if it's the same problem, but it sounds similar to the issues we
>> had a couple years ago getting FreeSwitch to work with Whistlephone.
>>  Whistlephone also used multiple servers for load balancing, and while
>> outgoing calls worked fine, incoming calls were unreliable.
>>
>> For more technical details, you can search this mailing list for
>> Whistlephone, about a year and a half ago.
>>
>> The solution was to add a separate gateway for each proxy server.
>>
>> So, in conf/sip_profiles/external, I added whistlephone.xml, with the
>> stuff below.  Pay attention to proxy vs proxy1 in the server names, and the
>> added "from-domain" in the secondary gateways.  Using the "realm" parameter
>> seemed to help me, but it took experimentation to find what worked.
>>
>> --------------------------------------------------------------------
>> <code>
>>
>> <include>
>> <gateway name="WhistlePhone">
>>    <param name="username" value="xxxxxxxxxxx"/>
>>    <param name="from-user" value="xxxxxxxxxx"/>
>>    <param name="proxy" value="proxy.WhistlePhone.com"/>
>>    <param name="realm" value="WhistlePhone"/>
>>    <param name="password" value="xxxx"/>
>>    <param name="expire-seconds" value="600"/>
>>    <param name="register" value="true"/>
>>    <param name="retry-seconds" value="30"/>
>> </gateway>
>>
>> <gateway name="WhistlePhoneProxy1">
>>    <param name="username" value="xxxxxxxxxxx"/>
>>    <param name="from-user" value="xxxxxxxxxx"/>
>>    <param name="proxy" value="proxy1.WhistlePhone.com"/>
>>    <param name="from-domain" value="proxy.Whistlephone.com"/>
>>    <param name="password" value="xxxx"/>
>>    <param name="expire-seconds" value="600"/>
>>    <param name="register" value="true"/>
>>    <param name="retry-seconds" value="30"/>
>> </gateway>
>>
>>  repeat for each proxy server.....
>>
>> </include>
>>
>> </code>
>> ------------------------------------------------------------------------
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From iam at onnet.su  Mon Jun 11 15:44:49 2012
From: iam at onnet.su (Kirill Sysoev)
Date: Mon, 11 Jun 2012 15:44:49 +0400
Subject: [Freeswitch-users] Cluecon 2012 accommodation
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se><1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
Message-ID: <55A07435360D47ADAA0D95ADC2B8846E@OnNet>

Hi! 

Sorry for non-technical question on the list, but I just do not know where to ask it.
Regarding to the nightly rates at Wyndham. It gets much more cheaper if we'll cooperate and book double or may be triple room.

Is there any attendees that want to save some money?

Best regards,
Kirill 
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From alex at thewinelake.com  Mon Jun 11 17:52:36 2012
From: alex at thewinelake.com (Alex Lake)
Date: Mon, 11 Jun 2012 14:52:36 +0100
Subject: [Freeswitch-users] Only calling the first element in the list
 in this mode
In-Reply-To: <CAKV-jFwX7qV73-VCNhzco0Jyb-zmCE8C4AFQjPnc9uKYCZvkZw@mail.gmail.com>
References: <4FD5DAEA.4000601@digitalmail.com>
	<CAKV-jFwX7qV73-VCNhzco0Jyb-zmCE8C4AFQjPnc9uKYCZvkZw@mail.gmail.com>
Message-ID: <4FD5F824.1070705@thewinelake.com>

Yes, that already is set to true - but I think that's what might be 
causing the problem!

Let me explain the scenario a little better:

A DDI is linked to a dialplan script a bit like this:

<include>
<extension name="0044302">
<condition field="destination_number" expression="^(302|441316110347)$">
<action application="set" 
data="sb_routing=${sb_routing}&441316110347_302_PersonalExtension_Rob 
Darwin two"/>
<action application="set" data="tenant_id=0044"/>
<action application="set" data="b_ext=302"/>
<action application="export" 
data="alert_info=http://lnhnov11.dmclub.net/generic/tones/Personal.wav"/>
<action application="set" data="call_timeout=60"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="bridge_generate_comfort_noise=true"/>
<action application="set" data="group_confirm_key=exec"/>
<action application="export" 
data="whisper_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/302_whisper.wav"/>
<action application="export" 
data="accept_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/accept1.wav"/>
<action application="set" data="ringback=$${uk-ring}"/>
<action application="set" data="transfer_ringback_=$${uk-ring}"/>
<action application="set" data="ignore_early_media=true"/>
<action application="set" data="group_confirm_file=lua confirmcall.lua"/>
<action application="bridge" 
data="[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@${domain_name},[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020${ani}]sofia/internal/898000000006207855360320 at a.b.c.d"/>
<action application="set" data="comment=pers_inhours_VM" />
<action application="answer" />
<action application="sleep" data="1000"/>
<action application="voicemail" data="default ${domain_name} 0044302"/>
</condition>
</extension>
</include>

You'll see from the bridge command that one of the outbound legs is to a 
user URL and the other to a sofia URL. In actuality that is 1 SIP 
handset and 1 PSTN destination (via a PSTN gateway)

I believe we have only 1 sofia profile in the whole system - which is 
"internal", and that certainly has multiple-registrations set to "true" 
and I can see that's working by attempting to register simultaneously 
from lots of SIP handsets and then "show registrations" or 
"sofia_contact" from fs_cli.

I'm wondering if I disable multiple SIP registrations, that it might 
then allow multiple-destination bridge commands.

> Set multiple-registrations to true in the affected Sofia profile if 
> you haven't already done so and report back.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
>
> On Jun 11, 2012 7:49 AM, "Alex Lake" <alex at digitalmail.com 
> <mailto:alex at digitalmail.com>> wrote:
>
>     Having upgraded to the latest build (we were previously on a build
>     from
>     2011), I'm finding that multiple SIP registrations are now
>     working, but
>     that multi-destination bridge commands are not.
>
>     Getting messages
>     2012-06-11 11:45:00.565775 [WARNING] switch_ivr_originate.c:2353 Only
>     calling the first element in the list in this mode.
>
>     Is there some special "allow multi destination" setting that I
>     need to add?
>
>     Rgds,
>     Alex
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2433/5062 - Release Date: 06/11/12
>


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From bdfoster at endigotech.com  Mon Jun 11 18:29:10 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 11 Jun 2012 10:29:10 -0400
Subject: [Freeswitch-users] Cluecon 2012 accommodation
In-Reply-To: <55A07435360D47ADAA0D95ADC2B8846E@OnNet>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
	<55A07435360D47ADAA0D95ADC2B8846E@OnNet>
Message-ID: <CAKV-jFzhuSy20J20Q27+5762MJr=rffB=mGXK9Ww0Npe0at68g@mail.gmail.com>

Email marketing at cluecon.com if you don't get a response here. There are
some pretty hefty discounts already in place for booking at that hotel
(since this years cluecon will be held in the same hotel). I can't remember
what the figures are on that bit its safe to say that I don't think you'll
get any more $$ discounted than what is currently available. I believe you
will have to call the hotel and book your rooms with a slight discount, but
you'll end up saving money through the actual cluecon tickets if you book
there. I could be wrong on that. We had this discussion a few weeks ago on
the community call, so my memory is a little fuzzy.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 11, 2012 9:58 AM, "Kirill Sysoev" <iam at onnet.su> wrote:

> **
> Hi!
>
> Sorry for non-technical question on the list, but I just do not know where
> to ask it.
> Regarding to the nightly rates at Wyndham. It gets much more cheaper if
> we'll cooperate and book double or may be triple room.
>
> Is there any attendees that want to save some money?
>
> Best regards,
> Kirill
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Mon Jun 11 18:33:54 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 11 Jun 2012 10:33:54 -0400
Subject: [Freeswitch-users] Only calling the first element in the list
 in this mode
In-Reply-To: <4FD5F824.1070705@thewinelake.com>
References: <4FD5DAEA.4000601@digitalmail.com>
	<CAKV-jFwX7qV73-VCNhzco0Jyb-zmCE8C4AFQjPnc9uKYCZvkZw@mail.gmail.com>
	<4FD5F824.1070705@thewinelake.com>
Message-ID: <CAKV-jFwM1iT-bNExCukZ68+_xJbBKbeVJ_ebT7m7w+zoC0cQzQ@mail.gmail.com>

I don't really understand your dialplan since I don't see everything going
on, but make sure you are using user/<EXTNUMBER>@domain.tld for your
bridges.

Otherwise, ill have to defer to someone else.more.knowledgeable on that
subject. I will note that it does work fine for me and I am on latest as of
last night.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 11, 2012 9:58 AM, "Alex Lake" <alex at thewinelake.com> wrote:

>  Yes, that already is set to true - but I think that's what might be
> causing the problem!
>
> Let me explain the scenario a little better:
>
> A DDI is linked to a dialplan script a bit like this:
>
> <include>
>  <extension name="0044302">
>   <condition field="destination_number" expression="^(302|441316110347)$">
>    <action application="set"
> data="sb_routing=${sb_routing}&441316110347_302_PersonalExtension_Rob
> Darwin two"/>
>    <action application="set" data="tenant_id=0044"/>
>    <action application="set" data="b_ext=302"/>
>    <action application="export" data="alert_info=
> http://lnhnov11.dmclub.net/generic/tones/Personal.wav"/>
>    <action application="set" data="call_timeout=60"/>
>    <action application="set" data="hangup_after_bridge=true"/>
>    <action application="set" data="continue_on_fail=true"/>
>    <action application="set" data="bridge_generate_comfort_noise=true"/>
>    <action application="set" data="group_confirm_key=exec"/>
>    <action application="export"
> data="whisper_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/302_whisper.wav"/>
>    <action application="export"
> data="accept_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/accept1.wav"/>
>    <action application="set" data="ringback=$${uk-ring}"/>
>    <action application="set" data="transfer_ringback_=$${uk-ring}"/>
>    <action application="set" data="ignore_early_media=true"/>
>    <action application="set" data="group_confirm_file=lua
> confirmcall.lua"/>
>    <action application="bridge" data=
> "[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@${domain_name},[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020${ani}]sofia/internal/898000000006207855360320 at a.b.c.d"<[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@$%7Bdomain_name%7D,[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020$%7Bani%7D]sofia/internal/898000000006207855360320 at a.b.c.d>
> />
>    <action application="set" data="comment=pers_inhours_VM" />
>    <action application="answer" />
>    <action application="sleep" data="1000"/>
>    <action application="voicemail" data="default ${domain_name} 0044302"/>
>   </condition>
>  </extension>
> </include>
>
> You'll see from the bridge command that one of the outbound legs is to a
> user URL and the other to a sofia URL. In actuality that is 1 SIP handset
> and 1 PSTN destination (via a PSTN gateway)
>
> I believe we have only 1 sofia profile in the whole system - which is
> "internal", and that certainly has multiple-registrations set to "true" and
> I can see that's working by attempting to register simultaneously from lots
> of SIP handsets and then "show registrations" or "sofia_contact" from
> fs_cli.
>
> I'm wondering if I disable multiple SIP registrations, that it might then
> allow multiple-destination bridge commands.
>
>  Set multiple-registrations to true in the affected Sofia profile if you
> haven't already done so and report back.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 11, 2012 7:49 AM, "Alex Lake" <alex at digitalmail.com> wrote:
>
>> Having upgraded to the latest build (we were previously on a build from
>> 2011), I'm finding that multiple SIP registrations are now working, but
>> that multi-destination bridge commands are not.
>>
>> Getting messages
>> 2012-06-11 11:45:00.565775 [WARNING] switch_ivr_originate.c:2353 Only
>> calling the first element in the list in this mode.
>>
>> Is there some special "allow multi destination" setting that I need to
>> add?
>>
>> Rgds,
>> Alex
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2177 / Virus Database: 2433/5062 - Release Date: 06/11/12
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mitch.capper at gmail.com  Mon Jun 11 18:37:05 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 11 Jun 2012 07:37:05 -0700
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <CAKV-jFxHEYwM8cj7tNPmPytRf9_41N6z2gH4xuk2dCctnP-sDQ@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
	<1339414395.62082.YahooMailNeo@web120105.mail.ne1.yahoo.com>
	<30DFAA60-ADB7-4B3A-A5FE-9085491FD83E@opencsta.org>
	<CAKV-jFxHEYwM8cj7tNPmPytRf9_41N6z2gH4xuk2dCctnP-sDQ@mail.gmail.com>
Message-ID: <CALn233oSkLZowVb+FMcFySMrtWfi18dW7ovNMP2p0MsFvQ+xcg@mail.gmail.com>

Of course if you are faxing over g729 gl:)   as said if the phone
supports G729 then you just need to enable pass through mode for it,
check the wiki for pass through.

~Mitch

On Mon, Jun 11, 2012 at 5:14 AM, Brian Foster <bdfoster at endigotech.com> wrote:
> Adding to comments by Chris, this includes voicemail, faxing, basically
> anything media related that freeswitch would have to accomplish on it's own.
> If your getting that message then you are in fact doing something media
> relates inside freeswitch and its telling you that you've been a naughty,
> naughty boy.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
>
> On Jun 11, 2012 7:41 AM, "Chris Mylonas" <chris at opencsta.org> wrote:
>>
>> Are you bridging a call to this gateway or are you playing a soundfile to
>> it?
>>
>> "passthrough" means no file based stuff is to happen on the FS server -
>> only the codec can pass through the switch.
>> in order to play "files" or record to a file, you will need a g729
>> license.
>>
>> HTH
>> Chris
>>
>>
>>
>> On 11/06/2012, at 9:33 PM, Samira Mh wrote:
>>
>> yes, my voipgateway support G729 and i set "G729" on freeswitch but when
>> making call the error occure:
>> G729 only for passthrough !!
>>
>> ________________________________
>> From: Nandy Dagondon <gcd at i.ph>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, June 11, 2012 3:47 PM
>> Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
>>
>> samira,
>>
>> if your IP phone, ATA or gateway supports G729, then you can use FS free
>> G729 codec. just make sure the codec is set in the dialplan.
>>
>> -nandy
>>
>> On Mon, Jun 11, 2012 at 6:13 PM, Samira Mh <saami_mh at ymail.com> wrote:
>>
>> but for making call from one country to another need to set codec to
>> G729(no passthrough) because the carrier support only G729 ?and i want the
>> free one not licensing ,
>> is it posiible to use media proxy so that use G729 without need to
>> use??commercial license?
>>
>> ________________________________
>> From: Peter Olsson <peter.olsson at visionutveckling.se>
>> To: Free SWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, June 11, 2012 2:21 PM
>> Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
>>
>> The passthrough codec for G.729 is built by default, just make sure to
>> load the module mod_g729. If you need transcoding, or use IVR?s, voicemail
>> etc, you must purchase commercial licenses for G729.
>>
>> /Peter
>>
>>
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh
>> Skickat: den 11 juni 2012 10:16
>> Till: Free SWITCH Users Help
>> ?mne: [Freeswitch-users] how to use codec g729 on freeswitch ?
>>
>> hi guys,
>> i am going to make call using codec g729,
>> because the carrier ?support only codec g729
>> is it possible to get free that?
>> in freeswitch implement only g729 passthrough .
>> but how can i get it ?
>> !DSPAM:4fd5a7d032762000267653!
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Mon Jun 11 18:46:50 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Mon, 11 Jun 2012 16:46:50 +0200
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch
In-Reply-To: <CAFLdx6OmybxeAxDpPrbCjeBTou5beLemzsZYCRhGnsPS-DgV0A@mail.gmail.com>
References: <1339334863709-7579670.post@n2.nabble.com>
	<CAO7OxTwFdpNS9Yjpbg2Do3gN9QA3J9JRDyHn_us1bhyx4Kay_Q@mail.gmail.com>
	<CAFLdx6MqmuUHvKxwn2x+qszKVx_24ytHNp0Hr375hp2byikDcw@mail.gmail.com>
	<CAFLdx6OmybxeAxDpPrbCjeBTou5beLemzsZYCRhGnsPS-DgV0A@mail.gmail.com>
Message-ID: <4FD604DA.10701@puzzled.xs4all.nl>

On 11-06-12 11:09, Patrick D?rsteler wrote:
> Hello
>
> I just got some information form the Provider, they use DNS
> for Loadbalancing, is there a way to tell Freeswitch to don't lookup agin.

Maybe use the IP addresses instead of FQDNs?

Regards,
Patrick



From alex at thewinelake.com  Mon Jun 11 18:37:49 2012
From: alex at thewinelake.com (Alex Lake)
Date: Mon, 11 Jun 2012 15:37:49 +0100
Subject: [Freeswitch-users] Only calling the first element in the list
 in this mode
In-Reply-To: <4FD5F824.1070705@thewinelake.com>
References: <4FD5DAEA.4000601@digitalmail.com>
	<CAKV-jFwX7qV73-VCNhzco0Jyb-zmCE8C4AFQjPnc9uKYCZvkZw@mail.gmail.com>
	<4FD5F824.1070705@thewinelake.com>
Message-ID: <4FD602BD.8070708@thewinelake.com>

By the way, user/0044302 is set up with this file:

<include>
<user id="0044301">
<params>
<param name="password" value="AAAAAAAAAA" />
<param name="vm-password" value="1234" />
<param name="vm-alternate-greet-id" value="301" />
<param name="vm-mailto" value="fred.smith at fitclub.net" />
<param name="vm-mailfrom" value="vm-ext301 at sb0044.dmclub.net" />
<param name="vm-attach-file" value="true" />
<param name="vm-email-all-messages" value="true" />
<param name="vm-keep-local-after-email" value="false" />
<param name="vm-disk-quota" value="3600" />
</params>
<variables>
<variable name="tenant_id" value="0044"/>
<variable name="a_ext" value="301"/>
<variable name="accountcode" value="301"/>
<variable name="user_context" value="dp0044"/>
<variable name="effective_caller_id_name" value="Fred Smith one"/>
<variable name="effective_caller_id_number" value="01316110346"/>
<variable name="outbound_caller_id_name" value="Fred Smith one testing 
co 44"/>
<variable name="outbound_caller_id_number" value="01316110346"/>
<variable name="ani" value="0131123456"/>
<variable name="default_ani_prefix" value="00443010"/>
<variable name="callgroup" value="deferred"/>
<variable name="mailbox" value="0044301"/>
</variables>
</user>
</include>


And dp0044 is set up like this:
<?xml version="1.0" encoding="utf-8"?>
<!--
     NOTICE:

     This context is usually accessed via authenticated callers on the 
sip profile on port 5060
     or transfered callers from the public context which arrived via the 
sip profile on port 5080.

     Authenticated users will use the user_context variable on the user 
to determine what context
     they can access.  You can also add a user in the directory with the 
cidr= attribute acl.conf.xml
     will build the domains ACL using this value.
-->
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="dp0044">
<X-PRE-PROCESS cmd="include" data="0044/*.xml" />
<extension name="Outbound_call">
<action application="log" data="WARNING 
default_ani_prefix=${default_ani_prefix}"/>
<action application="log" data="WARNING 
sip_auth_username=${sip_auth_username}"/>
<condition field="default_ani_prefix" expression=""/>
<condition field="sip_auth_username" expression="">
<action application="set" data="default_ani_prefix=${tenant_id}${b_ext}1"/>
</condition>
<condition field="default_ani_prefix" expression="" break="never">
<action application="set" data="ani_prefix=${sip_auth_username}0"/>
<anti-action application="set" data="ani_prefix=${default_ani_prefix}"/>
</condition>
<condition field="ani_prefix" expression="^(0)$" break="never">
<action application="set" data="ani_prefix=00000000"/>
</condition>
<condition field="destination_number" expression="^(\d{8}.*)">
<action application="log" data="WARNING 2 ani_prefix=${ani_prefix}"/>
<action application="log" data="WARNING 2 
default_ani_prefix=${default_ani_prefix}"/>
<action application="bridge" 
data="[tenant_id=${tenant_id},a_ext=${a_ext},b_ext=${b_ext},origination_callee_id_number=$1,origination_caller_id_number=${ani_prefix}${ani}]sofia/internal/8980000000062$1 at 193.1.5.1"/>
</condition>
</extension>
</context>
</include>



From alex at thewinelake.com  Mon Jun 11 18:51:39 2012
From: alex at thewinelake.com (Alex Lake)
Date: Mon, 11 Jun 2012 15:51:39 +0100
Subject: [Freeswitch-users] Only calling the first element in the list
 in this mode
In-Reply-To: <CAKV-jFwM1iT-bNExCukZ68+_xJbBKbeVJ_ebT7m7w+zoC0cQzQ@mail.gmail.com>
References: <4FD5DAEA.4000601@digitalmail.com>
	<CAKV-jFwX7qV73-VCNhzco0Jyb-zmCE8C4AFQjPnc9uKYCZvkZw@mail.gmail.com>
	<4FD5F824.1070705@thewinelake.com>
	<CAKV-jFwM1iT-bNExCukZ68+_xJbBKbeVJ_ebT7m7w+zoC0cQzQ@mail.gmail.com>
Message-ID: <4FD605FB.4030503@thewinelake.com>

Brian, when you say "it" (as in "it does work fine for me") are you 
doing a bridge to several different destinations, or are you just 
talking about bridging to a user that happens to have multiple SIP 
registrations?
Thanks for the encouragement, anyway...
Alex
>
> I don't really understand your dialplan since I don't see everything 
> going on, but make sure you are using user/<EXTNUMBER>@domain.tld for 
> your bridges.
>
> Otherwise, ill have to defer to someone else.more.knowledgeable on 
> that subject. I will note that it does work fine for me and I am on 
> latest as of last night.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
>
> On Jun 11, 2012 9:58 AM, "Alex Lake" <alex at thewinelake.com 
> <mailto:alex at thewinelake.com>> wrote:
>
>     Yes, that already is set to true - but I think that's what might
>     be causing the problem!
>
>     Let me explain the scenario a little better:
>
>     A DDI is linked to a dialplan script a bit like this:
>
>     <include>
>     <extension name="0044302">
>     <condition field="destination_number"
>     expression="^(302|441316110347)$">
>     <action application="set"
>     data="sb_routing=${sb_routing}&441316110347_302_PersonalExtension_Rob
>     Darwin two"/>
>     <action application="set" data="tenant_id=0044"/>
>     <action application="set" data="b_ext=302"/>
>     <action application="export"
>     data="alert_info=http://lnhnov11.dmclub.net/generic/tones/Personal.wav"/>
>     <action application="set" data="call_timeout=60"/>
>     <action application="set" data="hangup_after_bridge=true"/>
>     <action application="set" data="continue_on_fail=true"/>
>     <action application="set" data="bridge_generate_comfort_noise=true"/>
>     <action application="set" data="group_confirm_key=exec"/>
>     <action application="export"
>     data="whisper_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/302_whisper.wav"/>
>     <action application="export"
>     data="accept_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/accept1.wav"/>
>     <action application="set" data="ringback=$${uk-ring}"/>
>     <action application="set" data="transfer_ringback_=$${uk-ring}"/>
>     <action application="set" data="ignore_early_media=true"/>
>     <action application="set" data="group_confirm_file=lua
>     confirmcall.lua"/>
>     <action application="bridge"
>     data="[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@${domain_name},[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020${ani}]sofia/internal/898000000006207855360320 at a.b.c.d"
>     <mailto:[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@$%7Bdomain_name%7D,[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020$%7Bani%7D]sofia/internal/898000000006207855360320 at a.b.c.d>/>
>     <action application="set" data="comment=pers_inhours_VM" />
>     <action application="answer" />
>     <action application="sleep" data="1000"/>
>     <action application="voicemail" data="default ${domain_name}
>     0044302"/>
>     </condition>
>     </extension>
>     </include>
>
>     You'll see from the bridge command that one of the outbound legs
>     is to a user URL and the other to a sofia URL. In actuality that
>     is 1 SIP handset and 1 PSTN destination (via a PSTN gateway)
>
>     I believe we have only 1 sofia profile in the whole system - which
>     is "internal", and that certainly has multiple-registrations set
>     to "true" and I can see that's working by attempting to register
>     simultaneously from lots of SIP handsets and then "show
>     registrations" or "sofia_contact" from fs_cli.
>
>     I'm wondering if I disable multiple SIP registrations, that it
>     might then allow multiple-destination bridge commands.
>
>>     Set multiple-registrations to true in the affected Sofia profile
>>     if you haven't already done so and report back.
>>
>>     Brian Foster
>>     Endigo Computer LLC
>>
>>     Sent from a mobile device.
>>
>>     On Jun 11, 2012 7:49 AM, "Alex Lake" <alex at digitalmail.com
>>     <mailto:alex at digitalmail.com>> wrote:
>>
>>         Having upgraded to the latest build (we were previously on a
>>         build from
>>         2011), I'm finding that multiple SIP registrations are now
>>         working, but
>>         that multi-destination bridge commands are not.
>>
>>         Getting messages
>>         2012-06-11 11:45:00.565775 [WARNING]
>>         switch_ivr_originate.c:2353 Only
>>         calling the first element in the list in this mode.
>>
>>         Is there some special "allow multi destination" setting that
>>         I need to add?
>>
>>         Rgds,
>>         Alex
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.com
>>
>>         Join Us At ClueCon - Aug 7-9, 2012
>>
>>         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
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.com
>>
>>     Join Us At ClueCon - Aug 7-9, 2012
>>
>>     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
>>
>>
>>     No virus found in this message.
>>     Checked by AVG - www.avg.com <http://www.avg.com>
>>     Version: 2012.0.2177 / Virus Database: 2433/5062 - Release Date:
>>     06/11/12
>>
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2433/5062 - Release Date: 06/11/12
>

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From robert.longfield at klinsight.com  Mon Jun 11 19:00:02 2012
From: robert.longfield at klinsight.com (Robert Longfield)
Date: Mon, 11 Jun 2012 11:00:02 -0400
Subject: [Freeswitch-users] Time needed for ringing a call group
In-Reply-To: <-8988759419527339202@unknownmsgid>
References: <D3BC5C093B2E4150AF352558D6E077BE@KITPC003><-6559196381009927779@unknownmsgid><20120608223010.2d079108@anubis.defcon1>
	<-8988759419527339202@unknownmsgid>
Message-ID: <14C24E97FBF044208146E7D8511CA914@KITPC003>

Thanks for catching that. I have no idea where 2000 came from. I'd like to 
say I was thinking milliseconds but that doesn't work either :)
I could also blame it on my team member that hosed the entire system which I 
had to pull backups to restore except 2000 was in the backups....
Well thanks for catching my goof, the system works much better already!

-Rob

-----Original Message----- 
From: Brian West
Sent: Saturday, June 09, 2012 2:49 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Time needed for ringing a call group

I alway kid, but my question real.  2000 seconds was exce???v?!

(^-^)/

Sent from my iPad

On Jun 8, 2012, at 3:30 PM, Bzzz <lazyvirus at gmx.com> wrote:

> Hmm, either you're kidding or you have a keyboard|board problem

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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



From bdfoster at endigotech.com  Mon Jun 11 19:03:23 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 11 Jun 2012 11:03:23 -0400
Subject: [Freeswitch-users] Only calling the first element in the list
 in this mode
In-Reply-To: <4FD605FB.4030503@thewinelake.com>
References: <4FD5DAEA.4000601@digitalmail.com>
	<CAKV-jFwX7qV73-VCNhzco0Jyb-zmCE8C4AFQjPnc9uKYCZvkZw@mail.gmail.com>
	<4FD5F824.1070705@thewinelake.com>
	<CAKV-jFwM1iT-bNExCukZ68+_xJbBKbeVJ_ebT7m7w+zoC0cQzQ@mail.gmail.com>
	<4FD605FB.4030503@thewinelake.com>
Message-ID: <CAKV-jFy+=pzVS8u0kXFB1YD8Leo+VaXQbKRNHSb5htcWY6NJog@mail.gmail.com>

At the end of your bridge string there is a Sofia/. Check that, could be by
design but I'm not understanding what you are doing there.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 11, 2012 10:57 AM, "Alex Lake" <alex at thewinelake.com> wrote:

>  Brian, when you say "it" (as in "it does work fine for me") are you doing
> a bridge to several different destinations, or are you just talking about
> bridging to a user that happens to have multiple SIP registrations?
> Thanks for the encouragement, anyway...
> Alex
>
> I don't really understand your dialplan since I don't see everything going
> on, but make sure you are using user/<EXTNUMBER>@domain.tld for your
> bridges.
>
> Otherwise, ill have to defer to someone else.more.knowledgeable on that
> subject. I will note that it does work fine for me and I am on latest as of
> last night.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 11, 2012 9:58 AM, "Alex Lake" <alex at thewinelake.com> wrote:
>
>>  Yes, that already is set to true - but I think that's what might be
>> causing the problem!
>>
>> Let me explain the scenario a little better:
>>
>> A DDI is linked to a dialplan script a bit like this:
>>
>> <include>
>>  <extension name="0044302">
>>   <condition field="destination_number" expression="^(302|441316110347)$">
>>    <action application="set"
>> data="sb_routing=${sb_routing}&441316110347_302_PersonalExtension_Rob
>> Darwin two"/>
>>    <action application="set" data="tenant_id=0044"/>
>>    <action application="set" data="b_ext=302"/>
>>    <action application="export" data="alert_info=
>> http://lnhnov11.dmclub.net/generic/tones/Personal.wav"/>
>>    <action application="set" data="call_timeout=60"/>
>>    <action application="set" data="hangup_after_bridge=true"/>
>>    <action application="set" data="continue_on_fail=true"/>
>>    <action application="set" data="bridge_generate_comfort_noise=true"/>
>>    <action application="set" data="group_confirm_key=exec"/>
>>    <action application="export"
>> data="whisper_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/302_whisper.wav"/>
>>    <action application="export"
>> data="accept_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/accept1.wav"/>
>>    <action application="set" data="ringback=$${uk-ring}"/>
>>    <action application="set" data="transfer_ringback_=$${uk-ring}"/>
>>    <action application="set" data="ignore_early_media=true"/>
>>    <action application="set" data="group_confirm_file=lua
>> confirmcall.lua"/>
>>    <action application="bridge" data=
>> "[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@${domain_name},[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020${ani}]sofia/internal/898000000006207855360320 at a.b.c.d"<[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@$%7Bdomain_name%7D,[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020$%7Bani%7D]sofia/internal/898000000006207855360320 at a.b.c.d>
>> />
>>    <action application="set" data="comment=pers_inhours_VM" />
>>    <action application="answer" />
>>    <action application="sleep" data="1000"/>
>>    <action application="voicemail" data="default ${domain_name} 0044302"/>
>>   </condition>
>>  </extension>
>> </include>
>>
>> You'll see from the bridge command that one of the outbound legs is to a
>> user URL and the other to a sofia URL. In actuality that is 1 SIP handset
>> and 1 PSTN destination (via a PSTN gateway)
>>
>> I believe we have only 1 sofia profile in the whole system - which is
>> "internal", and that certainly has multiple-registrations set to "true" and
>> I can see that's working by attempting to register simultaneously from lots
>> of SIP handsets and then "show registrations" or "sofia_contact" from
>> fs_cli.
>>
>> I'm wondering if I disable multiple SIP registrations, that it might then
>> allow multiple-destination bridge commands.
>>
>>  Set multiple-registrations to true in the affected Sofia profile if you
>> haven't already done so and report back.
>>
>> Brian Foster
>> Endigo Computer LLC
>>
>> Sent from a mobile device.
>> On Jun 11, 2012 7:49 AM, "Alex Lake" <alex at digitalmail.com> wrote:
>>
>>> Having upgraded to the latest build (we were previously on a build from
>>> 2011), I'm finding that multiple SIP registrations are now working, but
>>> that multi-destination bridge commands are not.
>>>
>>> Getting messages
>>> 2012-06-11 11:45:00.565775 [WARNING] switch_ivr_originate.c:2353 Only
>>> calling the first element in the list in this mode.
>>>
>>> Is there some special "allow multi destination" setting that I need to
>>> add?
>>>
>>> Rgds,
>>> Alex
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> No virus found in this message.
>> Checked by AVG - www.avg.com
>> Version: 2012.0.2177 / Virus Database: 2433/5062 - Release Date: 06/11/12
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2177 / Virus Database: 2433/5062 - Release Date: 06/11/12
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From colins at succinct.co.za  Mon Jun 11 19:07:05 2012
From: colins at succinct.co.za (Colin Sindle)
Date: Mon, 11 Jun 2012 17:07:05 +0200
Subject: [Freeswitch-users] Garbled Leg Recording G729 with Asynchronous
	ptime
In-Reply-To: <CAKbxfG-g5fqC+ZQA5ZJzXwwzoy-zS71JB0MjuWkEaYcvN+bUbw@mail.gmail.com>
References: <CAFV278N8ShLqxz5x3ZJhc6OE5kxme1Vqaa8NTbscom_E5QCGjQ@mail.gmail.com>
	<CAKbxfG-g5fqC+ZQA5ZJzXwwzoy-zS71JB0MjuWkEaYcvN+bUbw@mail.gmail.com>
Message-ID: <CAFV278MKNTdXJ-MWn2qPAFbQb9GbOYtoJOBEWipb-fiPmZVkwQ@mail.gmail.com>

Hi Anthony,

Thank you very much for your response.

Sorry, I meant that we are using the _latest_ git HEAD at time of writing.

I have now tried passthru_ptime_mismatch=true, and it fixed the recording
of some of the cases of asynchronous ptime, but also broke other previously
"working" cases.


Kind regards,


Colin



On 8 June 2012 21:03, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> have you tried updating lately?
> also try global_setvar passthru_ptime_mismatch=true
>
>
> On Fri, Jun 8, 2012 at 3:29 AM, <colins at succinct.co.za> wrote:
> > Hi,
> >
> > Please, I was wondering if there is anything obvious that we are missing
> > when trying to record a G729 call with asynchronous pimes (in: 20ms, out:
> > 60ms).
> >
> > The problem case call flow is:  inbound leg (ptime 20 ms), start
> recording,
> > playback a short message, bridge to outbound (destination can only
> handle a
> > ptime of 60 ms).   We are using mod_com_g729's codec.   The audio while
> on
> > the call in normally fine (rarely there is one way speech depending on
> > inbound/outbound provider), but the recording has a garbled outbound
> > (Callee) leg (inbound/Caller voice is fine.).
> >
> > We have tried all the combinations of  G729 at 20i, G729 at 60i,
> > rtp-autofix-timing, and default-ptimes to no avail.
> >
> > We are using git HEAD.
> >
> > I do see this on fs_cli:
> >
> > 2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> > 0xa8894898 (nil)
> > 2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:79 DECODER DESTROYX -
> > 0xa8894898 (nil)
> > 2012-06-07 16:56:45.583270 [WARNING] mod_sofia.c:1158 Asynchronous PTIME
> not
> > supported, changing our end from 20 to 60
> > 2012-06-07 16:56:45.583270 [DEBUG] sofia_glue.c:2931 Changing Codec from
> > G729 at 20ms@8000hz to G729 at 60ms@8000hz
> > 2012-06-07 16:56:45.623271 [INFO] mod_com_g729.c:78 ENCODER DESTROYX -
> > 0xa86fe340 0xa86f1e78
> >
> > Should this be working, are we missing something obvious?
> >
> >
> > Thank you,
> >
> >
> > Kind regards,
> >
> >
> > Colin
>
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From bdfoster at endigotech.com  Mon Jun 11 19:21:55 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 11 Jun 2012 11:21:55 -0400
Subject: [Freeswitch-users] Only calling the first element in the list
 in this mode
In-Reply-To: <CAKV-jFy+=pzVS8u0kXFB1YD8Leo+VaXQbKRNHSb5htcWY6NJog@mail.gmail.com>
References: <4FD5DAEA.4000601@digitalmail.com>
	<CAKV-jFwX7qV73-VCNhzco0Jyb-zmCE8C4AFQjPnc9uKYCZvkZw@mail.gmail.com>
	<4FD5F824.1070705@thewinelake.com>
	<CAKV-jFwM1iT-bNExCukZ68+_xJbBKbeVJ_ebT7m7w+zoC0cQzQ@mail.gmail.com>
	<4FD605FB.4030503@thewinelake.com>
	<CAKV-jFy+=pzVS8u0kXFB1YD8Leo+VaXQbKRNHSb5htcWY6NJog@mail.gmail.com>
Message-ID: <CAKV-jFzbuOMXFBU-hF7vmWsWZbVXtxBivx9D1SLO5zAJ2Vo2tA@mail.gmail.com>

This is going into stuff I haven't played with, but I'm trying my best :-)

In your bridge string, instead of using a comma have you tried using :_: ?

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 11, 2012 11:03 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> At the end of your bridge string there is a Sofia/. Check that, could be
> by design but I'm not understanding what you are doing there.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 11, 2012 10:57 AM, "Alex Lake" <alex at thewinelake.com> wrote:
>
>>  Brian, when you say "it" (as in "it does work fine for me") are you
>> doing a bridge to several different destinations, or are you just talking
>> about bridging to a user that happens to have multiple SIP registrations?
>> Thanks for the encouragement, anyway...
>> Alex
>>
>> I don't really understand your dialplan since I don't see everything
>> going on, but make sure you are using user/<EXTNUMBER>@domain.tld for your
>> bridges.
>>
>> Otherwise, ill have to defer to someone else.more.knowledgeable on that
>> subject. I will note that it does work fine for me and I am on latest as of
>> last night.
>>
>> Brian Foster
>> Endigo Computer LLC
>>
>> Sent from a mobile device.
>> On Jun 11, 2012 9:58 AM, "Alex Lake" <alex at thewinelake.com> wrote:
>>
>>>  Yes, that already is set to true - but I think that's what might be
>>> causing the problem!
>>>
>>> Let me explain the scenario a little better:
>>>
>>> A DDI is linked to a dialplan script a bit like this:
>>>
>>> <include>
>>>  <extension name="0044302">
>>>   <condition field="destination_number"
>>> expression="^(302|441316110347)$">
>>>    <action application="set"
>>> data="sb_routing=${sb_routing}&441316110347_302_PersonalExtension_Rob
>>> Darwin two"/>
>>>    <action application="set" data="tenant_id=0044"/>
>>>    <action application="set" data="b_ext=302"/>
>>>    <action application="export" data="alert_info=
>>> http://lnhnov11.dmclub.net/generic/tones/Personal.wav"/>
>>>    <action application="set" data="call_timeout=60"/>
>>>    <action application="set" data="hangup_after_bridge=true"/>
>>>    <action application="set" data="continue_on_fail=true"/>
>>>    <action application="set" data="bridge_generate_comfort_noise=true"/>
>>>    <action application="set" data="group_confirm_key=exec"/>
>>>    <action application="export"
>>> data="whisper_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/302_whisper.wav"/>
>>>    <action application="export"
>>> data="accept_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/accept1.wav"/>
>>>    <action application="set" data="ringback=$${uk-ring}"/>
>>>    <action application="set" data="transfer_ringback_=$${uk-ring}"/>
>>>    <action application="set" data="ignore_early_media=true"/>
>>>    <action application="set" data="group_confirm_file=lua
>>> confirmcall.lua"/>
>>>    <action application="bridge" data=
>>> "[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@${domain_name},[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020${ani}]sofia/internal/898000000006207855360320 at a.b.c.d"<[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@$%7Bdomain_name%7D,[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020$%7Bani%7D]sofia/internal/898000000006207855360320 at a.b.c.d>
>>> />
>>>    <action application="set" data="comment=pers_inhours_VM" />
>>>    <action application="answer" />
>>>    <action application="sleep" data="1000"/>
>>>    <action application="voicemail" data="default ${domain_name}
>>> 0044302"/>
>>>   </condition>
>>>  </extension>
>>> </include>
>>>
>>> You'll see from the bridge command that one of the outbound legs is to a
>>> user URL and the other to a sofia URL. In actuality that is 1 SIP handset
>>> and 1 PSTN destination (via a PSTN gateway)
>>>
>>> I believe we have only 1 sofia profile in the whole system - which is
>>> "internal", and that certainly has multiple-registrations set to "true" and
>>> I can see that's working by attempting to register simultaneously from lots
>>> of SIP handsets and then "show registrations" or "sofia_contact" from
>>> fs_cli.
>>>
>>> I'm wondering if I disable multiple SIP registrations, that it might
>>> then allow multiple-destination bridge commands.
>>>
>>>  Set multiple-registrations to true in the affected Sofia profile if
>>> you haven't already done so and report back.
>>>
>>> Brian Foster
>>> Endigo Computer LLC
>>>
>>> Sent from a mobile device.
>>> On Jun 11, 2012 7:49 AM, "Alex Lake" <alex at digitalmail.com> wrote:
>>>
>>>> Having upgraded to the latest build (we were previously on a build from
>>>> 2011), I'm finding that multiple SIP registrations are now working, but
>>>> that multi-destination bridge commands are not.
>>>>
>>>> Getting messages
>>>> 2012-06-11 11:45:00.565775 [WARNING] switch_ivr_originate.c:2353 Only
>>>> calling the first element in the list in this mode.
>>>>
>>>> Is there some special "allow multi destination" setting that I need to
>>>> add?
>>>>
>>>> Rgds,
>>>> Alex
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> 
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>>
>>> No virus found in this message.
>>> Checked by AVG - www.avg.com
>>> Version: 2012.0.2177 / Virus Database: 2433/5062 - Release Date: 06/11/12
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> No virus found in this message.
>> Checked by AVG - www.avg.com
>> Version: 2012.0.2177 / Virus Database: 2433/5062 - Release Date: 06/11/12
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
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From patrick at sunsus.net  Mon Jun 11 19:16:15 2012
From: patrick at sunsus.net (sunsus)
Date: Mon, 11 Jun 2012 08:16:15 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch
In-Reply-To: <4FD604DA.10701@puzzled.xs4all.nl>
References: <1339334863709-7579670.post@n2.nabble.com>
	<CAO7OxTwFdpNS9Yjpbg2Do3gN9QA3J9JRDyHn_us1bhyx4Kay_Q@mail.gmail.com>
	<CAFLdx6MqmuUHvKxwn2x+qszKVx_24ytHNp0Hr375hp2byikDcw@mail.gmail.com>
	<CAFLdx6OmybxeAxDpPrbCjeBTou5beLemzsZYCRhGnsPS-DgV0A@mail.gmail.com>
	<4FD604DA.10701@puzzled.xs4all.nl>
Message-ID: <1339427775648-7579710.post@n2.nabble.com>

Hi Patrick 

I've tried this, but FreeSWITCH still switch to the other server sometimes.
It seams it use the url from the sip header.

Any idea on how to solve this? Or is it a Bug? 

regards

Patrick 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-304-Unauthorized-tp7579670p7579710.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Mon Jun 11 19:29:04 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jun 2012 08:29:04 -0700
Subject: [Freeswitch-users] Cluecon 2012 accommodation
In-Reply-To: <CAKV-jFzhuSy20J20Q27+5762MJr=rffB=mGXK9Ww0Npe0at68g@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
	<55A07435360D47ADAA0D95ADC2B8846E@OnNet>
	<CAKV-jFzhuSy20J20Q27+5762MJr=rffB=mGXK9Ww0Npe0at68g@mail.gmail.com>
Message-ID: <CAKzWOxXSUGyuj+PnOhOjnNnA8dp1CU1_TB86QARfwRkJtMg1SA@mail.gmail.com>

The community conf call and IRC channels are the best place to get in touch
w/ other ClueCon attendees.
-MC

On Mon, Jun 11, 2012 at 7:29 AM, Brian Foster <bdfoster at endigotech.com>wrote:

> Email marketing at cluecon.com if you don't get a response here. There are
> some pretty hefty discounts already in place for booking at that hotel
> (since this years cluecon will be held in the same hotel). I can't remember
> what the figures are on that bit its safe to say that I don't think you'll
> get any more $$ discounted than what is currently available. I believe you
> will have to call the hotel and book your rooms with a slight discount, but
> you'll end up saving money through the actual cluecon tickets if you book
> there. I could be wrong on that. We had this discussion a few weeks ago on
> the community call, so my memory is a little fuzzy.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 11, 2012 9:58 AM, "Kirill Sysoev" <iam at onnet.su> wrote:
>
>> **
>> Hi!
>>
>> Sorry for non-technical question on the list, but I just do not know
>> where to ask it.
>> Regarding to the nightly rates at Wyndham. It gets much more cheaper if
>> we'll cooperate and book double or may be triple room.
>>
>> Is there any attendees that want to save some money?
>>
>> Best regards,
>> Kirill
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun 11 19:33:51 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jun 2012 08:33:51 -0700
Subject: [Freeswitch-users] XML_CURL and static XML (loaded in memory)
In-Reply-To: <CAC-LwPNO7JkC4-=VOh6ebbSEOdXtHub9jMUCA4XMHL6k=7fpVA@mail.gmail.com>
References: <CAC-LwPNO7JkC4-=VOh6ebbSEOdXtHub9jMUCA4XMHL6k=7fpVA@mail.gmail.com>
Message-ID: <CAKzWOxWU4w3h7M==KFST8j8nwK9g4TAOWfmwtdnXfJH37PZ35Q@mail.gmail.com>

This sounds like the curl request is timing out. I'd look outside FS to see
what's wrong.
-MC

On Thu, Jun 7, 2012 at 5:56 AM, Vik Killa <vipkilla at gmail.com> wrote:

> Is it possible to load freeswitch with static XML and then load
> mod_xml_curl?
> It seems FreeSWITCH does not work correctly when this is done.
> Here is an example:
> In freeswitch.xml, I have:
>  <section name="directory" description="User Directory">
>    <X-PRE-PROCESS cmd="include" data="static/directory.xml"/>
>  </section>
>
> This loads my directory configuration statically in memory.
> After I start FreeSWITCH, I load mod_xml_curl with 'fs_cli> load
> mod_xml_curl'
> (mod_xml_curl is not loaded when freeswitch starts)
>
> Here is my XML_CURL configuration:
>
>    <configuration name="xml_curl.conf" description="cURL XML Gateway">
>        <bindings>
>            <binding name="default">
>                <param name="gateway-url" value="192.168.0.158"
> bindings="directory" />
>            </binding>
>        </bindings>
>    </configuration>
>
> FreeSWITCH does not seem to function properly after XML_CURL is loaded.
> If I run the 'user_data' API, FS hangs there for a long time and
> eventually returns the password from memory (it does not make an
> XML_CURL request)
> Here is example:
> fs_cli> user_data 1000 at domain.com param password
>
> I guess the ultimate question is this:
> Is there any way to setup XML_CURL to failover to static XML (loaded in
> memory)?
>
> Thanks!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From alex at thewinelake.com  Mon Jun 11 19:51:01 2012
From: alex at thewinelake.com (Alex Lake)
Date: Mon, 11 Jun 2012 16:51:01 +0100
Subject: [Freeswitch-users] Only calling the first element in the list
 in this mode
In-Reply-To: <CAKV-jFzbuOMXFBU-hF7vmWsWZbVXtxBivx9D1SLO5zAJ2Vo2tA@mail.gmail.com>
References: <4FD5DAEA.4000601@digitalmail.com>
	<CAKV-jFwX7qV73-VCNhzco0Jyb-zmCE8C4AFQjPnc9uKYCZvkZw@mail.gmail.com>
	<4FD5F824.1070705@thewinelake.com>
	<CAKV-jFwM1iT-bNExCukZ68+_xJbBKbeVJ_ebT7m7w+zoC0cQzQ@mail.gmail.com>
	<4FD605FB.4030503@thewinelake.com>
	<CAKV-jFy+=pzVS8u0kXFB1YD8Leo+VaXQbKRNHSb5htcWY6NJog@mail.gmail.com>
	<CAKV-jFzbuOMXFBU-hF7vmWsWZbVXtxBivx9D1SLO5zAJ2Vo2tA@mail.gmail.com>
Message-ID: <4FD613E5.20105@thewinelake.com>

I have not!
I presume that's for the bits between the destinations (and not the use 
of commas between destination-specific channel variables).
<action application="bridge" 
data="[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@${domain_name}:_:[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020${ani}]sofia/internal/898000000006207855360320 at a.b.c.d" 
<mailto:[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@$%7Bdomain_name%7D,[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020$%7Bani%7D]sofia/internal/898000000006207855360320 at a.b.c.d>/>
I'll give it a go.
>
> This is going into stuff I haven't played with, but I'm trying my best :-)
>
> In your bridge string, instead of using a comma have you tried using :_: ?
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
>
> On Jun 11, 2012 11:03 AM, "Brian Foster" <bdfoster at endigotech.com 
> <mailto:bdfoster at endigotech.com>> wrote:
>
>     At the end of your bridge string there is a Sofia/. Check that,
>     could be by design but I'm not understanding what you are doing there.
>
>     Brian Foster
>     Endigo Computer LLC
>
>     Sent from a mobile device.
>
>     On Jun 11, 2012 10:57 AM, "Alex Lake" <alex at thewinelake.com
>     <mailto:alex at thewinelake.com>> wrote:
>
>         Brian, when you say "it" (as in "it does work fine for me")
>         are you doing a bridge to several different destinations, or
>         are you just talking about bridging to a user that happens to
>         have multiple SIP registrations?
>         Thanks for the encouragement, anyway...
>         Alex
>>
>>         I don't really understand your dialplan since I don't see
>>         everything going on, but make sure you are using
>>         user/<EXTNUMBER>@domain.tld for your bridges.
>>
>>         Otherwise, ill have to defer to someone
>>         else.more.knowledgeable on that subject. I will note that it
>>         does work fine for me and I am on latest as of last night.
>>
>>         Brian Foster
>>         Endigo Computer LLC
>>
>>         Sent from a mobile device.
>>
>>         On Jun 11, 2012 9:58 AM, "Alex Lake" <alex at thewinelake.com
>>         <mailto:alex at thewinelake.com>> wrote:
>>
>>             Yes, that already is set to true - but I think that's
>>             what might be causing the problem!
>>
>>             Let me explain the scenario a little better:
>>
>>             A DDI is linked to a dialplan script a bit like this:
>>
>>             <include>
>>             <extension name="0044302">
>>             <condition field="destination_number"
>>             expression="^(302|441316110347)$">
>>             <action application="set"
>>             data="sb_routing=${sb_routing}&441316110347_302_PersonalExtension_Rob
>>             Darwin two"/>
>>             <action application="set" data="tenant_id=0044"/>
>>             <action application="set" data="b_ext=302"/>
>>             <action application="export"
>>             data="alert_info=http://lnhnov11.dmclub.net/generic/tones/Personal.wav"/>
>>             <action application="set" data="call_timeout=60"/>
>>             <action application="set" data="hangup_after_bridge=true"/>
>>             <action application="set" data="continue_on_fail=true"/>
>>             <action application="set"
>>             data="bridge_generate_comfort_noise=true"/>
>>             <action application="set" data="group_confirm_key=exec"/>
>>             <action application="export"
>>             data="whisper_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/302_whisper.wav"/>
>>             <action application="export"
>>             data="accept_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/accept1.wav"/>
>>             <action application="set" data="ringback=$${uk-ring}"/>
>>             <action application="set"
>>             data="transfer_ringback_=$${uk-ring}"/>
>>             <action application="set" data="ignore_early_media=true"/>
>>             <action application="set" data="group_confirm_file=lua
>>             confirmcall.lua"/>
>>             <action application="bridge"
>>             data="[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@${domain_name},[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020${ani}]sofia/internal/898000000006207855360320 at a.b.c.d"
>>             <mailto:[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@$%7Bdomain_name%7D,[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020$%7Bani%7D]sofia/internal/898000000006207855360320 at a.b.c.d>/>
>>             <action application="set" data="comment=pers_inhours_VM" />
>>             <action application="answer" />
>>             <action application="sleep" data="1000"/>
>>             <action application="voicemail" data="default
>>             ${domain_name} 0044302"/>
>>             </condition>
>>             </extension>
>>             </include>
>>
>>             You'll see from the bridge command that one of the
>>             outbound legs is to a user URL and the other to a sofia
>>             URL. In actuality that is 1 SIP handset and 1 PSTN
>>             destination (via a PSTN gateway)
>>
>>             I believe we have only 1 sofia profile in the whole
>>             system - which is "internal", and that certainly has
>>             multiple-registrations set to "true" and I can see that's
>>             working by attempting to register simultaneously from
>>             lots of SIP handsets and then "show registrations" or
>>             "sofia_contact" from fs_cli.
>>
>>             I'm wondering if I disable multiple SIP registrations,
>>             that it might then allow multiple-destination bridge
>>             commands.
>>
>>>             Set multiple-registrations to true in the affected Sofia
>>>             profile if you haven't already done so and report back.
>>>
>>>             Brian Foster
>>>             Endigo Computer LLC
>>>
>>>             Sent from a mobile device.
>>>
>>>             On Jun 11, 2012 7:49 AM, "Alex Lake"
>>>             <alex at digitalmail.com <mailto:alex at digitalmail.com>> wrote:
>>>
>>>                 Having upgraded to the latest build (we were
>>>                 previously on a build from
>>>                 2011), I'm finding that multiple SIP registrations
>>>                 are now working, but
>>>                 that multi-destination bridge commands are not.
>>>
>>>                 Getting messages
>>>                 2012-06-11 11:45:00.565775 [WARNING]
>>>                 switch_ivr_originate.c:2353 Only
>>>                 calling the first element in the list in this mode.
>>>
>>>                 Is there some special "allow multi destination"
>>>                 setting that I need to add?
>>>
>>>                 Rgds,
>>>                 Alex
>>>
>>>                 _________________________________________________________________________
>>>                 Professional FreeSWITCH Consulting Services:
>>>                 consulting at freeswitch.org
>>>                 <mailto:consulting at freeswitch.org>
>>>                 http://www.freeswitchsolutions.com
>>>
>>>                 FreeSWITCH-powered IP PBX: The CudaTel Communication
>>>                 Server
>>>                 
>>>
>>>                 Official FreeSWITCH Sites
>>>                 http://www.freeswitch.org
>>>                 http://wiki.freeswitch.org
>>>                 http://www.cluecon.com
>>>
>>>                 Join Us At ClueCon - Aug 7-9, 2012
>>>
>>>                 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
>>>
>>>
>>>
>>>             _________________________________________________________________________
>>>             Professional FreeSWITCH Consulting Services:
>>>             consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>>             http://www.freeswitchsolutions.com
>>>
>>>             
>>>             
>>>
>>>             Official FreeSWITCH Sites
>>>             http://www.freeswitch.org
>>>             http://wiki.freeswitch.org
>>>             http://www.cluecon.com
>>>
>>>             Join Us At ClueCon - Aug 7-9, 2012
>>>
>>>             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
>>>
>>>
>>>             No virus found in this message.
>>>             Checked by AVG - www.avg.com <http://www.avg.com>
>>>             Version: 2012.0.2177 / Virus Database: 2433/5062 -
>>>             Release Date: 06/11/12
>>>
>>
>>
>>
>>             _________________________________________________________________________
>>             Professional FreeSWITCH Consulting Services:
>>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>             http://www.freeswitchsolutions.com
>>
>>             
>>             
>>
>>             Official FreeSWITCH Sites
>>             http://www.freeswitch.org
>>             http://wiki.freeswitch.org
>>             http://www.cluecon.com
>>
>>             Join Us At ClueCon - Aug 7-9, 2012
>>
>>             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
>>
>>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.com
>>
>>         Join Us At ClueCon - Aug 7-9, 2012
>>
>>         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
>>
>>
>>         No virus found in this message.
>>         Checked by AVG - www.avg.com <http://www.avg.com>
>>         Version: 2012.0.2177 / Virus Database: 2433/5062 - Release
>>         Date: 06/11/12
>>
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>
>         Join Us At ClueCon - Aug 7-9, 2012
>
>         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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2433/5062 - Release Date: 06/11/12
>

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From fvillarroel at yahoo.com  Mon Jun 11 19:52:08 2012
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Mon, 11 Jun 2012 08:52:08 -0700 (PDT)
Subject: [Freeswitch-users] application set variable problem
Message-ID: <1339429928.32332.YahooMailClassic@web160305.mail.bf1.yahoo.com>

Hi All.

i am using mod_lcr


  <action application="lcr" data="$1 pg_prefix2"/>
  <action application="set" data="nibble_account=${accountcode}"/>
  <action application="set" data="nibble_rate=${lcr_rate_1}"/>
  <action application="set" data="foo=11"/>
  <action application="set" data="accounts_tarifa=${tarifa}"/>

My custom SQL:


      <param name="custom_sql" value="
SELECT l.digits AS lcr_digits,
       c.carrier_name AS lcr_carrier_name,
       l.${lcr_rate_field} as lcr_rate_field,
       cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix,
       l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip,
       l.prefix AS lcr_prefix, l.suffix AS lcr_suffix,
       a.id AS nibble_account,
       l.${lcr_rate_field} AS nibble_rate,
       a.warn_limit AS lowbal_amt,
       a.nobal_limit AS nobal_amt,
       a.id_tarifa AS tarifa
  FROM lcr l
    JOIN carriers c ON l.carrier_id=c.id
    JOIN carrier_gateway cg ON c.id=cg.carrier_id
    JOIN accounts a ON a.id = '${accountcode}'
    JOIN tarifas t ON a.id_tarifa = t.id 
  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
    AND l.digits @> '${lcr_query_digits}'
    AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end
  ORDER BY digits DESC, ${lcr_rate_field}, random();


Log :


2012-06-11 11:33:52.336613 [DEBUG] mod_lcr.c:940 SQL:  SELECT l.digits AS lcr_digits,        c.carrier_name AS lcr_carrier_name,        l.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,        a.id AS nibble_account,        l.rate AS nibble_rate,        a.warn_limit AS lowbal_amt,        a.nobal_limit AS nobal_amt,        a.id_tarifa AS tarifa   FROM lcr l     JOIN carriers c ON l.carrier_id=c.id     JOIN carrier_gateway cg ON c.id=cg.carrier_id     JOIN accounts a ON a.id = '1004'     JOIN tarifas t ON a.id_tarifa = t.id    WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'     AND l.digits @> '1006'     AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end   ORDER BY digits DESC, rate, random();       
2012-06-11 11:33:52.356514 [DEBUG] mod_lcr.c:338 Returning Dialstring [lcr_carrier=carrier2,lcr_rate=0.14000,nibble_account=1004,nibble_rate=0.14000,lowbal_amt=0,nobal_amt=0,tarifa=1]user/gateway/0006
2012-06-11 11:33:52.356514 [DEBUG] mod_lcr.c:677 Adding carrier2 to head of list
2012-06-11 11:33:52.356514 [DEBUG] mod_lcr.c:338 Returning Dialstring [lcr_carrier=carrier1,lcr_rate=0.15000,nibble_account=1004,nibble_rate=0.15000,lowbal_amt=0,nobal_amt=0,tarifa=1]user/1006
2012-06-11 11:33:52.356514 [DEBUG] mod_lcr.c:705 Adding carrier1 to end of list
EXECUTE sofia/internal/1004 at 192.168.1.108 set(nibble_account=1004)
2012-06-11 11:33:52.356514 [DEBUG] mod_dptools.c:1294 sofia/internal/1004 at 192.168.1.108 SET [nibble_account]=[1004]
EXECUTE sofia/internal/1004 at 192.168.1.108 set(nibble_rate=0.14000)
2012-06-11 11:33:52.356514 [DEBUG] mod_dptools.c:1294 sofia/internal/1004 at 192.168.1.108 SET [nibble_rate]=[0.14000]
EXECUTE sofia/internal/1004 at 192.168.1.108 set(foo=11)
2012-06-11 11:33:52.356514 [DEBUG] mod_dptools.c:1294 sofia/internal/1004 at 192.168.1.108 SET [foo]=[11]
EXECUTE sofia/internal/1004 at 192.168.1.108 set(accounts_tarifa=)
2012-06-11 11:33:52.356514 [DEBUG] mod_dptools.c:1294 sofia/internal/1004 at 192.168.1.108 SET [accounts_tarifa]=[UNDEF]


My accounts_tarifa variable is UNDEF

What i am doing wrong?

Regard


From vipkilla at gmail.com  Mon Jun 11 20:04:06 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 11 Jun 2012 12:04:06 -0400
Subject: [Freeswitch-users] XML_CURL and static XML (loaded in memory)
In-Reply-To: <CAKzWOxWU4w3h7M==KFST8j8nwK9g4TAOWfmwtdnXfJH37PZ35Q@mail.gmail.com>
References: <CAC-LwPNO7JkC4-=VOh6ebbSEOdXtHub9jMUCA4XMHL6k=7fpVA@mail.gmail.com>
	<CAKzWOxWU4w3h7M==KFST8j8nwK9g4TAOWfmwtdnXfJH37PZ35Q@mail.gmail.com>
Message-ID: <CAC-LwPOssDC2=M=Jqif5hhaZ-ncdaCZPHpjPA7J31Xbx7BomGQ@mail.gmail.com>

> This sounds like the curl request is timing out. I'd look outside FS to see
> what's wrong.
> -MC

I have multiple logs setup to record every XML_CURL request.
FreeSWITCH never makes an XML_CURL request in my example setup. It
seems FreeSWITCH will not function if it has static XML (dialplan,
directory, and/or configuration) loaded as well as XML_CURL loaded
(and binding to dialplan, directory, and/or configuration).


From msc at freeswitch.org  Mon Jun 11 20:11:10 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jun 2012 09:11:10 -0700
Subject: [Freeswitch-users] XML_CURL and static XML (loaded in memory)
In-Reply-To: <CAC-LwPOssDC2=M=Jqif5hhaZ-ncdaCZPHpjPA7J31Xbx7BomGQ@mail.gmail.com>
References: <CAC-LwPNO7JkC4-=VOh6ebbSEOdXtHub9jMUCA4XMHL6k=7fpVA@mail.gmail.com>
	<CAKzWOxWU4w3h7M==KFST8j8nwK9g4TAOWfmwtdnXfJH37PZ35Q@mail.gmail.com>
	<CAC-LwPOssDC2=M=Jqif5hhaZ-ncdaCZPHpjPA7J31Xbx7BomGQ@mail.gmail.com>
Message-ID: <CAKzWOxWQdQc4Vvu3ZS5z57P+aCiWHh3EHjKPf80DGykopNu0_A@mail.gmail.com>

I would check Ray's recipes from the cookbook. I think he explicitly talks
about dynamic w/ static failover. I don't know that he specifically
discusses the directory, but I don't know if that is different than
dialplan or other configs.
-MC

On Mon, Jun 11, 2012 at 9:04 AM, Vik Killa <vipkilla at gmail.com> wrote:

> > This sounds like the curl request is timing out. I'd look outside FS to
> see
> > what's wrong.
> > -MC
>
> I have multiple logs setup to record every XML_CURL request.
> FreeSWITCH never makes an XML_CURL request in my example setup. It
> seems FreeSWITCH will not function if it has static XML (dialplan,
> directory, and/or configuration) loaded as well as XML_CURL loaded
> (and binding to dialplan, directory, and/or configuration).
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vipkilla at gmail.com  Mon Jun 11 20:18:57 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 11 Jun 2012 12:18:57 -0400
Subject: [Freeswitch-users] XML_CURL and static XML (loaded in memory)
In-Reply-To: <CAKzWOxWQdQc4Vvu3ZS5z57P+aCiWHh3EHjKPf80DGykopNu0_A@mail.gmail.com>
References: <CAC-LwPNO7JkC4-=VOh6ebbSEOdXtHub9jMUCA4XMHL6k=7fpVA@mail.gmail.com>
	<CAKzWOxWU4w3h7M==KFST8j8nwK9g4TAOWfmwtdnXfJH37PZ35Q@mail.gmail.com>
	<CAC-LwPOssDC2=M=Jqif5hhaZ-ncdaCZPHpjPA7J31Xbx7BomGQ@mail.gmail.com>
	<CAKzWOxWQdQc4Vvu3ZS5z57P+aCiWHh3EHjKPf80DGykopNu0_A@mail.gmail.com>
Message-ID: <CAC-LwPMnF5eBi=_g0WcK0ALo=Vm4TC2N6Agz5QU+J-k+Ce+oww@mail.gmail.com>

I don't see anything related to XML_CURL in the cookbook. Would you
know what page or chapter to look in?

On Mon, Jun 11, 2012 at 12:11 PM, Michael Collins <msc at freeswitch.org> wrote:
> I would check Ray's recipes from the cookbook. I think he explicitly talks
> about dynamic w/ static failover. I don't know that he specifically
> discusses the directory, but I don't know if that is different than dialplan
> or other configs.
> -MC


From freeswitch-list at puzzled.xs4all.nl  Mon Jun 11 20:30:25 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Mon, 11 Jun 2012 18:30:25 +0200
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch
In-Reply-To: <1339427775648-7579710.post@n2.nabble.com>
References: <1339334863709-7579670.post@n2.nabble.com>
	<CAO7OxTwFdpNS9Yjpbg2Do3gN9QA3J9JRDyHn_us1bhyx4Kay_Q@mail.gmail.com>
	<CAFLdx6MqmuUHvKxwn2x+qszKVx_24ytHNp0Hr375hp2byikDcw@mail.gmail.com>
	<CAFLdx6OmybxeAxDpPrbCjeBTou5beLemzsZYCRhGnsPS-DgV0A@mail.gmail.com>
	<4FD604DA.10701@puzzled.xs4all.nl>
	<1339427775648-7579710.post@n2.nabble.com>
Message-ID: <4FD61D21.30100@puzzled.xs4all.nl>

On 11-06-12 17:16, sunsus wrote:
> Hi Patrick
>
> I've tried this, but FreeSWITCH still switch to the other server sometimes.
> It seams it use the url from the sip header.
>
> Any idea on how to solve this? Or is it a Bug?

I'm afraid I don't know. If you can't figure it out perhaps some 
professional help from consulting at freeswitch.org could offer a solution.

Regards,
Patrick


From vipkilla at gmail.com  Mon Jun 11 20:35:11 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 11 Jun 2012 12:35:11 -0400
Subject: [Freeswitch-users] checking voicemail without ID
Message-ID: <CAC-LwPMDDUvjb-6EkN9GwTRR6VUgnp7PfU9DDseZqFLJnp+eQA@mail.gmail.com>

Why is it that a caller can enter a non-existent voicemail box and
made up PIN and login to the non-existent mailbox?
I've tested this with *98 and entering random numbers followed by a
password of random numbers.


From msc at freeswitch.org  Mon Jun 11 20:40:01 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jun 2012 09:40:01 -0700
Subject: [Freeswitch-users] XML_CURL and static XML (loaded in memory)
In-Reply-To: <CAC-LwPMnF5eBi=_g0WcK0ALo=Vm4TC2N6Agz5QU+J-k+Ce+oww@mail.gmail.com>
References: <CAC-LwPNO7JkC4-=VOh6ebbSEOdXtHub9jMUCA4XMHL6k=7fpVA@mail.gmail.com>
	<CAKzWOxWU4w3h7M==KFST8j8nwK9g4TAOWfmwtdnXfJH37PZ35Q@mail.gmail.com>
	<CAC-LwPOssDC2=M=Jqif5hhaZ-ncdaCZPHpjPA7J31Xbx7BomGQ@mail.gmail.com>
	<CAKzWOxWQdQc4Vvu3ZS5z57P+aCiWHh3EHjKPf80DGykopNu0_A@mail.gmail.com>
	<CAC-LwPMnF5eBi=_g0WcK0ALo=Vm4TC2N6Agz5QU+J-k+Ce+oww@mail.gmail.com>
Message-ID: <CAKzWOxX-YzrkU1gsgaaphqounNC7dBSk9gyK1Yg2F8f-8HEGcg@mail.gmail.com>

D'oh! I'm thinking of the recipes that Ray sent to me for *possible *inclusion
in the cookbook. They never actually made it in. Never mind. All I can say
is make sure that "accept-blind-reg" and "auth-calls" are not set to false
in your SIP profile, otherwise you won't get xml_curl calls for those
operations.
-MC

On Mon, Jun 11, 2012 at 9:18 AM, Vik Killa <vipkilla at gmail.com> wrote:

> I don't see anything related to XML_CURL in the cookbook. Would you
> know what page or chapter to look in?
>
> On Mon, Jun 11, 2012 at 12:11 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > I would check Ray's recipes from the cookbook. I think he explicitly
> talks
> > about dynamic w/ static failover. I don't know that he specifically
> > discusses the directory, but I don't know if that is different than
> dialplan
> > or other configs.
> > -MC
>
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From jayesh.voip at gmail.com  Mon Jun 11 20:44:10 2012
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Mon, 11 Jun 2012 22:14:10 +0530
Subject: [Freeswitch-users] mod_callcenter missing the variable
	cc_queue_terminated_epoch in xml-cdr
Message-ID: <CALvF6vC8GwEVHTM3y6i3FD=XQUJfaDSePdoWOhOjnmGAsAt6ag@mail.gmail.com>

Hi,
I am trying to generate call-center reports using the variables generated
on channels by mod_callcenter. I am using xml-cdr for the same. I can see
in the xml-cdr all the variables documented in the mod_callcenter wiki page
except "cc_queue_terminated_epoch". When I checked the code it seemed that
if cc_queue_answer_epoch is set, cc_queue_terminated has to be set and
should be visible in the xml-cdr but it doesn't.
Now to troubleshoot further, I entered the fs_cli and executed /events
plain all to check the variables generated in the events. To my surprize I
could see the variable "cc_queue_terminated_epoch"  in the
"CHANNEL_DESTROY" event but it still does not show in the xml-cdr. I was
talking to Sevet on IRC who recommended to pastebin the event output and
the xml-cdr output for the same call.
I have copied it in the Pastebin here:
http://pastebin.freeswitch.org/19258

Any ideas on why this variable would not be visible in the xml-cdr !! I was
expecting to get the exact call duration between the member and the agent
with this variable.

Thanks in advance.

--- Jayesh
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From saami_mh at ymail.com  Mon Jun 11 20:54:45 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Mon, 11 Jun 2012 09:54:45 -0700 (PDT)
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <30DFAA60-ADB7-4B3A-A5FE-9085491FD83E@opencsta.org>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
	<1339414395.62082.YahooMailNeo@web120105.mail.ne1.yahoo.com>
	<30DFAA60-ADB7-4B3A-A5FE-9085491FD83E@opencsta.org>
Message-ID: <1339433685.46946.YahooMailNeo@web120106.mail.ne1.yahoo.com>

hi guys,
thanks alot for your great replies,
yes Chris,
i want to bridge call using my VOIPgateway so that making calls to another countries..
but the carrier only support G729 codec and the FS send G722 (set in vars.xml) to myVoipGateway that is set as an gateway in /usr/local/freeswitch/sip-profile/external/
and when FS send media to Gateway(using bridge application) the error occure:unacceptable media,then check VOIPGW and find out the only codec that?
can be pass through VOIPgw is G729, but FS only send G711,G722,... not G729



________________________________
 From: Chris Mylonas <chris at opencsta.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Monday, June 11, 2012 4:10 PM
Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
 

Are you bridging a call to this gateway or are you playing a soundfile to it?

"passthrough" means no file based stuff is to happen on the FS server - only the codec can pass through the switch.
in order to play "files" or record to a file, you will need a g729 license.

HTH
Chris




On 11/06/2012, at 9:33 PM, Samira Mh wrote:

yes, my voipgateway support G729 and i set "G729" on freeswitch but when making call the error occure:
>G729 only for passthrough !!
>
>
>
>________________________________
> From: Nandy Dagondon <gcd at i.ph>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Monday, June 11, 2012 3:47 PM
>Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
> 
>
>samira,
>
>if your IP phone, ATA or gateway supports G729, then you can use FS free G729 codec. just make sure the codec is set in the dialplan.
>
>-nandy
>
>
>On Mon, Jun 11, 2012 at 6:13 PM, Samira Mh <saami_mh at ymail.com> wrote:
>
>but for making call from one country to another need to set codec to G729(no passthrough) because the carrier support only G729 ?and i want the free one not licensing ,
>>is it posiible to use media proxy so that use G729 without need to use??commercial license?
>>
>>
>>
>>________________________________
>> From: Peter Olsson <peter.olsson at visionutveckling.se>
>>To: Free SWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>>Sent: Monday, June 11, 2012 2:21 PM
>>Subject: Re: [Freeswitch-users] how to use  codec g729 on freeswitch ?
>> 
>>
>>
>>The passthrough codec for G.729 is built by default, just make sure to load the module mod_g729. If you need transcoding, or use IVR?s, voicemail etc, you must purchase commercial licenses for G729.
>>?
>>/Peter
>>?
>>?
>>Fr?n:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh
>>Skickat: den 11 juni 2012 10:16
>>Till: Free SWITCH Users Help
>>?mne: [Freeswitch-users] how to use codec g729 on freeswitch ?
>>?
>>hi guys,
>>i am going to make call using codec g729,
>>because the carrier ?support only codec g729
>>is it possible to get free that?
>>in freeswitch implement only g729 passthrough .
>>but how can i get it ?
>>!DSPAM:4fd5a7d032762000267653! 
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
>>consulting at freeswitch.org
>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
>>http://www.freeswitch.org
>>http://wiki.freeswitch.org
>>http://www.cluecon.com
>>
>>Join Us At ClueCon - Aug 7-9, 2012
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
>>consulting at freeswitch.org
>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
>>http://www.freeswitch.org
>>http://wiki.freeswitch.org
>>http://www.cluecon.com
>>
>>Join Us At ClueCon - Aug 7-9, 2012
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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  Mon Jun 11 21:12:47 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 11 Jun 2012 10:12:47 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxWpyoyteEcXMw8mv0p0k0_v9FEAs63MByfXHy_BWZRSqQ@mail.gmail.com>

Happy Monday to you all!

Last Wednesday we had a nice discussion on the weekly conference
call<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls?cc12-0611>about
mod_httapi <http://wiki.freeswitch.org/wiki/Mod_httapi?cc12-0611>. It turns
out that there is a fair amount of interest in the great module but that
there are some questions about how to handle certain scenarios, such as
"session tracking." Thanks to Raymond Chandler for being available to
answer a lot of those questions. We will be doing a followup discussion
that focuses on Raymond's PHP examples for using mod_httapi. These examples
do a good job of demonstrating the power and ease of HTTAPI.

ClueCon <http://www.cluecon.com?cc12-0611> season is upon us and our list
of speakers and sponsors continues to grow. Packt
Publishing<http://packtpub.com?cc12-0611>is now on board as a media
sponsor and we hope to have some nice items from
them to give away at the conference. Keep checking the
schedule<http://www.cluecon.com/schedule?cc12-0611>for updates as we
are confirming new speakers each week. We are also happy
to report that the Wyndham has completed its changeover and is officially
known as the Hyatt Chicago Magnificent
Mile<http://www.chicagomagnificentmile.hyatt.com?cc12-0611>.
(As of this writing their web site is undergoing maintenance, so be
patient.) We are excited to see what's in store at this revamped hotel and
can't wait to see everyone there this August!


-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com?cc12-0611
877-7-4ACLUE
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From gabe at gundy.org  Mon Jun 11 21:25:13 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 11 Jun 2012 11:25:13 -0600
Subject: [Freeswitch-users] using APNS with diaplan : throttle a call
 until a certain condition changes
In-Reply-To: <CACU5aF4rP_03cmKXCp5LgJ5mU=Ynxkz1zqLbLwCKKD3b-TJR8Q@mail.gmail.com>
References: <CACU5aF6ma-W+VM98FzhS6i=kWHQhJPehpvPt1oS9E8662CaC2A@mail.gmail.com>
	<CAMwZ-tLtXYEWbxo+R9ooudh+PZ-F_bnpPrtawZLMFjK13Z2ceQ@mail.gmail.com>
	<CACU5aF4rP_03cmKXCp5LgJ5mU=Ynxkz1zqLbLwCKKD3b-TJR8Q@mail.gmail.com>
Message-ID: <CAMwZ-tJLAFhZ8tnRWUxa3OWksaBPU_JW=bn=jtnUcHQa=VSGnw@mail.gmail.com>

On Sun, Jun 10, 2012 at 1:48 PM, John Gathm <john.gathm at gmail.com> wrote:
> so to summarize, I should (I did tome tries)

Keep in mind, I was just sharing ideas about a general approach, not a
specific solution... so you may have to tinker a bit.



> 1) kind of park if failure - kind of replace the voicemail

More or less.



> 2) subscribe to "event xml CHANNEL_PARK"
> ==> which UDID is the one I should watch to be able to unpark the call ?

Check this against the 'show channels'



> ==> which command from mod_command should I then use to "resume" the call ?
> or transfer/bridge again to extension 1234

There are a number of ways you could do it.  Bridging once you have
both UUIDs is an option.

http://wiki.freeswitch.org/wiki/Mod_commands#uuid_bridge


Good luck,
Gabe


From roger.castaldo at gmail.com  Mon Jun 11 22:06:35 2012
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Mon, 11 Jun 2012 14:06:35 -0400
Subject: [Freeswitch-users] Event Socket
In-Reply-To: <CAAYf2LtEyqjrpzaw-x-7j=OLdAMcfxmpZ3Hr1-yfwjc7FzP0vQ@mail.gmail.com>
References: <CAAYf2LsypUhejdzsR8phUFvi0dbjBzyi-FPeYJ8iQnTdWq55-w@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD0225507272@NY1-EXMB-01.ip-soft.net>
	<CAAYf2LtEyqjrpzaw-x-7j=OLdAMcfxmpZ3Hr1-yfwjc7FzP0vQ@mail.gmail.com>
Message-ID: <CAAYf2LuO1_iO+W7KXpo7LR+OvTjZ8qEp1o6fA47-Ne951hakdg@mail.gmail.com>

I have updated the code base with more functionality as well as added in a
fix to handle a memory leak issue within mono if you had not noticed it.

On Thu, Dec 22, 2011 at 12:04 PM, Roger Castaldo
<roger.castaldo at gmail.com>wrote:

> Okay thanks for the info I will just have to do it somewhat differently in
> my code then, as i said I was just curious thinking that there might be
> some obscure command to list them
>
> On Thu, Dec 22, 2011 at 11:19 AM, Hector Geraldino <
> Hector.Geraldino at ip-soft.net> wrote:
>
>> Hi Roger,****
>>
>> ** **
>>
>> If the set of available events is constant (any of the existing events
>> can be received by the ESL application at any time), I don?t see why you
>> need to query FS to get this list. Let?s suppose this api command exists:
>> issuing this command will always return a list with the same values, which
>> are the values already listed on the Events list page:****
>>
>> ** **
>>
>> http://wiki.freeswitch.org/wiki/Event_List****
>>
>> ** **
>>
>> Can you do that? I?m pretty sure you can?t. But, if you **really** want
>> to get this list of events after issuing an api command to freeswitch
>> (non-negotiable feature), you can always send an ?api echo STRING_LIST?
>> with the list of possible values:****
>>
>> ** **
>>
>> api echo CHANNEL_CALLSTATE CHANNEL_CREATE CHANNEL_DESTROY CHANNEL_STATE
>> CHANNEL_ANSWER CHANNEL_HANGUP CHANNEL_HANGUP_COMPLETE CHANNEL_EXECUTE
>> CHANNEL_EXECUTE_COMPLETE****
>>
>> ** **
>>
>> Content-Type: api/response****
>>
>> Content-Length: 157****
>>
>> ** **
>>
>> CHANNEL_CALLSTATE CHANNEL_CREATE CHANNEL_DESTROY CHANNEL_STATE
>> CHANNEL_ANSWER CHANNEL_HANGUP CHANNEL_HANGUP_COMPLETE CHANNEL_EXECUTE
>> CHANNEL_EXECUTE_COMPLETE ****
>>
>> ** **
>>
>> ** **
>>
>> PS: I?m kidding****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Roger
>> Castaldo
>> *Sent:* Thursday, December 22, 2011 10:41 AM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* [Freeswitch-users] Event Socket****
>>
>> ** **
>>
>> Hi everyone i have  a simple question, is there an api command that will
>> list all the available events that can be listened on by the event socket?
>> i realize there is a wiki page containing a list but for my purposes an api
>> command to list the available event names would be much better.****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 paul at cupis.co.uk  Mon Jun 11 22:21:17 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Mon, 11 Jun 2012 19:21:17 +0100
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <1339433685.46946.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
	<1339414395.62082.YahooMailNeo@web120105.mail.ne1.yahoo.com>
	<30DFAA60-ADB7-4B3A-A5FE-9085491FD83E@opencsta.org>
	<1339433685.46946.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <4FD6371D.1010307@cupis.co.uk>

On 11/06/12 17:54, Samira Mh wrote:
> i want to bridge call using my VOIPgateway so that making calls to
> another countries..
> but the carrier only support G729 codec and the FS send G722 (set in
> vars.xml) to myVoipGateway that is set as an gateway in
> /usr/local/freeswitch/sip-profile/external/
> and when FS send media to Gateway(using bridge application) the error
> occure:unacceptable media,then check VOIPGW and find out the only codec
> that
> can be pass through VOIPgw is G729, but FS only send G711,G722,... not G729

Can you provide a SIP or FreeSWITCH trace of a call, please?

Do you have the following enabled in your SIP profile?

   <param name="inbound-late-negotiation" value="true"/>

Do you have mod_g729 loaded and codec G729 enabled in your vars.xml?

Regards,


From djbinter at gmail.com  Mon Jun 11 23:11:28 2012
From: djbinter at gmail.com (DJB International)
Date: Mon, 11 Jun 2012 12:11:28 -0700
Subject: [Freeswitch-users] Polycom caller ID problem on 4.0.2
Message-ID: <CAAB3jWsQPFyrr1d=d6gfA0G9Q+kyM0v7VyvDoOYrq=h6LaZ9rA@mail.gmail.com>

I've recently updated my Polycom 650 phone to use 4.0.2.

I am not sure what broke the caller id for received calls, it kept showing
as Unknown Party; however, the weird thing is that the Missed Calls show
the correct caller id.

Anyone experienced this issue.  Please advise.

FreeSWITCH Version 1.2.0-rc2

Thank you.
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From royj at yandex.ru  Mon Jun 11 23:31:23 2012
From: royj at yandex.ru (royj)
Date: Mon, 11 Jun 2012 23:31:23 +0400
Subject: [Freeswitch-users] mod_gsmopen on FreeBSD
Message-ID: <20120611233123.ebed9bee.royj@yandex.ru>


Hi all
Does anyone use mod_gsmopen on FreeBSD? I have tried to make according to the instructions - http://wiki.freeswitch.org/wiki/GSMopen#Building, and got an error when compiling libctb-0.16 which is only for Linux and Win32 (https://iftools.com/download/ctb/0.14/refman.pdf). Is there any solution other than changing OS )
-- 
royj


From bdfoster at endigotech.com  Mon Jun 11 23:59:53 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 11 Jun 2012 15:59:53 -0400
Subject: [Freeswitch-users] Polycom caller ID problem on 4.0.2
In-Reply-To: <CAAB3jWsQPFyrr1d=d6gfA0G9Q+kyM0v7VyvDoOYrq=h6LaZ9rA@mail.gmail.com>
References: <CAAB3jWsQPFyrr1d=d6gfA0G9Q+kyM0v7VyvDoOYrq=h6LaZ9rA@mail.gmail.com>
Message-ID: <CAKV-jFwHHbFRY4i-NiLen8b4b5n6VL9k+gqbEO1_mys5nPGxpQ@mail.gmail.com>

Pastebin a log with sip trace where this happens and we'll go from there.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 11, 2012 3:13 PM, "DJB International" <djbinter at gmail.com> wrote:

> I've recently updated my Polycom 650 phone to use 4.0.2.
>
> I am not sure what broke the caller id for received calls, it kept showing
> as Unknown Party; however, the weird thing is that the Missed Calls show
> the correct caller id.
>
> Anyone experienced this issue.  Please advise.
>
> FreeSWITCH Version 1.2.0-rc2
>
> Thank you.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue Jun 12 01:02:04 2012
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Mon, 11 Jun 2012 14:02:04 -0700 (PDT)
Subject: [Freeswitch-users] application set variable problem
In-Reply-To: <1339429928.32332.YahooMailClassic@web160305.mail.bf1.yahoo.com>
Message-ID: <1339448524.78109.YahooMailClassic@web160303.mail.bf1.yahoo.com>

Hi.

Does anyone could help me.




--- On Mon, 6/11/12, FERNANDO VILLARROEL <fvillarroel at yahoo.com> wrote:

> From: FERNANDO VILLARROEL <fvillarroel at yahoo.com>
> Subject: [Freeswitch-users] application set variable problem
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, June 11, 2012, 12:52 PM
> Hi All.
> 
> i am using mod_lcr
> 
> 
> ? <action application="lcr" data="$1
> pg_prefix2"/>
> ? <action application="set"
> data="nibble_account=${accountcode}"/>
> ? <action application="set"
> data="nibble_rate=${lcr_rate_1}"/>
> ? <action application="set" data="foo=11"/>
> ? <action application="set"
> data="accounts_tarifa=${tarifa}"/>
> 
> My custom SQL:
> 
> 
> ? ? ? <param name="custom_sql" value="
> SELECT l.digits AS lcr_digits,
> ? ? ???c.carrier_name AS
> lcr_carrier_name,
> ? ? ???l.${lcr_rate_field} as
> lcr_rate_field,
> ? ? ???cg.prefix AS lcr_gw_prefix,
> cg.suffix AS lcr_gw_suffix,
> ? ? ???l.lead_strip AS
> lcr_lead_strip, l.trail_strip AS lcr_trail_strip,
> ? ? ???l.prefix AS lcr_prefix,
> l.suffix AS lcr_suffix,
> ? ? ???a.id AS nibble_account,
> ? ? ???l.${lcr_rate_field} AS
> nibble_rate,
> ? ? ???a.warn_limit AS lowbal_amt,
> ? ? ???a.nobal_limit AS nobal_amt,
> ? ? ???a.id_tarifa AS tarifa
> ? FROM lcr l
> ? ? JOIN carriers c ON l.carrier_id=c.id
> ? ? JOIN carrier_gateway cg ON c.id=cg.carrier_id
> ? ? JOIN accounts a ON a.id = '${accountcode}'
> ? ? JOIN tarifas t ON a.id_tarifa = t.id 
> ? WHERE c.enabled = '1' AND cg.enabled = '1' AND
> l.enabled = '1'
> ? ? AND l.digits @> '${lcr_query_digits}'
> ? ? AND CURRENT_TIMESTAMP BETWEEN date_start AND
> date_end
> ? ORDER BY digits DESC, ${lcr_rate_field}, random();
> 
> 
> Log :
> 
> 
> 2012-06-11 11:33:52.336613 [DEBUG] mod_lcr.c:940 SQL:?
> SELECT l.digits AS lcr_digits,? ? ? ?
> c.carrier_name AS lcr_carrier_name,? ? ?
> ? l.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,? ? ? ? a.id AS
> nibble_account,? ? ? ? l.rate AS
> nibble_rate,? ? ? ? a.warn_limit AS
> lowbal_amt,? ? ? ? a.nobal_limit AS
> nobal_amt,? ? ? ? a.id_tarifa AS
> tarifa???FROM lcr l?
> ???JOIN carriers c ON l.carrier_id=c.id?
> ???JOIN carrier_gateway cg ON
> c.id=cg.carrier_id? ???JOIN accounts a
> ON a.id = '1004'? ???JOIN tarifas t ON
> a.id_tarifa = t.id? ? WHERE c.enabled = '1' AND
> cg.enabled = '1' AND l.enabled = '1'?
> ???AND l.digits @> '1006'?
> ???AND CURRENT_TIMESTAMP BETWEEN date_start
> AND date_end???ORDER BY digits DESC, rate,
> random();? ? ???
> 2012-06-11 11:33:52.356514 [DEBUG] mod_lcr.c:338 Returning
> Dialstring
> [lcr_carrier=carrier2,lcr_rate=0.14000,nibble_account=1004,nibble_rate=0.14000,lowbal_amt=0,nobal_amt=0,tarifa=1]user/gateway/0006
> 2012-06-11 11:33:52.356514 [DEBUG] mod_lcr.c:677 Adding
> carrier2 to head of list
> 2012-06-11 11:33:52.356514 [DEBUG] mod_lcr.c:338 Returning
> Dialstring
> [lcr_carrier=carrier1,lcr_rate=0.15000,nibble_account=1004,nibble_rate=0.15000,lowbal_amt=0,nobal_amt=0,tarifa=1]user/1006
> 2012-06-11 11:33:52.356514 [DEBUG] mod_lcr.c:705 Adding
> carrier1 to end of list
> EXECUTE sofia/internal/1004 at 192.168.1.108
> set(nibble_account=1004)
> 2012-06-11 11:33:52.356514 [DEBUG] mod_dptools.c:1294
> sofia/internal/1004 at 192.168.1.108 SET
> [nibble_account]=[1004]
> EXECUTE sofia/internal/1004 at 192.168.1.108
> set(nibble_rate=0.14000)
> 2012-06-11 11:33:52.356514 [DEBUG] mod_dptools.c:1294
> sofia/internal/1004 at 192.168.1.108 SET
> [nibble_rate]=[0.14000]
> EXECUTE sofia/internal/1004 at 192.168.1.108 set(foo=11)
> 2012-06-11 11:33:52.356514 [DEBUG] mod_dptools.c:1294
> sofia/internal/1004 at 192.168.1.108 SET [foo]=[11]
> EXECUTE sofia/internal/1004 at 192.168.1.108
> set(accounts_tarifa=)
> 2012-06-11 11:33:52.356514 [DEBUG] mod_dptools.c:1294
> sofia/internal/1004 at 192.168.1.108 SET
> [accounts_tarifa]=[UNDEF]
> 
> 
> My accounts_tarifa variable is UNDEF
> 
> What i am doing wrong?
> 
> Regard
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From alex at thewinelake.com  Tue Jun 12 01:16:20 2012
From: alex at thewinelake.com (Alex Lake)
Date: Mon, 11 Jun 2012 22:16:20 +0100
Subject: [Freeswitch-users] Only calling the first element in the list
	in this mode
In-Reply-To: <CAKV-jFzbuOMXFBU-hF7vmWsWZbVXtxBivx9D1SLO5zAJ2Vo2tA@mail.gmail.com>
References: <4FD5DAEA.4000601@digitalmail.com>
	<CAKV-jFwX7qV73-VCNhzco0Jyb-zmCE8C4AFQjPnc9uKYCZvkZw@mail.gmail.com>
	<4FD5F824.1070705@thewinelake.com>
	<CAKV-jFwM1iT-bNExCukZ68+_xJbBKbeVJ_ebT7m7w+zoC0cQzQ@mail.gmail.com>
	<4FD605FB.4030503@thewinelake.com>
	<CAKV-jFy+=pzVS8u0kXFB1YD8Leo+VaXQbKRNHSb5htcWY6NJog@mail.gmail.com>
	<CAKV-jFzbuOMXFBU-hF7vmWsWZbVXtxBivx9D1SLO5zAJ2Vo2tA@mail.gmail.com>
Message-ID: <E2880F24-CEC6-4D98-8996-0D56D8726F60@digitalmail.com>

Well, whaddyaknow - using :_: fixed it!
Thanks a million...


On 11 Jun 2012, at 16:21, Brian Foster <bdfoster at endigotech.com> wrote:

> This is going into stuff I haven't played with, but I'm trying my best :-)
> 
> In your bridge string, instead of using a comma have you tried using :_: ?
> 
> Brian Foster
> Endigo Computer LLC
> 
> Sent from a mobile device.
> 
> On Jun 11, 2012 11:03 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:
> At the end of your bridge string there is a Sofia/. Check that, could be by design but I'm not understanding what you are doing there.
> 
> Brian Foster
> Endigo Computer LLC
> 
> Sent from a mobile device.
> 
> On Jun 11, 2012 10:57 AM, "Alex Lake" <alex at thewinelake.com> wrote:
> Brian, when you say "it" (as in "it does work fine for me") are you doing a bridge to several different destinations, or are you just talking about bridging to a user that happens to have multiple SIP registrations?
> Thanks for the encouragement, anyway...
> Alex
>> I don't really understand your dialplan since I don't see everything going on, but make sure you are using user/<EXTNUMBER>@domain.tld for your bridges.
>> 
>> Otherwise, ill have to defer to someone else.more.knowledgeable on that subject. I will note that it does work fine for me and I am on latest as of last night.
>> 
>> Brian Foster
>> Endigo Computer LLC
>> 
>> Sent from a mobile device.
>> 
>> On Jun 11, 2012 9:58 AM, "Alex Lake" <alex at thewinelake.com> wrote:
>> Yes, that already is set to true - but I think that's what might be causing the problem!
>> 
>> Let me explain the scenario a little better:
>> 
>> A DDI is linked to a dialplan script a bit like this:
>> 
>> <include>
>>  <extension name="0044302">
>>   <condition field="destination_number" expression="^(302|441316110347)$">
>>    <action application="set" data="sb_routing=${sb_routing}&441316110347_302_PersonalExtension_Rob Darwin two"/>
>>    <action application="set" data="tenant_id=0044"/>
>>    <action application="set" data="b_ext=302"/>
>>    <action application="export" data="alert_info=http://lnhnov11.dmclub.net/generic/tones/Personal.wav"/>
>>    <action application="set" data="call_timeout=60"/>
>>    <action application="set" data="hangup_after_bridge=true"/>
>>    <action application="set" data="continue_on_fail=true"/>
>>    <action application="set" data="bridge_generate_comfort_noise=true"/>
>>    <action application="set" data="group_confirm_key=exec"/>
>>    <action application="export" data="whisper_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/302_whisper.wav"/>
>>    <action application="export" data="accept_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/accept1.wav"/>
>>    <action application="set" data="ringback=$${uk-ring}"/>
>>    <action application="set" data="transfer_ringback_=$${uk-ring}"/>
>>    <action application="set" data="ignore_early_media=true"/>
>>    <action application="set" data="group_confirm_file=lua confirmcall.lua"/>
>>    <action application="bridge" data="[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@${domain_name},[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020${ani}]sofia/internal/898000000006207855360320 at a.b.c.d"/>
>>    <action application="set" data="comment=pers_inhours_VM" />
>>    <action application="answer" />
>>    <action application="sleep" data="1000"/>
>>    <action application="voicemail" data="default ${domain_name} 0044302"/>
>>   </condition>
>>  </extension>
>> </include>
>> 
>> You'll see from the bridge command that one of the outbound legs is to a user URL and the other to a sofia URL. In actuality that is 1 SIP handset and 1 PSTN destination (via a PSTN gateway)
>> 
>> I believe we have only 1 sofia profile in the whole system - which is "internal", and that certainly has             multiple-registrations set to "true" and I can see that's working by attempting to register simultaneously from lots of SIP handsets and then "show registrations" or "sofia_contact" from fs_cli.
>> 
>> I'm wondering if I disable multiple SIP registrations, that it might then allow multiple-destination bridge commands.
>> 
>>> Set multiple-registrations to true in the affected Sofia profile if you haven't already done so and report back.
>>> 
>>> Brian Foster
>>> Endigo Computer LLC
>>> 
>>> Sent from a mobile device.
>>> 
>>> On Jun 11, 2012 7:49 AM, "Alex Lake" <alex at digitalmail.com> wrote:
>>> Having upgraded to the latest build (we were previously on a build from
>>> 2011), I'm finding that multiple SIP registrations are now working, but
>>> that multi-destination bridge commands are not.
>>> 
>>> Getting messages
>>> 2012-06-11 11:45:00.565775 [WARNING] switch_ivr_originate.c:2353 Only
>>> calling the first element in the list in this mode.
>>> 
>>> Is there some special "allow multi destination" setting that I need to add?
>>> 
>>> Rgds,
>>> Alex
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> Join Us At ClueCon - Aug 7-9, 2012
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> Join Us At ClueCon - Aug 7-9, 2012
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> No virus found in this message.
>>> Checked by AVG - www.avg.com
>>> Version: 2012.0.2177 / Virus Database: 2433/5062 - Release Date: 06/11/12
>>> 
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> No virus found in this message.
>> Checked by AVG - www.avg.com
>> Version: 2012.0.2177 / Virus Database: 2433/5062 - Release Date: 06/11/12
>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun 12 01:23:29 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jun 2012 16:23:29 -0500
Subject: [Freeswitch-users] Only calling the first element in the list
 in this mode
In-Reply-To: <E2880F24-CEC6-4D98-8996-0D56D8726F60@digitalmail.com>
References: <4FD5DAEA.4000601@digitalmail.com>
	<CAKV-jFwX7qV73-VCNhzco0Jyb-zmCE8C4AFQjPnc9uKYCZvkZw@mail.gmail.com>
	<4FD5F824.1070705@thewinelake.com>
	<CAKV-jFwM1iT-bNExCukZ68+_xJbBKbeVJ_ebT7m7w+zoC0cQzQ@mail.gmail.com>
	<4FD605FB.4030503@thewinelake.com>
	<CAKV-jFy+=pzVS8u0kXFB1YD8Leo+VaXQbKRNHSb5htcWY6NJog@mail.gmail.com>
	<CAKV-jFzbuOMXFBU-hF7vmWsWZbVXtxBivx9D1SLO5zAJ2Vo2tA@mail.gmail.com>
	<E2880F24-CEC6-4D98-8996-0D56D8726F60@digitalmail.com>
Message-ID: <CAKbxfG8RMtfWP-Gj8og3tzZD=ShG8Psn4SfXkk4XseKSMxL8vA@mail.gmail.com>

This is because you cannot put a forked dial as the dial string for a
/user and then use that /user in another forked dial, this is a double
forked dial.
Enterprise originate (ie :_: ) solves this with multi-threaded
simultaneous originates.




On Mon, Jun 11, 2012 at 4:16 PM, Alex Lake <alex at thewinelake.com> wrote:
> Well, whaddyaknow - using :_: fixed it!
> Thanks a million...
>
>
> On 11 Jun 2012, at 16:21, Brian Foster <bdfoster at endigotech.com> wrote:
>
> This is going into stuff I haven't played with, but I'm trying my best :-)
>
> In your bridge string, instead of using a comma have you tried using :_: ?
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
>
> On Jun 11, 2012 11:03 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>>
>> At the end of your bridge string there is a Sofia/. Check that, could be
>> by design but I'm not understanding what you are doing there.
>>
>> Brian Foster
>> Endigo Computer LLC
>>
>> Sent from a mobile device.
>>
>> On Jun 11, 2012 10:57 AM, "Alex Lake" <alex at thewinelake.com> wrote:
>>>
>>> Brian, when you say "it" (as in "it does work fine for me") are you doing
>>> a bridge to several different destinations, or are you just talking about
>>> bridging to a user that happens to have multiple SIP registrations?
>>> Thanks for the encouragement, anyway...
>>> Alex
>>>
>>> I don't really understand your dialplan since I don't see everything
>>> going on, but make sure you are using user/<EXTNUMBER>@domain.tld for your
>>> bridges.
>>>
>>> Otherwise, ill have to defer to someone else.more.knowledgeable on that
>>> subject. I will note that it does work fine for me and I am on latest as of
>>> last night.
>>>
>>> Brian Foster
>>> Endigo Computer LLC
>>>
>>> Sent from a mobile device.
>>>
>>> On Jun 11, 2012 9:58 AM, "Alex Lake" <alex at thewinelake.com> wrote:
>>>>
>>>> Yes, that already is set to true - but I think that's what might be
>>>> causing the problem!
>>>>
>>>> Let me explain the scenario a little better:
>>>>
>>>> A DDI is linked to a dialplan script a bit like this:
>>>>
>>>> <include>
>>>> ?<extension name="0044302">
>>>> ? <condition field="destination_number"
>>>> expression="^(302|441316110347)$">
>>>> ?? <action application="set"
>>>> data="sb_routing=${sb_routing}&441316110347_302_PersonalExtension_Rob Darwin
>>>> two"/>
>>>> ?? <action application="set" data="tenant_id=0044"/>
>>>> ?? <action application="set" data="b_ext=302"/>
>>>> ?? <action application="export"
>>>> data="alert_info=http://lnhnov11.dmclub.net/generic/tones/Personal.wav"/>
>>>> ?? <action application="set" data="call_timeout=60"/>
>>>> ?? <action application="set" data="hangup_after_bridge=true"/>
>>>> ?? <action application="set" data="continue_on_fail=true"/>
>>>> ?? <action application="set" data="bridge_generate_comfort_noise=true"/>
>>>> ?? <action application="set" data="group_confirm_key=exec"/>
>>>> ?? <action application="export"
>>>> data="whisper_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/302_whisper.wav"/>
>>>> ?? <action application="export"
>>>> data="accept_msg=/home/pabx/004-0044/20120611-114725-00001/recordings/tts/accept1.wav"/>
>>>> ?? <action application="set" data="ringback=$${uk-ring}"/>
>>>> ?? <action application="set" data="transfer_ringback_=$${uk-ring}"/>
>>>> ?? <action application="set" data="ignore_early_media=true"/>
>>>> ?? <action application="set" data="group_confirm_file=lua
>>>> confirmcall.lua"/>
>>>> ?? <action application="bridge"
>>>> data="[tenant_id=0044,b_ext=302,accept_mode=Direct]user/0044302@${domain_name},[tenant_id=0044,b_ext=302,leg_delay_start=60,accept_mode=Direct,origination_caller_id_number=00443020${ani}]sofia/internal/898000000006207855360320 at a.b.c.d"/>
>>>> ?? <action application="set" data="comment=pers_inhours_VM" />
>>>> ?? <action application="answer" />
>>>> ?? <action application="sleep" data="1000"/>
>>>> ?? <action application="voicemail" data="default ${domain_name}
>>>> 0044302"/>
>>>> ? </condition>
>>>> ?</extension>
>>>> </include>
>>>>
>>>> You'll see from the bridge command that one of the outbound legs is to a
>>>> user URL and the other to a sofia URL. In actuality that is 1 SIP handset
>>>> and 1 PSTN destination (via a PSTN gateway)
>>>>
>>>> I believe we have only 1 sofia profile in the whole system - which is
>>>> "internal", and that certainly has multiple-registrations set to "true" and
>>>> I can see that's working by attempting to register simultaneously from lots
>>>> of SIP handsets and then "show registrations" or "sofia_contact" from
>>>> fs_cli.
>>>>
>>>> I'm wondering if I disable multiple SIP registrations, that it might
>>>> then allow multiple-destination bridge commands.
>>>>
>>>> Set multiple-registrations to true in the affected Sofia profile if you
>>>> haven't already done so and report back.
>>>>
>>>> Brian Foster
>>>> Endigo Computer LLC
>>>>
>>>> Sent from a mobile device.
>>>>
>>>> On Jun 11, 2012 7:49 AM, "Alex Lake" <alex at digitalmail.com> wrote:
>>>>>
>>>>> Having upgraded to the latest build (we were previously on a build from
>>>>> 2011), I'm finding that multiple SIP registrations are now working, but
>>>>> that multi-destination bridge commands are not.
>>>>>
>>>>> Getting messages
>>>>> 2012-06-11 11:45:00.565775 [WARNING] switch_ivr_originate.c:2353 Only
>>>>> calling the first element in the list in this mode.
>>>>>
>>>>> Is there some special "allow multi destination" setting that I need to
>>>>> add?
>>>>>
>>>>> Rgds,
>>>>> Alex
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> No virus found in this message.
>>>> Checked by AVG - www.avg.com
>>>> Version: 2012.0.2177 / Virus Database: 2433/5062 - Release Date:
>>>> 06/11/12
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> No virus found in this message.
>>> Checked by AVG - www.avg.com
>>> Version: 2012.0.2177 / Virus Database: 2433/5062 - Release Date: 06/11/12
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From lloyd.aloysius at gmail.com  Tue Jun 12 05:17:35 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Mon, 11 Jun 2012 21:17:35 -0400
Subject: [Freeswitch-users] NAT Router Recommendation For Large no of IP
	Phones Behind the Router
Message-ID: <CAGvuWYaqsqO6N0y+5+pJ2zyA0UqS0iAFjQiXuaSxssH52GcdVQ@mail.gmail.com>

Hi All:

Currently I use pfSense for NAT implementations. But I found there are few
bugs in the firmware cause Presence not working reliably.

What are best routers out there with good support for SIP. I am looking for
a NAT router that support 20-30 phones . Any good recommendations?

Thanks
Lloyd
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From sos at sokhapkin.dyndns.org  Tue Jun 12 05:27:46 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 11 Jun 2012 21:27:46 -0400
Subject: [Freeswitch-users] NAT Router Recommendation For Large no of IP
	Phones Behind the Router
In-Reply-To: <CAGvuWYaqsqO6N0y+5+pJ2zyA0UqS0iAFjQiXuaSxssH52GcdVQ@mail.gmail.com>
References: <CAGvuWYaqsqO6N0y+5+pJ2zyA0UqS0iAFjQiXuaSxssH52GcdVQ@mail.gmail.com>
Message-ID: <1421146.EMNVR6yscs@sos>

 The best router is a router which knows nothing about SIP. I'm not kidding. 
Let SIP provider's server handle NAT-related issues.

On Monday 11 June 2012 21:17:35 Lloyd Aloysius wrote:
> Hi All:
> 
> Currently I use pfSense for NAT implementations. But I found there are few
> bugs in the firmware cause Presence not working reliably.
> 
> What are best routers out there with good support for SIP. I am looking for
> a NAT router that support 20-30 phones . Any good recommendations?
> 
> Thanks
> Lloyd


From gabe at gundy.org  Tue Jun 12 07:16:58 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 11 Jun 2012 21:16:58 -0600
Subject: [Freeswitch-users] NAT Router Recommendation For Large no of IP
 Phones Behind the Router
In-Reply-To: <1421146.EMNVR6yscs@sos>
References: <CAGvuWYaqsqO6N0y+5+pJ2zyA0UqS0iAFjQiXuaSxssH52GcdVQ@mail.gmail.com>
	<1421146.EMNVR6yscs@sos>
Message-ID: <CAMwZ-tLYqY32LR+6maDeiapetbymU-M+CGTDrbfpQfFdRdvXHA@mail.gmail.com>

On Mon, Jun 11, 2012 at 7:27 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> The best router is a router which knows nothing about SIP. I'm not kidding.
> Let SIP provider's server handle NAT-related issues.

I agree. If you monkey with the SIP messages, it makes their job much harder.


Gabe


From yehavi.bourvine at gmail.com  Tue Jun 12 07:56:49 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 12 Jun 2012 06:56:49 +0300
Subject: [Freeswitch-users] Polycom caller ID problem on 4.0.2
In-Reply-To: <CAAB3jWsQPFyrr1d=d6gfA0G9Q+kyM0v7VyvDoOYrq=h6LaZ9rA@mail.gmail.com>
References: <CAAB3jWsQPFyrr1d=d6gfA0G9Q+kyM0v7VyvDoOYrq=h6LaZ9rA@mail.gmail.com>
Message-ID: <CAAtWKeFcgvVbW03ece3yGFLGoXxeP5uiDuTTUWeQ7s+5wmqs2g@mail.gmail.com>

You will soon notice another problem with them: If you use shared lines
then all outgoing calls are listed as incoming, and all unanswered outgoing
calls are listed as incoming missed calls...

                     __Yehavi:

2012/6/11 DJB International <djbinter at gmail.com>

> I've recently updated my Polycom 650 phone to use 4.0.2.
>
> I am not sure what broke the caller id for received calls, it kept showing
> as Unknown Party; however, the weird thing is that the Missed Calls show
> the correct caller id.
>
> Anyone experienced this issue.  Please advise.
>
> FreeSWITCH Version 1.2.0-rc2
>
> Thank you.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From saami_mh at ymail.com  Tue Jun 12 08:06:35 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Mon, 11 Jun 2012 21:06:35 -0700 (PDT)
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <4FD6371D.1010307@cupis.co.uk>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
	<1339414395.62082.YahooMailNeo@web120105.mail.ne1.yahoo.com>
	<30DFAA60-ADB7-4B3A-A5FE-9085491FD83E@opencsta.org>
	<1339433685.46946.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<4FD6371D.1010307@cupis.co.uk>
Message-ID: <1339473995.42452.YahooMailNeo@web120106.mail.ne1.yahoo.com>

thansk for your reply,
it is kind of you to help me..
please let me paste myconfigurations files here;
1-the configuration ?file /usr/local/freeswitch/conf/dialplan/default/001_luacallduration.xml ?is like this:

<include>
? <extension name="mainmenuvodsl">

? ? ? ? <condition field="destination_number" expression="^(00|\+)?(\d{5}.*)$" break="never">
? ? ??? ? ? ? ? ? ?<action application="odbc_query" data="select cash as cashvalue from accounts where contractid like '${nibble_account}';"/>
? ? ? ? ? ? ? ? <action application="log" data="INFO The value of cashvalue is ${cashvalue}" />

?? ? ? ? ? ? ? ? <action application="lua" data="checkcash.lua ${cashvalue}" />

? ? ? ? ? ? ? ? <action application="log" data="INFO The value of nibble_account is ?${nibble_account}"/>
? ? ? ? ? ? ? ? <action application="log" data="INFO The value of nibble_rate [before] is ?${nibble_rate}"/>
? ? ? ? ? ? ? ? <!-- RateList Context -->
? ? ? ? ? ? ? ? <action application="lua" data="checkzeroplus.lua ${destination_number:0:2} ${destination_number:0:1}" />
? ? ? ? ? ? ? ? <action application="execute_extension" data="${destination_number} XML ratelist"/>
? ? ? ? ? ? ? ? <action application="log" data="INFO The value of nibble_rate [after] is ${nibble_rate}"/>
? ? ? ? ? ? ? ? ?<!-- Check Nibble_rate -->
? ? ? ? ? ? ? ? <action application="lua" data="checknibblerate.lua ${nibble_rate}" />
? ? ? ? ? ? ? ? <action application="set" data="divvalue=${expr(floor((${cashvalue}/${nibble_rate}))}" />
? ? ? ? ? ? ? ? <action application="set" data="modvalue=${expr(mod(${cashvalue},${nibble_rate}))}" />
? ? ? ? ? ? ? ? <action application="lua" data="checktime.lua ${divvalue} ${modvalue}" />
? ? ? ? ? ? ? ?? ? ?<!-- ?Check ZeroZero , Plus ?-->
? ? ? ? ? ? ?? ?<action application="lua" data="checkzeroplus.lua ${destination_number:0:2} ${destination_number:0:1}" />?
? ? ? ? ? ? ? ? <!-- Making Calls ? -->
? ? ? ? ? ? ? ? <action application="odbc_query" data="select callerid ?as effective_caller_id_number from accounts where contractid like '${nibble_account}';"/>
? ? ? ? ? ? ? ? <action application="log" data="INFO ?callerid for Outbound calls ${effective_caller_id_number}"/>
? ? ? ? ? ? ? ? <!-- <action application="set" data="ignore_early_media=true"/>
? ? ? ? ? ? ? ? <action application="answer"/> ?-->
? ? ? ? ? ? ? ? <action application="enable_heartbeat"/>
? ? ? ?? ? ? ? ??
<!-- <param name="disable-transcoding" value="true"/> -->
? ? ? ? <!-- ? ?<action application="export" data="nolocal:absolute_codec_string=G729,PCMU"/> -->
<!-- ?<action application="set" data="bridge_early_media=true"/> ?-->
? ? ? ? <!-- <action application="set" data="proxy_media=true"/> -->
? ? ? ? ? ? ? ? <action application="bridge" data="sofia/gateway/cisco/140112${destination_number}"/>

? ? ? ? ? ? ? ? <!-- <action application="bridge" data="sofia/gateway/mainasterisk/${destination_number}"/> ?-->
<!-- ?<action application="bridge" data="sofia/gateway/test/${destination_number}"/> ?-->
? ? ? ? </condition>

?</extension>
</include>

2-yes, i have enabled ?"inbound-late-negotiation"?in the (/usr/local/freeswitch/conf/sip_profiles/internal.xml) as follow:
?<param name="inbound-late-negotiation" value="true"/>



3-the issue of sofia status:
?external::cisco ? ? ? gateway ? ? ? ? ? ? sip:register:false at 85.15.0.154 ? ? ?NOREG



4-also , the configuration file for codecs are as follow
:/usr/local/freeswitch/conf/vars.xml

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA,G7221 at 32000h,G7221 at 16000h,G722,GSM"/>?

<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>

5- the mod_g729 was loaded?

6-i have enabled the siptrace:
?sofia profile external siptrace on:

the siptrace outpout as follow:

send 1042 bytes to udp/[85.15.0.154]:5060 at 03:53:07.448136:
? ?------------------------------------------------------------------------
? ?INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
? ?Max-Forwards: 69
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
? ?To: <sip:140112971507247227 at 85.15.0.154>
? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
? ?CSeq: 29400529 INVITE
? ?Contact: <sip:gw+cisco at 192.168.10.70:5080;transport=udp;gw=cisco>
? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
? ?Supported: timer, precondition, path, replaces
? ?Allow-Events: talk, hold, refer
? ?Content-Type: application/sdp
? ?Content-Disposition: session
? ?Content-Length: 234
? ?X-FS-Support: update_display,send_info
? ?Remote-Party-ID: "1000" <sip:1000 at 85.15.0.154>;party=calling;screen=yes;privacy=off

? ?v=0
? ?o=FreeSWITCH 1339446571 1339446572 IN IP4 192.168.10.70
? ?s=FreeSWITCH
? ?c=IN IP4 192.168.10.70
? ?t=0 0
? ?m=audio 26616 RTP/AVP 9 0 8 18 3 101 13
? ?a=fmtp:18 annexb=yes
? ?a=rtpmap:101 telephone-event/8000
? ?a=fmtp:101 0-16
? ?a=ptime:20
? ?------------------------------------------------------------------------
recv 410 bytes from udp/[85.15.0.154]:5060 at 03:53:07.463921:
? ?------------------------------------------------------------------------
? ?SIP/2.0 100 Trying
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
? ?Server: Cisco-SIPGateway/IOS-12.x
? ?CSeq: 29400529 INVITE
? ?Allow-Events: telephone-event
? ?Content-Length: 0

? ?------------------------------------------------------------------------
recv 927 bytes from udp/[85.15.0.154]:5060 at 03:53:11.004804:
? ?------------------------------------------------------------------------
? ?SIP/2.0 183 Session Progress
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
? ?Server: Cisco-SIPGateway/IOS-12.x
? ?CSeq: 29400529 INVITE
? ?Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
? ?Allow-Events: telephone-event
? ?Contact: <sip:140112971507247227 at 85.15.0.154:5060>
? ?Content-Disposition: session;handling=required
? ?Content-Type: application/sdp
? ?Content-Length: 268

? ?v=0
? ?o=CiscoSystemsSIP-GW-UserAgent 7252 5649 IN IP4 85.15.0.154
? ?s=SIP Call
? ?c=IN IP4 85.15.0.154
? ?t=0 0
? ?m=audio 18218 RTP/AVP 0 13 101
? ?c=IN IP4 85.15.0.154
? ?a=rtpmap:0 PCMU/8000
? ?a=rtpmap:13 CN/8000
? ?a=rtpmap:101 telephone-event/8000
? ?a=fmtp:101 0-15
? ?a=ptime:20
? ?------------------------------------------------------------------------
recv 425 bytes from udp/[85.15.0.154]:5060 at 03:53:11.005144:
? ?------------------------------------------------------------------------
? ?SIP/2.0 500 Internal Server Error
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
? ?Server: Cisco-SIPGateway/IOS-12.x
? ?CSeq: 29400529 INVITE
? ?Allow-Events: telephone-event
? ?Content-Length: 0

? ?------------------------------------------------------------------------
send 350 bytes to udp/[85.15.0.154]:5060 at 03:53:11.005333:
? ?------------------------------------------------------------------------
? ?ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
? ?Max-Forwards: 69
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
? ?CSeq: 29400529 ACK
? ?Content-Length: 0


------------------------------------------------------------------------------------------------------------------
when change the configuration file the below:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>?


the siptrace is like this:

send 1034 bytes to udp/[85.15.0.154]:5060 at 04:01:16.202342:
? ?------------------------------------------------------------------------
? ?INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
? ?Max-Forwards: 69
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
? ?To: <sip:140112971507247227 at 85.15.0.154>
? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
? ?CSeq: 29400774 INVITE
? ?Contact: <sip:gw+cisco at 192.168.10.70:5080;transport=udp;gw=cisco>
? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
? ?Supported: timer, precondition, path, replaces
? ?Allow-Events: talk, hold, refer
? ?Content-Type: application/sdp
? ?Content-Disposition: session
? ?Content-Length: 226
? ?X-FS-Support: update_display,send_info
? ?Remote-Party-ID: "1000" <sip:1000 at 85.15.0.154>;party=calling;screen=yes;privacy=off

? ?v=0
? ?o=FreeSWITCH 1339447862 1339447863 IN IP4 192.168.10.70
? ?s=FreeSWITCH
? ?c=IN IP4 192.168.10.70
? ?t=0 0
? ?m=audio 25814 RTP/AVP 18 101 13
? ?a=fmtp:18 annexb=yes
? ?a=rtpmap:101 telephone-event/8000
? ?a=fmtp:101 0-16
? ?a=ptime:20
? ?------------------------------------------------------------------------
recv 423 bytes from udp/[85.15.0.154]:5060 at 04:01:16.208118:
? ?------------------------------------------------------------------------
? ?SIP/2.0 488 Not Acceptable Media
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=457FC664-6A6
? ?Date: Tue, 12 Jun 2012 04:01:24 GMT
? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
? ?Server: Cisco-SIPGateway/IOS-12.x
? ?CSeq: 29400774 INVITE
? ?Allow-Events: telephone-event
? ?Content-Length: 0

? ?------------------------------------------------------------------------
send 349 bytes to udp/[85.15.0.154]:5060 at 04:01:16.208201:
? ?------------------------------------------------------------------------
? ?ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
? ?Max-Forwards: 69
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=457FC664-6A6
? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
? ?CSeq: 29400774 ACK
? ?Content-Length: 0



plz help,thanks so much



________________________________
 From: Paul Cupis <paul at cupis.co.uk>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Monday, June 11, 2012 10:51 PM
Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
 
On 11/06/12 17:54, Samira Mh wrote:
> i want to bridge call using my VOIPgateway so that making calls to
> another countries..
> but the carrier only support G729 codec and the FS send G722 (set in
> vars.xml) to myVoipGateway that is set as an gateway in
> /usr/local/freeswitch/sip-profile/external/
> and when FS send media to Gateway(using bridge application) the error
> occure:unacceptable media,then check VOIPGW and find out the only codec
> that
> can be pass through VOIPgw is G729, but FS only send G711,G722,... not G729

Can you provide a SIP or FreeSWITCH trace of a call, please?

Do you have the following enabled in your SIP profile?

?  <param name="inbound-late-negotiation" value="true"/>

Do you have mod_g729 loaded and codec G729 enabled in your vars.xml?

Regards,

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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Join Us At ClueCon - Aug 7-9, 2012

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From saami_mh at ymail.com  Tue Jun 12 08:13:10 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Mon, 11 Jun 2012 21:13:10 -0700 (PDT)
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <1339473995.42452.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
	<1339414395.62082.YahooMailNeo@web120105.mail.ne1.yahoo.com>
	<30DFAA60-ADB7-4B3A-A5FE-9085491FD83E@opencsta.org>
	<1339433685.46946.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<4FD6371D.1010307@cupis.co.uk>
	<1339473995.42452.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <1339474390.46866.YahooMailNeo@web120106.mail.ne1.yahoo.com>

sorry i forgot to paste codec on internal and external:

freeswitch at internal> sofia status profile internal
=================================================================================================
Name ? ? ? ? ? ? ? ? ? ?internal
Domain Name ? ? ? ? ? ? N/A
Auto-NAT ? ? ? ? ? ? ? ?false
DBName ? ? ? ? ? ? ? ? ?sofia_reg_internal
Pres Hosts ? ? ? ? ? ? ?192.168.10.70,192.168.10.70
Dialplan ? ? ? ? ? ? ? ?XML
Context ? ? ? ? ? ? ? ? public
Challenge Realm ? ? ? ? auto_from
RTP-IP ? ? ? ? ? ? ? ? ?192.168.10.70
SIP-IP ? ? ? ? ? ? ? ? ?192.168.10.70
URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.10.70:5060
BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.10.70:5060
HOLD-MUSIC ? ? ? ? ? ? ?local_stream://moh
OUTBOUND-PROXY ? ? ? ? ?N/A
CODECS IN ? ? ? ? ? ? ? G729
CODECS OUT ? ? ? ? ? ? ?G729
TEL-EVENT ? ? ? ? ? ? ? 101
DTMF-MODE ? ? ? ? ? ? ? rfc2833
CNG ? ? ? ? ? ? ? ? ? ? 13
SESSION-TO ? ? ? ? ? ? ?0
MAX-DIALOG ? ? ? ? ? ? ?0
NOMEDIA ? ? ? ? ? ? ? ? false
LATE-NEG ? ? ? ? ? ? ? ?true
PROXY-MEDIA ? ? ? ? ? ? false
ZRTP-PASSTHRU ? ? ? ? ? false
AGGRESSIVENAT ? ? ? ? ? false
STUN-ENABLED ? ? ? ? ? ?true
STUN-AUTO-DISABLE ? ? ? false
CALLS-IN ? ? ? ? ? ? ? ?0
FAILED-CALLS-IN ? ? ? ? 0
CALLS-OUT ? ? ? ? ? ? ? 0
FAILED-CALLS-OUT ? ? ? ?0
REGISTRATIONS ? ? ? ? ? 3

---------------------------------------------------------------------------------
freeswitch at internal> sofia status profile external
=================================================================================================
Name ? ? ? ? ? ? ? ? ? ?external
Domain Name ? ? ? ? ? ? N/A
Auto-NAT ? ? ? ? ? ? ? ?false
DBName ? ? ? ? ? ? ? ? ?sofia_reg_external
Pres Hosts
Dialplan ? ? ? ? ? ? ? ?XML
Context ? ? ? ? ? ? ? ? public
Challenge Realm ? ? ? ? auto_to
RTP-IP ? ? ? ? ? ? ? ? ?192.168.10.70
Ext-RTP-IP ? ? ? ? ? ? ?192.168.10.70
SIP-IP ? ? ? ? ? ? ? ? ?192.168.10.70
Ext-SIP-IP ? ? ? ? ? ? ?192.168.10.70
URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.10.70:5080
BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.10.70:5080;maddr=192.168.10.70
HOLD-MUSIC ? ? ? ? ? ? ?local_stream://moh
OUTBOUND-PROXY ? ? ? ? ?N/A
CODECS IN ? ? ? ? ? ? ? G729
CODECS OUT ? ? ? ? ? ? ?PCMU
TEL-EVENT ? ? ? ? ? ? ? 101
DTMF-MODE ? ? ? ? ? ? ? rfc2833
CNG ? ? ? ? ? ? ? ? ? ? 13
SESSION-TO ? ? ? ? ? ? ?0
MAX-DIALOG ? ? ? ? ? ? ?0
NOMEDIA ? ? ? ? ? ? ? ? false
LATE-NEG ? ? ? ? ? ? ? ?false
PROXY-MEDIA ? ? ? ? ? ? false
ZRTP-PASSTHRU ? ? ? ? ? false
AGGRESSIVENAT ? ? ? ? ? false
STUN-ENABLED ? ? ? ? ? ?true
STUN-AUTO-DISABLE ? ? ? false
CALLS-IN ? ? ? ? ? ? ? ?0
FAILED-CALLS-IN ? ? ? ? 0
CALLS-OUT ? ? ? ? ? ? ? 0
FAILED-CALLS-OUT ? ? ? ?0
REGISTRATIONS ? ? ? ? ? 0



________________________________
 From: Samira Mh <saami_mh at ymail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Tuesday, June 12, 2012 8:36 AM
Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
 

thansk for your reply,
it is kind of you to help me..
please let me paste myconfigurations files here;
1-the configuration ?file /usr/local/freeswitch/conf/dialplan/default/001_luacallduration.xml ?is like this:

<include>
? <extension name="mainmenuvodsl">

? ? ? ? <condition field="destination_number" expression="^(00|\+)?(\d{5}.*)$" break="never">
? ? ??? ? ? ? ? ? ?<action application="odbc_query" data="select cash as cashvalue from accounts where contractid like '${nibble_account}';"/>
? ? ? ? ? ? ? ? <action application="log" data="INFO The value of cashvalue is ${cashvalue}" />

?? ? ? ? ? ? ? ? <action application="lua" data="checkcash.lua ${cashvalue}" />

? ? ? ? ? ? ? ? <action application="log" data="INFO The value of nibble_account is ?${nibble_account}"/>
? ? ? ? ? ? ? ? <action application="log" data="INFO The value of nibble_rate [before] is ?${nibble_rate}"/>
? ? ? ? ? ? ? ? <!-- RateList Context -->
? ? ? ? ? ? ? ? <action application="lua" data="checkzeroplus.lua ${destination_number:0:2} ${destination_number:0:1}" />
? ? ? ? ? ? ? ? <action application="execute_extension" data="${destination_number} XML ratelist"/>
? ? ? ? ? ? ? ? <action application="log" data="INFO The value of nibble_rate [after] is ${nibble_rate}"/>
? ? ? ? ? ? ? ? ?<!-- Check Nibble_rate -->
? ? ? ? ? ? ? ? <action application="lua" data="checknibblerate.lua ${nibble_rate}" />
? ? ? ? ? ? ? ? <action application="set" data="divvalue=${expr(floor((${cashvalue}/${nibble_rate}))}" />
? ? ? ? ? ? ? ? <action application="set" data="modvalue=${expr(mod(${cashvalue},${nibble_rate}))}" />
? ? ? ? ? ? ? ? <action application="lua" data="checktime.lua ${divvalue} ${modvalue}" />
? ? ? ? ? ? ? ?? ? ?<!-- ?Check ZeroZero , Plus ?-->
? ? ? ? ? ? ?? ?<action application="lua" data="checkzeroplus.lua ${destination_number:0:2} ${destination_number:0:1}" />?
? ? ? ? ? ? ? ? <!-- Making Calls ? -->
? ? ? ? ? ? ? ? <action application="odbc_query" data="select callerid ?as effective_caller_id_number from accounts where contractid like '${nibble_account}';"/>
? ? ? ? ? ? ? ? <action application="log" data="INFO ?callerid for Outbound calls ${effective_caller_id_number}"/>
? ? ? ? ? ? ? ? <!-- <action application="set" data="ignore_early_media=true"/>
? ? ? ? ? ? ? ? <action application="answer"/> ?-->
? ? ? ? ? ? ? ? <action application="enable_heartbeat"/>
? ? ? ?? ? ? ? ??
<!-- <param name="disable-transcoding" value="true"/> -->
? ? ? ? <!-- ? ?<action application="export" data="nolocal:absolute_codec_string=G729,PCMU"/> -->
<!-- ?<action application="set" data="bridge_early_media=true"/> ?-->
? ? ? ? <!-- <action application="set" data="proxy_media=true"/> -->
? ? ? ? ? ? ? ? <action application="bridge" data="sofia/gateway/cisco/140112${destination_number}"/>

? ? ? ? ? ? ? ? <!-- <action application="bridge" data="sofia/gateway/mainasterisk/${destination_number}"/> ?-->
<!-- ?<action application="bridge" data="sofia/gateway/test/${destination_number}"/> ?-->
? ? ? ? </condition>

?</extension>
</include>

2-yes, i have enabled ?"inbound-late-negotiation"?in the (/usr/local/freeswitch/conf/sip_profiles/internal.xml) as follow:
?<param name="inbound-late-negotiation" value="true"/>



3-the issue of sofia status:
?external::cisco ? ? ? gateway ? ? ? ? ? ? sip:register:false at 85.15.0.154 ? ? ?NOREG



4-also , the configuration file for codecs are as follow
:/usr/local/freeswitch/conf/vars.xml

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA,G7221 at 32000h,G7221 at 16000h,G722,GSM"/>?

<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>

5- the mod_g729 was loaded?

6-i have enabled the siptrace:
?sofia profile external siptrace on:

the siptrace outpout as follow:

send 1042 bytes to udp/[85.15.0.154]:5060 at 03:53:07.448136:
? ?------------------------------------------------------------------------
? ?INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
? ?Max-Forwards: 69
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
? ?To: <sip:140112971507247227 at 85.15.0.154>
? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
? ?CSeq: 29400529 INVITE
? ?Contact: <sip:gw+cisco at 192.168.10.70:5080;transport=udp;gw=cisco>
? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
? ?Supported: timer, precondition, path, replaces
? ?Allow-Events: talk, hold, refer
? ?Content-Type: application/sdp
? ?Content-Disposition: session
? ?Content-Length: 234
? ?X-FS-Support: update_display,send_info
? ?Remote-Party-ID: "1000" <sip:1000 at 85.15.0.154>;party=calling;screen=yes;privacy=off

? ?v=0
? ?o=FreeSWITCH 1339446571 1339446572 IN IP4 192.168.10.70
? ?s=FreeSWITCH
? ?c=IN IP4 192.168.10.70
? ?t=0 0
? ?m=audio 26616 RTP/AVP 9 0 8 18 3 101 13
? ?a=fmtp:18 annexb=yes
? ?a=rtpmap:101 telephone-event/8000
? ?a=fmtp:101 0-16
? ?a=ptime:20
? ?------------------------------------------------------------------------
recv 410 bytes from udp/[85.15.0.154]:5060 at 03:53:07.463921:
? ?------------------------------------------------------------------------
? ?SIP/2.0 100 Trying
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
? ?Server: Cisco-SIPGateway/IOS-12.x
? ?CSeq: 29400529 INVITE
? ?Allow-Events: telephone-event
? ?Content-Length: 0

? ?------------------------------------------------------------------------
recv 927 bytes from udp/[85.15.0.154]:5060 at 03:53:11.004804:
? ?------------------------------------------------------------------------
? ?SIP/2.0 183 Session Progress
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
? ?Server: Cisco-SIPGateway/IOS-12.x
? ?CSeq: 29400529 INVITE
? ?Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
? ?Allow-Events: telephone-event
? ?Contact: <sip:140112971507247227 at 85.15.0.154:5060>
? ?Content-Disposition: session;handling=required
? ?Content-Type: application/sdp
? ?Content-Length: 268

? ?v=0
? ?o=CiscoSystemsSIP-GW-UserAgent 7252 5649 IN IP4 85.15.0.154
? ?s=SIP Call
? ?c=IN IP4 85.15.0.154
? ?t=0 0
? ?m=audio 18218 RTP/AVP 0 13 101
? ?c=IN IP4 85.15.0.154
? ?a=rtpmap:0 PCMU/8000
? ?a=rtpmap:13 CN/8000
? ?a=rtpmap:101 telephone-event/8000
? ?a=fmtp:101 0-15
? ?a=ptime:20
? ?------------------------------------------------------------------------
recv 425 bytes from udp/[85.15.0.154]:5060 at 03:53:11.005144:
? ?------------------------------------------------------------------------
? ?SIP/2.0 500 Internal Server Error
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
? ?Server: Cisco-SIPGateway/IOS-12.x
? ?CSeq: 29400529 INVITE
? ?Allow-Events: telephone-event
? ?Content-Length: 0

? ?------------------------------------------------------------------------
send 350 bytes to udp/[85.15.0.154]:5060 at 03:53:11.005333:
? ?------------------------------------------------------------------------
? ?ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
? ?Max-Forwards: 69
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
? ?CSeq: 29400529 ACK
? ?Content-Length: 0


------------------------------------------------------------------------------------------------------------------
when change the configuration file the below:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>?


the siptrace is like this:

send 1034 bytes to udp/[85.15.0.154]:5060 at 04:01:16.202342:
? ?------------------------------------------------------------------------
? ?INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
? ?Max-Forwards: 69
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
? ?To: <sip:140112971507247227 at 85.15.0.154>
? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
? ?CSeq: 29400774 INVITE
? ?Contact: <sip:gw+cisco at 192.168.10.70:5080;transport=udp;gw=cisco>
? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
? ?Supported: timer, precondition, path, replaces
? ?Allow-Events: talk, hold, refer
? ?Content-Type: application/sdp
? ?Content-Disposition: session
? ?Content-Length: 226
? ?X-FS-Support: update_display,send_info
? ?Remote-Party-ID: "1000" <sip:1000 at 85.15.0.154>;party=calling;screen=yes;privacy=off

? ?v=0
? ?o=FreeSWITCH 1339447862 1339447863 IN IP4 192.168.10.70
? ?s=FreeSWITCH
? ?c=IN IP4 192.168.10.70
? ?t=0 0
? ?m=audio 25814 RTP/AVP 18 101 13
? ?a=fmtp:18 annexb=yes
? ?a=rtpmap:101 telephone-event/8000
? ?a=fmtp:101 0-16
? ?a=ptime:20
? ?------------------------------------------------------------------------
recv 423 bytes from udp/[85.15.0.154]:5060 at 04:01:16.208118:
? ?------------------------------------------------------------------------
? ?SIP/2.0 488 Not Acceptable Media
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=457FC664-6A6
? ?Date: Tue, 12 Jun 2012 04:01:24 GMT
? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
? ?Server: Cisco-SIPGateway/IOS-12.x
? ?CSeq: 29400774 INVITE
? ?Allow-Events: telephone-event
? ?Content-Length: 0

? ?------------------------------------------------------------------------
send 349 bytes to udp/[85.15.0.154]:5060 at 04:01:16.208201:
? ?------------------------------------------------------------------------
? ?ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
? ?Max-Forwards: 69
? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=457FC664-6A6
? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
? ?CSeq: 29400774 ACK
? ?Content-Length: 0



plz help,thanks so much



________________________________
 From: Paul Cupis <paul at cupis.co.uk>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Monday, June 11, 2012 10:51 PM
Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
 
On 11/06/12 17:54, Samira Mh wrote:
> i want to bridge call using my VOIPgateway so that making calls to
> another countries..
> but the carrier only support G729 codec and the FS send G722 (set in
> vars.xml) to myVoipGateway that is set as an gateway in
> /usr/local/freeswitch/sip-profile/external/
> and when FS send media to Gateway(using bridge application) the error
> occure:unacceptable media,then check VOIPGW and find out the only codec
> that
> can be pass through VOIPgw is G729, but FS only send G711,G722,... not G729

Can you provide a SIP or FreeSWITCH trace of a call, please?

Do you have the following enabled in your SIP profile?

?  <param name="inbound-late-negotiation" value="true"/>

Do you have mod_g729 loaded and codec G729 enabled in your
 vars.xml?

Regards,

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From chris at gonumina.com  Tue Jun 12 08:24:10 2012
From: chris at gonumina.com (Chris Ferreira)
Date: Tue, 12 Jun 2012 00:24:10 -0400
Subject: [Freeswitch-users] NAT Router Recommendation For Large no of IP
 Phones Behind the Router
In-Reply-To: <CAGvuWYaqsqO6N0y+5+pJ2zyA0UqS0iAFjQiXuaSxssH52GcdVQ@mail.gmail.com>
References: <CAGvuWYaqsqO6N0y+5+pJ2zyA0UqS0iAFjQiXuaSxssH52GcdVQ@mail.gmail.com>
Message-ID: <8488172921530077229@unknownmsgid>

What version of pfsense are you running?

___________________
Mobile Reply

On Jun 11, 2012, at 9:24 PM, Lloyd Aloysius <lloyd.aloysius at gmail.com>
wrote:

Hi All:

Currently I use pfSense for NAT implementations. But I found there are few
bugs in the firmware cause Presence not working reliably.

What are best routers out there with good support for SIP. I am looking for
a NAT router that support 20-30 phones . Any good recommendations?

Thanks
Lloyd

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From sdevoy at bizfocused.com  Tue Jun 12 08:45:56 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 12 Jun 2012 00:45:56 -0400
Subject: [Freeswitch-users] Moving up to better PC - Multi-tenant died
Message-ID: <05a501cd4856$41711610$c4534230$@bizfocused.com>

HI,

 

I have a nice working multi-tenant configuration working on my development
box - P4 3.0Ghz 4GB RAM Raid 1 Centos 5.7 Freeswitch (make current).

 

I built a nice clean Centos 5.7 install on a Dell Power Edge 2850 Dual Xeon
3.0 Ghs 4GB RAM and current Freeswitch.

I saved the entire "/usr/local/freeswitch/conf" tree from new server.

                I copied the entire "/usr/local/freeswitch/conf" tree from
devel box to new server.

                I shutdown development box.

                I set new server to same IP as old Devel boix.

                Rebooted.

 

I have a couple of issues:

                1. Freeswitch does not startup at boot.  I tried to follow
the wiki.  I have a freeswitch file in /;etc/init.d but no startup!

 

                2. Only my primary domain (tenant) users can login.  All of
my secondary domain phones have proxy=nnn at fs_mbri.bizfocused.com but the
errors on FS_CLI say it can't find nnn at fs_bfis.bizfocused.com (MY PRIMARY
DOMAIN).  It is though freeswitch thinks these registrations are coming to
the primary domain even though they are going to a secondry with the same IP
address.

 

So, how do I get FS to start on boot up in CENTOS 5.7?  I swear I tried
what's in the wiki.  Or maybe I should say how do I tell wait is failing?

 

What am I missing in the multi-tenant domain definition (that is not in the
/conf folder)?

 

Thanks in advance.

 

Sean

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From gmaruzz at gmail.com  Tue Jun 12 12:03:46 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 12 Jun 2012 10:03:46 +0200
Subject: [Freeswitch-users] mod_gsmopen on FreeBSD
In-Reply-To: <20120611233123.ebed9bee.royj@yandex.ru>
References: <20120611233123.ebed9bee.royj@yandex.ru>
Message-ID: <CALXCt0qNKnn9UB2ONf+X3effOW3OyQhU18MU2XXzrrgA6K=z2Q@mail.gmail.com>

On Mon, Jun 11, 2012 at 9:31 PM, royj <royj at yandex.ru> wrote:

>
> Hi all
> Does anyone use mod_gsmopen on FreeBSD? I have tried to make according to
> the instructions - http://wiki.freeswitch.org/wiki/GSMopen#Building, and
> got an error when compiling libctb-0.16 which is only for Linux and Win32 (
> https://iftools.com/download/ctb/0.14/refman.pdf). Is there any solution
> other than changing OS )
>

I believe would be easy to adapt libctb to compile on FreeBSD. Why don't
you do that, and contribute it back with a patch on jira.freeswitch.org ? I
would happily integrate your code into mainline.
If you're not able to, maybe you can ask some of the FreeBSD guys in the
mailing list or in IRC, will probably take 15mins or so...

-giovanni




> --
> royj
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From paul at cupis.co.uk  Tue Jun 12 12:12:52 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Tue, 12 Jun 2012 09:12:52 +0100
Subject: [Freeswitch-users] Moving up to better PC - Multi-tenant died
In-Reply-To: <05a501cd4856$41711610$c4534230$@bizfocused.com>
References: <05a501cd4856$41711610$c4534230$@bizfocused.com>
Message-ID: <20120612081252.GA13156@eagle.cupis.co.uk>

On Tue, Jun 12, 2012 at 12:45:56AM -0400, Sean Devoy wrote:
>    I built a nice clean Centos 5.7 install on a Dell Power Edge 2850 Dual
>    Xeon 3.0 Ghs 4GB RAM and current Freeswitch.

>                    1. Freeswitch does not startup at boot.  I tried to
>    follow the wiki.  I have a freeswitch file in /;etc/init.d but no
>    startup!

Can you run:

  chkconfig --list freeswitch

and check the output?

You may need to do:

  chkconfig --add freeswitch

(where 'freeswitch' is the name of your init-script in /etc/init.d).

Regards,



From steveayre at gmail.com  Tue Jun 12 18:00:13 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 12 Jun 2012 15:00:13 +0100
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <1339473995.42452.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
	<1339414395.62082.YahooMailNeo@web120105.mail.ne1.yahoo.com>
	<30DFAA60-ADB7-4B3A-A5FE-9085491FD83E@opencsta.org>
	<1339433685.46946.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<4FD6371D.1010307@cupis.co.uk>
	<1339473995.42452.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <CAFiqYunZkNfb7YK8SakmQ4D=Ft5dj5QOtOJ6SHm_-3HrckQYhg@mail.gmail.com>

The problem is on your Cisco device, not FreeSWITCH.

You're sending an invite with G729:
   m=audio 25814 RTP/AVP 18 101 13
18=G729 codec  101=RFC2833 DTMF  13=Comfort Noise
This gets the 488 Not Acceptable response from the Cisco device.

So you're offering G729 to the Cisco but having it refused. This either
means the Cisco device doesn't have G729 enabled or the Cisco SIP
implementation doesn't like the lack of the optional a=rtpmap line. If it
is the latter use <action application="set" data"verbose_sdp=true"/> before
the bridge to workaround that issue.




On 12 June 2012 05:06, Samira Mh <saami_mh at ymail.com> wrote:

> thansk for your reply,
> it is kind of you to help me..
> please let me paste myconfigurations files here;
> 1-the configuration  file
> /usr/local/freeswitch/conf/dialplan/default/001_luacallduration.xml  is
> like this:
>
> <include>
>   <extension name="mainmenuvodsl">
>
>         <condition field="destination_number"
> expression="^(00|\+)?(\d{5}.*)$" break="never">
>                    <action application="odbc_query" data="select cash as
> cashvalue from accounts where contractid like '${nibble_account}';"/>
>                 <action application="log" data="INFO The value of
> cashvalue is ${cashvalue}" />
>                  <action application="lua" data="checkcash.lua
> ${cashvalue}" />
>                 <action application="log" data="INFO The value of
> nibble_account is  ${nibble_account}"/>
>                 <action application="log" data="INFO The value of
> nibble_rate [before] is  ${nibble_rate}"/>
>                 <!-- RateList Context -->
>                 <action application="lua" data="checkzeroplus.lua
> ${destination_number:0:2} ${destination_number:0:1}" />
>                 <action application="execute_extension"
> data="${destination_number} XML ratelist"/>
>                 <action application="log" data="INFO The value of
> nibble_rate [after] is ${nibble_rate}"/>
>                  <!-- Check Nibble_rate -->
>                 <action application="lua" data="checknibblerate.lua
> ${nibble_rate}" />
>                 <action application="set"
> data="divvalue=${expr(floor((${cashvalue}/${nibble_rate}))}" />
>                 <action application="set"
> data="modvalue=${expr(mod(${cashvalue},${nibble_rate}))}" />
>                 <action application="lua" data="checktime.lua ${divvalue}
> ${modvalue}" />
>                     <!--  Check ZeroZero , Plus  -->
>                 <action application="lua" data="checkzeroplus.lua
> ${destination_number:0:2} ${destination_number:0:1}" />
>                 <!-- Making Calls   -->
>                 <action application="odbc_query" data="select callerid  as
> effective_caller_id_number from accounts where contractid like
> '${nibble_account}';"/>
>                 <action application="log" data="INFO  callerid for
> Outbound calls ${effective_caller_id_number}"/>
>                 <!-- <action application="set"
> data="ignore_early_media=true"/>
>                 <action application="answer"/>  -->
>                 <action application="enable_heartbeat"/>
>
> <!-- <param name="disable-transcoding" value="true"/> -->
>         <!--    <action application="export"
> data="nolocal:absolute_codec_string=G729,PCMU"/> -->
> <!--  <action application="set" data="bridge_early_media=true"/>  -->
>         <!-- <action application="set" data="proxy_media=true"/> -->
>                 <action application="bridge"
> data="sofia/gateway/cisco/140112${destination_number}"/>
>                 <!-- <action application="bridge"
> data="sofia/gateway/mainasterisk/${destination_number}"/>  -->
> <!--  <action application="bridge"
> data="sofia/gateway/test/${destination_number}"/>  -->
>         </condition>
>
>  </extension>
> </include>
>
> 2-yes, i have enabled  "inbound-late-negotiation" in the
> (/usr/local/freeswitch/conf/sip_profiles/internal.xml) as follow:
>  <param name="inbound-late-negotiation" value="true"/>
>
>
> 3-the issue of sofia status:
>  external::cisco       gateway             sip:register:false at 85.15.0.154     NOREG
>
>
> 4-also , the configuration file for codecs are as follow
> :/usr/local/freeswitch/conf/vars.xml
>
> <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G729,PCMU,PCMA,G7221 at 32000h,G7221 at 16000h
> ,G722,GSM"/>
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>
>
> 5- the mod_g729 was loaded
>
> 6-i have enabled the siptrace:
>  sofia profile external siptrace on:
> the siptrace outpout as follow:
>
> send 1042 bytes to udp/[85.15.0.154]:5060 at 03:53:07.448136:
>    ------------------------------------------------------------------------
>    INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>    Max-Forwards: 69
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>    To: <sip:140112971507247227 at 85.15.0.154>
>    Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>    CSeq: 29400529 INVITE
>    Contact: <sip:gw+cisco at 192.168.10.70:5080;transport=udp;gw=cisco>
>    User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 234
>    X-FS-Support: update_display,send_info
>    Remote-Party-ID: "1000" <sip:1000 at 85.15.0.154
> >;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=FreeSWITCH 1339446571 1339446572 IN IP4 192.168.10.70
>    s=FreeSWITCH
>    c=IN IP4 192.168.10.70
>    t=0 0
>    m=audio 26616 RTP/AVP 9 0 8 18 3 101 13
>    a=fmtp:18 annexb=yes
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>    ------------------------------------------------------------------------
> recv 410 bytes from udp/[85.15.0.154]:5060 at 03:53:07.463921:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>    To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
>    Date: Tue, 12 Jun 2012 03:53:15 GMT
>    Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>    Server: Cisco-SIPGateway/IOS-12.x
>    CSeq: 29400529 INVITE
>    Allow-Events: telephone-event
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 927 bytes from udp/[85.15.0.154]:5060 at 03:53:11.004804:
>    ------------------------------------------------------------------------
>    SIP/2.0 183 Session Progress
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>    To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
>    Date: Tue, 12 Jun 2012 03:53:15 GMT
>    Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>    Server: Cisco-SIPGateway/IOS-12.x
>    CSeq: 29400529 INVITE
>    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
>    Allow-Events: telephone-event
>    Contact: <sip:140112971507247227 at 85.15.0.154:5060>
>    Content-Disposition: session;handling=required
>    Content-Type: application/sdp
>    Content-Length: 268
>
>    v=0
>    o=CiscoSystemsSIP-GW-UserAgent 7252 5649 IN IP4 85.15.0.154
>    s=SIP Call
>    c=IN IP4 85.15.0.154
>    t=0 0
>    m=audio 18218 RTP/AVP 0 13 101
>    c=IN IP4 85.15.0.154
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:13 CN/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-15
>    a=ptime:20
>    ------------------------------------------------------------------------
> recv 425 bytes from udp/[85.15.0.154]:5060 at 03:53:11.005144:
>    ------------------------------------------------------------------------
>    SIP/2.0 500 Internal Server Error
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>    To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
>    Date: Tue, 12 Jun 2012 03:53:15 GMT
>    Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>    Server: Cisco-SIPGateway/IOS-12.x
>    CSeq: 29400529 INVITE
>    Allow-Events: telephone-event
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 350 bytes to udp/[85.15.0.154]:5060 at 03:53:11.005333:
>    ------------------------------------------------------------------------
>    ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>    Max-Forwards: 69
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>    To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
>    Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>    CSeq: 29400529 ACK
>    Content-Length: 0
>
>
>
> ------------------------------------------------------------------------------------------------------------------
> when change the configuration file the below:
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>
>
> the siptrace is like this:
>
> send 1034 bytes to udp/[85.15.0.154]:5060 at 04:01:16.202342:
>    ------------------------------------------------------------------------
>    INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
>    Max-Forwards: 69
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
>    To: <sip:140112971507247227 at 85.15.0.154>
>    Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
>    CSeq: 29400774 INVITE
>    Contact: <sip:gw+cisco at 192.168.10.70:5080;transport=udp;gw=cisco>
>    User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 226
>    X-FS-Support: update_display,send_info
>    Remote-Party-ID: "1000" <sip:1000 at 85.15.0.154
> >;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=FreeSWITCH 1339447862 1339447863 IN IP4 192.168.10.70
>    s=FreeSWITCH
>    c=IN IP4 192.168.10.70
>    t=0 0
>    m=audio 25814 RTP/AVP 18 101 13
>    a=fmtp:18 annexb=yes
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>    ------------------------------------------------------------------------
> recv 423 bytes from udp/[85.15.0.154]:5060 at 04:01:16.208118:
>    ------------------------------------------------------------------------
>    SIP/2.0 488 Not Acceptable Media
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
>    To: <sip:140112971507247227 at 85.15.0.154>;tag=457FC664-6A6
>    Date: Tue, 12 Jun 2012 04:01:24 GMT
>    Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
>    Server: Cisco-SIPGateway/IOS-12.x
>    CSeq: 29400774 INVITE
>    Allow-Events: telephone-event
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 349 bytes to udp/[85.15.0.154]:5060 at 04:01:16.208201:
>    ------------------------------------------------------------------------
>    ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
>    Max-Forwards: 69
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
>    To: <sip:140112971507247227 at 85.15.0.154>;tag=457FC664-6A6
>    Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
>    CSeq: 29400774 ACK
>    Content-Length: 0
>
>
>
> plz help,thanks so much
>
>
>   ------------------------------
> *From:* Paul Cupis <paul at cupis.co.uk>
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, June 11, 2012 10:51 PM
>
> *Subject:* Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
>
> On 11/06/12 17:54, Samira Mh wrote:
> > i want to bridge call using my VOIPgateway so that making calls to
> > another countries..
> > but the carrier only support G729 codec and the FS send G722 (set in
> > vars.xml) to myVoipGateway that is set as an gateway in
> > /usr/local/freeswitch/sip-profile/external/
> > and when FS send media to Gateway(using bridge application) the error
> > occure:unacceptable media,then check VOIPGW and find out the only codec
> > that
> > can be pass through VOIPgw is G729, but FS only send G711,G722,... not
> G729
>
> Can you provide a SIP or FreeSWITCH trace of a call, please?
>
> Do you have the following enabled in your SIP profile?
>
>   <param name="inbound-late-negotiation" value="true"/>
>
> Do you have mod_g729 loaded and codec G729 enabled in your vars.xml?
>
> Regards,
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Jun 12 18:03:11 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 12 Jun 2012 15:03:11 +0100
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <1339474390.46866.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
	<1339414395.62082.YahooMailNeo@web120105.mail.ne1.yahoo.com>
	<30DFAA60-ADB7-4B3A-A5FE-9085491FD83E@opencsta.org>
	<1339433685.46946.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<4FD6371D.1010307@cupis.co.uk>
	<1339473995.42452.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<1339474390.46866.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <CAFiqYuk5ViEtJX_hSfKcP5ifCnNXNsib5-6OOmcVV7LL7ZdhZw@mail.gmail.com>

Also in your first example you say you set the codec preferences in
vars.conf.xml to G729, but only PCMU was offered. CODECS OUT only lists
PCMU on your external profile. That means you've either not reloaded
mod_sofia since editing the file, or you have PCMU listed in the
sip_profiles/external.xml

Steve


On 12 June 2012 05:13, Samira Mh <saami_mh at ymail.com> wrote:

> sorry i forgot to paste codec on internal and external:
>
> freeswitch at internal> sofia status profile internal
>
> =================================================================================================
> Name                    internal
> Domain Name             N/A
> Auto-NAT                false
> DBName                  sofia_reg_internal
> Pres Hosts              192.168.10.70,192.168.10.70
> Dialplan                XML
> Context                 public
> Challenge Realm         auto_from
> RTP-IP                  192.168.10.70
> SIP-IP                  192.168.10.70
> URL                     sip:mod_sofia at 192.168.10.70:5060
> BIND-URL                sip:mod_sofia at 192.168.10.70:5060
> HOLD-MUSIC              local_stream://moh
> OUTBOUND-PROXY          N/A
> CODECS IN               G729
> CODECS OUT              G729
> TEL-EVENT               101
> DTMF-MODE               rfc2833
> CNG                     13
> SESSION-TO              0
> MAX-DIALOG              0
> NOMEDIA                 false
> LATE-NEG                true
> PROXY-MEDIA             false
> ZRTP-PASSTHRU           false
> AGGRESSIVENAT           false
> STUN-ENABLED            true
> STUN-AUTO-DISABLE       false
> CALLS-IN                0
> FAILED-CALLS-IN         0
> CALLS-OUT               0
> FAILED-CALLS-OUT        0
> REGISTRATIONS           3
>
>
> ---------------------------------------------------------------------------------
> freeswitch at internal> sofia status profile external
>
> =================================================================================================
> Name                    external
> Domain Name             N/A
> Auto-NAT                false
> DBName                  sofia_reg_external
> Pres Hosts
> Dialplan                XML
> Context                 public
> Challenge Realm         auto_to
> RTP-IP                  192.168.10.70
> Ext-RTP-IP              192.168.10.70
> SIP-IP                  192.168.10.70
> Ext-SIP-IP              192.168.10.70
> URL                     sip:mod_sofia at 192.168.10.70:5080
> BIND-URL                sip:mod_sofia at 192.168.10.70:5080
> ;maddr=192.168.10.70
> HOLD-MUSIC              local_stream://moh
> OUTBOUND-PROXY          N/A
> CODECS IN               G729
> CODECS OUT              PCMU
> TEL-EVENT               101
> DTMF-MODE               rfc2833
> CNG                     13
> SESSION-TO              0
> MAX-DIALOG              0
> NOMEDIA                 false
> LATE-NEG                false
> PROXY-MEDIA             false
> ZRTP-PASSTHRU           false
> AGGRESSIVENAT           false
> STUN-ENABLED            true
> STUN-AUTO-DISABLE       false
> CALLS-IN                0
> FAILED-CALLS-IN         0
> CALLS-OUT               0
> FAILED-CALLS-OUT        0
> REGISTRATIONS           0
>
>
>   ------------------------------
> *From:* Samira Mh <saami_mh at ymail.com>
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, June 12, 2012 8:36 AM
>
> *Subject:* Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
>
> thansk for your reply,
> it is kind of you to help me..
> please let me paste myconfigurations files here;
> 1-the configuration  file
> /usr/local/freeswitch/conf/dialplan/default/001_luacallduration.xml  is
> like this:
>
> <include>
>   <extension name="mainmenuvodsl">
>
>         <condition field="destination_number"
> expression="^(00|\+)?(\d{5}.*)$" break="never">
>                    <action application="odbc_query" data="select cash as
> cashvalue from accounts where contractid like '${nibble_account}';"/>
>                 <action application="log" data="INFO The value of
> cashvalue is ${cashvalue}" />
>                  <action application="lua" data="checkcash.lua
> ${cashvalue}" />
>                 <action application="log" data="INFO The value of
> nibble_account is  ${nibble_account}"/>
>                 <action application="log" data="INFO The value of
> nibble_rate [before] is  ${nibble_rate}"/>
>                 <!-- RateList Context -->
>                 <action application="lua" data="checkzeroplus.lua
> ${destination_number:0:2} ${destination_number:0:1}" />
>                 <action application="execute_extension"
> data="${destination_number} XML ratelist"/>
>                 <action application="log" data="INFO The value of
> nibble_rate [after] is ${nibble_rate}"/>
>                  <!-- Check Nibble_rate -->
>                 <action application="lua" data="checknibblerate.lua
> ${nibble_rate}" />
>                 <action application="set"
> data="divvalue=${expr(floor((${cashvalue}/${nibble_rate}))}" />
>                 <action application="set"
> data="modvalue=${expr(mod(${cashvalue},${nibble_rate}))}" />
>                 <action application="lua" data="checktime.lua ${divvalue}
> ${modvalue}" />
>                     <!--  Check ZeroZero , Plus  -->
>                 <action application="lua" data="checkzeroplus.lua
> ${destination_number:0:2} ${destination_number:0:1}" />
>                 <!-- Making Calls   -->
>                 <action application="odbc_query" data="select callerid  as
> effective_caller_id_number from accounts where contractid like
> '${nibble_account}';"/>
>                 <action application="log" data="INFO  callerid for
> Outbound calls ${effective_caller_id_number}"/>
>                 <!-- <action application="set"
> data="ignore_early_media=true"/>
>                 <action application="answer"/>  -->
>                 <action application="enable_heartbeat"/>
>
> <!-- <param name="disable-transcoding" value="true"/> -->
>         <!--    <action application="export"
> data="nolocal:absolute_codec_string=G729,PCMU"/> -->
> <!--  <action application="set" data="bridge_early_media=true"/>  -->
>         <!-- <action application="set" data="proxy_media=true"/> -->
>                 <action application="bridge"
> data="sofia/gateway/cisco/140112${destination_number}"/>
>                 <!-- <action application="bridge"
> data="sofia/gateway/mainasterisk/${destination_number}"/>  -->
> <!--  <action application="bridge"
> data="sofia/gateway/test/${destination_number}"/>  -->
>         </condition>
>
>  </extension>
> </include>
>
> 2-yes, i have enabled  "inbound-late-negotiation" in the
> (/usr/local/freeswitch/conf/sip_profiles/internal.xml) as follow:
>  <param name="inbound-late-negotiation" value="true"/>
>
>
> 3-the issue of sofia status:
>  external::cisco       gateway             sip:register:false at 85.15.0.154     NOREG
>
>
> 4-also , the configuration file for codecs are as follow
> :/usr/local/freeswitch/conf/vars.xml
>
> <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G729,PCMU,PCMA,G7221 at 32000h,G7221 at 16000h
> ,G722,GSM"/>
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>
>
> 5- the mod_g729 was loaded
>
> 6-i have enabled the siptrace:
>  sofia profile external siptrace on:
> the siptrace outpout as follow:
>
> send 1042 bytes to udp/[85.15.0.154]:5060 at 03:53:07.448136:
>    ------------------------------------------------------------------------
>    INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>    Max-Forwards: 69
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>    To: <sip:140112971507247227 at 85.15.0.154>
>    Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>    CSeq: 29400529 INVITE
>    Contact: <sip:gw+cisco at 192.168.10.70:5080;transport=udp;gw=cisco>
>    User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 234
>    X-FS-Support: update_display,send_info
>    Remote-Party-ID: "1000" <sip:1000 at 85.15.0.154
> >;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=FreeSWITCH 1339446571 1339446572 IN IP4 192.168.10.70
>    s=FreeSWITCH
>    c=IN IP4 192.168.10.70
>    t=0 0
>    m=audio 26616 RTP/AVP 9 0 8 18 3 101 13
>    a=fmtp:18 annexb=yes
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>    ------------------------------------------------------------------------
> recv 410 bytes from udp/[85.15.0.154]:5060 at 03:53:07.463921:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>    To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
>    Date: Tue, 12 Jun 2012 03:53:15 GMT
>    Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>    Server: Cisco-SIPGateway/IOS-12.x
>    CSeq: 29400529 INVITE
>    Allow-Events: telephone-event
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 927 bytes from udp/[85.15.0.154]:5060 at 03:53:11.004804:
>    ------------------------------------------------------------------------
>    SIP/2.0 183 Session Progress
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>    To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
>    Date: Tue, 12 Jun 2012 03:53:15 GMT
>    Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>    Server: Cisco-SIPGateway/IOS-12.x
>    CSeq: 29400529 INVITE
>    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
>    Allow-Events: telephone-event
>    Contact: <sip:140112971507247227 at 85.15.0.154:5060>
>    Content-Disposition: session;handling=required
>    Content-Type: application/sdp
>    Content-Length: 268
>
>    v=0
>    o=CiscoSystemsSIP-GW-UserAgent 7252 5649 IN IP4 85.15.0.154
>    s=SIP Call
>    c=IN IP4 85.15.0.154
>    t=0 0
>    m=audio 18218 RTP/AVP 0 13 101
>    c=IN IP4 85.15.0.154
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:13 CN/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-15
>    a=ptime:20
>    ------------------------------------------------------------------------
> recv 425 bytes from udp/[85.15.0.154]:5060 at 03:53:11.005144:
>    ------------------------------------------------------------------------
>    SIP/2.0 500 Internal Server Error
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>    To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
>    Date: Tue, 12 Jun 2012 03:53:15 GMT
>    Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>    Server: Cisco-SIPGateway/IOS-12.x
>    CSeq: 29400529 INVITE
>    Allow-Events: telephone-event
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 350 bytes to udp/[85.15.0.154]:5060 at 03:53:11.005333:
>    ------------------------------------------------------------------------
>    ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>    Max-Forwards: 69
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>    To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
>    Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>    CSeq: 29400529 ACK
>    Content-Length: 0
>
>
>
> ------------------------------------------------------------------------------------------------------------------
> when change the configuration file the below:
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>
>
> the siptrace is like this:
>
> send 1034 bytes to udp/[85.15.0.154]:5060 at 04:01:16.202342:
>    ------------------------------------------------------------------------
>    INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
>    Max-Forwards: 69
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
>    To: <sip:140112971507247227 at 85.15.0.154>
>    Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
>    CSeq: 29400774 INVITE
>    Contact: <sip:gw+cisco at 192.168.10.70:5080;transport=udp;gw=cisco>
>    User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 226
>    X-FS-Support: update_display,send_info
>    Remote-Party-ID: "1000" <sip:1000 at 85.15.0.154
> >;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=FreeSWITCH 1339447862 1339447863 IN IP4 192.168.10.70
>    s=FreeSWITCH
>    c=IN IP4 192.168.10.70
>    t=0 0
>    m=audio 25814 RTP/AVP 18 101 13
>    a=fmtp:18 annexb=yes
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>    ------------------------------------------------------------------------
> recv 423 bytes from udp/[85.15.0.154]:5060 at 04:01:16.208118:
>    ------------------------------------------------------------------------
>    SIP/2.0 488 Not Acceptable Media
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
>    To: <sip:140112971507247227 at 85.15.0.154>;tag=457FC664-6A6
>    Date: Tue, 12 Jun 2012 04:01:24 GMT
>    Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
>    Server: Cisco-SIPGateway/IOS-12.x
>    CSeq: 29400774 INVITE
>    Allow-Events: telephone-event
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 349 bytes to udp/[85.15.0.154]:5060 at 04:01:16.208201:
>    ------------------------------------------------------------------------
>    ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
>    Max-Forwards: 69
>    From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
>    To: <sip:140112971507247227 at 85.15.0.154>;tag=457FC664-6A6
>    Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
>    CSeq: 29400774 ACK
>    Content-Length: 0
>
>
>
> plz help,thanks so much
>
>
>   ------------------------------
> *From:* Paul Cupis <paul at cupis.co.uk>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, June 11, 2012 10:51 PM
> *Subject:* Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
>
> On 11/06/12 17:54, Samira Mh wrote:
> > i want to bridge call using my VOIPgateway so that making calls to
> > another countries..
> > but the carrier only support G729 codec and the FS send G722 (set in
> > vars.xml) to myVoipGateway that is set as an gateway in
> > /usr/local/freeswitch/sip-profile/external/
> > and when FS send media to Gateway(using bridge application) the error
> > occure:unacceptable media,then check VOIPGW and find out the only codec
> > that
> > can be pass through VOIPgw is G729, but FS only send G711,G722,... not
> G729
>
> Can you provide a SIP or FreeSWITCH trace of a call, please?
>
> Do you have the following enabled in your SIP profile?
>
>   <param name="inbound-late-negotiation" value="true"/>
>
> Do you have mod_g729 loaded and codec G729 enabled in your vars.xml?
>
> Regards,
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
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From george.d at telesoftlabs.com  Tue Jun 12 18:07:55 2012
From: george.d at telesoftlabs.com (George D'mithrov)
Date: Tue, 12 Jun 2012 19:37:55 +0530
Subject: [Freeswitch-users] Openzap bridge and DTMF issue
Message-ID: <ABCC4BED9408E84E91D9348B2DA819C6061BE6@tslsrv.TSL.LOCAL>

Hi,

I wants to bridge calls with particular DID, which landing on one PRI
span (Span_1) to another PRI span (Span_2). 




I'm able to bridge the calls between PRI span and able to talk to each
other.




My problem is that sometimes DTMF which received on Span_1 is not
sending to Span2 and vice-verse. DTMF tones are hearing partially on
receving end.




Sometimes after a some delay (30-60 seconds) am getting the DTMF, but
that's too some dtmf are missing.




If I hangup the line while the dtmf is in queue then the dtmf are
generating on the next call on the same line.




But when I bridge the call on PRI with sip channel then am not getting
any problem.

I think the following log in freeswitch is usefull

 
2012-06-12 18:58:38.379506 [DEBUG] mod_openzap.c:725 queue DTMF [1]
2012-06-12 18:58:38.019513 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]

after this  it takes a while to generate dtmf
 
2012-06-12 18:58:38.019513 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [1]
 




Any suggestions on places to look or things to try would be most
appreciated.




See the log below.




2012-06-12 18:58:32.017050 [DEBUG] ozmod_isdn.c:1115 READ 44
------------------------------------------------------------------------
--------
[08 02 1d 81 05 a1 04 03 80 90 a3 18 03 a1 83 82 1e 02 80 83 6c 0c 21 81
38 30 34 30 33 32 37 33 32 34 70 04 c1 33 32 35 7d 02 91 81]


2012-06-12 18:58:32.017050 [DEBUG] ozmod_isdn.c:584 Yay I got an event!
Type:[05] Size:[187] CRV: 7553 (0x1d81, CTX: Originator)
2012-06-12 18:58:32.017050 [DEBUG] ozmod_isdn.c:616 zchan 0 (-1:-1)
source isdn_data->channels_remote_crv[0x1d81]
2012-06-12 18:58:32.017050 [DEBUG] ozmod_isdn.c:954 Changing state on
1:2 from DOWN to RING
2012-06-12 18:58:32.117048 [DEBUG] ozmod_isdn.c:1163 1:2 STATE [RING]
2012-06-12 18:58:32.117048 [DEBUG] mod_openzap.c:1937 got clear channel
sig [START]
2012-06-12 18:58:32.117048 [DEBUG] mod_openzap.c:407 Set codec PCMA 20ms
2012-06-12 18:58:32.117048 [DEBUG] mod_openzap.c:1440 Connect inbound
channel OpenZAP/1:2/325
2012-06-12 18:58:32.117048 [NOTICE] switch_channel.c:816 New Channel
OpenZAP/1:2/325 [e74d979f-6689-4345-836c-c7dc087e210a]
2012-06-12 18:58:32.117048 [DEBUG] mod_openzap.c:1451 (OpenZAP/1:2/325)
State Change CS_NEW -> CS_INIT
2012-06-12 18:58:32.117048 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/1:2/325 [BREAK]
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:325
(OpenZAP/1:2/325) Running State Change CS_INIT
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:361
(OpenZAP/1:2/325) State INIT
2012-06-12 18:58:32.117048 [DEBUG] mod_openzap.c:435 (OpenZAP/1:2/325)
State Change CS_INIT -> CS_ROUTING
2012-06-12 18:58:32.117048 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/1:2/325 [BREAK]
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:361
(OpenZAP/1:2/325) State INIT going to sleep
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:325
(OpenZAP/1:2/325) Running State Change CS_ROUTING
2012-06-12 18:58:32.117048 [DEBUG] switch_channel.c:1687
(OpenZAP/1:2/325) Callstate Change DOWN -> RINGING
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:364
(OpenZAP/1:2/325) State ROUTING
2012-06-12 18:58:32.117048 [DEBUG] mod_openzap.c:458 OpenZAP/1:2/325
CHANNEL ROUTING
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:77
OpenZAP/1:2/325 Standard ROUTING
2012-06-12 18:58:32.117048 [INFO] mod_dialplan_xml.c:331 Processing
8040327324 <8040327324>->325 in context default
Dialplan: OpenZAP/1:2/325 parsing [default->Span2] continue=false
Dialplan: OpenZAP/1:2/325 Regex (PASS) [Span2] destination_number(325)
=~ /^325$/ break=on-false
Dialplan: OpenZAP/1:2/325 Action set(call_timeout=45) 
Dialplan: OpenZAP/1:2/325 Action set(hangup_after_bridge=true) 
Dialplan: OpenZAP/1:2/325 Action set(continue_on_fail=true) 
Dialplan: OpenZAP/1:2/325 Action set(inherit_codec=true) 
Dialplan: OpenZAP/1:2/325 Action bridge(Openzap/2/a/8040327325) 
Dialplan: OpenZAP/1:2/325 Action hangup() 
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:119
(OpenZAP/1:2/325) State Change CS_ROUTING -> CS_EXECUTE
2012-06-12 18:58:32.117048 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/1:2/325 [BREAK]
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:364
(OpenZAP/1:2/325) State ROUTING going to sleep
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:325
(OpenZAP/1:2/325) Running State Change CS_EXECUTE
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:371
(OpenZAP/1:2/325) State EXECUTE
2012-06-12 18:58:32.117048 [DEBUG] mod_openzap.c:475 OpenZAP/1:2/325
CHANNEL EXECUTE
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:157
OpenZAP/1:2/325 Standard EXECUTE
EXECUTE OpenZAP/1:2/325 set(call_timeout=45)
2012-06-12 18:58:32.117048 [DEBUG] mod_dptools.c:1060 OpenZAP/1:2/325
SET [call_timeout]=[45]
EXECUTE OpenZAP/1:2/325 set(hangup_after_bridge=true)
2012-06-12 18:58:32.117048 [DEBUG] mod_dptools.c:1060 OpenZAP/1:2/325
SET [hangup_after_bridge]=[true]
EXECUTE OpenZAP/1:2/325 set(continue_on_fail=true)
2012-06-12 18:58:32.117048 [DEBUG] mod_dptools.c:1060 OpenZAP/1:2/325
SET [continue_on_fail]=[true]
EXECUTE OpenZAP/1:2/325 set(inherit_codec=true)
2012-06-12 18:58:32.117048 [DEBUG] mod_dptools.c:1060 OpenZAP/1:2/325
SET [inherit_codec]=[true]
EXECUTE OpenZAP/1:2/325 bridge(Openzap/2/a/8040327325)
2012-06-12 18:58:32.117048 [DEBUG] switch_ivr_originate.c:1873 Parsing
global variables
2012-06-12 18:58:32.117048 [INFO] ozmod_zt.c:638 Setting echo cancel to
64 taps for 2:1
2012-06-12 18:58:32.117048 [INFO] ozmod_zt.c:638 Setting echo cancel to
64 taps for 2:1
2012-06-12 18:58:32.117048 [DEBUG] mod_openzap.c:407 Set codec PCMA 20ms
2012-06-12 18:58:32.117048 [DEBUG] mod_openzap.c:1319 Connect outbound
channel OpenZAP/2:1/8040327325
2012-06-12 18:58:32.117048 [NOTICE] switch_channel.c:816 New Channel
OpenZAP/2:1/8040327325 [668e6166-40a0-4b3d-a575-16de89668282]
2012-06-12 18:58:32.117048 [DEBUG] mod_openzap.c:1332
(OpenZAP/2:1/8040327325) State Change CS_NEW -> CS_INIT
2012-06-12 18:58:32.117048 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:58:32.117048 [DEBUG] ozmod_isdn.c:299 Changing state on
2:1 from DOWN to DIALING
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:325
(OpenZAP/2:1/8040327325) Running State Change CS_INIT
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:361
(OpenZAP/2:1/8040327325) State INIT
2012-06-12 18:58:32.117048 [DEBUG] mod_openzap.c:435
(OpenZAP/2:1/8040327325) State Change CS_INIT -> CS_ROUTING
2012-06-12 18:58:32.117048 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:361
(OpenZAP/2:1/8040327325) State INIT going to sleep
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:325
(OpenZAP/2:1/8040327325) Running State Change CS_ROUTING
2012-06-12 18:58:32.117048 [DEBUG] switch_channel.c:1687
(OpenZAP/2:1/8040327325) Callstate Change DOWN -> RINGING
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:364
(OpenZAP/2:1/8040327325) State ROUTING
2012-06-12 18:58:32.117048 [DEBUG] mod_openzap.c:458
OpenZAP/2:1/8040327325 CHANNEL ROUTING
2012-06-12 18:58:32.117048 [DEBUG] switch_ivr_originate.c:66
(OpenZAP/2:1/8040327325) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2012-06-12 18:58:32.117048 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:364
(OpenZAP/2:1/8040327325) State ROUTING going to sleep
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:325
(OpenZAP/2:1/8040327325) Running State Change CS_CONSUME_MEDIA
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:383
(OpenZAP/2:1/8040327325) State CONSUME_MEDIA
2012-06-12 18:58:32.117048 [DEBUG] switch_core_state_machine.c:383
(OpenZAP/2:1/8040327325) State CONSUME_MEDIA going to sleep
2012-06-12 18:58:32.196997 [DEBUG] ozmod_isdn.c:1163 2:1 STATE [DIALING]
2012-06-12 18:58:32.196997 [DEBUG] ozmod_isdn.c:1938 WRITE 62
------------------------------------------------------------------------
--------
[08 02 00 05 05 04 03 80 90 a3 18 03 a1 83 81 1e 02 80 83 28 0a 38 30 34
30 33 32 37 33 32 34 6c 0c 01 80 38 30 34 30 33 32 37 33 32 34 70 0b 81
38 30 34 30 33 32 37 33 32 35 7d 02 91 81]

2012-06-12 18:58:32.236939 [DEBUG] ozmod_isdn.c:1115 READ 10
------------------------------------------------------------------------
--------
[08 02 80 05 02 18 03 a9 83 81]


2012-06-12 18:58:32.236939 [DEBUG] ozmod_isdn.c:584 Yay I got an event!
Type:[02] Size:[115] CRV: 5 (0x5, CTX: Terminator)
2012-06-12 18:58:32.236939 [DEBUG] ozmod_isdn.c:616 zchan b750c4f0 (2:1)
source isdn_data->channels_local_crv[0x5]
2012-06-12 18:58:32.236939 [CRIT] ozmod_isdn.c:963 Received CALL
PROCEEDING message for channel 1
2012-06-12 18:58:32.236939 [DEBUG] ozmod_isdn.c:964 Changing state on
2:1 from DIALING to PROGRESS
2012-06-12 18:58:32.236939 [DEBUG] ozmod_isdn.c:1163 2:1 STATE
[PROGRESS]
2012-06-12 18:58:32.236939 [DEBUG] mod_openzap.c:1937 got clear channel
sig [PROGRESS]
2012-06-12 18:58:32.236939 [NOTICE] mod_openzap.c:1996 Ring-Ready
OpenZAP/2:1/8040327325!
2012-06-12 18:58:32.236939 [DEBUG] ozmod_isdn.c:1115 READ 5
------------------------------------------------------------------------
--------
[08 02 80 05 01]


2012-06-12 18:58:32.236939 [DEBUG] ozmod_isdn.c:584 Yay I got an event!
Type:[01] Size:[103] CRV: 5 (0x5, CTX: Terminator)
2012-06-12 18:58:32.236939 [DEBUG] ozmod_isdn.c:616 zchan b750c4f0 (2:1)
source isdn_data->channels_local_crv[0x5]
2012-06-12 18:58:32.236939 [DEBUG] ozmod_isdn.c:728 Changing state on
2:1 from PROGRESS to PROGRESS_MEDIA
2012-06-12 18:58:32.337864 [DEBUG] ozmod_isdn.c:1163 1:2 STATE
[PROGRESS]
2012-06-12 18:58:32.337864 [DEBUG] ozmod_isdn.c:1938 WRITE 23
------------------------------------------------------------------------
--------
[08 02 9d 81 02 04 03 80 90 a3 18 03 a1 83 82 1e 02 80 83 7d 02 91 81]

2012-06-12 18:58:32.337864 [DEBUG] switch_core_session.c:707 Send signal
OpenZAP/1:2/325 [BREAK]
2012-06-12 18:58:32.337864 [NOTICE] switch_ivr_originate.c:479 Ring
Ready OpenZAP/1:2/325!
2012-06-12 18:58:32.337864 [NOTICE] switch_ivr_originate.c:479
Ring-Ready OpenZAP/1:2/325!
2012-06-12 18:58:32.337864 [DEBUG] ozmod_isdn.c:1163 2:1 STATE
[PROGRESS_MEDIA]
2012-06-12 18:58:32.337864 [DEBUG] mod_openzap.c:1937 got clear channel
sig [PROGRESS_MEDIA]
2012-06-12 18:58:32.337864 [NOTICE] mod_openzap.c:1982 Pre-Answer
OpenZAP/2:1/8040327325!
2012-06-12 18:58:32.337864 [DEBUG] switch_channel.c:2668
(OpenZAP/2:1/8040327325) Callstate Change RINGING -> EARLY
2012-06-12 18:58:32.337864 [DEBUG] switch_channel.c:2707 Send signal
OpenZAP/1:2/325 [BREAK]
2012-06-12 18:58:32.337864 [DEBUG] switch_ivr_originate.c:404 Codec
string PCMA at 8000h@20i not supported on OpenZAP/1:2/325, skipping
inheritance
2012-06-12 18:58:32.357863 [DEBUG] ozmod_isdn.c:1163 1:2 STATE
[PROGRESS_MEDIA]
2012-06-12 18:58:32.357863 [INFO] ozmod_zt.c:638 Setting echo cancel to
64 taps for 1:2
2012-06-12 18:58:32.357863 [DEBUG] ozmod_isdn.c:1938 WRITE 23
------------------------------------------------------------------------
--------
[08 02 9d 81 01 04 03 80 90 a3 18 03 a1 83 82 1e 02 80 83 7d 02 91 81]

2012-06-12 18:58:32.378867 [DEBUG] switch_core_session.c:707 Send signal
OpenZAP/1:2/325 [BREAK]
2012-06-12 18:58:32.378867 [NOTICE] switch_ivr_originate.c:3163
Pre-Answer OpenZAP/1:2/325!
2012-06-12 18:58:32.378867 [DEBUG] switch_channel.c:2668
(OpenZAP/1:2/325) Callstate Change RINGING -> EARLY
2012-06-12 18:58:32.378867 [DEBUG] switch_ivr_originate.c:3206 Originate
Resulted in Success: [OpenZAP/2:1/8040327325]
2012-06-12 18:58:32.378867 [DEBUG] switch_core_session.c:707 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:58:32.378867 [DEBUG] switch_core_session.c:707 Send signal
OpenZAP/1:2/325 [BREAK]
2012-06-12 18:58:32.378867 [DEBUG] switch_ivr_bridge.c:1239
(OpenZAP/2:1/8040327325) State Change CS_CONSUME_MEDIA ->
CS_EXCHANGE_MEDIA
2012-06-12 18:58:32.378867 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:58:32.378867 [DEBUG] switch_core_state_machine.c:325
(OpenZAP/2:1/8040327325) Running State Change CS_EXCHANGE_MEDIA
2012-06-12 18:58:32.378867 [DEBUG] switch_core_state_machine.c:374
(OpenZAP/2:1/8040327325) State EXCHANGE_MEDIA
2012-06-12 18:58:32.378867 [DEBUG] mod_openzap.c:599 CHANNEL
EXCHANGE_MEDIA
2012-06-12 18:58:32.438740 [DEBUG] ozmod_isdn.c:1115 READ 22
------------------------------------------------------------------------
--------
[08 02 80 04 03 04 03 80 90 a3 1e 02 80 83 1e 02 80 83 7d 02 91 81]


2012-06-12 18:58:32.438740 [DEBUG] ozmod_isdn.c:584 Yay I got an event!
Type:[03] Size:[155] CRV: 4 (0x4, CTX: Terminator)
2012-06-12 18:58:32.438740 [DEBUG] ozmod_isdn.c:616 zchan 8e082e8 (1:1)
source isdn_data->channels_local_crv[0x4]
2012-06-12 18:58:32.438740 [WARNING] ozmod_isdn.c:737 Why bother
changing state on 1:1 from PROGRESS to PROGRESS
2012-06-12 18:58:32.458799 [DEBUG] ozmod_isdn.c:1115 READ 17
------------------------------------------------------------------------
--------
[08 02 80 04 01 1e 02 80 83 1e 02 82 88 7d 02 91 81]


2012-06-12 18:58:32.458799 [DEBUG] ozmod_isdn.c:584 Yay I got an event!
Type:[01] Size:[121] CRV: 4 (0x4, CTX: Terminator)
2012-06-12 18:58:32.458799 [DEBUG] ozmod_isdn.c:616 zchan 8e082e8 (1:1)
source isdn_data->channels_local_crv[0x4]
2012-06-12 18:58:32.458799 [DEBUG] ozmod_isdn.c:728 Changing state on
1:1 from PROGRESS to PROGRESS_MEDIA
2012-06-12 18:58:32.558735 [DEBUG] ozmod_isdn.c:1163 1:1 STATE
[PROGRESS_MEDIA]
2012-06-12 18:58:32.558735 [DEBUG] mod_openzap.c:1937 got clear channel
sig [PROGRESS_MEDIA]
2012-06-12 18:58:32.558735 [NOTICE] mod_openzap.c:1982 Pre-Answer
OpenZAP/1:1/8040327325!
2012-06-12 18:58:32.558735 [DEBUG] switch_channel.c:2668
(OpenZAP/1:1/8040327325) Callstate Change RINGING -> EARLY
2012-06-12 18:58:32.558735 [DEBUG] switch_channel.c:2707 Send signal
sofia/internal/324 at 192.168.1.155:5060 [BREAK]
2012-06-12 18:58:32.578743 [DEBUG] switch_core_codec.c:141
sofia/internal/324 at 192.168.1.155:5060 Restore previous codec PCMU:0.
2012-06-12 18:58:32.578743 [DEBUG] switch_ivr_originate.c:3206 Originate
Resulted in Success: [OpenZAP/1:1/8040327325]
2012-06-12 18:58:32.578743 [DEBUG] switch_core_session.c:707 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 18:58:32.578743 [DEBUG] switch_core_session.c:707 Send signal
sofia/internal/324 at 192.168.1.155:5060 [BREAK]
2012-06-12 18:58:32.578743 [DEBUG] switch_ivr_bridge.c:1239
(OpenZAP/1:1/8040327325) State Change CS_CONSUME_MEDIA ->
CS_EXCHANGE_MEDIA
2012-06-12 18:58:32.578743 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 18:58:32.578743 [DEBUG] switch_core_state_machine.c:325
(OpenZAP/1:1/8040327325) Running State Change CS_EXCHANGE_MEDIA
2012-06-12 18:58:32.578743 [DEBUG] switch_core_state_machine.c:374
(OpenZAP/1:1/8040327325) State EXCHANGE_MEDIA
2012-06-12 18:58:32.578743 [DEBUG] mod_openzap.c:599 CHANNEL
EXCHANGE_MEDIA
2012-06-12 18:58:33.758609 [DEBUG] ozmod_isdn.c:1115 READ 5
------------------------------------------------------------------------
--------
[08 02 80 05 07]


2012-06-12 18:58:33.758609 [DEBUG] ozmod_isdn.c:584 Yay I got an event!
Type:[07] Size:[103] CRV: 5 (0x5, CTX: Terminator)
2012-06-12 18:58:33.758609 [DEBUG] ozmod_isdn.c:616 zchan b750c4f0 (2:1)
source isdn_data->channels_local_crv[0x5]
2012-06-12 18:58:33.758609 [DEBUG] ozmod_isdn.c:746 Changing state on
2:1 from PROGRESS_MEDIA to UP
2012-06-12 18:58:33.758609 [DEBUG] ozmod_isdn.c:1938 WRITE 5
------------------------------------------------------------------------
--------
[08 02 00 05 0f]

2012-06-12 18:58:33.758609 [DEBUG] ozmod_isdn.c:1163 2:1 STATE [UP]
2012-06-12 18:58:33.758609 [DEBUG] mod_openzap.c:1937 got clear channel
sig [UP]
2012-06-12 18:58:33.758609 [DEBUG] switch_channel.c:2859
(OpenZAP/2:1/8040327325) Callstate Change EARLY -> ACTIVE
2012-06-12 18:58:33.758609 [DEBUG] switch_channel.c:2871 Send signal
OpenZAP/1:2/325 [BREAK]
2012-06-12 18:58:33.758609 [NOTICE] mod_openzap.c:1968 Channel
[OpenZAP/2:1/8040327325] has been answered
2012-06-12 18:58:33.818609 [DEBUG] ozmod_isdn.c:1163 1:2 STATE [UP]
2012-06-12 18:58:33.818609 [DEBUG] ozmod_isdn.c:1938 WRITE 18
------------------------------------------------------------------------
--------
[08 02 9d 81 07 18 03 a1 83 82 1e 02 80 83 7d 02 91 81]

2012-06-12 18:58:33.838610 [DEBUG] switch_core_session.c:707 Send signal
OpenZAP/1:2/325 [BREAK]
2012-06-12 18:58:33.838610 [DEBUG] switch_channel.c:2859
(OpenZAP/1:2/325) Callstate Change EARLY -> ACTIVE
2012-06-12 18:58:33.838610 [NOTICE] switch_ivr_bridge.c:417 Channel
[OpenZAP/1:2/325] has been answered
2012-06-12 18:58:33.918607 [DEBUG] ozmod_isdn.c:1115 READ 26
------------------------------------------------------------------------
--------
[08 02 80 04 07 04 03 80 90 a3 1e 02 80 83 29 06 0c 06 0c 13 0d 2a 7d 02
91 81]


2012-06-12 18:58:33.918607 [DEBUG] ozmod_isdn.c:584 Yay I got an event!
Type:[07] Size:[159] CRV: 4 (0x4, CTX: Terminator)
2012-06-12 18:58:33.918607 [DEBUG] ozmod_isdn.c:616 zchan 8e082e8 (1:1)
source isdn_data->channels_local_crv[0x4]
2012-06-12 18:58:33.918607 [DEBUG] ozmod_isdn.c:746 Changing state on
1:1 from PROGRESS_MEDIA to UP
2012-06-12 18:58:33.918607 [DEBUG] ozmod_isdn.c:1938 WRITE 5
------------------------------------------------------------------------
--------
[08 02 00 04 0f]

2012-06-12 18:58:33.918607 [DEBUG] ozmod_isdn.c:1163 1:1 STATE [UP]
2012-06-12 18:58:33.918607 [DEBUG] mod_openzap.c:1937 got clear channel
sig [UP]
2012-06-12 18:58:33.918607 [DEBUG] switch_channel.c:2859
(OpenZAP/1:1/8040327325) Callstate Change EARLY -> ACTIVE
2012-06-12 18:58:33.918607 [DEBUG] switch_channel.c:2871 Send signal
sofia/internal/324 at 192.168.1.155:5060 [BREAK]
2012-06-12 18:58:33.918607 [NOTICE] mod_openzap.c:1968 Channel
[OpenZAP/1:1/8040327325] has been answered
2012-06-12 18:58:33.939272 [DEBUG] mod_sofia.c:683 Local SDP
sofia/internal/324 at 192.168.1.155:5060:
v=0
o=FreeSWITCH 1339477129 1339477131 IN IP4 192.168.1.155
s=FreeSWITCH
c=IN IP4 192.168.1.155
t=0 0
m=audio 30582 RTP/AVP 0 110
a=rtpmap:0 PCMU/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2012-06-12 18:58:33.939272 [DEBUG] switch_core_session.c:707 Send signal
sofia/internal/324 at 192.168.1.155:5060 [BREAK]
2012-06-12 18:58:33.939272 [DEBUG] switch_channel.c:2859
(sofia/internal/324 at 192.168.1.155:5060) Callstate Change EARLY -> ACTIVE
2012-06-12 18:58:33.939272 [NOTICE] switch_ivr_bridge.c:417 Channel
[sofia/internal/324 at 192.168.1.155:5060] has been answered
2012-06-12 18:58:33.939272 [DEBUG] sofia.c:4770 Channel
sofia/internal/324 at 192.168.1.155:5060 entering state [completed][200]
2012-06-12 18:58:33.939272 [DEBUG] ozmod_isdn.c:1115 READ 5
------------------------------------------------------------------------
--------
[08 02 1d 81 0f]


2012-06-12 18:58:33.939272 [DEBUG] ozmod_isdn.c:584 Yay I got an event!
Type:[0f] Size:[103] CRV: 7553 (0x1d81, CTX: Originator)
2012-06-12 18:58:33.939272 [DEBUG] ozmod_isdn.c:616 zchan 8e0dc38 (1:2)
source isdn_data->channels_remote_crv[0x1d81]
2012-06-12 18:58:33.939272 [DEBUG] ozmod_isdn.c:973 Received CONNECT_ACK
message for channel 0
2012-06-12 18:58:33.959605 [DEBUG] switch_core_media_bug.c:360 Attaching
BUG to sofia/internal/324 at 192.168.1.155:5060
2012-06-12 18:58:33.999606 [DEBUG] switch_rtp.c:3104 Correct ip/port
confirmed.
2012-06-12 18:58:34.039603 [DEBUG] sofia.c:4770 Channel
sofia/internal/324 at 192.168.1.155:5060 entering state [ready][200]
2012-06-12 18:58:38.019513 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
1:2240
2012-06-12 18:58:38.019513 [DEBUG] zap_io.c:1469 Created DTMF Buffer!
2012-06-12 18:58:38.019513 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 18:58:38.019513 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [1]
2012-06-12 18:58:38.079514 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 2012-06-12 18:58:38.079514 [DEBUG] zap_io.c:1469 Created DTMF
Buffer!
1
2012-06-12 18:58:38.079514 [DEBUG] mod_openzap.c:725 queue DTMF [1]
2012-06-12 18:58:38.079514 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:58:38.099512 [DEBUG] zap_io.c:2137 2:1 GENERATE DTMF [1]
2012-06-12 18:58:38.299508 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
2:2240
2012-06-12 18:58:38.299508 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 18:58:38.299508 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [2]
2012-06-12 18:58:38.379506 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 2
2012-06-12 18:58:38.379506 [DEBUG] mod_openzap.c:725 queue DTMF [2]
2012-06-12 18:58:38.379506 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:58:40.819450 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
3:2240
2012-06-12 18:58:40.819450 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 18:58:40.819450 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [3]
2012-06-12 18:58:40.879452 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 3
2012-06-12 18:58:40.879452 [DEBUG] mod_openzap.c:725 queue DTMF [3]
2012-06-12 18:58:40.879452 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:58:41.099444 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
4:2240
2012-06-12 18:58:41.099444 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 18:58:41.099444 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [4]
2012-06-12 18:58:41.199442 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 4
2012-06-12 18:58:41.199442 [DEBUG] mod_openzap.c:725 queue DTMF [4]
2012-06-12 18:58:41.199442 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:58:41.379438 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
5:2240
2012-06-12 18:58:41.379438 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 18:58:41.379438 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [5]
2012-06-12 18:58:41.479436 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 5
2012-06-12 18:58:41.479436 [DEBUG] mod_openzap.c:725 queue DTMF [5]
2012-06-12 18:58:41.479436 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:58:41.659431 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
6:2240
2012-06-12 18:58:41.659431 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 18:58:41.659431 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [6]
2012-06-12 18:58:41.799429 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 6
2012-06-12 18:58:41.799429 [DEBUG] mod_openzap.c:725 queue DTMF [6]
2012-06-12 18:58:41.799429 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:58:41.939425 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
7:2240
2012-06-12 18:58:41.939425 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 18:58:41.939425 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [7]
2012-06-12 18:58:42.079422 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 7
2012-06-12 18:58:42.079422 [DEBUG] mod_openzap.c:725 queue DTMF [7]
2012-06-12 18:58:42.079422 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:58:42.159421 [WARNING] sofia_reg.c:1241 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [324 at 192.168.1.155] from ip
192.168.1.32
2012-06-12 18:58:42.219421 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
8:2240
2012-06-12 18:58:42.219421 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 18:58:42.219421 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [8]
2012-06-12 18:58:42.400415 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 8
2012-06-12 18:58:42.400415 [DEBUG] mod_openzap.c:725 queue DTMF [8]
2012-06-12 18:58:42.400415 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:58:42.500413 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
9:2240
2012-06-12 18:58:42.500413 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 18:58:42.500413 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [9]
2012-06-12 18:58:42.680409 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 9
2012-06-12 18:58:42.680409 [DEBUG] mod_openzap.c:725 queue DTMF [9]
2012-06-12 18:58:42.680409 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:58:50.242296 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
*:2240
2012-06-12 18:58:50.242296 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 18:58:50.242296 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [*]
2012-06-12 18:58:50.304238 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF *
2012-06-12 18:58:50.304238 [DEBUG] mod_openzap.c:725 queue DTMF [*]
2012-06-12 18:58:50.304238 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:59:22.692383 [DEBUG] zap_io.c:2137 2:1 GENERATE DTMF
[23456789*]
2012-06-12 18:59:22.892378 [DEBUG] mod_openzap.c:784 Dropping frame!
(write not ready)
2012-06-12 18:59:23.332368 [DEBUG] mod_openzap.c:784 Dropping frame!
(write not ready)
2012-06-12 18:59:23.832357 [DEBUG] mod_openzap.c:784 Dropping frame!
(write not ready)
2012-06-12 18:59:32.440788 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
*:2240
2012-06-12 18:59:32.440788 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 18:59:32.440788 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [*]
2012-06-12 18:59:32.501788 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF *
2012-06-12 18:59:32.501788 [DEBUG] mod_openzap.c:725 queue DTMF [*]
2012-06-12 18:59:32.521799 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 18:59:32.563816 [DEBUG] zap_io.c:2137 2:1 GENERATE DTMF [*]
2012-06-12 18:59:34.845735 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
1:2240
2012-06-12 18:59:34.845735 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 18:59:34.845735 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [1]
2012-06-12 18:59:34.885760 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 1
2012-06-12 18:59:34.885760 [DEBUG] mod_openzap.c:725 queue DTMF [1]
2012-06-12 18:59:34.885760 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:25.244664 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
1:2240
2012-06-12 19:00:25.244664 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:25.244664 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [1]
2012-06-12 19:00:25.284665 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 1
2012-06-12 19:00:25.284665 [DEBUG] mod_openzap.c:725 queue DTMF [1]
2012-06-12 19:00:25.284665 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:25.524657 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
2:2240
2012-06-12 19:00:25.524657 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:25.524657 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [2]
2012-06-12 19:00:25.604656 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 2
2012-06-12 19:00:25.604656 [DEBUG] mod_openzap.c:725 queue DTMF [2]
2012-06-12 19:00:25.604656 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:25.804651 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
3:2240
2012-06-12 19:00:25.804651 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:25.804651 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [3]
2012-06-12 19:00:25.884663 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 3
2012-06-12 19:00:25.884663 [DEBUG] mod_openzap.c:725 queue DTMF [3]
2012-06-12 19:00:25.884663 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:26.084646 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
4:2240
2012-06-12 19:00:26.084646 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:26.084646 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [4]
2012-06-12 19:00:26.204663 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 4
2012-06-12 19:00:26.204663 [DEBUG] mod_openzap.c:725 queue DTMF [4]
2012-06-12 19:00:26.204663 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:26.364640 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
5:2240
2012-06-12 19:00:26.364640 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:26.364640 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [5]
2012-06-12 19:00:26.484637 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 5
2012-06-12 19:00:26.484637 [DEBUG] mod_openzap.c:725 queue DTMF [5]
2012-06-12 19:00:26.484637 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:26.644633 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
6:2240
2012-06-12 19:00:26.644633 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:26.644633 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [6]
2012-06-12 19:00:26.804630 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 6
2012-06-12 19:00:26.804630 [DEBUG] mod_openzap.c:725 queue DTMF [6]
2012-06-12 19:00:26.804630 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:26.924642 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
7:2240
2012-06-12 19:00:26.924642 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:26.924642 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [7]
2012-06-12 19:00:27.084623 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 7
2012-06-12 19:00:27.084623 [DEBUG] mod_openzap.c:725 queue DTMF [7]
2012-06-12 19:00:27.084623 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:27.204621 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
8:2240
2012-06-12 19:00:27.204621 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:27.204621 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [8]
2012-06-12 19:00:27.404626 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 8
2012-06-12 19:00:27.404626 [DEBUG] mod_openzap.c:725 queue DTMF [8]
2012-06-12 19:00:27.404626 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:27.484615 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
9:2240
2012-06-12 19:00:27.484615 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:27.484615 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [9]
2012-06-12 19:00:27.684610 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF 9
2012-06-12 19:00:27.684610 [DEBUG] mod_openzap.c:725 queue DTMF [9]
2012-06-12 19:00:27.684610 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:27.764608 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
*:2240
2012-06-12 19:00:27.764608 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:27.764608 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [*]
2012-06-12 19:00:28.004603 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF *
2012-06-12 19:00:28.004603 [DEBUG] mod_openzap.c:725 queue DTMF [*]
2012-06-12 19:00:28.004603 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:28.924597 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF
#:2240
2012-06-12 19:00:28.924597 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:28.924597 [DEBUG] zap_io.c:2137 1:1 GENERATE DTMF [#]
2012-06-12 19:00:28.984582 [DEBUG] zap_io.c:2031 [s1c2][1:2] Queuing
DTMF #
2012-06-12 19:00:28.984582 [DEBUG] mod_openzap.c:725 queue DTMF [#]
2012-06-12 19:00:28.984582 [DEBUG] switch_ivr_bridge.c:391 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:30.444548 [WARNING] sofia_reg.c:1241 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [324 at 192.168.1.155] from ip
192.168.1.32
2012-06-12 19:00:30.744532 [DEBUG] zap_io.c:2137 2:1 GENERATE DTMF
[1123456789*#]
2012-06-12 19:00:33.304329 [DEBUG] mod_openzap.c:784 Dropping frame!
(write not ready)
2012-06-12 19:00:33.504324 [DEBUG] mod_openzap.c:784 Dropping frame!
(write not ready)
2012-06-12 19:00:33.704320 [DEBUG] mod_openzap.c:784 Dropping frame!
(write not ready)
2012-06-12 19:00:33.904315 [DEBUG] mod_openzap.c:784 Dropping frame!
(write not ready)
2012-06-12 19:00:34.104311 [DEBUG] mod_openzap.c:784 Dropping frame!
(write not ready)
2012-06-12 19:00:34.304306 [DEBUG] mod_openzap.c:784 Dropping frame!
(write not ready)
2012-06-12 19:00:45.025232 [DEBUG] switch_channel.c:2592
(sofia/internal/324 at 192.168.1.155:5060) Callstate Change ACTIVE ->
HANGUP
2012-06-12 19:00:45.025232 [NOTICE] sofia.c:546 Hangup
sofia/internal/324 at 192.168.1.155:5060 [CS_EXECUTE] [NORMAL_CLEARING]
2012-06-12 19:00:45.025232 [DEBUG] switch_channel.c:2608 Send signal
sofia/internal/324 at 192.168.1.155:5060 [KILL]
2012-06-12 19:00:45.025232 [DEBUG] switch_core_session.c:1114 Send
signal sofia/internal/324 at 192.168.1.155:5060 [BREAK]
2012-06-12 19:00:45.045374 [DEBUG] switch_ivr_bridge.c:503
sofia/internal/324 at 192.168.1.155:5060 ending bridge by request from read
function
2012-06-12 19:00:45.045374 [DEBUG] switch_ivr_bridge.c:584 BRIDGE THREAD
DONE [sofia/internal/324 at 192.168.1.155:5060]
2012-06-12 19:00:45.045374 [DEBUG] switch_ivr_bridge.c:604 Send signal
OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:45.045374 [DEBUG] switch_ivr_bridge.c:497
sofia/internal/324 at 192.168.1.155:5060 ending bridge by request from
write function
2012-06-12 19:00:45.045374 [DEBUG] switch_ivr_bridge.c:584 BRIDGE THREAD
DONE [OpenZAP/1:1/8040327325]
2012-06-12 19:00:45.045374 [DEBUG] switch_ivr_bridge.c:604 Send signal
sofia/internal/324 at 192.168.1.155:5060 [BREAK]
2012-06-12 19:00:45.045374 [DEBUG] switch_channel.c:2592
(OpenZAP/1:1/8040327325) Callstate Change ACTIVE -> HANGUP
2012-06-12 19:00:45.045374 [NOTICE] switch_ivr_bridge.c:656 Hangup
OpenZAP/1:1/8040327325 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2012-06-12 19:00:45.045374 [DEBUG] switch_channel.c:2608 Send signal
OpenZAP/1:1/8040327325 [KILL]
2012-06-12 19:00:45.045374 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:374
(OpenZAP/1:1/8040327325) State EXCHANGE_MEDIA going to sleep
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:325
(OpenZAP/1:1/8040327325) Running State Change CS_HANGUP
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:565
(OpenZAP/1:1/8040327325) State HANGUP
2012-06-12 19:00:45.045374 [DEBUG] mod_openzap.c:544 Changing state on
1:1 from UP to HANGUP
2012-06-12 19:00:45.045374 [DEBUG] mod_openzap.c:560
OpenZAP/1:1/8040327325 CHANNEL HANGUP
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:46
OpenZAP/1:1/8040327325 Standard HANGUP, cause: NORMAL_CLEARING
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:565
(OpenZAP/1:1/8040327325) State HANGUP going to sleep
2012-06-12 19:00:45.045374 [DEBUG] switch_ivr_bridge.c:1310
OpenZAP/1:1/8040327325 skip receive message [UNBRIDGE] (channel is
hungup already)
2012-06-12 19:00:45.045374 [DEBUG] switch_ivr_bridge.c:1313
sofia/internal/324 at 192.168.1.155:5060 skip receive message [UNBRIDGE]
(channel is hungup already)
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:356
(OpenZAP/1:1/8040327325) State Change CS_HANGUP -> CS_REPORTING
2012-06-12 19:00:45.045374 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:325
(OpenZAP/1:1/8040327325) Running State Change CS_REPORTING
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:625
(OpenZAP/1:1/8040327325) State REPORTING
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:53
OpenZAP/1:1/8040327325 Standard REPORTING, cause: NORMAL_CLEARING
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:625
(OpenZAP/1:1/8040327325) State REPORTING going to sleep
2012-06-12 19:00:45.045374 [DEBUG] switch_core_session.c:2059
sofia/internal/324 at 192.168.1.155:5060 skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:350
(OpenZAP/1:1/8040327325) State Change CS_REPORTING -> CS_DESTROY
2012-06-12 19:00:45.045374 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/1:1/8040327325 [BREAK]
2012-06-12 19:00:45.045374 [DEBUG] switch_core_session.c:1286 Session 6
(OpenZAP/1:1/8040327325) Locked, Waiting on external entities
2012-06-12 19:00:45.045374 [NOTICE] switch_core_session.c:1304 Session 6
(OpenZAP/1:1/8040327325) Ended
2012-06-12 19:00:45.045374 [NOTICE] switch_core_session.c:1306 Close
Channel OpenZAP/1:1/8040327325 [CS_DESTROY]
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:454
(OpenZAP/1:1/8040327325) Callstate Change HANGUP -> DOWN
2012-06-12 19:00:45.045374 [DEBUG] switch_core_session.c:2059
sofia/internal/324 at 192.168.1.155:5060 skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:371
(sofia/internal/324 at 192.168.1.155:5060) State EXECUTE going to sleep
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/324 at 192.168.1.155:5060) Running State Change CS_HANGUP
2012-06-12 19:00:45.045374 [DEBUG] switch_ivr_async.c:936 Stop recording
file /usr/local/freeswitch/recordings/324/Jun-12/77-185333046.wav
2012-06-12 19:00:45.045374 [DEBUG] switch_core_media_bug.c:439 Removing
BUG from sofia/internal/324 at 192.168.1.155:5060
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:457
(OpenZAP/1:1/8040327325) Running State Change CS_DESTROY
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:467
(OpenZAP/1:1/8040327325) State DESTROY
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:565
(sofia/internal/324 at 192.168.1.155:5060) State HANGUP
2012-06-12 19:00:45.045374 [DEBUG] mod_sofia.c:453
sofia/internal/324 at 192.168.1.155:5060 Overriding SIP cause 480 with 200
from the other leg
2012-06-12 19:00:45.045374 [DEBUG] mod_sofia.c:459 Channel
sofia/internal/324 at 192.168.1.155:5060 hanging up, cause: NORMAL_CLEARING
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:46
sofia/internal/324 at 192.168.1.155:5060 Standard HANGUP, cause:
NORMAL_CLEARING
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:565
(sofia/internal/324 at 192.168.1.155:5060) State HANGUP going to sleep
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:356
(sofia/internal/324 at 192.168.1.155:5060) State Change CS_HANGUP ->
CS_REPORTING
2012-06-12 19:00:45.045374 [DEBUG] switch_core_session.c:1114 Send
signal sofia/internal/324 at 192.168.1.155:5060 [BREAK]
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/324 at 192.168.1.155:5060) Running State Change
CS_REPORTING
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:625
(sofia/internal/324 at 192.168.1.155:5060) State REPORTING
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:53
sofia/internal/324 at 192.168.1.155:5060 Standard REPORTING, cause:
NORMAL_CLEARING
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:625
(sofia/internal/324 at 192.168.1.155:5060) State REPORTING going to sleep
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:350
(sofia/internal/324 at 192.168.1.155:5060) State Change CS_REPORTING ->
CS_DESTROY
2012-06-12 19:00:45.045374 [DEBUG] switch_core_session.c:1114 Send
signal sofia/internal/324 at 192.168.1.155:5060 [BREAK]
2012-06-12 19:00:45.045374 [DEBUG] switch_core_session.c:1286 Session 5
(sofia/internal/324 at 192.168.1.155:5060) Locked, Waiting on external
entities
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:60
OpenZAP/1:1/8040327325 Standard DESTROY
2012-06-12 19:00:45.045374 [DEBUG] switch_core_state_machine.c:467
(OpenZAP/1:1/8040327325) State DESTROY going to sleep
2012-06-12 19:00:45.107225 [DEBUG] ozmod_isdn.c:1163 1:1 STATE [HANGUP]
2012-06-12 19:00:45.107225 [DEBUG] ozmod_isdn.c:1413 Hangup: Direction
Outbound
2012-06-12 19:00:45.107225 [DEBUG] ozmod_isdn.c:1938 WRITE 29
------------------------------------------------------------------------
--------
[08 02 00 04 45 08 02 81 90 1e 02 80 83 28 0e 44 65 76 20 54 65 61 6d 20
40 20 33 32 34]

2012-06-12 19:00:45.168168 [NOTICE] switch_core_session.c:1304 Session 5
(sofia/internal/324 at 192.168.1.155:5060) Ended
2012-06-12 19:00:45.168168 [NOTICE] switch_core_session.c:1306 Close
Channel sofia/internal/324 at 192.168.1.155:5060 [CS_DESTROY]
2012-06-12 19:00:45.168168 [DEBUG] switch_core_state_machine.c:454
(sofia/internal/324 at 192.168.1.155:5060) Callstate Change HANGUP -> DOWN
2012-06-12 19:00:45.168168 [DEBUG] switch_core_state_machine.c:457
(sofia/internal/324 at 192.168.1.155:5060) Running State Change CS_DESTROY
2012-06-12 19:00:45.168168 [DEBUG] switch_core_state_machine.c:467
(sofia/internal/324 at 192.168.1.155:5060) State DESTROY
2012-06-12 19:00:45.168168 [DEBUG] mod_sofia.c:364
sofia/internal/324 at 192.168.1.155:5060 SOFIA DESTROY
2012-06-12 19:00:45.168168 [DEBUG] switch_core_state_machine.c:60
sofia/internal/324 at 192.168.1.155:5060 Standard DESTROY
2012-06-12 19:00:45.168168 [DEBUG] switch_core_state_machine.c:467
(sofia/internal/324 at 192.168.1.155:5060) State DESTROY going to sleep
2012-06-12 19:00:45.229192 [DEBUG] ozmod_isdn.c:1115 READ 17
------------------------------------------------------------------------
--------
[08 02 1d 81 45 08 06 81 90 00 00 00 00 1e 02 80 83]


2012-06-12 19:00:45.229192 [DEBUG] ozmod_isdn.c:584 Yay I got an event!
Type:[45] Size:[115] CRV: 7553 (0x1d81, CTX: Originator)
2012-06-12 19:00:45.229192 [DEBUG] ozmod_isdn.c:616 zchan 8e0dc38 (1:2)
source isdn_data->channels_remote_crv[0x1d81]
2012-06-12 19:00:45.229192 [DEBUG] ozmod_isdn.c:719 Changing state on
1:2 from UP to TERMINATING
2012-06-12 19:00:45.289082 [DEBUG] ozmod_isdn.c:1163 1:2 STATE
[TERMINATING]
2012-06-12 19:00:45.289082 [DEBUG] ozmod_isdn.c:1473 Terminating:
Direction Inbound
2012-06-12 19:00:45.289082 [DEBUG] mod_openzap.c:1937 got clear channel
sig [STOP]
2012-06-12 19:00:45.289082 [DEBUG] switch_channel.c:2592
(OpenZAP/1:2/325) Callstate Change ACTIVE -> HANGUP
2012-06-12 19:00:45.289082 [NOTICE] mod_openzap.c:1958 Hangup
OpenZAP/1:2/325 [CS_EXECUTE] [NORMAL_CLEARING]
2012-06-12 19:00:45.289082 [DEBUG] switch_channel.c:2608 Send signal
OpenZAP/1:2/325 [KILL]
2012-06-12 19:00:45.289082 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/1:2/325 [BREAK]
2012-06-12 19:00:45.289082 [DEBUG] ozmod_isdn.c:1938 WRITE 5
------------------------------------------------------------------------
--------
[08 02 9d 81 4d]

2012-06-12 19:00:45.289082 [DEBUG] switch_ivr_bridge.c:584 BRIDGE THREAD
DONE [OpenZAP/1:2/325]
2012-06-12 19:00:45.289082 [DEBUG] switch_ivr_bridge.c:604 Send signal
OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:45.289082 [DEBUG] switch_ivr_bridge.c:584 BRIDGE THREAD
DONE [OpenZAP/2:1/8040327325]
2012-06-12 19:00:45.289082 [DEBUG] switch_ivr_bridge.c:604 Send signal
OpenZAP/1:2/325 [BREAK]
2012-06-12 19:00:45.289082 [DEBUG] switch_channel.c:2592
(OpenZAP/2:1/8040327325) Callstate Change ACTIVE -> HANGUP
2012-06-12 19:00:45.289082 [NOTICE] switch_ivr_bridge.c:656 Hangup
OpenZAP/2:1/8040327325 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2012-06-12 19:00:45.289082 [DEBUG] switch_channel.c:2608 Send signal
OpenZAP/2:1/8040327325 [KILL]
2012-06-12 19:00:45.289082 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:374
(OpenZAP/2:1/8040327325) State EXCHANGE_MEDIA going to sleep
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:325
(OpenZAP/2:1/8040327325) Running State Change CS_HANGUP
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:565
(OpenZAP/2:1/8040327325) State HANGUP
2012-06-12 19:00:45.289082 [DEBUG] mod_openzap.c:544 Changing state on
2:1 from UP to HANGUP
2012-06-12 19:00:45.289082 [DEBUG] mod_openzap.c:560
OpenZAP/2:1/8040327325 CHANNEL HANGUP
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:46
OpenZAP/2:1/8040327325 Standard HANGUP, cause: NORMAL_CLEARING
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:565
(OpenZAP/2:1/8040327325) State HANGUP going to sleep
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:356
(OpenZAP/2:1/8040327325) State Change CS_HANGUP -> CS_REPORTING
2012-06-12 19:00:45.289082 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:325
(OpenZAP/2:1/8040327325) Running State Change CS_REPORTING
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:625
(OpenZAP/2:1/8040327325) State REPORTING
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:53
OpenZAP/2:1/8040327325 Standard REPORTING, cause: NORMAL_CLEARING
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:625
(OpenZAP/2:1/8040327325) State REPORTING going to sleep
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:350
(OpenZAP/2:1/8040327325) State Change CS_REPORTING -> CS_DESTROY
2012-06-12 19:00:45.289082 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/2:1/8040327325 [BREAK]
2012-06-12 19:00:45.289082 [DEBUG] switch_core_session.c:1286 Session 8
(OpenZAP/2:1/8040327325) Locked, Waiting on external entities
2012-06-12 19:00:45.289082 [DEBUG] switch_ivr_bridge.c:1313
OpenZAP/1:2/325 skip receive message [UNBRIDGE] (channel is hungup
already)
2012-06-12 19:00:45.289082 [DEBUG] switch_core_session.c:2059
OpenZAP/1:2/325 skip receive message [APPLICATION_EXEC_COMPLETE]
(channel is hungup already)
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:371
(OpenZAP/1:2/325) State EXECUTE going to sleep
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:325
(OpenZAP/1:2/325) Running State Change CS_HANGUP
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:565
(OpenZAP/1:2/325) State HANGUP
2012-06-12 19:00:45.289082 [DEBUG] mod_openzap.c:560 OpenZAP/1:2/325
CHANNEL HANGUP
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:46
OpenZAP/1:2/325 Standard HANGUP, cause: NORMAL_CLEARING
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:565
(OpenZAP/1:2/325) State HANGUP going to sleep
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:356
(OpenZAP/1:2/325) State Change CS_HANGUP -> CS_REPORTING
2012-06-12 19:00:45.289082 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/1:2/325 [BREAK]
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:325
(OpenZAP/1:2/325) Running State Change CS_REPORTING
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:625
(OpenZAP/1:2/325) State REPORTING
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:53
OpenZAP/1:2/325 Standard REPORTING, cause: NORMAL_CLEARING
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:625
(OpenZAP/1:2/325) State REPORTING going to sleep
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:350
(OpenZAP/1:2/325) State Change CS_REPORTING -> CS_DESTROY
2012-06-12 19:00:45.289082 [DEBUG] switch_core_session.c:1114 Send
signal OpenZAP/1:2/325 [BREAK]
2012-06-12 19:00:45.289082 [DEBUG] switch_core_session.c:1286 Session 7
(OpenZAP/1:2/325) Locked, Waiting on external entities
2012-06-12 19:00:45.289082 [NOTICE] switch_core_session.c:1304 Session 7
(OpenZAP/1:2/325) Ended
2012-06-12 19:00:45.289082 [NOTICE] switch_core_session.c:1306 Close
Channel OpenZAP/1:2/325 [CS_DESTROY]
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:454
(OpenZAP/1:2/325) Callstate Change HANGUP -> DOWN
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:457
(OpenZAP/1:2/325) Running State Change CS_DESTROY
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:467
(OpenZAP/1:2/325) State DESTROY
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:60
OpenZAP/1:2/325 Standard DESTROY
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:467
(OpenZAP/1:2/325) State DESTROY going to sleep
2012-06-12 19:00:45.289082 [NOTICE] switch_core_session.c:1304 Session 8
(OpenZAP/2:1/8040327325) Ended
2012-06-12 19:00:45.289082 [NOTICE] switch_core_session.c:1306 Close
Channel OpenZAP/2:1/8040327325 [CS_DESTROY]
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:454
(OpenZAP/2:1/8040327325) Callstate Change HANGUP -> DOWN
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:457
(OpenZAP/2:1/8040327325) Running State Change CS_DESTROY
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:467
(OpenZAP/2:1/8040327325) State DESTROY
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:60
OpenZAP/2:1/8040327325 Standard DESTROY
2012-06-12 19:00:45.289082 [DEBUG] switch_core_state_machine.c:467
(OpenZAP/2:1/8040327325) State DESTROY going to sleep
2012-06-12 19:00:45.289082 [DEBUG] ozmod_isdn.c:1115 READ 5
------------------------------------------------------------------------
--------
[08 02 80 04 4d]


2012-06-12 19:00:45.289082 [DEBUG] ozmod_isdn.c:584 Yay I got an event!
Type:[4d] Size:[103] CRV: 4 (0x4, CTX: Terminator)
2012-06-12 19:00:45.289082 [DEBUG] ozmod_isdn.c:616 zchan 8e082e8 (1:1)
source isdn_data->channels_local_crv[0x4]
2012-06-12 19:00:45.289082 [DEBUG] ozmod_isdn.c:680 Changing state on
1:1 from HANGUP to HANGUP_COMPLETE
2012-06-12 19:00:45.289082 [DEBUG] ozmod_isdn.c:1938 WRITE 5
------------------------------------------------------------------------
--------
[08 02 00 04 5a]

2012-06-12 19:00:45.289082 [DEBUG] ozmod_isdn.c:1120 931 parse error [1]
[Q931E_NO_ERROR]
2012-06-12 19:00:45.329082 [DEBUG] ozmod_isdn.c:1163 1:1 STATE
[HANGUP_COMPLETE]
2012-06-12 19:00:45.329082 [DEBUG] ozmod_isdn.c:1406 Changing state on
1:1 from HANGUP_COMPLETE to DOWN
2012-06-12 19:00:45.329082 [DEBUG] ozmod_isdn.c:1163 2:1 STATE [HANGUP]
2012-06-12 19:00:45.329082 [DEBUG] ozmod_isdn.c:1413 Hangup: Direction
Outbound
2012-06-12 19:00:45.329082 [DEBUG] ozmod_isdn.c:1938 WRITE 25
------------------------------------------------------------------------
--------
[08 02 00 05 45 08 02 81 90 1e 02 80 83 28 0a 38 30 34 30 33 32 37 33 32
34]

2012-06-12 19:00:45.349081 [DEBUG] ozmod_isdn.c:1115 READ 5
------------------------------------------------------------------------
--------
[08 02 80 05 4d]


2012-06-12 19:00:45.349081 [DEBUG] ozmod_isdn.c:584 Yay I got an event!
Type:[4d] Size:[103] CRV: 5 (0x5, CTX: Terminator)
2012-06-12 19:00:45.349081 [DEBUG] ozmod_isdn.c:616 zchan b750c4f0 (2:1)
source isdn_data->channels_local_crv[0x5]
2012-06-12 19:00:45.349081 [DEBUG] ozmod_isdn.c:680 Changing state on
2:1 from HANGUP to HANGUP_COMPLETE
2012-06-12 19:00:45.349081 [DEBUG] ozmod_isdn.c:1938 WRITE 5
------------------------------------------------------------------------
--------
[08 02 00 05 5a]

2012-06-12 19:00:45.349081 [DEBUG] ozmod_isdn.c:1120 931 parse error [1]
[Q931E_NO_ERROR]
2012-06-12 19:00:45.368996 [DEBUG] ozmod_isdn.c:1163 2:1 STATE
[HANGUP_COMPLETE]
2012-06-12 19:00:45.368996 [DEBUG] ozmod_isdn.c:1406 Changing state on
2:1 from HANGUP_COMPLETE to DOWN
2012-06-12 19:00:45.428780 [DEBUG] ozmod_isdn.c:1163 1:1 STATE [DOWN]
2012-06-12 19:00:45.428780 [DEBUG] zap_io.c:1415 channel done 1:1
2012-06-12 19:00:45.428780 [DEBUG] ozmod_isdn.c:1115 READ 5
------------------------------------------------------------------------
--------
[08 02 1d 81 5a]


2012-06-12 19:00:45.428780 [DEBUG] ozmod_isdn.c:584 Yay I got an event!
Type:[5a] Size:[103] CRV: 7553 (0x1d81, CTX: Originator)
2012-06-12 19:00:45.428780 [DEBUG] ozmod_isdn.c:616 zchan 8e0dc38 (1:2)
source isdn_data->channels_remote_crv[0x1d81]
2012-06-12 19:00:45.428780 [DEBUG] ozmod_isdn.c:682 Changing state on
1:2 from TERMINATING to DOWN
2012-06-12 19:00:45.468782 [DEBUG] ozmod_isdn.c:1163 2:1 STATE [DOWN]
2012-06-12 19:00:45.468782 [DEBUG] zap_io.c:1415 channel done 2:1
2012-06-12 19:00:45.528782 [DEBUG] ozmod_isdn.c:1163 1:2 STATE [DOWN]
2012-06-12 19:00:45.528782 [DEBUG] zap_io.c:1415 channel done 1:2
 
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From sdevoy at bizfocused.com  Tue Jun 12 19:18:19 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 12 Jun 2012 11:18:19 -0400
Subject: [Freeswitch-users] Moving up to better PC - Multi-tenant died
In-Reply-To: <20120612081252.GA13156@eagle.cupis.co.uk>
References: <05a501cd4856$41711610$c4534230$@bizfocused.com>
	<20120612081252.GA13156@eagle.cupis.co.uk>
Message-ID: <076d01cd48ae$991ff450$cb5fdcf0$@bizfocused.com>

Thanks for the reply Paul.

Results:
[root at FreeSwitch1 bin]#   chkconfig --list freeswitch
freeswitch      0:off   1:off   2:off   3:on    4:on    5:on    6:off

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul
Cupis
Sent: Tuesday, June 12, 2012 4:13 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Moving up to better PC - Multi-tenant died

On Tue, Jun 12, 2012 at 12:45:56AM -0400, Sean Devoy wrote:
>    I built a nice clean Centos 5.7 install on a Dell Power Edge 2850 Dual
>    Xeon 3.0 Ghs 4GB RAM and current Freeswitch.

>                    1. Freeswitch does not startup at boot.  I tried to
>    follow the wiki.  I have a freeswitch file in /;etc/init.d but no
>    startup!

Can you run:

  chkconfig --list freeswitch

and check the output?

You may need to do:

  chkconfig --add freeswitch

(where 'freeswitch' is the name of your init-script in /etc/init.d).

Regards,


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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




From th982a at googlemail.com  Tue Jun 12 20:10:59 2012
From: th982a at googlemail.com (Tamer Higazi)
Date: Tue, 12 Jun 2012 18:10:59 +0200
Subject: [Freeswitch-users] freeswitch and mISDN
In-Reply-To: <4FD29902.3080702@freeswitch.org>
References: <4FD13E02.4040202@googlemail.com> <4FD29902.3080702@freeswitch.org>
Message-ID: <4FD76A13.1000306@googlemail.com>

Thank you!


Tamer


Am 09.06.2012 02:29, schrieb Stefan Knoblich:
> On 06/08/12 01:49, Tamer Higazi wrote:
>> How do I configure mISDN in freetdm, and what are the channel drivers, with it's dial-plan respectively.
> 
> 
> The initial commit of ftmod_misdn contains the freetdm.conf/mISDN-specific setup instructions:
> 
>    http://git.freeswitch.org/git/freeswitch/commit/?id=09a61f5025b1c7dee05259db0e169f39a565b62d
> 
>    (Complete history of the module: http://git.freeswitch.org/git/freeswitch/log/libs/freetdm/src/ftmod/ftmod_misdn/ftmod_misdn.c )
> 
> This covers only the I/O part (ftmod_misdn, like ftmod_zap or ftmod_wanpipe, is an I/O module),
> for the signalling part you'll have to (build, install and) configure ftmod_libpri in freetdm.conf.xml:
> 
>    http://stkn.techmage.de/archives/191
> 
> is (very) a short summary (ignore the dahdi/zaptel specific configuration parts).
> 
> 
> ftmod_libpri span parameters:
> 
>       context  -  FreeSWITCH dialplan context to send incoming calls to
> 
>       dialplan -  FreeSWITCH dialplan (e.g. "XML")
> 
> 
>       mode    -  "user" or "net", the latter being unsupported with ftmod_misdn,
>                  due to restrictions caused by the way FreeTDM handles configuration of I/O modules.
> 
>       dialect -  Q.931 protocol dialect used for signalling: "euroisdn"(/"q931"),
>                  (other possible values: "ni1", "ni2", "dms100", "5ess", "4ess", "gr303eoc", "gr303tmc")
> 
>       opts    -  Comma separated list of flags:
> 
>                      "suggest_channel"         : Not sure what this one does, ignore this one.
> 
>                      "omit_display"            : Disables sending callerid name, some switches don't like this.
> 
>                      "omit_redirecting_number" : The same for REDIRECTING NUMBER information elements.
> 
>                      "aoc"                     : Enables some Advice of Charge bits in ftmod_libpri, supposed to
>                                                  handle incoming AOC messages and output some details at NOTICE loglevel
>                                                  (experimental).
> 
>       ton     -  Default Type-of-Number for outgoing calls: "international", "national", "local", "private", "unknown" (default).
>                  Better not touch this unless you really have to.
> 
>       layer1  -  B-Channel Layer1 protocol, "alaw" or "ulaw"   NOTE: ftmod_libpri will auto-select the right one
>                  based on the the chosen dialect (a-law for EuroISDN/Q.931, u-law for everything else).
> 
>       overlapdial - "yes" (or "both"), "receive", "send", "no"
> 
>       debug   -  Comma separated list of debug flags for libpri:
> 
>                      "q921_state" + "q921_dump" + "q921_raw"     = "q921_all" : Enables Q.921 logging (State changes, decoded messages, raw messages)
>                      "q931_state" + "q931_dump" + "q931_anomaly" = "q931_all" : Enables Q.931 logging (State changes, decoded messages, message anomalies)
> 
>                      "aoc"    : AOC debug messages
>                      "apdu"   : Q.932 / ROSE debug messages(?)
>                      "config" : libpri configuration debug messages
> 
>                      "all"  : Enable all of the above
>                      "none" : Disables all
> 
> 
>       service_message_support   - Enables support for maintenance / restart / service messages  (not EuroISDN/Q,931)
> 
> 
>       And there are a couple of new features (MSN filter, Numer prefix/ToN autoselection) for ftmod_libpri sitting in my post-1.2.0 queue.
> 
> 
> Dialplan:  Incoming calls should be sent to their own ISDN-specific dialplan context, makes handling them easier
>            (like the public dialplan for the external SIP profile).
> 
> 
> I'll stop here now, it's past 2 AM and i can't concentrate anymore.
> 
> stkn
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mario_fs at mgtech.com  Tue Jun 12 20:20:05 2012
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 12 Jun 2012 09:20:05 -0700
Subject: [Freeswitch-users] NAT Router Recommendation For Large no of IP
	Phones Behind the Router
In-Reply-To: <CAGvuWYaqsqO6N0y+5+pJ2zyA0UqS0iAFjQiXuaSxssH52GcdVQ@mail.gmail.com>
References: <CAGvuWYaqsqO6N0y+5+pJ2zyA0UqS0iAFjQiXuaSxssH52GcdVQ@mail.gmail.com>
Message-ID: <729B9245-CBCA-4F7F-9B71-9142E0ADA448@mgtech.com>

I looked for a year to find something with no NAT problems, dual WAN, and auto fallback with load/balancing without causing FS any hiccups. I highly recommend Zyxel, I use a USG100 with SPI ALG turned ON (I read some people hate it) but it has been bullet proof. And the best part is there is practically nothing to tell FS about! And when one WAN (DSL) goes down the SIP traffic goes out the other one, even switches back automatically! Pair this up with FS and you'll never look back. Hope this helps.
Mario G

On Jun 11, 2012, at 6:17 PM, Lloyd Aloysius wrote:

> Hi All:
> 
> Currently I use pfSense for NAT implementations. But I found there are few bugs in the firmware cause Presence not working reliably.
> 
> What are best routers out there with good support for SIP. I am looking for a NAT router that support 20-30 phones . Any good recommendations?
> 
> Thanks
> Lloyd
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue Jun 12 21:03:41 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Tue, 12 Jun 2012 19:03:41 +0200
Subject: [Freeswitch-users] NAT Router Recommendation For Large no of IP
 Phones Behind the Router
In-Reply-To: <729B9245-CBCA-4F7F-9B71-9142E0ADA448@mgtech.com>
References: <CAGvuWYaqsqO6N0y+5+pJ2zyA0UqS0iAFjQiXuaSxssH52GcdVQ@mail.gmail.com>
	<729B9245-CBCA-4F7F-9B71-9142E0ADA448@mgtech.com>
Message-ID: <4FD7766D.8040208@puzzled.xs4all.nl>

On 12-06-12 18:20, Mario G wrote:
> I looked for a year to find something with no NAT problems, dual WAN,
> and auto fallback with load/balancing without causing FS any hiccups. I
> highly recommend Zyxel, I use a USG100 with SPI ALG turned ON (I read
> some people hate it) but it has been bullet proof. And the best part is
> there is practically nothing to tell FS about! And when one WAN (DSL)
> goes down the SIP traffic goes out the other one, even switches back
> automatically! Pair this up with FS and you'll never look back. Hope
> this helps.

Thanks for the info. Have you tried any Draytek Vigor routers like the 
2920? I have no experience with them but was told they work fine in dual 
WAN setups.

Regards,
Patrick


From djbinter at gmail.com  Tue Jun 12 21:08:18 2012
From: djbinter at gmail.com (DJB International)
Date: Tue, 12 Jun 2012 10:08:18 -0700
Subject: [Freeswitch-users] Polycom caller ID problem on 4.0.2
In-Reply-To: <CAAtWKeFcgvVbW03ece3yGFLGoXxeP5uiDuTTUWeQ7s+5wmqs2g@mail.gmail.com>
References: <CAAB3jWsQPFyrr1d=d6gfA0G9Q+kyM0v7VyvDoOYrq=h6LaZ9rA@mail.gmail.com>
	<CAAtWKeFcgvVbW03ece3yGFLGoXxeP5uiDuTTUWeQ7s+5wmqs2g@mail.gmail.com>
Message-ID: <CAAB3jWtVAso3T74b5jPqFEXJtV-jXFoMu27DkbWK-GBxy_SxRw@mail.gmail.com>

Maybe I should downgrade the firmware back to 3.3.x

-djbinter

On Mon, Jun 11, 2012 at 8:56 PM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> You will soon notice another problem with them: If you use shared lines
> then all outgoing calls are listed as incoming, and all unanswered outgoing
> calls are listed as incoming missed calls...
>
>                      __Yehavi:
>
> 2012/6/11 DJB International <djbinter at gmail.com>
>
>> I've recently updated my Polycom 650 phone to use 4.0.2.
>>
>> I am not sure what broke the caller id for received calls, it kept
>> showing as Unknown Party; however, the weird thing is that the Missed Calls
>> show the correct caller id.
>>
>> Anyone experienced this issue.  Please advise.
>>
>> FreeSWITCH Version 1.2.0-rc2
>>
>> Thank you.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sdevoy at bizfocused.com  Tue Jun 12 21:29:54 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 12 Jun 2012 13:29:54 -0400
Subject: [Freeswitch-users] Moving up to better PC - Multi-tenant died
In-Reply-To: <20120612081252.GA13156@eagle.cupis.co.uk>
References: <05a501cd4856$41711610$c4534230$@bizfocused.com>
	<20120612081252.GA13156@eagle.cupis.co.uk>
Message-ID: <00cf01cd48c0$fb279b50$f176d1f0$@bizfocused.com>

UPDATE:
If I watch the console during startup I see it trying to start freeswitch
.... [FAILED]

How do I determine why it failed?

Sean

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul
Cupis
Sent: Tuesday, June 12, 2012 4:13 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Moving up to better PC - Multi-tenant died

On Tue, Jun 12, 2012 at 12:45:56AM -0400, Sean Devoy wrote:
>    I built a nice clean Centos 5.7 install on a Dell Power Edge 2850 Dual
>    Xeon 3.0 Ghs 4GB RAM and current Freeswitch.

>                    1. Freeswitch does not startup at boot.  I tried to
>    follow the wiki.  I have a freeswitch file in /;etc/init.d but no
>    startup!

Can you run:

  chkconfig --list freeswitch

and check the output?

You may need to do:

  chkconfig --add freeswitch

(where 'freeswitch' is the name of your init-script in /etc/init.d).

Regards,


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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




From sdevoy at bizfocused.com  Tue Jun 12 21:57:15 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 12 Jun 2012 13:57:15 -0400
Subject: [Freeswitch-users] Moving up to better PC - Multi-tenant died
In-Reply-To: <00cf01cd48c0$fb279b50$f176d1f0$@bizfocused.com>
References: <05a501cd4856$41711610$c4534230$@bizfocused.com>	<20120612081252.GA13156@eagle.cupis.co.uk>
	<00cf01cd48c0$fb279b50$f176d1f0$@bizfocused.com>
Message-ID: <00f601cd48c4$cd7502c0$685f0840$@bizfocused.com>

Startup problem resolved.  The file /etc/init.d/freeswitch has syntax I do
not understand at the beginning but I changed it to this and it worked fine:

	PROG_NAME=freeswitch
	PID_FILE=/usr/local/freeswitch/run/freeswitch.pid
	FS_USER=freeswitch
	FS_FILE=/usr/local/freeswitch/bin/freeswitch 
	FS_HOME=/usr/local/freeswitch
	LOCK_FILE=/var/lock/subsys/freeswitch
	FREESWITCH_ARGS="-nc"
	RETVAL=0

Hope that helps someone.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Devoy
Sent: Tuesday, June 12, 2012 1:30 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Moving up to better PC - Multi-tenant died

UPDATE:
If I watch the console during startup I see it trying to start freeswitch
.... [FAILED]

How do I determine why it failed?

Sean

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul
Cupis
Sent: Tuesday, June 12, 2012 4:13 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Moving up to better PC - Multi-tenant died

On Tue, Jun 12, 2012 at 12:45:56AM -0400, Sean Devoy wrote:
>    I built a nice clean Centos 5.7 install on a Dell Power Edge 2850 Dual
>    Xeon 3.0 Ghs 4GB RAM and current Freeswitch.

>                    1. Freeswitch does not startup at boot.  I tried to
>    follow the wiki.  I have a freeswitch file in /;etc/init.d but no
>    startup!

Can you run:

  chkconfig --list freeswitch

and check the output?

You may need to do:

  chkconfig --add freeswitch

(where 'freeswitch' is the name of your init-script in /etc/init.d).

Regards,


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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




From patrick at sunsus.net  Tue Jun 12 22:08:03 2012
From: patrick at sunsus.net (sunsus)
Date: Tue, 12 Jun 2012 11:08:03 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch 304 Unauthorized
In-Reply-To: <1339334863709-7579670.post@n2.nabble.com>
References: <1339334863709-7579670.post@n2.nabble.com>
Message-ID: <1339524483645-7579753.post@n2.nabble.com>

working sip trace with debug: 


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-304-Unauthorized-tp7579670p7579753.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From bdfoster at endigotech.com  Tue Jun 12 22:10:21 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 12 Jun 2012 14:10:21 -0400
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch 304 Unauthorized
In-Reply-To: <1339524483645-7579753.post@n2.nabble.com>
References: <1339334863709-7579670.post@n2.nabble.com>
	<1339524483645-7579753.post@n2.nabble.com>
Message-ID: <CAKV-jFwWLygHJVozz41=5rsjXMVuTrDsek98Rty2DL4zF1AC1w@mail.gmail.com>

No link, try again.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 12, 2012 2:08 PM, "sunsus" <patrick at sunsus.net> wrote:

> working sip trace with debug:
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-304-Unauthorized-tp7579670p7579753.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From patrick at sunsus.net  Tue Jun 12 22:16:35 2012
From: patrick at sunsus.net (sunsus)
Date: Tue, 12 Jun 2012 11:16:35 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch 304 Unauthorized
In-Reply-To: <1339524483645-7579753.post@n2.nabble.com>
References: <1339334863709-7579670.post@n2.nabble.com>
	<1339524483645-7579753.post@n2.nabble.com>
Message-ID: <1339524995793-7579754.post@n2.nabble.com>

not working sip trace with debug: 



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From patrick at sunsus.net  Tue Jun 12 22:18:47 2012
From: patrick at sunsus.net (sunsus)
Date: Tue, 12 Jun 2012 11:18:47 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch 304 Unauthorized
In-Reply-To: <1339524995793-7579754.post@n2.nabble.com>
References: <1339334863709-7579670.post@n2.nabble.com>
	<1339524483645-7579753.post@n2.nabble.com>
	<1339524995793-7579754.post@n2.nabble.com>
Message-ID: <1339525127651-7579755.post@n2.nabble.com>

gateway configuration 


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From sdevoy at bizfocused.com  Tue Jun 12 22:38:45 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 12 Jun 2012 14:38:45 -0400
Subject: [Freeswitch-users] Domain name lost in SIP registration (w/ SIP
	TRACE)
Message-ID: <011e01cd48ca$99953aa0$ccbfafe0$@bizfocused.com>

HI All,

 

In the pastebin:   <http://pastebin.com/KyGXMTP0>
http://pastebin.com/KyGXMTP0 is a sip trace.  

 

The phone is on the lan and uses proxy: fs_test.bizfocused.com which is
10.10.40.189.  In the SIP messages you can see the domain name
"fs_test.bizfocused.com".

 

I have a directory that includes <domain name="fs_test.bizfocused.com"> and
a user for this subscription.

 

But the SIP response is "SIP/2.0 401 Unauthorized" and the message displayed
in FS_CLI is:

[WARNING] sofia_reg.c:2376 Can't find user [202 at 10.10.40.189]

You must define a domain called '10.10.40.189' in your directory and add a
user with the id="202" attribute

and you must configure your device to use the proper domain in its
authentication credentials.

 

Why isn't it using the credentials in the domain "fs_test.bizfocused.com"?

 

Thanks,

Sean

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From patrick at sunsus.net  Tue Jun 12 22:40:33 2012
From: patrick at sunsus.net (sunsus)
Date: Tue, 12 Jun 2012 11:40:33 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch 304 Unauthorized
In-Reply-To: <CAKV-jFwWLygHJVozz41=5rsjXMVuTrDsek98Rty2DL4zF1AC1w@mail.gmail.com>
References: <1339334863709-7579670.post@n2.nabble.com>
	<1339524483645-7579753.post@n2.nabble.com>
	<CAKV-jFwWLygHJVozz41=5rsjXMVuTrDsek98Rty2DL4zF1AC1w@mail.gmail.com>
Message-ID: <1339526433720-7579758.post@n2.nabble.com>


Non working sip trace
http://pastebin.freeswitch.org/19271 http://pastebin.freeswitch.org/19271 
XML Config
http://pastebin.freeswitch.org/19272 http://pastebin.freeswitch.org/19272 
Working sip trace
http://pastebin.freeswitch.org/19273 http://pastebin.freeswitch.org/19273 

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From toddb at toddbailey.net  Tue Jun 12 23:05:38 2012
From: toddb at toddbailey.net (toddb at toddbailey.net)
Date: Tue, 12 Jun 2012 12:05:38 -0700
Subject: [Freeswitch-users] Updated FS & Spa3102 setup how to
Message-ID: <20120612120538.33e327b490679d2282e332758c73b55b.ff4401fd01.wbe@email14.secureserver.net>

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From paul at cupis.co.uk  Tue Jun 12 23:12:57 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Tue, 12 Jun 2012 20:12:57 +0100
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <1339473995.42452.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
	<1339414395.62082.YahooMailNeo@web120105.mail.ne1.yahoo.com>
	<30DFAA60-ADB7-4B3A-A5FE-9085491FD83E@opencsta.org>
	<1339433685.46946.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<4FD6371D.1010307@cupis.co.uk>
	<1339473995.42452.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <4FD794B9.70003@cupis.co.uk>

First call:

FreeSWITCH 192.168.10.70 send invite to 140112971507247227 at 85.15.0.154
  offers: m=audio 26616 RTP/AVP 9 0 8 18 3 101 13
Cisco 85.15.0.154 replies with 18/SDP: m=audio 18218 RTP/AVP 0 13 101
Cisco 85.15.0.154 replies with 500 Internal Server Error

Second call:

FreeSWITCH 192.168.10.70 send invite to 140112971507247227 at 85.15.0.154
  offers: m=audio 25814 RTP/AVP 18 101 13
Cisco 85.15.0.154 replied with 488 Not Acceptable Media

Your Cisco is either misconfigured or is rejecting the INVITE/SDP from 
FreeSWITCH - you'll need to check the logs on the Cisco to see why.

You should try Stevens suggestion of verbose_sdp=true and do the same 
two tests, but it looks like your Cisco should be happy to set the call 
up as G711 (PCMU) or G729 if you can get it to accept the INVITE/SDP.

Regards,


From paul at cupis.co.uk  Tue Jun 12 23:19:10 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Tue, 12 Jun 2012 20:19:10 +0100
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch 304 Unauthorized
In-Reply-To: <1339526433720-7579758.post@n2.nabble.com>
References: <1339334863709-7579670.post@n2.nabble.com>
	<1339524483645-7579753.post@n2.nabble.com>
	<CAKV-jFwWLygHJVozz41=5rsjXMVuTrDsek98Rty2DL4zF1AC1w@mail.gmail.com>
	<1339526433720-7579758.post@n2.nabble.com>
Message-ID: <4FD7962E.9070401@cupis.co.uk>

On 12/06/12 19:40, sunsus wrote:
>
> Non working sip trace
> http://pastebin.freeswitch.org/19271 http://pastebin.freeswitch.org/19271
> XML Config
> http://pastebin.freeswitch.org/19272 http://pastebin.freeswitch.org/19272
> Working sip trace
> http://pastebin.freeswitch.org/19273 http://pastebin.freeswitch.org/19273

Try sending the call to 62.65.137.114 or to 62.65.137.113 instead of to 
sip.netvoip.ch and see if that works - it is does then the "issue" could 
be resolved by changes to the suppliers load balancer andor your DNS 
setup/cache.

Changing the XML file above is not going to help you at the moment - it 
looks like your XML dialplan is still sending the call to 
sip.netvoip.ch, not to the IP address:

Dialplan: sofia/internal/1003 at pbx.domain.tld Action 
bridge([leg_progress_timeout=15]sofia/gateway/netvoip-41440000026/0041440000069 at sip.netvoip.ch)

I'd possibly suggest that your bridge command above looks wrong as well 
- you are using '/gateway' AND '@sip.netvoip.ch'...

Regards,


From lloyd.aloysius at gmail.com  Tue Jun 12 23:23:51 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Tue, 12 Jun 2012 15:23:51 -0400
Subject: [Freeswitch-users] NAT Router Recommendation For Large no of IP
 Phones Behind the Router
In-Reply-To: <4FD7766D.8040208@puzzled.xs4all.nl>
References: <CAGvuWYaqsqO6N0y+5+pJ2zyA0UqS0iAFjQiXuaSxssH52GcdVQ@mail.gmail.com>
	<729B9245-CBCA-4F7F-9B71-9142E0ADA448@mgtech.com>
	<4FD7766D.8040208@puzzled.xs4all.nl>
Message-ID: <CAGvuWYZnw05oKVcAArT9HaeqBagBanaeiG5ngdwNP4Mv79fXBQ@mail.gmail.com>

I try with the pfsense 2.0. Specially BLF causing lots of problems.

====

In the past .. I try the following

1. Linksys WRT54GL + Tomato ... Good for a small office < 5 Extension
2. Monowall with PC Engine Alix


Does any one test with a Cisco ASA 5505?



On Tue, Jun 12, 2012 at 1:03 PM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 12-06-12 18:20, Mario G wrote:
> > I looked for a year to find something with no NAT problems, dual WAN,
> > and auto fallback with load/balancing without causing FS any hiccups. I
> > highly recommend Zyxel, I use a USG100 with SPI ALG turned ON (I read
> > some people hate it) but it has been bullet proof. And the best part is
> > there is practically nothing to tell FS about! And when one WAN (DSL)
> > goes down the SIP traffic goes out the other one, even switches back
> > automatically! Pair this up with FS and you'll never look back. Hope
> > this helps.
>
> Thanks for the info. Have you tried any Draytek Vigor routers like the
> 2920? I have no experience with them but was told they work fine in dual
> WAN setups.
>
> Regards,
> Patrick
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sdevoy at bizfocused.com  Tue Jun 12 23:57:49 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 12 Jun 2012 15:57:49 -0400
Subject: [Freeswitch-users] Updated FS & Spa3102 setup how to
In-Reply-To: <20120612120538.33e327b490679d2282e332758c73b55b.ff4401fd01.wbe@email14.secureserver.net>
References: <20120612120538.33e327b490679d2282e332758c73b55b.ff4401fd01.wbe@email14.secureserver.net>
Message-ID: <01b101cd48d5$a54f1630$efed4290$@bizfocused.com>

HI Todd,

 

I don?t have an SPA3102, but I use many SPA504Gs and PAP2s and RPT300s.  I think the SPA3102 is VERY much like a PAP2.

 

I need to understand where you are in the process.  Let?s focus on the SPA3102 and not so much on FS config first.

Do you have Admin Web Access to the SPA3102 VOICE configuration pages?

If so, have you configured Line 1 to connect to user you have defined in FS?

Is the SPA3102 on a local LAN with the FS server or does it use NAT?

Does it register with FS (ie get a Phone 1 light on)?

 

Let me know that much and I can start feeding you configuration information for the SPA3102 and for FS.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of toddb at toddbailey.net
Sent: Tuesday, June 12, 2012 3:06 PM
To: FreeSWITCH user group Help
Subject: [Freeswitch-users] Updated FS & Spa3102 setup how to

 

Hi all,

 

 

I'm using the latest FS build and I'm trying to get a SPA3102 to cooperate.

The latest postings on configuration for both components appear out of date.

 

can someone please provide a link to recent how to on getting the fs server to place a call to the 3102?

 

The guides I've used so far don't seem to work.  Perhaps a crucial step is missing, Or maybe I just need to key in a magic number to connect to the 3102.

 

Additionally,  is there any printed fs guides, or downloadable documentation.

pref written for an audience that doesn't know what they're doing?

 

thanks...

 

 

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From patrick at sunsus.net  Tue Jun 12 23:58:58 2012
From: patrick at sunsus.net (sunsus)
Date: Tue, 12 Jun 2012 12:58:58 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch 304 Unauthorized
In-Reply-To: <4FD7962E.9070401@cupis.co.uk>
References: <1339334863709-7579670.post@n2.nabble.com>
	<1339524483645-7579753.post@n2.nabble.com>
	<CAKV-jFwWLygHJVozz41=5rsjXMVuTrDsek98Rty2DL4zF1AC1w@mail.gmail.com>
	<1339526433720-7579758.post@n2.nabble.com>
	<4FD7962E.9070401@cupis.co.uk>
Message-ID: <1339531138888-7579764.post@n2.nabble.com>

Hello Paul 

Thanks for you answer, but now freeswitch is canceling the call: 
http://pastebin.freeswitch.org/19277 http://pastebin.freeswitch.org/19277 

regards

Patrick 

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From covici at ccs.covici.com  Wed Jun 13 00:40:53 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 12 Jun 2012 16:40:53 -0400
Subject: [Freeswitch-users] Updated FS & Spa3102 setup how to
In-Reply-To: <20120612120538.33e327b490679d2282e332758c73b55b.ff4401fd01.wbe@email14.secureserver.net>
References: <20120612120538.33e327b490679d2282e332758c73b55b.ff4401fd01.wbe@email14.secureserver.net>
Message-ID: <22370.1339533653@ccs.covici.com>

One thing -- at least with the default configs on the 3102, in your
bridge statement, you have to have :5061 after the ip address of the
3102 -- at least that is where I got stuck.

<toddb at toddbailey.net> wrote:

>    Hi all,
> 
> 
> 
> 
> 
>    I'm using the latest FS build and I'm trying to get a SPA3102 to
>    cooperate.
> 
>    The latest postings on configuration for both components appear out of
>    date.
> 
> 
> 
>    can someone please provide a link to recent how to on getting the fs
>    server to place a call to the 3102?
> 
> 
> 
>    The guides I've used so far don't seem to work.  Perhaps a crucial step
>    is missing, Or maybe I just need to key in a magic number to connect to
>    the 3102.
> 
> 
> 
>    Additionally,  is there any printed fs guides, or downloadable
>    documentation.
> 
>    pref written for an audience that doesn't know what they're doing?
> 
> 
> 
>    thanks...
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From chris at gonumina.com  Wed Jun 13 01:02:25 2012
From: chris at gonumina.com (Chris Ferreira)
Date: Tue, 12 Jun 2012 17:02:25 -0400
Subject: [Freeswitch-users] Domain name lost in SIP registration (w/ SIP
	TRACE)
In-Reply-To: <011e01cd48ca$99953aa0$ccbfafe0$@bizfocused.com>
References: <011e01cd48ca$99953aa0$ccbfafe0$@bizfocused.com>
Message-ID: <-5702489430043259544@unknownmsgid>

Verify that your TURN server settings are correct on the phone.

___________________
Mobile Reply

On Jun 12, 2012, at 2:41 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

HI All,



In the pastebin:  http://pastebin.com/KyGXMTP0 is a sip trace.



The phone is on the lan and uses proxy: fs_test.bizfocused.com which is
10.10.40.189.  In the SIP messages you can see the domain name ?
fs_test.bizfocused.com?.



I have a directory that includes <domain name="fs_test.bizfocused.com"> and
a user for this subscription.



But the SIP response is ?SIP/2.0 401 Unauthorized? and the message
displayed in FS_CLI is:

[WARNING] sofia_reg.c:2376 Can't find user [202 at 10.10.40.189]

You must define a domain called '10.10.40.189' in your directory and add a
user with the id="202" attribute

and you must configure your device to use the proper domain in its
authentication credentials.



Why isn?t it using the credentials in the domain "fs_test.bizfocused.com"?



Thanks,

Sean

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From chris at gonumina.com  Wed Jun 13 01:02:54 2012
From: chris at gonumina.com (Chris Ferreira)
Date: Tue, 12 Jun 2012 17:02:54 -0400
Subject: [Freeswitch-users] Domain name lost in SIP registration (w/ SIP
	TRACE)
In-Reply-To: <011e01cd48ca$99953aa0$ccbfafe0$@bizfocused.com>
References: <011e01cd48ca$99953aa0$ccbfafe0$@bizfocused.com>
Message-ID: <6856752153193153969@unknownmsgid>

Oops, I mean STUN Server.

___________________
Mobile Reply

On Jun 12, 2012, at 2:41 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

HI All,



In the pastebin:  http://pastebin.com/KyGXMTP0 is a sip trace.



The phone is on the lan and uses proxy: fs_test.bizfocused.com which is
10.10.40.189.  In the SIP messages you can see the domain name ?
fs_test.bizfocused.com?.



I have a directory that includes <domain name="fs_test.bizfocused.com"> and
a user for this subscription.



But the SIP response is ?SIP/2.0 401 Unauthorized? and the message
displayed in FS_CLI is:

[WARNING] sofia_reg.c:2376 Can't find user [202 at 10.10.40.189]

You must define a domain called '10.10.40.189' in your directory and add a
user with the id="202" attribute

and you must configure your device to use the proper domain in its
authentication credentials.



Why isn?t it using the credentials in the domain "fs_test.bizfocused.com"?



Thanks,

Sean

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From kris at kriskinc.com  Wed Jun 13 01:10:02 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 12 Jun 2012 17:10:02 -0400
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <1339473995.42452.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
	<1339414395.62082.YahooMailNeo@web120105.mail.ne1.yahoo.com>
	<30DFAA60-ADB7-4B3A-A5FE-9085491FD83E@opencsta.org>
	<1339433685.46946.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<4FD6371D.1010307@cupis.co.uk>
	<1339473995.42452.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <CAKDfjgfbfTgfb0t0d6oVV4trT+zH-D=eS-FuzcomDqKzp_7g0g@mail.gmail.com>

Your Cisco gateway doesn't support G729.  Check your codec-list
configuration in IOS.

On Tue, Jun 12, 2012 at 12:06 AM, Samira Mh <saami_mh at ymail.com> wrote:
> thansk for your reply,
> it is kind of you to help me..
> please let me paste myconfigurations files here;
> 1-the configuration ?file
> /usr/local/freeswitch/conf/dialplan/default/001_luacallduration.xml ?is like
> this:
>
> <include>
> ? <extension name="mainmenuvodsl">
>
> ? ? ? ? <condition field="destination_number"
> expression="^(00|\+)?(\d{5}.*)$" break="never">
> ? ? ??? ? ? ? ? ? ?<action application="odbc_query" data="select cash as
> cashvalue from accounts where contractid like '${nibble_account}';"/>
> ? ? ? ? ? ? ? ? <action application="log" data="INFO The value of cashvalue
> is ${cashvalue}" />
> ?? ? ? ? ? ? ? ? <action application="lua" data="checkcash.lua ${cashvalue}"
> />
> ? ? ? ? ? ? ? ? <action application="log" data="INFO The value of
> nibble_account is ?${nibble_account}"/>
> ? ? ? ? ? ? ? ? <action application="log" data="INFO The value of
> nibble_rate [before] is ?${nibble_rate}"/>
> ? ? ? ? ? ? ? ? <!-- RateList Context -->
> ? ? ? ? ? ? ? ? <action application="lua" data="checkzeroplus.lua
> ${destination_number:0:2} ${destination_number:0:1}" />
> ? ? ? ? ? ? ? ? <action application="execute_extension"
> data="${destination_number} XML ratelist"/>
> ? ? ? ? ? ? ? ? <action application="log" data="INFO The value of
> nibble_rate [after] is ${nibble_rate}"/>
> ? ? ? ? ? ? ? ? ?<!-- Check Nibble_rate -->
> ? ? ? ? ? ? ? ? <action application="lua" data="checknibblerate.lua
> ${nibble_rate}" />
> ? ? ? ? ? ? ? ? <action application="set"
> data="divvalue=${expr(floor((${cashvalue}/${nibble_rate}))}" />
> ? ? ? ? ? ? ? ? <action application="set"
> data="modvalue=${expr(mod(${cashvalue},${nibble_rate}))}" />
> ? ? ? ? ? ? ? ? <action application="lua" data="checktime.lua ${divvalue}
> ${modvalue}" />
> ? ? ? ? ? ? ? ?? ? ?<!-- ?Check ZeroZero , Plus ?-->
> ? ? ? ? ? ? ?? ?<action application="lua" data="checkzeroplus.lua
> ${destination_number:0:2} ${destination_number:0:1}" />
> ? ? ? ? ? ? ? ? <!-- Making Calls ? -->
> ? ? ? ? ? ? ? ? <action application="odbc_query" data="select callerid ?as
> effective_caller_id_number from accounts where contractid like
> '${nibble_account}';"/>
> ? ? ? ? ? ? ? ? <action application="log" data="INFO ?callerid for Outbound
> calls ${effective_caller_id_number}"/>
> ? ? ? ? ? ? ? ? <!-- <action application="set"
> data="ignore_early_media=true"/>
> ? ? ? ? ? ? ? ? <action application="answer"/> ?-->
> ? ? ? ? ? ? ? ? <action application="enable_heartbeat"/>
>
> <!-- <param name="disable-transcoding" value="true"/> -->
> ? ? ? ? <!-- ? ?<action application="export"
> data="nolocal:absolute_codec_string=G729,PCMU"/> -->
> <!-- ?<action application="set" data="bridge_early_media=true"/> ?-->
> ? ? ? ? <!-- <action application="set" data="proxy_media=true"/> -->
> ? ? ? ? ? ? ? ? <action application="bridge"
> data="sofia/gateway/cisco/140112${destination_number}"/>
> ? ? ? ? ? ? ? ? <!-- <action application="bridge"
> data="sofia/gateway/mainasterisk/${destination_number}"/> ?-->
> <!-- ?<action application="bridge"
> data="sofia/gateway/test/${destination_number}"/> ?-->
> ? ? ? ? </condition>
>
> ?</extension>
> </include>
>
> 2-yes, i have enabled ?"inbound-late-negotiation"?in the
> (/usr/local/freeswitch/conf/sip_profiles/internal.xml) as follow:
> ?<param name="inbound-late-negotiation" value="true"/>
>
>
> 3-the issue of sofia status:
> ?external::cisco ? ? ? gateway ? ? ? ? ? ? sip:register:false at 85.15.0.154
> ? ?NOREG
>
>
> 4-also , the configuration file for codecs are as follow
> :/usr/local/freeswitch/conf/vars.xml
>
> <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G729,PCMU,PCMA,G7221 at 32000h,G7221 at 16000h,G722,GSM"/>
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>
>
> 5- the mod_g729 was loaded
>
> 6-i have enabled the siptrace:
> ?sofia profile external siptrace on:
> the siptrace outpout as follow:
>
> send 1042 bytes to udp/[85.15.0.154]:5060 at 03:53:07.448136:
> ? ?------------------------------------------------------------------------
> ? ?INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
> ? ?Max-Forwards: 69
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
> ? ?To: <sip:140112971507247227 at 85.15.0.154>
> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
> ? ?CSeq: 29400529 INVITE
> ? ?Contact: <sip:gw+cisco at 192.168.10.70:5080;transport=udp;gw=cisco>
> ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ? ?Supported: timer, precondition, path, replaces
> ? ?Allow-Events: talk, hold, refer
> ? ?Content-Type: application/sdp
> ? ?Content-Disposition: session
> ? ?Content-Length: 234
> ? ?X-FS-Support: update_display,send_info
> ? ?Remote-Party-ID: "1000"
> <sip:1000 at 85.15.0.154>;party=calling;screen=yes;privacy=off
>
> ? ?v=0
> ? ?o=FreeSWITCH 1339446571 1339446572 IN IP4 192.168.10.70
> ? ?s=FreeSWITCH
> ? ?c=IN IP4 192.168.10.70
> ? ?t=0 0
> ? ?m=audio 26616 RTP/AVP 9 0 8 18 3 101 13
> ? ?a=fmtp:18 annexb=yes
> ? ?a=rtpmap:101 telephone-event/8000
> ? ?a=fmtp:101 0-16
> ? ?a=ptime:20
> ? ?------------------------------------------------------------------------
> recv 410 bytes from udp/[85.15.0.154]:5060 at 03:53:07.463921:
> ? ?------------------------------------------------------------------------
> ? ?SIP/2.0 100 Trying
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
> ? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
> ? ?Server: Cisco-SIPGateway/IOS-12.x
> ? ?CSeq: 29400529 INVITE
> ? ?Allow-Events: telephone-event
> ? ?Content-Length: 0
>
> ? ?------------------------------------------------------------------------
> recv 927 bytes from udp/[85.15.0.154]:5060 at 03:53:11.004804:
> ? ?------------------------------------------------------------------------
> ? ?SIP/2.0 183 Session Progress
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
> ? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
> ? ?Server: Cisco-SIPGateway/IOS-12.x
> ? ?CSeq: 29400529 INVITE
> ? ?Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
> NOTIFY, INFO, UPDATE, REGISTER
> ? ?Allow-Events: telephone-event
> ? ?Contact: <sip:140112971507247227 at 85.15.0.154:5060>
> ? ?Content-Disposition: session;handling=required
> ? ?Content-Type: application/sdp
> ? ?Content-Length: 268
>
> ? ?v=0
> ? ?o=CiscoSystemsSIP-GW-UserAgent 7252 5649 IN IP4 85.15.0.154
> ? ?s=SIP Call
> ? ?c=IN IP4 85.15.0.154
> ? ?t=0 0
> ? ?m=audio 18218 RTP/AVP 0 13 101
> ? ?c=IN IP4 85.15.0.154
> ? ?a=rtpmap:0 PCMU/8000
> ? ?a=rtpmap:13 CN/8000
> ? ?a=rtpmap:101 telephone-event/8000
> ? ?a=fmtp:101 0-15
> ? ?a=ptime:20
> ? ?------------------------------------------------------------------------
> recv 425 bytes from udp/[85.15.0.154]:5060 at 03:53:11.005144:
> ? ?------------------------------------------------------------------------
> ? ?SIP/2.0 500 Internal Server Error
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
> ? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
> ? ?Server: Cisco-SIPGateway/IOS-12.x
> ? ?CSeq: 29400529 INVITE
> ? ?Allow-Events: telephone-event
> ? ?Content-Length: 0
>
> ? ?------------------------------------------------------------------------
> send 350 bytes to udp/[85.15.0.154]:5060 at 03:53:11.005333:
> ? ?------------------------------------------------------------------------
> ? ?ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
> ? ?Max-Forwards: 69
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
> ? ?CSeq: 29400529 ACK
> ? ?Content-Length: 0
>
>
> ------------------------------------------------------------------------------------------------------------------
> when change the configuration file the below:
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>
>
> the siptrace is like this:
>
> send 1034 bytes to udp/[85.15.0.154]:5060 at 04:01:16.202342:
> ? ?------------------------------------------------------------------------
> ? ?INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
> ? ?Max-Forwards: 69
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
> ? ?To: <sip:140112971507247227 at 85.15.0.154>
> ? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
> ? ?CSeq: 29400774 INVITE
> ? ?Contact: <sip:gw+cisco at 192.168.10.70:5080;transport=udp;gw=cisco>
> ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ? ?Supported: timer, precondition, path, replaces
> ? ?Allow-Events: talk, hold, refer
> ? ?Content-Type: application/sdp
> ? ?Content-Disposition: session
> ? ?Content-Length: 226
> ? ?X-FS-Support: update_display,send_info
> ? ?Remote-Party-ID: "1000"
> <sip:1000 at 85.15.0.154>;party=calling;screen=yes;privacy=off
>
> ? ?v=0
> ? ?o=FreeSWITCH 1339447862 1339447863 IN IP4 192.168.10.70
> ? ?s=FreeSWITCH
> ? ?c=IN IP4 192.168.10.70
> ? ?t=0 0
> ? ?m=audio 25814 RTP/AVP 18 101 13
> ? ?a=fmtp:18 annexb=yes
> ? ?a=rtpmap:101 telephone-event/8000
> ? ?a=fmtp:101 0-16
> ? ?a=ptime:20
> ? ?------------------------------------------------------------------------
> recv 423 bytes from udp/[85.15.0.154]:5060 at 04:01:16.208118:
> ? ?------------------------------------------------------------------------
> ? ?SIP/2.0 488 Not Acceptable Media
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=457FC664-6A6
> ? ?Date: Tue, 12 Jun 2012 04:01:24 GMT
> ? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
> ? ?Server: Cisco-SIPGateway/IOS-12.x
> ? ?CSeq: 29400774 INVITE
> ? ?Allow-Events: telephone-event
> ? ?Content-Length: 0
>
> ? ?------------------------------------------------------------------------
> send 349 bytes to udp/[85.15.0.154]:5060 at 04:01:16.208201:
> ? ?------------------------------------------------------------------------
> ? ?ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
> ? ?Max-Forwards: 69
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=457FC664-6A6
> ? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
> ? ?CSeq: 29400774 ACK
> ? ?Content-Length: 0
>
>
>
> plz help,thanks so much
>
>
> ________________________________
> From: Paul Cupis <paul at cupis.co.uk>
>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, June 11, 2012 10:51 PM
>
> Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
>
> On 11/06/12 17:54, Samira Mh wrote:
>> i want to bridge call using my VOIPgateway so that making calls to
>> another countries..
>> but the carrier only support G729 codec and the FS send G722 (set in
>> vars.xml) to myVoipGateway that is set as an gateway in
>> /usr/local/freeswitch/sip-profile/external/
>> and when FS send media to Gateway(using bridge application) the error
>> occure:unacceptable media,then check VOIPGW and find out the only codec
>> that
>> can be pass through VOIPgw is G729, but FS only send G711,G722,... not
>> G729
>
> Can you provide a SIP or FreeSWITCH trace of a call, please?
>
> Do you have the following enabled in your SIP profile?
>
> ? <param name="inbound-late-negotiation" value="true"/>
>
> Do you have mod_g729 loaded and codec G729 enabled in your vars.xml?
>
> Regards,
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From kris at kriskinc.com  Wed Jun 13 01:12:29 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 12 Jun 2012 17:12:29 -0400
Subject: [Freeswitch-users] Moving up to better PC - Multi-tenant died
In-Reply-To: <05a501cd4856$41711610$c4534230$@bizfocused.com>
References: <05a501cd4856$41711610$c4534230$@bizfocused.com>
Message-ID: <CAKDfjgco_8K_mLW7Ekzm+s2Ut0L_74RBEVYhUEEaLT0hT5XHBw@mail.gmail.com>

What is your hostname?  How are the various domain values set in the
FreeSWITCH Sofia profiles?

On Tue, Jun 12, 2012 at 12:45 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> HI,
>
>
>
> I have a nice working multi-tenant configuration working on my development
> box ? P4 3.0Ghz 4GB RAM Raid 1 Centos 5.7 Freeswitch (make current).
>
>
>
> I built a nice clean Centos 5.7 install on a Dell Power Edge 2850 Dual Xeon
> 3.0 Ghs 4GB RAM and current Freeswitch.
>
> I saved the entire ?/usr/local/freeswitch/conf? tree from new server.
>
> ??????????????? I copied the entire ?/usr/local/freeswitch/conf? tree from
> devel box to new server.
>
> ??????????????? I shutdown development box.
>
> ??????????????? I set new server to same IP as old Devel boix.
>
> ??????????????? Rebooted.
>
>
>
> I have a couple of issues:
>
> ??????????????? 1. Freeswitch does not startup at boot.? I tried to follow
> the wiki.? I have a freeswitch file in /;etc/init.d but no startup!
>
>
>
> ??????????????? 2. Only my primary domain (tenant) users can login.? All of
> my secondary domain phones have proxy=nnn at fs_mbri.bizfocused.com but the
> errors on FS_CLI say it can?t find nnn at fs_bfis.bizfocused.com (MY PRIMARY
> DOMAIN).? It is though freeswitch thinks these registrations are coming to
> the primary domain even though they are going to a secondry with the same IP
> address.
>
>
>
> So, how do I get FS to start on boot up in CENTOS 5.7?? I swear I tried
> what?s in the wiki.? Or maybe I should say how do I tell wait is failing?
>
>
>
> What am I missing in the multi-tenant domain definition (that is not in the
> /conf folder)?
>
>
>
> Thanks in advance.
>
>
>
> Sean
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From paul at cupis.co.uk  Wed Jun 13 01:29:18 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Tue, 12 Jun 2012 22:29:18 +0100
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch 304 Unauthorized
In-Reply-To: <1339531138888-7579764.post@n2.nabble.com>
References: <1339334863709-7579670.post@n2.nabble.com>
	<1339524483645-7579753.post@n2.nabble.com>
	<CAKV-jFwWLygHJVozz41=5rsjXMVuTrDsek98Rty2DL4zF1AC1w@mail.gmail.com>
	<1339526433720-7579758.post@n2.nabble.com>
	<4FD7962E.9070401@cupis.co.uk>
	<1339531138888-7579764.post@n2.nabble.com>
Message-ID: <4FD7B4AE.6030805@cupis.co.uk>

On 12/06/12 20:58, sunsus wrote:
> Hello Paul
>
> Thanks for you answer, but now freeswitch is canceling the call:
> http://pastebin.freeswitch.org/19277 http://pastebin.freeswitch.org/19277

Looks like 1003 at pbx is hanging up - can you adjust your logging to 
include the SIP signalling between (I presume) the handset and 
FreeSWITCH (the "internal" SIP profile)? We seem to be seeing 
FreeSWITCH<>Provider but not Handset<>FreeSWITCH?

2012-06-12 21:50:47.393989 [NOTICE] sofia.c:6618 Hangup 
sofia/internal/1003 at pbx.domain.tld [CS_EXECUTE] [ORIGINATOR_CANCEL]

Regards,


From alexanderjp at thinksimplicity.net  Wed Jun 13 01:41:35 2012
From: alexanderjp at thinksimplicity.net (Perovich Alexander)
Date: Tue, 12 Jun 2012 17:41:35 -0400
Subject: [Freeswitch-users] Caller ID on Polycom phones
Message-ID: <BED10B57-D8DA-470F-A61C-D8BD5CC831B6@thinksimplicity.net>

Hello,

I am having a strange issue with Polycom phones and caller ID.

When the polycom receives a call it shows the caller id name fine but the caller id number is 1234 at domain, however when I pickup the phone it shows the caller id number as 1234 without the domain name.  

This happens on both extension to extension calls as well as external calls coming in.

Any help would be greatly appreciated.



-------
Alexander
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From fvillarroel at yahoo.com  Wed Jun 13 04:00:39 2012
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Tue, 12 Jun 2012 17:00:39 -0700 (PDT)
Subject: [Freeswitch-users] lcr sql variable
Message-ID: <1339545639.80187.YahooMailClassic@web160306.mail.bf1.yahoo.com>

Hi all.

I need setup some variable like foo to my cdr from lcr :

<param name="export_fields" value="tarifa"/>


<param name="custom_sql" value="select a.tarifa as tarifa ......


Returning Dialstring [lcr_carrier=carrier1,lcr_rate=0.15000,nibble_account=1004,nibble_rate=0.15000,lowbal_amt=0,nobal_amt=0,tarifa=1]user/1006

My dialplan:

<extension name="Outbound to PSTN 11 Digits">
                <condition field="destination_number" expression="^(10[01][0-9])$">
                        <action application="set" data="hangup_after_bridge=false"/>
                        <action application="set" data="continue_on_fail=true"/>
                        <action application="lcr" data="$1 pg_prefix2"/>

                        <action application="set"  data="tarifa=${tarifa}"/>
                        <action application="bridge" data="${lcr_auto_route}"/>
                </condition>
        </extension>

But the log show :

mod_dptools.c:1294 sofia/internal/1004 at 192.168.1.108 SET [tarifa]=[UNDEF]


So how i can get the value of the variable "tarifa" and set for my CDR?

Regards.



From saami_mh at ymail.com  Wed Jun 13 07:40:40 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Tue, 12 Jun 2012 20:40:40 -0700 (PDT)
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <CAKDfjgfbfTgfb0t0d6oVV4trT+zH-D=eS-FuzcomDqKzp_7g0g@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAHKqMb76wmvwzcNpCT5aHNShK1wZepSv-24fBRmCTrfi+yDBMg@mail.gmail.com>
	<1339414395.62082.YahooMailNeo@web120105.mail.ne1.yahoo.com>
	<30DFAA60-ADB7-4B3A-A5FE-9085491FD83E@opencsta.org>
	<1339433685.46946.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<4FD6371D.1010307@cupis.co.uk>
	<1339473995.42452.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<CAKDfjgfbfTgfb0t0d6oVV4trT+zH-D=eS-FuzcomDqKzp_7g0g@mail.gmail.com>
Message-ID: <1339558840.7099.YahooMailNeo@web120106.mail.ne1.yahoo.com>

bu i have set the codec in dialpeer of VOIP gateway



________________________________
 From: Kristian Kielhofner <kris at kriskinc.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Wednesday, June 13, 2012 1:40 AM
Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
 
Your Cisco gateway doesn't support G729.? Check your codec-list
configuration in IOS.

On Tue, Jun 12, 2012 at 12:06 AM, Samira Mh <saami_mh at ymail.com> wrote:
> thansk for your reply,
> it is kind of you to help me..
> please let me paste myconfigurations files here;
> 1-the configuration ?file
> /usr/local/freeswitch/conf/dialplan/default/001_luacallduration.xml ?is like
> this:
>
> <include>
> ? <extension name="mainmenuvodsl">
>
> ? ? ? ? <condition field="destination_number"
> expression="^(00|\+)?(\d{5}.*)$" break="never">
> ? ? ??? ? ? ? ? ? ?<action application="odbc_query" data="select cash as
> cashvalue from accounts where contractid like '${nibble_account}';"/>
> ? ? ? ? ? ? ? ? <action application="log" data="INFO The value of cashvalue
> is ${cashvalue}" />
> ?? ? ? ? ? ? ? ? <action application="lua" data="checkcash.lua ${cashvalue}"
> />
> ? ? ? ? ? ? ? ? <action application="log" data="INFO The value of
> nibble_account is ?${nibble_account}"/>
> ? ? ? ? ? ? ? ? <action application="log" data="INFO The value of
> nibble_rate [before] is ?${nibble_rate}"/>
> ? ? ? ? ? ? ? ? <!-- RateList Context -->
> ? ? ? ? ? ? ? ? <action application="lua" data="checkzeroplus.lua
> ${destination_number:0:2} ${destination_number:0:1}" />
> ? ? ? ? ? ? ? ? <action application="execute_extension"
> data="${destination_number} XML ratelist"/>
> ? ? ? ? ? ? ? ? <action application="log" data="INFO The value of
> nibble_rate [after] is ${nibble_rate}"/>
> ? ? ? ? ? ? ? ? ?<!-- Check Nibble_rate -->
> ? ? ? ? ? ? ? ? <action application="lua" data="checknibblerate.lua
> ${nibble_rate}" />
> ? ? ? ? ? ? ? ? <action application="set"
> data="divvalue=${expr(floor((${cashvalue}/${nibble_rate}))}" />
> ? ? ? ? ? ? ? ? <action application="set"
> data="modvalue=${expr(mod(${cashvalue},${nibble_rate}))}" />
> ? ? ? ? ? ? ? ? <action application="lua" data="checktime.lua ${divvalue}
> ${modvalue}" />
> ? ? ? ? ? ? ? ?? ? ?<!-- ?Check ZeroZero , Plus ?-->
> ? ? ? ? ? ? ?? ?<action application="lua" data="checkzeroplus.lua
> ${destination_number:0:2} ${destination_number:0:1}" />
> ? ? ? ? ? ? ? ? <!-- Making Calls ? -->
> ? ? ? ? ? ? ? ? <action application="odbc_query" data="select callerid ?as
> effective_caller_id_number from accounts where contractid like
> '${nibble_account}';"/>
> ? ? ? ? ? ? ? ? <action application="log" data="INFO ?callerid for Outbound
> calls ${effective_caller_id_number}"/>
> ? ? ? ? ? ? ? ? <!-- <action application="set"
> data="ignore_early_media=true"/>
> ? ? ? ? ? ? ? ? <action application="answer"/> ?-->
> ? ? ? ? ? ? ? ? <action application="enable_heartbeat"/>
>
> <!-- <param name="disable-transcoding" value="true"/> -->
> ? ? ? ? <!-- ? ?<action application="export"
> data="nolocal:absolute_codec_string=G729,PCMU"/> -->
> <!-- ?<action application="set" data="bridge_early_media=true"/> ?-->
> ? ? ? ? <!-- <action application="set" data="proxy_media=true"/> -->
> ? ? ? ? ? ? ? ? <action application="bridge"
> data="sofia/gateway/cisco/140112${destination_number}"/>
> ? ? ? ? ? ? ? ? <!-- <action application="bridge"
> data="sofia/gateway/mainasterisk/${destination_number}"/> ?-->
> <!-- ?<action application="bridge"
> data="sofia/gateway/test/${destination_number}"/> ?-->
> ? ? ? ? </condition>
>
> ?</extension>
> </include>
>
> 2-yes, i have enabled ?"inbound-late-negotiation"?in the
> (/usr/local/freeswitch/conf/sip_profiles/internal.xml) as follow:
> ?<param name="inbound-late-negotiation" value="true"/>
>
>
> 3-the issue of sofia status:
> ?external::cisco ? ? ? gateway ? ? ? ? ? ? sip:register:false at 85.15.0.154
> ? ?NOREG
>
>
> 4-also , the configuration file for codecs are as follow
> :/usr/local/freeswitch/conf/vars.xml
>
> <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G729,PCMU,PCMA,G7221 at 32000h,G7221 at 16000h,G722,GSM"/>
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>
>
> 5- the mod_g729 was loaded
>
> 6-i have enabled the siptrace:
> ?sofia profile external siptrace on:
> the siptrace outpout as follow:
>
> send 1042 bytes to udp/[85.15.0.154]:5060 at 03:53:07.448136:
> ? ?------------------------------------------------------------------------
> ? ?INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
> ? ?Max-Forwards: 69
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
> ? ?To: <sip:140112971507247227 at 85.15.0.154>
> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
> ? ?CSeq: 29400529 INVITE
> ? ?Contact: <sip:gw+cisco at 192.168.10.70:5080;transport=udp;gw=cisco>
> ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ? ?Supported: timer, precondition, path, replaces
> ? ?Allow-Events: talk, hold, refer
> ? ?Content-Type: application/sdp
> ? ?Content-Disposition: session
> ? ?Content-Length: 234
> ? ?X-FS-Support: update_display,send_info
> ? ?Remote-Party-ID: "1000"
> <sip:1000 at 85.15.0.154>;party=calling;screen=yes;privacy=off
>
> ? ?v=0
> ? ?o=FreeSWITCH 1339446571 1339446572 IN IP4 192.168.10.70
> ? ?s=FreeSWITCH
> ? ?c=IN IP4 192.168.10.70
> ? ?t=0 0
> ? ?m=audio 26616 RTP/AVP 9 0 8 18 3 101 13
> ? ?a=fmtp:18 annexb=yes
> ? ?a=rtpmap:101 telephone-event/8000
> ? ?a=fmtp:101 0-16
> ? ?a=ptime:20
> ? ?------------------------------------------------------------------------
> recv 410 bytes from udp/[85.15.0.154]:5060 at 03:53:07.463921:
> ? ?------------------------------------------------------------------------
> ? ?SIP/2.0 100 Trying
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
> ? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
> ? ?Server: Cisco-SIPGateway/IOS-12.x
> ? ?CSeq: 29400529 INVITE
> ? ?Allow-Events: telephone-event
> ? ?Content-Length: 0
>
> ? ?------------------------------------------------------------------------
> recv 927 bytes from udp/[85.15.0.154]:5060 at 03:53:11.004804:
> ? ?------------------------------------------------------------------------
> ? ?SIP/2.0 183 Session Progress
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
> ? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
> ? ?Server: Cisco-SIPGateway/IOS-12.x
> ? ?CSeq: 29400529 INVITE
> ? ?Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
> NOTIFY, INFO, UPDATE, REGISTER
> ? ?Allow-Events: telephone-event
> ? ?Contact: <sip:140112971507247227 at 85.15.0.154:5060>
> ? ?Content-Disposition: session;handling=required
> ? ?Content-Type: application/sdp
> ? ?Content-Length: 268
>
> ? ?v=0
> ? ?o=CiscoSystemsSIP-GW-UserAgent 7252 5649 IN IP4 85.15.0.154
> ? ?s=SIP Call
> ? ?c=IN IP4 85.15.0.154
> ? ?t=0 0
> ? ?m=audio 18218 RTP/AVP 0 13 101
> ? ?c=IN IP4 85.15.0.154
> ? ?a=rtpmap:0 PCMU/8000
> ? ?a=rtpmap:13 CN/8000
> ? ?a=rtpmap:101 telephone-event/8000
> ? ?a=fmtp:101 0-15
> ? ?a=ptime:20
> ? ?------------------------------------------------------------------------
> recv 425 bytes from udp/[85.15.0.154]:5060 at 03:53:11.005144:
> ? ?------------------------------------------------------------------------
> ? ?SIP/2.0 500 Internal Server Error
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
> ? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
> ? ?Server: Cisco-SIPGateway/IOS-12.x
> ? ?CSeq: 29400529 INVITE
> ? ?Allow-Events: telephone-event
> ? ?Content-Length: 0
>
> ? ?------------------------------------------------------------------------
> send 350 bytes to udp/[85.15.0.154]:5060 at 03:53:11.005333:
> ? ?------------------------------------------------------------------------
> ? ?ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
> ? ?Max-Forwards: 69
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
> ? ?CSeq: 29400529 ACK
> ? ?Content-Length: 0
>
>
> ------------------------------------------------------------------------------------------------------------------
> when change the configuration file the below:
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>
>
> the siptrace is like this:
>
> send 1034 bytes to udp/[85.15.0.154]:5060 at 04:01:16.202342:
> ? ?------------------------------------------------------------------------
> ? ?INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
> ? ?Max-Forwards: 69
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
> ? ?To: <sip:140112971507247227 at 85.15.0.154>
> ? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
> ? ?CSeq: 29400774 INVITE
> ? ?Contact: <sip:gw+cisco at 192.168.10.70:5080;transport=udp;gw=cisco>
> ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ? ?Supported: timer, precondition, path, replaces
> ? ?Allow-Events: talk, hold, refer
> ? ?Content-Type: application/sdp
> ? ?Content-Disposition: session
> ? ?Content-Length: 226
> ? ?X-FS-Support: update_display,send_info
> ? ?Remote-Party-ID: "1000"
> <sip:1000 at 85.15.0.154>;party=calling;screen=yes;privacy=off
>
> ? ?v=0
> ? ?o=FreeSWITCH 1339447862 1339447863 IN IP4 192.168.10.70
> ? ?s=FreeSWITCH
> ? ?c=IN IP4 192.168.10.70
> ? ?t=0 0
> ? ?m=audio 25814 RTP/AVP 18 101 13
> ? ?a=fmtp:18 annexb=yes
> ? ?a=rtpmap:101 telephone-event/8000
> ? ?a=fmtp:101 0-16
> ? ?a=ptime:20
> ? ?------------------------------------------------------------------------
> recv 423 bytes from udp/[85.15.0.154]:5060 at 04:01:16.208118:
> ? ?------------------------------------------------------------------------
> ? ?SIP/2.0 488 Not Acceptable Media
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=457FC664-6A6
> ? ?Date: Tue, 12 Jun 2012 04:01:24 GMT
> ? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
> ? ?Server: Cisco-SIPGateway/IOS-12.x
> ? ?CSeq: 29400774 INVITE
> ? ?Allow-Events: telephone-event
> ? ?Content-Length: 0
>
> ? ?------------------------------------------------------------------------
> send 349 bytes to udp/[85.15.0.154]:5060 at 04:01:16.208201:
> ? ?------------------------------------------------------------------------
> ? ?ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
> ? ?Max-Forwards: 69
> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=457FC664-6A6
> ? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
> ? ?CSeq: 29400774 ACK
> ? ?Content-Length: 0
>
>
>
> plz help,thanks so much
>
>
> ________________________________
> From: Paul Cupis <paul at cupis.co.uk>
>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, June 11, 2012 10:51 PM
>
> Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
>
> On 11/06/12 17:54, Samira Mh wrote:
>> i want to bridge call using my VOIPgateway so that making calls to
>> another countries..
>> but the carrier only support G729 codec and the FS send G722 (set in
>> vars.xml) to myVoipGateway that is set as an gateway in
>> /usr/local/freeswitch/sip-profile/external/
>> and when FS send media to Gateway(using bridge application) the error
>> occure:unacceptable media,then check VOIPGW and find out the only codec
>> that
>> can be pass through VOIPgw is G729, but FS only send G711,G722,... not
>> G729
>
> Can you provide a SIP or FreeSWITCH trace of a call, please?
>
> Do you have the following enabled in your SIP profile?
>
> ? <param name="inbound-late-negotiation" value="true"/>
>
> Do you have mod_g729 loaded and codec G729 enabled in your vars.xml?
>
> Regards,
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing 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 odermann at googlemail.com  Wed Jun 13 11:37:13 2012
From: odermann at googlemail.com (Dennis)
Date: Wed, 13 Jun 2012 09:37:13 +0200
Subject: [Freeswitch-users] Missing event for "180 Ringing"?
In-Reply-To: <CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
Message-ID: <CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>

yes, we register for all events... it's in jira now, thanks for the
hint to open a bug there.

kind regards
dennis


From alex at digitalmail.com  Wed Jun 13 14:05:37 2012
From: alex at digitalmail.com (Alex Lake)
Date: Wed, 13 Jun 2012 11:05:37 +0100
Subject: [Freeswitch-users] Conditions
In-Reply-To: <CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
Message-ID: <4FD865F1.3080600@digitalmail.com>

I've been struggling a bit with conditions. In particular the "break" 
attribute, which I couldn't really find any documentation on.

I think it would be nice to put some examples in the wiki that translate 
from the usual way of doing things (perhaps using PHP or JavaScript 
notation) to the dialplan format.

For example, I have a requirement that might be expressed like this:

<context name="dp0095">
<extension name="Outbound_call">
     if (${default_ani_prefix} == "" and sip_auth_username == "") {
<action application="log" data="WARNING default_ani_prefix and 
sip_auth_username are not set (A)"/>
<action application="set" data="default_ani_prefix=${tenant_id}${b_ext}1"/>
     }
     if (${default_ani_prefix} == "") {
<action application="log" data="WARNING default_ani_prefix is not set (B)"/>
<action application="set" data="ani_prefix=${sip_auth_username}0"/>
     } else {
<action application="set" data="ani_prefix=${default_ani_prefix}"/>
     }
     if ($destination_number == "^(\d{8}.*)") {
<action application="bridge" 
data="[tenant_id=${tenant_id},a_ext=${a_ext},b_ext=${b_ext},origination_callee_id_number=$1,origination_caller_id_number=${ani_prefix}${ani}]sofia/internal/8980000000002$1 at 193.105.54.10"/>
     }
</extension>
</context>

What is the nefarious combination of conditions, breaks, actions and 
anti-actions required to implement this?

I'm happy to try and put a newbie-perspective updates into some part of 
the wiki. Talking of which, maybe it would be good to have more 
example-based content in the wiki, but put it in a separate section so 
that it doesn't clutter up the more reference-orientated bits.

Alex


From alex at thewinelake.com  Wed Jun 13 14:14:15 2012
From: alex at thewinelake.com (Alex Lake)
Date: Wed, 13 Jun 2012 11:14:15 +0100
Subject: [Freeswitch-users] Conditions
In-Reply-To: <CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
Message-ID: <4FD867F7.6090701@thewinelake.com>

I've been struggling a bit with conditions. In particular the "break" 
attribute, which I couldn't really find any documentation on.

<action application="set" data="testvarA=hello/>
<action application="set" data="testvarB=hello/>
<condition field="${testvarA}" expression="^$" />
<condition field="${testvarB}" expression="^$" >
    <action application="log" data="WARNING TestVarsA and B are both 
empty"/>
    <anti-action application="log" data="WARNING TestVarsA and B are NOT 
both empty"/>
</condition>

The above gives "TestVarsA and B are both empty" which I don't understand!

I think it would be nice to put some examples in the wiki that translate 
from the usual way of doing things (perhaps using PHP or JavaScript 
notation) to the dialplan format.

For example, I have a requirement that might be expressed like this:

<context name="dp0095">
<extension name="Outbound_call">
     if (${default_ani_prefix} == "" and sip_auth_username == "") {
<action application="log" data="WARNING default_ani_prefix and 
sip_auth_username are not set (A)"/>
<action application="set" data="default_ani_prefix=${tenant_id}${b_ext}1"/>
     }
     if (${default_ani_prefix} == "") {
<action application="log" data="WARNING default_ani_prefix is not set (B)"/>
<action application="set" data="ani_prefix=${sip_auth_username}0"/>
     } else {
<action application="set" data="ani_prefix=${default_ani_prefix}"/>
     }
     if ($destination_number == "^(\d{8}.*)") {
<action application="bridge" 
data="[tenant_id=${tenant_id},a_ext=${a_ext},b_ext=${b_ext},origination_callee_id_number=$1,origination_caller_id_number=${ani_prefix}${ani}]sofia/internal/8980000000002$1 at 193.105.54.10"/>
     }
</extension>
</context>

What is the nefarious combination of conditions, breaks, actions and 
anti-actions required to implement this?

I'm happy to try and put a newbie-perspective updates into some part of 
the wiki. Talking of which, maybe it would be good to have more 
example-based content in the wiki, but put it in a separate section so 
that it doesn't clutter up the more reference-orientated bits.

Alex


From evgeniy at bestnet.kharkov.ua  Wed Jun 13 14:25:46 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Wed, 13 Jun 2012 13:25:46 +0300
Subject: [Freeswitch-users] mod_nibblebill problem
In-Reply-To: <4FD1A7FA.3090009@bestnet.kharkov.ua>
References: <4FD199DA.7080805@bestnet.kharkov.ua>
	<4FD1A7FA.3090009@bestnet.kharkov.ua>
Message-ID: <4FD86AAA.3030807@bestnet.kharkov.ua>

Any ideas?

08.06.2012 10:21, Evgeniy Movlyan ???????:
> Hi All, when i'm calling from one user to another freeswitch user
> mod works fine, but when i'm calling to external number (through
> gateway) billing doesn't work:
> 
> 2012-06-07 16:16:07.294840 [DEBUG] mod_nibblebill.c:612 Received
> request via SESSION_HEARTBEAT! 2012-06-07 16:16:07.294840 [DEBUG]
> mod_nibblebill.c:453 Attempting to bill at $1 per minute to account
> 7604504 2012-06-07 16:16:07.294840 [DEBUG] mod_nibblebill.c:465 Not
> billing 7604504 - call is not in answered state
> 
> I don't understand what this message means.
> 
> Sorry for my english:)

-- 
Evgeniy Movlyan,
BestNet Ltd.


From hkalyoncu at gmail.com  Wed Jun 13 14:26:46 2012
From: hkalyoncu at gmail.com (huseyin kalyoncu)
Date: Wed, 13 Jun 2012 13:26:46 +0300
Subject: [Freeswitch-users] Conditions
In-Reply-To: <4FD867F7.6090701@thewinelake.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
Message-ID: <CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>

hello alex,

you must set testvarA and testvarB as inline

check this  http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions

huseyin



On Wed, Jun 13, 2012 at 1:14 PM, Alex Lake <alex at thewinelake.com> wrote:

> I've been struggling a bit with conditions. In particular the "break"
> attribute, which I couldn't really find any documentation on.
>
> <action application="set" data="testvarA=hello/>
> <action application="set" data="testvarB=hello/>
> <condition field="${testvarA}" expression="^$" />
> <condition field="${testvarB}" expression="^$" >
>    <action application="log" data="WARNING TestVarsA and B are both
> empty"/>
>    <anti-action application="log" data="WARNING TestVarsA and B are NOT
> both empty"/>
> </condition>
>
> The above gives "TestVarsA and B are both empty" which I don't understand!
>
> I think it would be nice to put some examples in the wiki that translate
> from the usual way of doing things (perhaps using PHP or JavaScript
> notation) to the dialplan format.
>
> For example, I have a requirement that might be expressed like this:
>
> <context name="dp0095">
> <extension name="Outbound_call">
>     if (${default_ani_prefix} == "" and sip_auth_username == "") {
> <action application="log" data="WARNING default_ani_prefix and
> sip_auth_username are not set (A)"/>
> <action application="set" data="default_ani_prefix=${tenant_id}${b_ext}1"/>
>     }
>     if (${default_ani_prefix} == "") {
> <action application="log" data="WARNING default_ani_prefix is not set
> (B)"/>
> <action application="set" data="ani_prefix=${sip_auth_username}0"/>
>     } else {
> <action application="set" data="ani_prefix=${default_ani_prefix}"/>
>     }
>     if ($destination_number == "^(\d{8}.*)") {
> <action application="bridge"
>
> data="[tenant_id=${tenant_id},a_ext=${a_ext},b_ext=${b_ext},origination_callee_id_number=$1,origination_caller_id_number=${ani_prefix}${ani}]sofia/internal/8980000000002$
> 1 at 193.105.54.10"/>
>     }
> </extension>
> </context>
>
> What is the nefarious combination of conditions, breaks, actions and
> anti-actions required to implement this?
>
> I'm happy to try and put a newbie-perspective updates into some part of
> the wiki. Talking of which, maybe it would be good to have more
> example-based content in the wiki, but put it in a separate section so
> that it doesn't clutter up the more reference-orientated bits.
>
> Alex
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From alex at thewinelake.com  Wed Jun 13 14:28:35 2012
From: alex at thewinelake.com (Alex)
Date: Wed, 13 Jun 2012 11:28:35 +0100
Subject: [Freeswitch-users] Conditions
In-Reply-To: <4FD867F7.6090701@thewinelake.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
Message-ID: <4FD86B53.9070203@thewinelake.com>

...also struggling with the mailing list (which will teach me to have 
too many email addresses!). Apologies for multi-post
> I've been struggling a bit with conditions. In particular the "break"
> attribute, which I couldn't really find any documentation on.
>
> <action application="set" data="testvarA=hello/>
> <action application="set" data="testvarB=hello/>
> <condition field="${testvarA}" expression="^$" />
> <condition field="${testvarB}" expression="^$">
>      <action application="log" data="WARNING TestVarsA and B are both
> empty"/>
>      <anti-action application="log" data="WARNING TestVarsA and B are NOT
> both empty"/>
> </condition>
>
> The above gives "TestVarsA and B are both empty" which I don't understand!
>
> I think it would be nice to put some examples in the wiki that translate
> from the usual way of doing things (perhaps using PHP or JavaScript
> notation) to the dialplan format.
>
> For example, I have a requirement that might be expressed like this:
>
> <context name="dp0095">
> <extension name="Outbound_call">
>       if (${default_ani_prefix} == "" and sip_auth_username == "") {
> <action application="log" data="WARNING default_ani_prefix and
> sip_auth_username are not set (A)"/>
> <action application="set" data="default_ani_prefix=${tenant_id}${b_ext}1"/>
>       }
>       if (${default_ani_prefix} == "") {
> <action application="log" data="WARNING default_ani_prefix is not set (B)"/>
> <action application="set" data="ani_prefix=${sip_auth_username}0"/>
>       } else {
> <action application="set" data="ani_prefix=${default_ani_prefix}"/>
>       }
>       if ($destination_number == "^(\d{8}.*)") {
> <action application="bridge"
> data="[tenant_id=${tenant_id},a_ext=${a_ext},b_ext=${b_ext},origination_callee_id_number=$1,origination_caller_id_number=${ani_prefix}${ani}]sofia/internal/8980000000002$1 at 193.105.54.10"/>
>       }
> </extension>
> </context>
>
> What is the nefarious combination of conditions, breaks, actions and
> anti-actions required to implement this?
>
> I'm happy to try and put a newbie-perspective updates into some part of
> the wiki. Talking of which, maybe it would be good to have more
> example-based content in the wiki, but put it in a separate section so
> that it doesn't clutter up the more reference-orientated bits.
>
> Alex
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>
>



From alex at thewinelake.com  Wed Jun 13 14:38:40 2012
From: alex at thewinelake.com (Alex)
Date: Wed, 13 Jun 2012 11:38:40 +0100
Subject: [Freeswitch-users] Conditions
In-Reply-To: <CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
Message-ID: <4FD86DB0.4030508@thewinelake.com>

Mmm. Strangely enough, I stumbled on that just after posting, but it 
still doesn't work

<action inline="true" application="set" data="testvarA=hello"/>
<action inline="true" application="set" data="testvarB=hello"/>
<condition field="${testvarA}" expression="^$" />
<condition field="${testvarB}" expression="^$" >
<action application="log" data="WARNING TestVarsA and B are both empty"/>
<anti-action application="log" data="WARNING TestVarsA and B are NOT 
both empty"/>
</condition>

Result: "TestVarsA and B are both empty"

> hello alex,
>
> you must set testvarA and testvarB as inline
>
> check this http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>
> huseyin
>
>
>
> On Wed, Jun 13, 2012 at 1:14 PM, Alex Lake <alex at thewinelake.com 
> <mailto:alex at thewinelake.com>> wrote:
>
>     I've been struggling a bit with conditions. In particular the "break"
>     attribute, which I couldn't really find any documentation on.
>
>     <action application="set" data="testvarA=hello/>
>     <action application="set" data="testvarB=hello/>
>     <condition field="${testvarA}" expression="^$" />
>     <condition field="${testvarB}" expression="^$" >
>     <action application="log" data="WARNING TestVarsA and B are both
>     empty"/>
>     <anti-action application="log" data="WARNING TestVarsA and B are NOT
>     both empty"/>
>     </condition>
>
>     The above gives "TestVarsA and B are both empty" which I don't
>     understand!
>
>     I think it would be nice to put some examples in the wiki that
>     translate
>     from the usual way of doing things (perhaps using PHP or JavaScript
>     notation) to the dialplan format.
>
>     For example, I have a requirement that might be expressed like this:
>
>     <context name="dp0095">
>     <extension name="Outbound_call">
>         if (${default_ani_prefix} == "" and sip_auth_username == "") {
>     <action application="log" data="WARNING default_ani_prefix and
>     sip_auth_username are not set (A)"/>
>     <action application="set"
>     data="default_ani_prefix=${tenant_id}${b_ext}1"/>
>         }
>         if (${default_ani_prefix} == "") {
>     <action application="log" data="WARNING default_ani_prefix is not
>     set (B)"/>
>     <action application="set" data="ani_prefix=${sip_auth_username}0"/>
>         } else {
>     <action application="set" data="ani_prefix=${default_ani_prefix}"/>
>         }
>         if ($destination_number == "^(\d{8}.*)") {
>     <action application="bridge"
>     data="[tenant_id=${tenant_id},a_ext=${a_ext},b_ext=${b_ext},origination_callee_id_number=$1,origination_caller_id_number=${ani_prefix}${ani}]sofia/internal/8980000000002$1 at 193.105.54.10
>     <mailto:1 at 193.105.54.10>"/>
>         }
>     </extension>
>     </context>
>
>     What is the nefarious combination of conditions, breaks, actions and
>     anti-actions required to implement this?
>
>     I'm happy to try and put a newbie-perspective updates into some
>     part of
>     the wiki. Talking of which, maybe it would be good to have more
>     example-based content in the wiki, but put it in a separate section so
>     that it doesn't clutter up the more reference-orientated bits.
>
>     Alex
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>

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From alex at thewinelake.com  Wed Jun 13 15:00:06 2012
From: alex at thewinelake.com (Alex)
Date: Wed, 13 Jun 2012 12:00:06 +0100
Subject: [Freeswitch-users] Conditions
In-Reply-To: <4FD86DB0.4030508@thewinelake.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
	<4FD86DB0.4030508@thewinelake.com>
Message-ID: <4FD872B6.4020107@thewinelake.com>

OK... I think there's a fundamental misunderstanding about what code 
actually gets executed.

The setting never happened in my example. If I change it to:

<condition field="${ani}" expression="^(\d+)$" break="never" >
<action inline="true" application="set" data="testvarA=hello"/>
<action inline="true" application="set" data="testvarB=hello"/>
</condition>
<condition field="${testvarA}" expression="^$" break="never" />
<condition field="${testvarB}" expression="^$" break="never">
<action application="log" data="WARNING TestVarsA and B are both empty"/>
<anti-action application="log" data="WARNING TestVarsA and B are NOT 
both empty"/>
</condition>


Then it works. (Comparing ani to any number of digits).
What I really want is something like <condition expression="true">

> Mmm. Strangely enough, I stumbled on that just after posting, but it 
> still doesn't work
>
> <action inline="true" application="set" data="testvarA=hello"/>
> <action inline="true" application="set" data="testvarB=hello"/>
> <condition field="${testvarA}" expression="^$" />
> <condition field="${testvarB}" expression="^$" >
> <action application="log" data="WARNING TestVarsA and B are both empty"/>
> <anti-action application="log" data="WARNING TestVarsA and B are NOT 
> both empty"/>
> </condition>
>
> Result: "TestVarsA and B are both empty"
>
>> hello alex,
>>
>> you must set testvarA and testvarB as inline
>>
>> check this http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>>
>> huseyin
>>
>>
>>
>> On Wed, Jun 13, 2012 at 1:14 PM, Alex Lake <alex at thewinelake.com 
>> <mailto:alex at thewinelake.com>> wrote:
>>
>>     I've been struggling a bit with conditions. In particular the "break"
>>     attribute, which I couldn't really find any documentation on.
>>
>>     <action application="set" data="testvarA=hello/>
>>     <action application="set" data="testvarB=hello/>
>>     <condition field="${testvarA}" expression="^$" />
>>     <condition field="${testvarB}" expression="^$" >
>>     <action application="log" data="WARNING TestVarsA and B are both
>>     empty"/>
>>     <anti-action application="log" data="WARNING TestVarsA and B are NOT
>>     both empty"/>
>>     </condition>
>>
>>     The above gives "TestVarsA and B are both empty" which I don't
>>     understand!
>>
>>     I think it would be nice to put some examples in the wiki that
>>     translate
>>     from the usual way of doing things (perhaps using PHP or JavaScript
>>     notation) to the dialplan format.
>>
>>     For example, I have a requirement that might be expressed like this:
>>
>>     <context name="dp0095">
>>     <extension name="Outbound_call">
>>         if (${default_ani_prefix} == "" and sip_auth_username == "") {
>>     <action application="log" data="WARNING default_ani_prefix and
>>     sip_auth_username are not set (A)"/>
>>     <action application="set"
>>     data="default_ani_prefix=${tenant_id}${b_ext}1"/>
>>         }
>>         if (${default_ani_prefix} == "") {
>>     <action application="log" data="WARNING default_ani_prefix is not
>>     set (B)"/>
>>     <action application="set" data="ani_prefix=${sip_auth_username}0"/>
>>         } else {
>>     <action application="set" data="ani_prefix=${default_ani_prefix}"/>
>>         }
>>         if ($destination_number == "^(\d{8}.*)") {
>>     <action application="bridge"
>>     data="[tenant_id=${tenant_id},a_ext=${a_ext},b_ext=${b_ext},origination_callee_id_number=$1,origination_caller_id_number=${ani_prefix}${ani}]sofia/internal/8980000000002$1 at 193.105.54.10
>>     <mailto:1 at 193.105.54.10>"/>
>>         }
>>     </extension>
>>     </context>
>>
>>     What is the nefarious combination of conditions, breaks, actions and
>>     anti-actions required to implement this?
>>
>>     I'm happy to try and put a newbie-perspective updates into some
>>     part of
>>     the wiki. Talking of which, maybe it would be good to have more
>>     example-based content in the wiki, but put it in a separate
>>     section so
>>     that it doesn't clutter up the more reference-orientated bits.
>>
>>     Alex
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.com
>>
>>     Join Us At ClueCon - Aug 7-9, 2012
>>
>>     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
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> No virus found in this message.
>> Checked by AVG - www.avg.com <http://www.avg.com>
>> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>

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From hkalyoncu at gmail.com  Wed Jun 13 15:31:51 2012
From: hkalyoncu at gmail.com (huseyin kalyoncu)
Date: Wed, 13 Jun 2012 14:31:51 +0300
Subject: [Freeswitch-users] Conditions
In-Reply-To: <4FD872B6.4020107@thewinelake.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
	<4FD86DB0.4030508@thewinelake.com> <4FD872B6.4020107@thewinelake.com>
Message-ID: <CAPf+86XBW2gs2N2zidqMFN7bMV5VuraUk7QG=opYGNzrki+4nQ@mail.gmail.com>

yes action statements must be inside condition tags. i thought you wrote
them in condition in your first mail..

for the expression=true ; i not sure you can do that but there is an
example usage together with regex in the wiki.
again check the link i sent before..

huseyin



On Wed, Jun 13, 2012 at 2:00 PM, Alex <alex at thewinelake.com> wrote:

>  OK... I think there's a fundamental misunderstanding about what code
> actually gets executed.
>
> The setting never happened in my example. If I change it to:
>
>         <condition field="${ani}" expression="^(\d+)$" break="never" >
>
>         <action inline="true" application="set" data="testvarA=hello"/>
>         <action inline="true" application="set" data="testvarB=hello"/>
>         </condition>
>        <condition field="${testvarA}" expression="^$" break="never" />
>         <condition field="${testvarB}" expression="^$" break="never">
>
>                 <action application="log" data="WARNING TestVarsA and B
> are both empty"/>
>                 <anti-action application="log" data="WARNING TestVarsA and
> B are NOT both empty"/>
>        </condition>
>
>
> Then it works. (Comparing ani to any number of digits).
> What I really want is something like <condition expression="true">
>
>
>  Mmm. Strangely enough, I stumbled on that just after posting, but it
> still doesn't work
>
>         <action inline="true" application="set" data="testvarA=hello"/>
>         <action inline="true" application="set" data="testvarB=hello"/>
>        <condition field="${testvarA}" expression="^$" />
>         <condition field="${testvarB}" expression="^$" >
>                 <action application="log" data="WARNING TestVarsA and B
> are both empty"/>
>                 <anti-action application="log" data="WARNING TestVarsA and
> B are NOT both empty"/>
>        </condition>
>
> Result: "TestVarsA and B are both empty"
>
> hello alex,
>
> you must set testvarA and testvarB as inline
>
>  check this  http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>
>  huseyin
>
>
>
>  On Wed, Jun 13, 2012 at 1:14 PM, Alex Lake <alex at thewinelake.com> wrote:
>
>> I've been struggling a bit with conditions. In particular the "break"
>> attribute, which I couldn't really find any documentation on.
>>
>>  <action application="set" data="testvarA=hello/>
>> <action application="set" data="testvarB=hello/>
>> <condition field="${testvarA}" expression="^$" />
>> <condition field="${testvarB}" expression="^$" >
>>    <action application="log" data="WARNING TestVarsA and B are both
>> empty"/>
>>    <anti-action application="log" data="WARNING TestVarsA and B are NOT
>> both empty"/>
>> </condition>
>>
>> The above gives "TestVarsA and B are both empty" which I don't understand!
>>
>> I think it would be nice to put some examples in the wiki that translate
>> from the usual way of doing things (perhaps using PHP or JavaScript
>> notation) to the dialplan format.
>>
>> For example, I have a requirement that might be expressed like this:
>>
>> <context name="dp0095">
>> <extension name="Outbound_call">
>>     if (${default_ani_prefix} == "" and sip_auth_username == "") {
>> <action application="log" data="WARNING default_ani_prefix and
>> sip_auth_username are not set (A)"/>
>> <action application="set"
>> data="default_ani_prefix=${tenant_id}${b_ext}1"/>
>>     }
>>     if (${default_ani_prefix} == "") {
>> <action application="log" data="WARNING default_ani_prefix is not set
>> (B)"/>
>> <action application="set" data="ani_prefix=${sip_auth_username}0"/>
>>     } else {
>> <action application="set" data="ani_prefix=${default_ani_prefix}"/>
>>     }
>>     if ($destination_number == "^(\d{8}.*)") {
>> <action application="bridge"
>>
>> data="[tenant_id=${tenant_id},a_ext=${a_ext},b_ext=${b_ext},origination_callee_id_number=$1,origination_caller_id_number=${ani_prefix}${ani}]sofia/internal/8980000000002$
>> 1 at 193.105.54.10"/>
>>     }
>> </extension>
>> </context>
>>
>> What is the nefarious combination of conditions, breaks, actions and
>> anti-actions required to implement this?
>>
>> I'm happy to try and put a newbie-perspective updates into some part of
>> the wiki. Talking of which, maybe it would be good to have more
>> example-based content in the wiki, but put it in a separate section so
>> that it doesn't clutter up the more reference-orientated bits.
>>
>> Alex
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hkalyoncu at gmail.com  Wed Jun 13 15:38:27 2012
From: hkalyoncu at gmail.com (huseyin kalyoncu)
Date: Wed, 13 Jun 2012 14:38:27 +0300
Subject: [Freeswitch-users] Conditions
In-Reply-To: <CAPf+86XBW2gs2N2zidqMFN7bMV5VuraUk7QG=opYGNzrki+4nQ@mail.gmail.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
	<4FD86DB0.4030508@thewinelake.com> <4FD872B6.4020107@thewinelake.com>
	<CAPf+86XBW2gs2N2zidqMFN7bMV5VuraUk7QG=opYGNzrki+4nQ@mail.gmail.com>
Message-ID: <CAPf+86VpHXqnnJpCOmq7TNu4g1WBXyO8PpsuStftWYiwLwhVBA@mail.gmail.com>

i checked it myself again and i find the following example in the wiki...

 <extension name="9191">
   <condition field="destination_number" expression="^9191$"/>
   <condition field="${sip_authorized}" expression="true">
     <anti-action application="reject" data="407"/>
   </condition>

   <condition>
     <action application="playback" data="/tmp/itworked.wav"/>
   </condition>
 </extension>

i hope this helps..

On Wed, Jun 13, 2012 at 2:31 PM, huseyin kalyoncu <hkalyoncu at gmail.com>wrote:

> yes action statements must be inside condition tags. i thought you wrote
> them in condition in your first mail..
>
> for the expression=true ; i not sure you can do that but there is an
> example usage together with regex in the wiki.
> again check the link i sent before..
>
> huseyin
>
>
>
> On Wed, Jun 13, 2012 at 2:00 PM, Alex <alex at thewinelake.com> wrote:
>
>>  OK... I think there's a fundamental misunderstanding about what code
>> actually gets executed.
>>
>> The setting never happened in my example. If I change it to:
>>
>>         <condition field="${ani}" expression="^(\d+)$" break="never" >
>>
>>         <action inline="true" application="set" data="testvarA=hello"/>
>>         <action inline="true" application="set" data="testvarB=hello"/>
>>         </condition>
>>        <condition field="${testvarA}" expression="^$" break="never" />
>>         <condition field="${testvarB}" expression="^$" break="never">
>>
>>                 <action application="log" data="WARNING TestVarsA and B
>> are both empty"/>
>>                 <anti-action application="log" data="WARNING TestVarsA
>> and B are NOT both empty"/>
>>        </condition>
>>
>>
>> Then it works. (Comparing ani to any number of digits).
>> What I really want is something like <condition expression="true">
>>
>>
>>  Mmm. Strangely enough, I stumbled on that just after posting, but it
>> still doesn't work
>>
>>         <action inline="true" application="set" data="testvarA=hello"/>
>>         <action inline="true" application="set" data="testvarB=hello"/>
>>        <condition field="${testvarA}" expression="^$" />
>>         <condition field="${testvarB}" expression="^$" >
>>                 <action application="log" data="WARNING TestVarsA and B
>> are both empty"/>
>>                 <anti-action application="log" data="WARNING TestVarsA
>> and B are NOT both empty"/>
>>        </condition>
>>
>> Result: "TestVarsA and B are both empty"
>>
>> hello alex,
>>
>> you must set testvarA and testvarB as inline
>>
>>  check this  http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>>
>>  huseyin
>>
>>
>>
>>  On Wed, Jun 13, 2012 at 1:14 PM, Alex Lake <alex at thewinelake.com> wrote:
>>
>>> I've been struggling a bit with conditions. In particular the "break"
>>> attribute, which I couldn't really find any documentation on.
>>>
>>>  <action application="set" data="testvarA=hello/>
>>> <action application="set" data="testvarB=hello/>
>>> <condition field="${testvarA}" expression="^$" />
>>> <condition field="${testvarB}" expression="^$" >
>>>    <action application="log" data="WARNING TestVarsA and B are both
>>> empty"/>
>>>    <anti-action application="log" data="WARNING TestVarsA and B are NOT
>>> both empty"/>
>>> </condition>
>>>
>>> The above gives "TestVarsA and B are both empty" which I don't
>>> understand!
>>>
>>> I think it would be nice to put some examples in the wiki that translate
>>> from the usual way of doing things (perhaps using PHP or JavaScript
>>> notation) to the dialplan format.
>>>
>>> For example, I have a requirement that might be expressed like this:
>>>
>>> <context name="dp0095">
>>> <extension name="Outbound_call">
>>>     if (${default_ani_prefix} == "" and sip_auth_username == "") {
>>> <action application="log" data="WARNING default_ani_prefix and
>>> sip_auth_username are not set (A)"/>
>>> <action application="set"
>>> data="default_ani_prefix=${tenant_id}${b_ext}1"/>
>>>     }
>>>     if (${default_ani_prefix} == "") {
>>> <action application="log" data="WARNING default_ani_prefix is not set
>>> (B)"/>
>>> <action application="set" data="ani_prefix=${sip_auth_username}0"/>
>>>     } else {
>>> <action application="set" data="ani_prefix=${default_ani_prefix}"/>
>>>     }
>>>     if ($destination_number == "^(\d{8}.*)") {
>>> <action application="bridge"
>>>
>>> data="[tenant_id=${tenant_id},a_ext=${a_ext},b_ext=${b_ext},origination_callee_id_number=$1,origination_caller_id_number=${ani_prefix}${ani}]sofia/internal/8980000000002$
>>> 1 at 193.105.54.10"/>
>>>     }
>>> </extension>
>>> </context>
>>>
>>> What is the nefarious combination of conditions, breaks, actions and
>>> anti-actions required to implement this?
>>>
>>> I'm happy to try and put a newbie-perspective updates into some part of
>>> the wiki. Talking of which, maybe it would be good to have more
>>> example-based content in the wiki, but put it in a separate section so
>>> that it doesn't clutter up the more reference-orientated bits.
>>>
>>> Alex
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> No virus found in this message.
>> Checked by AVG - www.avg.com
>> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> No virus found in this message.
>> Checked by AVG - www.avg.com
>> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From sdevoy at bizfocused.com  Wed Jun 13 16:20:59 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 13 Jun 2012 08:20:59 -0400
Subject: [Freeswitch-users] Domain name lost in SIP registration (w/
	SIP	TRACE)
In-Reply-To: <6856752153193153969@unknownmsgid>
References: <011e01cd48ca$99953aa0$ccbfafe0$@bizfocused.com>
	<6856752153193153969@unknownmsgid>
Message-ID: <015001cd495e$fdacdbd0$f9069370$@bizfocused.com>

"TURN" settings ??? - LOL

 

Thanks for the reply, the problem has been solved.

 

For future reference, when the /etc/init.d script was corrected and was
starting Freeswitch properly, all the domains started working.  I assume it
was not loading all the config files.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris
Ferreira
Sent: Tuesday, June 12, 2012 5:03 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Domain name lost in SIP registration (w/ SIP
TRACE)

 

Oops, I mean STUN Server.

___________________

Mobile Reply


On Jun 12, 2012, at 2:41 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

HI All,

 

In the pastebin:   <http://pastebin.com/KyGXMTP0>
http://pastebin.com/KyGXMTP0 is a sip trace.  

 

The phone is on the lan and uses proxy: fs_test.bizfocused.com which is
10.10.40.189.  In the SIP messages you can see the domain name
"fs_test.bizfocused.com".

 

I have a directory that includes <domain name="fs_test.bizfocused.com"> and
a user for this subscription.

 

But the SIP response is "SIP/2.0 401 Unauthorized" and the message displayed
in FS_CLI is:

[WARNING] sofia_reg.c:2376 Can't find user [202 at 10.10.40.189]

You must define a domain called '10.10.40.189' in your directory and add a
user with the id="202" attribute

and you must configure your device to use the proper domain in its
authentication credentials.

 

Why isn't it using the credentials in the domain "fs_test.bizfocused.com"?

 

Thanks,

Sean

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

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From sdevoy at bizfocused.com  Wed Jun 13 16:22:24 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 13 Jun 2012 08:22:24 -0400
Subject: [Freeswitch-users] Moving up to better PC - Multi-tenant died
In-Reply-To: <CAKDfjgco_8K_mLW7Ekzm+s2Ut0L_74RBEVYhUEEaLT0hT5XHBw@mail.gmail.com>
References: <05a501cd4856$41711610$c4534230$@bizfocused.com>
	<CAKDfjgco_8K_mLW7Ekzm+s2Ut0L_74RBEVYhUEEaLT0hT5XHBw@mail.gmail.com>
Message-ID: <015501cd495f$308e8620$91ab9260$@bizfocused.com>

Thanks for the response.  The /etc/init.d script problems were the cause.
Once Freeswitch was starting properly, the domain problem was gone.

Sean

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: Tuesday, June 12, 2012 5:12 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Moving up to better PC - Multi-tenant died

What is your hostname?  How are the various domain values set in the
FreeSWITCH Sofia profiles?

On Tue, Jun 12, 2012 at 12:45 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> HI,
>
>
>
> I have a nice working multi-tenant configuration working on my 
> development box ? P4 3.0Ghz 4GB RAM Raid 1 Centos 5.7 Freeswitch (make
current).
>
>
>
> I built a nice clean Centos 5.7 install on a Dell Power Edge 2850 Dual 
> Xeon
> 3.0 Ghs 4GB RAM and current Freeswitch.
>
> I saved the entire ?/usr/local/freeswitch/conf? tree from new server.
>
> ??????????????? I copied the entire ?/usr/local/freeswitch/conf? tree 
> from devel box to new server.
>
> ??????????????? I shutdown development box.
>
> ??????????????? I set new server to same IP as old Devel boix.
>
> ??????????????? Rebooted.
>
>
>
> I have a couple of issues:
>
> ??????????????? 1. Freeswitch does not startup at boot.? I tried to 
> follow the wiki.? I have a freeswitch file in /;etc/init.d but no startup!
>
>
>
> ??????????????? 2. Only my primary domain (tenant) users can login.? 
> All of my secondary domain phones have 
> proxy=nnn at fs_mbri.bizfocused.com but the errors on FS_CLI say it can?t 
> find nnn at fs_bfis.bizfocused.com (MY PRIMARY DOMAIN).? It is though 
> freeswitch thinks these registrations are coming to the primary domain 
> even though they are going to a secondry with the same IP address.
>
>
>
> So, how do I get FS to start on boot up in CENTOS 5.7?? I swear I 
> tried what?s in the wiki.? Or maybe I should say how do I tell wait is
failing?
>
>
>
> What am I missing in the multi-tenant domain definition (that is not 
> in the /conf folder)?
>
>
>
> Thanks in advance.
>
>
>
> Sean
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users 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
>



--
Kristian Kielhofner

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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From sdevoy at bizfocused.com  Wed Jun 13 16:39:47 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 13 Jun 2012 08:39:47 -0400
Subject: [Freeswitch-users] Conditions
In-Reply-To: <4FD872B6.4020107@thewinelake.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>	<4FD867F7.6090701@thewinelake.com>	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>	<4FD86DB0.4030508@thewinelake.com>
	<4FD872B6.4020107@thewinelake.com>
Message-ID: <017501cd4961$9e515c80$daf41580$@bizfocused.com>

I was about to answer and say  <action> tags must be inside <condition> tags.

 

I think you can in fact remove you break=?never? everywhere except:

                       <condition field="${testvarA}" expression="^$" break="never" />



Your first ?condition? can just be ?<condition>?  which is always true.

 

And  <condition field="${testvarB}" expression="^$" break="never"> 

doesn?t need a break=?never? as it has <action> and <anti-action>s.

 

Hope that helps (and is correct!)

 

Sean

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Alex
Sent: Wednesday, June 13, 2012 7:00 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conditions

 

OK... I think there's a fundamental misunderstanding about what code actually gets executed.

The setting never happened in my example. If I change it to:

        <condition field="${ani}" expression="^(\d+)$" break="never" >
        <action inline="true" application="set" data="testvarA=hello"/>
        <action inline="true" application="set" data="testvarB=hello"/>
        </condition>
       <condition field="${testvarA}" expression="^$" break="never" />
        <condition field="${testvarB}" expression="^$" break="never">
                <action application="log" data="WARNING TestVarsA and B are both empty"/>
                <anti-action application="log" data="WARNING TestVarsA and B are NOT both empty"/>
       </condition>


Then it works. (Comparing ani to any number of digits).
What I really want is something like <condition expression="true">




Mmm. Strangely enough, I stumbled on that just after posting, but it still doesn't work

        <action inline="true" application="set" data="testvarA=hello"/>
        <action inline="true" application="set" data="testvarB=hello"/>
       <condition field="${testvarA}" expression="^$" />
        <condition field="${testvarB}" expression="^$" >
                <action application="log" data="WARNING TestVarsA and B are both empty"/>
                <anti-action application="log" data="WARNING TestVarsA and B are NOT both empty"/>
       </condition>

Result: "TestVarsA and B are both empty"




hello alex, 

 

you must set testvarA and testvarB as inline

 

check this  http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions

 

huseyin

 

 

On Wed, Jun 13, 2012 at 1:14 PM, Alex Lake <alex at thewinelake.com> wrote:

I've been struggling a bit with conditions. In particular the "break"
attribute, which I couldn't really find any documentation on.

<action application="set" data="testvarA=hello/>
<action application="set" data="testvarB=hello/>
<condition field="${testvarA}" expression="^$" />
<condition field="${testvarB}" expression="^$" >
   <action application="log" data="WARNING TestVarsA and B are both
empty"/>
   <anti-action application="log" data="WARNING TestVarsA and B are NOT
both empty"/>
</condition>

The above gives "TestVarsA and B are both empty" which I don't understand!


I think it would be nice to put some examples in the wiki that translate
from the usual way of doing things (perhaps using PHP or JavaScript
notation) to the dialplan format.

For example, I have a requirement that might be expressed like this:

<context name="dp0095">
<extension name="Outbound_call">
    if (${default_ani_prefix} == "" and sip_auth_username == "") {
<action application="log" data="WARNING default_ani_prefix and
sip_auth_username are not set (A)"/>
<action application="set" data="default_ani_prefix=${tenant_id}${b_ext}1"/>
    }
    if (${default_ani_prefix} == "") {
<action application="log" data="WARNING default_ani_prefix is not set (B)"/>
<action application="set" data="ani_prefix=${sip_auth_username}0"/>
    } else {
<action application="set" data="ani_prefix=${default_ani_prefix}"/>
    }
    if ($destination_number == "^(\d{8}.*)") {
<action application="bridge"
data="[tenant_id=${tenant_id},a_ext=${a_ext},b_ext=${b_ext},origination_callee_id_number=$1,origination_caller_id_number=${ani_prefix}${ani}]sofia/internal/8980000000002$1 at 193.105.54.10"/>
    }
</extension>
</context>

What is the nefarious combination of conditions, breaks, actions and
anti-actions required to implement this?

I'm happy to try and put a newbie-perspective updates into some part of
the wiki. Talking of which, maybe it would be good to have more
example-based content in the wiki, but put it in a separate section so
that it doesn't clutter up the more reference-orientated bits.

Alex

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com
 


 
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
 
Join Us At ClueCon - Aug 7-9, 2012
 
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






No virus found in this message.
Checked by AVG - www.avg.com
Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12







_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com
 


 
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
 
Join Us At ClueCon - Aug 7-9, 2012
 
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






No virus found in this message.
Checked by AVG - www.avg.com
Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12

 

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From alex at thewinelake.com  Wed Jun 13 17:18:40 2012
From: alex at thewinelake.com (Alex)
Date: Wed, 13 Jun 2012 14:18:40 +0100
Subject: [Freeswitch-users] Conditions
In-Reply-To: <CAPf+86VpHXqnnJpCOmq7TNu4g1WBXyO8PpsuStftWYiwLwhVBA@mail.gmail.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
	<4FD86DB0.4030508@thewinelake.com>
	<4FD872B6.4020107@thewinelake.com>
	<CAPf+86XBW2gs2N2zidqMFN7bMV5VuraUk7QG=opYGNzrki+4nQ@mail.gmail.com>
	<CAPf+86VpHXqnnJpCOmq7TNu4g1WBXyO8PpsuStftWYiwLwhVBA@mail.gmail.com>
Message-ID: <4FD89330.60402@thewinelake.com>

Ah so
<condition>
</condition>

is essentially an "unconditional condition" ;-)

That is useful...
> i checked it myself again and i find the following example in the wiki...
>
>   <extension name="9191">
>     <condition field="destination_number" expression="^9191$"/>
>     <condition field="${sip_authorized}" expression="true">
>       <anti-action application="reject" data="407"/>
>     </condition>
>
>     <condition>
>       <action application="playback" data="/tmp/itworked.wav"/>
>     </condition>
>   </extension>
> i hope this helps..
>
> On Wed, Jun 13, 2012 at 2:31 PM, huseyin kalyoncu <hkalyoncu at gmail.com 
> <mailto:hkalyoncu at gmail.com>> wrote:
>
>     yes action statements must be inside condition tags. i thought you
>     wrote them in condition in your first mail..
>
>     for the expression=true ; i not sure you can do that but there is
>     an example usage together with regex in the wiki.
>     again check the link i sent before..
>
>     huseyin
>
>
>     On Wed, Jun 13, 2012 at 2:00 PM, Alex <alex at thewinelake.com
>     <mailto:alex at thewinelake.com>> wrote:
>
>         OK... I think there's a fundamental misunderstanding about
>         what code actually gets executed.
>
>         The setting never happened in my example. If I change it to:
>
>         <condition field="${ani}" expression="^(\d+)$" break="never" >
>
>         <action inline="true" application="set" data="testvarA=hello"/>
>         <action inline="true" application="set" data="testvarB=hello"/>
>         </condition>
>         <condition field="${testvarA}" expression="^$" break="never" />
>         <condition field="${testvarB}" expression="^$" break="never">
>
>         <action application="log" data="WARNING TestVarsA and B are
>         both empty"/>
>         <anti-action application="log" data="WARNING TestVarsA and B
>         are NOT both empty"/>
>         </condition>
>
>
>         Then it works. (Comparing ani to any number of digits).
>         What I really want is something like <condition
>         expression="true">
>
>
>>         Mmm. Strangely enough, I stumbled on that just after posting,
>>         but it still doesn't work
>>
>>         <action inline="true" application="set" data="testvarA=hello"/>
>>         <action inline="true" application="set" data="testvarB=hello"/>
>>         <condition field="${testvarA}" expression="^$" />
>>         <condition field="${testvarB}" expression="^$" >
>>         <action application="log" data="WARNING TestVarsA and B are
>>         both empty"/>
>>         <anti-action application="log" data="WARNING TestVarsA and B
>>         are NOT both empty"/>
>>         </condition>
>>
>>         Result: "TestVarsA and B are both empty"
>>
>>>         hello alex,
>>>
>>>         you must set testvarA and testvarB as inline
>>>
>>>         check this
>>>         http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>>>
>>>         huseyin
>>>
>>>
>>>
>>>         On Wed, Jun 13, 2012 at 1:14 PM, Alex Lake
>>>         <alex at thewinelake.com <mailto:alex at thewinelake.com>> wrote:
>>>
>>>             I've been struggling a bit with conditions. In
>>>             particular the "break"
>>>             attribute, which I couldn't really find any
>>>             documentation on.
>>>
>>>             <action application="set" data="testvarA=hello/>
>>>             <action application="set" data="testvarB=hello/>
>>>             <condition field="${testvarA}" expression="^$" />
>>>             <condition field="${testvarB}" expression="^$" >
>>>             <action application="log" data="WARNING TestVarsA and B
>>>             are both
>>>             empty"/>
>>>             <anti-action application="log" data="WARNING TestVarsA
>>>             and B are NOT
>>>             both empty"/>
>>>             </condition>
>>>
>>>             The above gives "TestVarsA and B are both empty" which I
>>>             don't understand!
>>>
>>>             I think it would be nice to put some examples in the
>>>             wiki that translate
>>>             from the usual way of doing things (perhaps using PHP or
>>>             JavaScript
>>>             notation) to the dialplan format.
>>>
>>>             For example, I have a requirement that might be
>>>             expressed like this:
>>>
>>>             <context name="dp0095">
>>>             <extension name="Outbound_call">
>>>                 if (${default_ani_prefix} == "" and
>>>             sip_auth_username == "") {
>>>             <action application="log" data="WARNING
>>>             default_ani_prefix and
>>>             sip_auth_username are not set (A)"/>
>>>             <action application="set"
>>>             data="default_ani_prefix=${tenant_id}${b_ext}1"/>
>>>                 }
>>>                 if (${default_ani_prefix} == "") {
>>>             <action application="log" data="WARNING
>>>             default_ani_prefix is not set (B)"/>
>>>             <action application="set"
>>>             data="ani_prefix=${sip_auth_username}0"/>
>>>                 } else {
>>>             <action application="set"
>>>             data="ani_prefix=${default_ani_prefix}"/>
>>>                 }
>>>                 if ($destination_number == "^(\d{8}.*)") {
>>>             <action application="bridge"
>>>             data="[tenant_id=${tenant_id},a_ext=${a_ext},b_ext=${b_ext},origination_callee_id_number=$1,origination_caller_id_number=${ani_prefix}${ani}]sofia/internal/8980000000002$1 at 193.105.54.10
>>>             <mailto:1 at 193.105.54.10>"/>
>>>                 }
>>>             </extension>
>>>             </context>
>>>
>>>             What is the nefarious combination of conditions, breaks,
>>>             actions and
>>>             anti-actions required to implement this?
>>>
>>>             I'm happy to try and put a newbie-perspective updates
>>>             into some part of
>>>             the wiki. Talking of which, maybe it would be good to
>>>             have more
>>>             example-based content in the wiki, but put it in a
>>>             separate section so
>>>             that it doesn't clutter up the more reference-orientated
>>>             bits.
>>>
>>>             Alex
>>>
>>>             _________________________________________________________________________
>>>             Professional FreeSWITCH Consulting Services:
>>>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>             http://www.freeswitchsolutions.com
>>>
>>>             
>>>             
>>>
>>>             Official FreeSWITCH Sites
>>>             http://www.freeswitch.org
>>>             http://wiki.freeswitch.org
>>>             http://www.cluecon.com
>>>
>>>             Join Us At ClueCon - Aug 7-9, 2012
>>>
>>>             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
>>>
>>>
>>>
>>>
>>>         _________________________________________________________________________
>>>         Professional FreeSWITCH Consulting Services:
>>>         consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>>         http://www.freeswitchsolutions.com
>>>
>>>         
>>>         
>>>
>>>         Official FreeSWITCH Sites
>>>         http://www.freeswitch.org
>>>         http://wiki.freeswitch.org
>>>         http://www.cluecon.com
>>>
>>>         Join Us At ClueCon - Aug 7-9, 2012
>>>
>>>         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
>>>
>>>
>>>         No virus found in this message.
>>>         Checked by AVG - www.avg.com <http://www.avg.com>
>>>         Version: 2012.0.2177 / Virus Database: 2433/5065 - Release
>>>         Date: 06/12/12
>>>
>>
>>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.com
>>
>>         Join Us At ClueCon - Aug 7-9, 2012
>>
>>         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
>>
>>
>>         No virus found in this message.
>>         Checked by AVG - www.avg.com <http://www.avg.com>
>>         Version: 2012.0.2177 / Virus Database: 2433/5065 - Release
>>         Date: 06/12/12
>>
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>
>         Join Us At ClueCon - Aug 7-9, 2012
>
>         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
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>

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From alex at thewinelake.com  Wed Jun 13 17:30:14 2012
From: alex at thewinelake.com (Alex)
Date: Wed, 13 Jun 2012 14:30:14 +0100
Subject: [Freeswitch-users] call_timeout and enterprise originate
In-Reply-To: <CAPf+86VpHXqnnJpCOmq7TNu4g1WBXyO8PpsuStftWYiwLwhVBA@mail.gmail.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
	<4FD86DB0.4030508@thewinelake.com>
	<4FD872B6.4020107@thewinelake.com>
	<CAPf+86XBW2gs2N2zidqMFN7bMV5VuraUk7QG=opYGNzrki+4nQ@mail.gmail.com>
	<CAPf+86VpHXqnnJpCOmq7TNu4g1WBXyO8PpsuStftWYiwLwhVBA@mail.gmail.com>
Message-ID: <4FD895E6.7000101@thewinelake.com>

Is there anything I should know about the combination of these two?

Alex


From avi at avimarcus.net  Wed Jun 13 17:46:53 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 13 Jun 2012 16:46:53 +0300
Subject: [Freeswitch-users] call_timeout and enterprise originate
In-Reply-To: <4FD895E6.7000101@thewinelake.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
	<4FD86DB0.4030508@thewinelake.com> <4FD872B6.4020107@thewinelake.com>
	<CAPf+86XBW2gs2N2zidqMFN7bMV5VuraUk7QG=opYGNzrki+4nQ@mail.gmail.com>
	<CAPf+86VpHXqnnJpCOmq7TNu4g1WBXyO8PpsuStftWYiwLwhVBA@mail.gmail.com>
	<4FD895E6.7000101@thewinelake.com>
Message-ID: <CALNZuwnDT23d44oR8-jPfpu4dKh-mh9GSQJzBOFtO9bWSUa4Rw@mail.gmail.com>

Hmm. re: enterprise originate remember that each is a separate thread,
so {} vars in one thread won't work on the other. You need to do
super-global <> to make that happen.

I don't think we have a wiki page purely on building bridge strings yet...

-Avi Marcus

On Wed, Jun 13, 2012 at 4:30 PM, Alex <alex at thewinelake.com> wrote:
> Is there anything I should know about the combination of these two?
>
> Alex
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From kris at kriskinc.com  Wed Jun 13 19:15:50 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 13 Jun 2012 11:15:50 -0400
Subject: [Freeswitch-users] how to use codec g729 on freeswitch ?
In-Reply-To: <1339558840.7099.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1FFF97C269757C458224B7C895F35F1510248C@cantor.std.visionutv.se>
	<1339409611.83296.YahooMailNeo@web120101.mail.ne1.yahoo.com>
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	<4FD6371D.1010307@cupis.co.uk>
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Message-ID: <CAKDfjgc4y0D6j0wHLmVSCrr+uLjs6266b0wNsjePnvc2MeDt-w@mail.gmail.com>

It is very clear the Cisco is responding with a 488 when the only
codec offered is G729.  When you offer G729 and PCMU it accepts PCMU
because that is all it supports.  By that time the A leg between your
other endpoint and FreeSWITCH has already selected G729 and FreeSWITCH
tries to transcode and it can't because you don't have the codec
installed.

You need to support G729 on your Cisco gateway, install the G729 codec
in FreeSWITCH, or use PCMU on all of your devices.

On Tue, Jun 12, 2012 at 11:40 PM, Samira Mh <saami_mh at ymail.com> wrote:
> bu i have set the codec in dialpeer of VOIP gateway
>
> ________________________________
> From: Kristian Kielhofner <kris at kriskinc.com>
>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, June 13, 2012 1:40 AM
>
> Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
>
> Your Cisco gateway doesn't support G729.? Check your codec-list
> configuration in IOS.
>
> On Tue, Jun 12, 2012 at 12:06 AM, Samira Mh <saami_mh at ymail.com> wrote:
>> thansk for your reply,
>> it is kind of you to help me..
>> please let me paste myconfigurations files here;
>> 1-the configuration ?file
>> /usr/local/freeswitch/conf/dialplan/default/001_luacallduration.xml ?is
>> like
>> this:
>>
>> <include>
>> ? <extension name="mainmenuvodsl">
>>
>> ? ? ? ? <condition field="destination_number"
>> expression="^(00|\+)?(\d{5}.*)$" break="never">
>> ? ? ??? ? ? ? ? ? ?<action application="odbc_query" data="select cash as
>> cashvalue from accounts where contractid like '${nibble_account}';"/>
>> ? ? ? ? ? ? ? ? <action application="log" data="INFO The value of
>> cashvalue
>> is ${cashvalue}" />
>> ?? ? ? ? ? ? ? ? <action application="lua" data="checkcash.lua
>> ${cashvalue}"
>> />
>> ? ? ? ? ? ? ? ? <action application="log" data="INFO The value of
>> nibble_account is ?${nibble_account}"/>
>> ? ? ? ? ? ? ? ? <action application="log" data="INFO The value of
>> nibble_rate [before] is ?${nibble_rate}"/>
>> ? ? ? ? ? ? ? ? <!-- RateList Context -->
>> ? ? ? ? ? ? ? ? <action application="lua" data="checkzeroplus.lua
>> ${destination_number:0:2} ${destination_number:0:1}" />
>> ? ? ? ? ? ? ? ? <action application="execute_extension"
>> data="${destination_number} XML ratelist"/>
>> ? ? ? ? ? ? ? ? <action application="log" data="INFO The value of
>> nibble_rate [after] is ${nibble_rate}"/>
>> ? ? ? ? ? ? ? ? ?<!-- Check Nibble_rate -->
>> ? ? ? ? ? ? ? ? <action application="lua" data="checknibblerate.lua
>> ${nibble_rate}" />
>> ? ? ? ? ? ? ? ? <action application="set"
>> data="divvalue=${expr(floor((${cashvalue}/${nibble_rate}))}" />
>> ? ? ? ? ? ? ? ? <action application="set"
>> data="modvalue=${expr(mod(${cashvalue},${nibble_rate}))}" />
>> ? ? ? ? ? ? ? ? <action application="lua" data="checktime.lua ${divvalue}
>> ${modvalue}" />
>> ? ? ? ? ? ? ? ?? ? ?<!-- ?Check ZeroZero , Plus ?-->
>> ? ? ? ? ? ? ?? ?<action application="lua" data="checkzeroplus.lua
>> ${destination_number:0:2} ${destination_number:0:1}" />
>> ? ? ? ? ? ? ? ? <!-- Making Calls ? -->
>> ? ? ? ? ? ? ? ? <action application="odbc_query" data="select callerid ?as
>> effective_caller_id_number from accounts where contractid like
>> '${nibble_account}';"/>
>> ? ? ? ? ? ? ? ? <action application="log" data="INFO ?callerid for
>> Outbound
>> calls ${effective_caller_id_number}"/>
>> ? ? ? ? ? ? ? ? <!-- <action application="set"
>> data="ignore_early_media=true"/>
>> ? ? ? ? ? ? ? ? <action application="answer"/> ?-->
>> ? ? ? ? ? ? ? ? <action application="enable_heartbeat"/>
>>
>> <!-- <param name="disable-transcoding" value="true"/> -->
>> ? ? ? ? <!-- ? ?<action application="export"
>> data="nolocal:absolute_codec_string=G729,PCMU"/> -->
>> <!-- ?<action application="set" data="bridge_early_media=true"/> ?-->
>> ? ? ? ? <!-- <action application="set" data="proxy_media=true"/> -->
>> ? ? ? ? ? ? ? ? <action application="bridge"
>> data="sofia/gateway/cisco/140112${destination_number}"/>
>> ? ? ? ? ? ? ? ? <!-- <action application="bridge"
>> data="sofia/gateway/mainasterisk/${destination_number}"/> ?-->
>> <!-- ?<action application="bridge"
>> data="sofia/gateway/test/${destination_number}"/> ?-->
>> ? ? ? ? </condition>
>>
>> ?</extension>
>> </include>
>>
>> 2-yes, i have enabled ?"inbound-late-negotiation"?in the
>> (/usr/local/freeswitch/conf/sip_profiles/internal.xml) as follow:
>> ?<param name="inbound-late-negotiation" value="true"/>
>>
>>
>> 3-the issue of sofia status:
>> ?external::cisco ? ? ? gateway ? ? ? ? ? ? sip:register:false at 85.15.0.154
>> ? ?NOREG
>>
>>
>> 4-also , the configuration file for codecs are as follow
>> :/usr/local/freeswitch/conf/vars.xml
>>
>> <X-PRE-PROCESS cmd="set"
>>
>> data="global_codec_prefs=G729,PCMU,PCMA,G7221 at 32000h,G7221 at 16000h,G722,GSM"/>
>>
>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>
>>
>> 5- the mod_g729 was loaded
>>
>> 6-i have enabled the siptrace:
>> ?sofia profile external siptrace on:
>> the siptrace outpout as follow:
>>
>> send 1042 bytes to udp/[85.15.0.154]:5060 at 03:53:07.448136:
>>
>> ?------------------------------------------------------------------------
>> ? ?INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
>> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>> ? ?Max-Forwards: 69
>> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>> ? ?To: <sip:140112971507247227 at 85.15.0.154>
>> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>> ? ?CSeq: 29400529 INVITE
>> ? ?Contact: <sip:gw+cisco at 192.168.10.70:5080;transport=udp;gw=cisco>
>> ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
>> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>> ? ?Supported: timer, precondition, path, replaces
>> ? ?Allow-Events: talk, hold, refer
>> ? ?Content-Type: application/sdp
>> ? ?Content-Disposition: session
>> ? ?Content-Length: 234
>> ? ?X-FS-Support: update_display,send_info
>> ? ?Remote-Party-ID: "1000"
>> <sip:1000 at 85.15.0.154>;party=calling;screen=yes;privacy=off
>>
>> ? ?v=0
>> ? ?o=FreeSWITCH 1339446571 1339446572 IN IP4 192.168.10.70
>> ? ?s=FreeSWITCH
>> ? ?c=IN IP4 192.168.10.70
>> ? ?t=0 0
>> ? ?m=audio 26616 RTP/AVP 9 0 8 18 3 101 13
>> ? ?a=fmtp:18 annexb=yes
>> ? ?a=rtpmap:101 telephone-event/8000
>> ? ?a=fmtp:101 0-16
>> ? ?a=ptime:20
>>
>> ?------------------------------------------------------------------------
>> recv 410 bytes from udp/[85.15.0.154]:5060 at 03:53:07.463921:
>>
>> ?------------------------------------------------------------------------
>> ? ?SIP/2.0 100 Trying
>> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
>> ? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
>> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>> ? ?Server: Cisco-SIPGateway/IOS-12.x
>> ? ?CSeq: 29400529 INVITE
>> ? ?Allow-Events: telephone-event
>> ? ?Content-Length: 0
>>
>>
>> ?------------------------------------------------------------------------
>> recv 927 bytes from udp/[85.15.0.154]:5060 at 03:53:11.004804:
>>
>> ?------------------------------------------------------------------------
>> ? ?SIP/2.0 183 Session Progress
>> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
>> ? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
>> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>> ? ?Server: Cisco-SIPGateway/IOS-12.x
>> ? ?CSeq: 29400529 INVITE
>> ? ?Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
>> SUBSCRIBE,
>> NOTIFY, INFO, UPDATE, REGISTER
>> ? ?Allow-Events: telephone-event
>> ? ?Contact: <sip:140112971507247227 at 85.15.0.154:5060>
>> ? ?Content-Disposition: session;handling=required
>> ? ?Content-Type: application/sdp
>> ? ?Content-Length: 268
>>
>> ? ?v=0
>> ? ?o=CiscoSystemsSIP-GW-UserAgent 7252 5649 IN IP4 85.15.0.154
>> ? ?s=SIP Call
>> ? ?c=IN IP4 85.15.0.154
>> ? ?t=0 0
>> ? ?m=audio 18218 RTP/AVP 0 13 101
>> ? ?c=IN IP4 85.15.0.154
>> ? ?a=rtpmap:0 PCMU/8000
>> ? ?a=rtpmap:13 CN/8000
>> ? ?a=rtpmap:101 telephone-event/8000
>> ? ?a=fmtp:101 0-15
>> ? ?a=ptime:20
>>
>> ?------------------------------------------------------------------------
>> recv 425 bytes from udp/[85.15.0.154]:5060 at 03:53:11.005144:
>>
>> ?------------------------------------------------------------------------
>> ? ?SIP/2.0 500 Internal Server Error
>> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
>> ? ?Date: Tue, 12 Jun 2012 03:53:15 GMT
>> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>> ? ?Server: Cisco-SIPGateway/IOS-12.x
>> ? ?CSeq: 29400529 INVITE
>> ? ?Allow-Events: telephone-event
>> ? ?Content-Length: 0
>>
>>
>> ?------------------------------------------------------------------------
>> send 350 bytes to udp/[85.15.0.154]:5060 at 03:53:11.005333:
>>
>> ?------------------------------------------------------------------------
>> ? ?ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
>> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
>> ? ?Max-Forwards: 69
>> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
>> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=45785134-1BDE
>> ? ?Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
>> ? ?CSeq: 29400529 ACK
>> ? ?Content-Length: 0
>>
>>
>>
>> ------------------------------------------------------------------------------------------------------------------
>> when change the configuration file the below:
>> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729"/>
>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>
>>
>> the siptrace is like this:
>>
>> send 1034 bytes to udp/[85.15.0.154]:5060 at 04:01:16.202342:
>>
>> ?------------------------------------------------------------------------
>> ? ?INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
>> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
>> ? ?Max-Forwards: 69
>> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
>> ? ?To: <sip:140112971507247227 at 85.15.0.154>
>> ? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
>> ? ?CSeq: 29400774 INVITE
>> ? ?Contact: <sip:gw+cisco at 192.168.10.70:5080;transport=udp;gw=cisco>
>> ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
>> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>> ? ?Supported: timer, precondition, path, replaces
>> ? ?Allow-Events: talk, hold, refer
>> ? ?Content-Type: application/sdp
>> ? ?Content-Disposition: session
>> ? ?Content-Length: 226
>> ? ?X-FS-Support: update_display,send_info
>> ? ?Remote-Party-ID: "1000"
>> <sip:1000 at 85.15.0.154>;party=calling;screen=yes;privacy=off
>>
>> ? ?v=0
>> ? ?o=FreeSWITCH 1339447862 1339447863 IN IP4 192.168.10.70
>> ? ?s=FreeSWITCH
>> ? ?c=IN IP4 192.168.10.70
>> ? ?t=0 0
>> ? ?m=audio 25814 RTP/AVP 18 101 13
>> ? ?a=fmtp:18 annexb=yes
>> ? ?a=rtpmap:101 telephone-event/8000
>> ? ?a=fmtp:101 0-16
>> ? ?a=ptime:20
>>
>> ?------------------------------------------------------------------------
>> recv 423 bytes from udp/[85.15.0.154]:5060 at 04:01:16.208118:
>>
>> ?------------------------------------------------------------------------
>> ? ?SIP/2.0 488 Not Acceptable Media
>> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
>> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
>> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=457FC664-6A6
>> ? ?Date: Tue, 12 Jun 2012 04:01:24 GMT
>> ? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
>> ? ?Server: Cisco-SIPGateway/IOS-12.x
>> ? ?CSeq: 29400774 INVITE
>> ? ?Allow-Events: telephone-event
>> ? ?Content-Length: 0
>>
>>
>> ?------------------------------------------------------------------------
>> send 349 bytes to udp/[85.15.0.154]:5060 at 04:01:16.208201:
>>
>> ?------------------------------------------------------------------------
>> ? ?ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
>> ? ?Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
>> ? ?Max-Forwards: 69
>> ? ?From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
>> ? ?To: <sip:140112971507247227 at 85.15.0.154>;tag=457FC664-6A6
>> ? ?Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
>> ? ?CSeq: 29400774 ACK
>> ? ?Content-Length: 0
>>
>>
>>
>> plz help,thanks so much
>>
>>
>> ________________________________
>> From: Paul Cupis <paul at cupis.co.uk>
>>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, June 11, 2012 10:51 PM
>>
>> Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
>>
>> On 11/06/12 17:54, Samira Mh wrote:
>>> i want to bridge call using my VOIPgateway so that making calls to
>>> another countries..
>>> but the carrier only support G729 codec and the FS send G722 (set in
>>> vars.xml) to myVoipGateway that is set as an gateway in
>>> /usr/local/freeswitch/sip-profile/external/
>>> and when FS send media to Gateway(using bridge application) the error
>>> occure:unacceptable media,then check VOIPGW and find out the only codec
>>> that
>>> can be pass through VOIPgw is G729, but FS only send G711,G722,... not
>>> G729
>>
>> Can you provide a SIP or FreeSWITCH trace of a call, please?
>>
>> Do you have the following enabled in your SIP profile?
>>
>> ? <param name="inbound-late-negotiation" value="true"/>
>>
>> Do you have mod_g729 loaded and codec G729 enabled in your vars.xml?
>>
>> Regards,
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Kristian Kielhofner
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From saami_mh at ymail.com  Wed Jun 13 20:49:23 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Wed, 13 Jun 2012 09:49:23 -0700 (PDT)
Subject: [Freeswitch-users] problem on playinback " .wav "files on
	freeswitch using asterisk with Codec G729 ?
Message-ID: <1339606163.79781.YahooMailNeo@web120104.mail.ne1.yahoo.com>

hi guys,
as you know on freeswitch the codec G729 only use fro pass through,
so we couldn't play .wav file using codec G729,
so i have installed asterisk to everytime i want to play file with that codec?
play them on asterisk using gateway on freeswitch ,
but in asterisk when i issue the command (exten => 999,1,playback(hello-world)) the file play successfully using freeswitch that is connect to asterisk
but i don't want to answer channel at this situation so issue the command (exten => 999,1,playback(hello-world,noanswer) or?
(exten => 999,1,background(hello-world,n) -- n used to don't answer channel )) i don't hear the sound of .wav file

on the debug of sip on asterisk the file is played and command was executed but i coudn't hear sound
notice:because of using nibble_account on freeswitch i don't want to answer channel to reduce the cash?
plz help
thanks?
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From alex at thewinelake.com  Wed Jun 13 20:53:53 2012
From: alex at thewinelake.com (Alex)
Date: Wed, 13 Jun 2012 17:53:53 +0100
Subject: [Freeswitch-users] call_timeout and enterprise originate
In-Reply-To: <CALNZuwnDT23d44oR8-jPfpu4dKh-mh9GSQJzBOFtO9bWSUa4Rw@mail.gmail.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
	<4FD86DB0.4030508@thewinelake.com>
	<4FD872B6.4020107@thewinelake.com>
	<CAPf+86XBW2gs2N2zidqMFN7bMV5VuraUk7QG=opYGNzrki+4nQ@mail.gmail.com>
	<CAPf+86VpHXqnnJpCOmq7TNu4g1WBXyO8PpsuStftWYiwLwhVBA@mail.gmail.com>
	<4FD895E6.7000101@thewinelake.com>
	<CALNZuwnDT23d44oR8-jPfpu4dKh-mh9GSQJzBOFtO9bWSUa4Rw@mail.gmail.com>
Message-ID: <4FD8C5A1.8050905@thewinelake.com>

Just 'cos it's a separate thread doesn't mean that it couldn't have been 
initialised with channel variables from the spawning thread.
I presume that exports don't work either.
How about group_confirm?
This is looking awfully like writing our own everything!
> Hmm. re: enterprise originate remember that each is a separate thread,
> so {} vars in one thread won't work on the other. You need to do
> super-global<>  to make that happen.
>
> I don't think we have a wiki page purely on building bridge strings yet...
>
> -Avi Marcus
>
> On Wed, Jun 13, 2012 at 4:30 PM, Alex<alex at thewinelake.com>  wrote:
>> Is there anything I should know about the combination of these two?
>>
>> Alex
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>
>



From peter.olsson at visionutveckling.se  Wed Jun 13 21:03:25 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 13 Jun 2012 17:03:25 +0000
Subject: [Freeswitch-users] problem on playinback " .wav "files
	on	freeswitch using asterisk with Codec G729 ?
In-Reply-To: <1339606163.79781.YahooMailNeo@web120104.mail.ne1.yahoo.com>
References: <1339606163.79781.YahooMailNeo@web120104.mail.ne1.yahoo.com>
Message-ID: <31E64CAD-2886-4B99-A04F-5456E198414F@visionutveckling.se>

I'd suggest getting a valid g729 licenses for FS, and skip the extra work...

/Peter

13 jun 2012 kl. 18:58 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com>>:

hi guys,
as you know on freeswitch the codec G729 only use fro pass through,
so we couldn't play .wav file using codec G729,
so i have installed asterisk to everytime i want to play file with that codec
play them on asterisk using gateway on freeswitch ,
but in asterisk when i issue the command (exten => 999,1,playback(hello-world)) the file play successfully using freeswitch that is connect to asterisk
but i don't want to answer channel at this situation so issue the command (exten => 999,1,playback(hello-world,noanswer) or
(exten => 999,1,background(hello-world,n) -- n used to don't answer channel )) i don't hear the sound of .wav file
on the debug of sip on asterisk the file is played and command was executed but i coudn't hear sound
notice:because of using nibble_account on freeswitch i don't want to answer channel to reduce the cash
plz help
thanks
!DSPAM:4fd8c38432761914914593!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


!DSPAM:4fd8c38432761914914593!


From avi at avimarcus.net  Wed Jun 13 21:34:07 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 13 Jun 2012 20:34:07 +0300
Subject: [Freeswitch-users] call_timeout and enterprise originate
In-Reply-To: <4FD8C5A1.8050905@thewinelake.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
	<4FD86DB0.4030508@thewinelake.com> <4FD872B6.4020107@thewinelake.com>
	<CAPf+86XBW2gs2N2zidqMFN7bMV5VuraUk7QG=opYGNzrki+4nQ@mail.gmail.com>
	<CAPf+86VpHXqnnJpCOmq7TNu4g1WBXyO8PpsuStftWYiwLwhVBA@mail.gmail.com>
	<4FD895E6.7000101@thewinelake.com>
	<CALNZuwnDT23d44oR8-jPfpu4dKh-mh9GSQJzBOFtO9bWSUa4Rw@mail.gmail.com>
	<4FD8C5A1.8050905@thewinelake.com>
Message-ID: <CALNZuw=17NZK6oR65Yg=hnPmdMjBQMX2XWbvHcYB8xNZtO4Rnw@mail.gmail.com>

bridge seems to be fairly manual for assembling complicated bridge strings.

I think export works to all threads, but afaik, is deprecated.

I've used group_confirm on these threads, it's fine. Just make sure
the group_confirm variable is in the right place...

-Avi


On Wed, Jun 13, 2012 at 7:53 PM, Alex <alex at thewinelake.com> wrote:
> Just 'cos it's a separate thread doesn't mean that it couldn't have been
> initialised with channel variables from the spawning thread.
> I presume that exports don't work either.
> How about group_confirm?
> This is looking awfully like writing our own everything!
>> Hmm. re: enterprise originate remember that each is a separate thread,
>> so {} vars in one thread won't work on the other. You need to do
>> super-global<> ?to make that happen.
>>
>> I don't think we have a wiki page purely on building bridge strings yet...
>>
>> -Avi Marcus
>>
>> On Wed, Jun 13, 2012 at 4:30 PM, Alex<alex at thewinelake.com> ?wrote:
>>> Is there anything I should know about the combination of these two?
>>>
>>> Alex
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> -----
>> No virus found in this message.
>> Checked by AVG - www.avg.com
>> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From jnankin at gmail.com  Wed Jun 13 21:50:39 2012
From: jnankin at gmail.com (Joshua Nankin)
Date: Wed, 13 Jun 2012 12:50:39 -0500
Subject: [Freeswitch-users] Spandsp events not being fired
Message-ID: <CAAYWZ8mUFBF1rcU_ErEURDgjCpmr8bGeZA8CCCfeQKPj2sOfBg@mail.gmail.com>

I'm sending a fax with freeswitch, but the events listed at
http://wiki.freeswitch.org/wiki/Mod_spandsp#Events do not seem to be fired.
 Do I need to enable them somehow?
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From vipkilla at gmail.com  Wed Jun 13 23:50:08 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Wed, 13 Jun 2012 15:50:08 -0400
Subject: [Freeswitch-users] Polycom caller ID problem on 4.0.2
In-Reply-To: <CAAB3jWtVAso3T74b5jPqFEXJtV-jXFoMu27DkbWK-GBxy_SxRw@mail.gmail.com>
References: <CAAB3jWsQPFyrr1d=d6gfA0G9Q+kyM0v7VyvDoOYrq=h6LaZ9rA@mail.gmail.com>
	<CAAtWKeFcgvVbW03ece3yGFLGoXxeP5uiDuTTUWeQ7s+5wmqs2g@mail.gmail.com>
	<CAAB3jWtVAso3T74b5jPqFEXJtV-jXFoMu27DkbWK-GBxy_SxRw@mail.gmail.com>
Message-ID: <CAC-LwPO+sB3ig739LrqkKObVnDmredm_nZpKsnZ9C_HRYWjHhQ@mail.gmail.com>

Don't mean to hijack this thread but do you have BLF working with the
4.0 firmware?


From vipkilla at gmail.com  Wed Jun 13 23:56:44 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Wed, 13 Jun 2012 15:56:44 -0400
Subject: [Freeswitch-users] Polycom caller ID problem on 4.0.2
In-Reply-To: <CAC-LwPO+sB3ig739LrqkKObVnDmredm_nZpKsnZ9C_HRYWjHhQ@mail.gmail.com>
References: <CAAB3jWsQPFyrr1d=d6gfA0G9Q+kyM0v7VyvDoOYrq=h6LaZ9rA@mail.gmail.com>
	<CAAtWKeFcgvVbW03ece3yGFLGoXxeP5uiDuTTUWeQ7s+5wmqs2g@mail.gmail.com>
	<CAAB3jWtVAso3T74b5jPqFEXJtV-jXFoMu27DkbWK-GBxy_SxRw@mail.gmail.com>
	<CAC-LwPO+sB3ig739LrqkKObVnDmredm_nZpKsnZ9C_HRYWjHhQ@mail.gmail.com>
Message-ID: <CAC-LwPNieX8omk5XcoR_=NXeOBQpz+-k0KL-Cc4AnRv6ZpvB9A@mail.gmail.com>

Nevermind, i got BLF working, had to add the line:
feature.presence.enabled="1"


I now have Polycoms on firmware 4.02b working perfectly with FS.

On Wed, Jun 13, 2012 at 3:50 PM, Vik Killa <vipkilla at gmail.com> wrote:
> Don't mean to hijack this thread but do you have BLF working with the
> 4.0 firmware?


From sdevoy at bizfocused.com  Thu Jun 14 00:05:12 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 13 Jun 2012 16:05:12 -0400
Subject: [Freeswitch-users] Audio Connection Delay with SCA
In-Reply-To: <CAKbxfG_a5QfK+FhqqMWmtRZhogw47rwyufnJRDOPk9unWyXxgQ@mail.gmail.com>
References: <018101cd4599$f3b2b320$db181960$@bizfocused.com>	<CAKbxfG-RiMFttXuZQjZSbyG-Uk6gZLWT-XzFYZfqUGdhdPJL-Q@mail.gmail.com>	<029601cd45b0$38d29d60$aa77d820$@bizfocused.com>
	<CAKbxfG_a5QfK+FhqqMWmtRZhogw47rwyufnJRDOPk9unWyXxgQ@mail.gmail.com>
Message-ID: <046701cd499f$d7a16a00$86e43e00$@bizfocused.com>

FYI: New System w/ 2 Hyperthreaded Dual core 3.0Ghz XEON processors (vs P4
3Ghx) appears to have solved this delay problem.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Friday, June 08, 2012 4:00 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Audio Connection Delay with SCA

from FS point of view there should be little different from SCA and not SCA
besides the presence signaling, you should try some on-site testing with the
same phones.


On Fri, Jun 8, 2012 at 2:52 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> Anthony,
> Thanks for the reply.
>
> We use all CSICO SPA504G phones.
> I did have the jitter buffer set up "HIGH" (choices: Low, Med, High, 
> Very High, Extremely High). ?I dropped them down to Low now for 
> testing. ?They do have pretty good network speeds (business level cable
modem).
>
> There is also something labeled: ?" Jitter Buffer Adjustment: ?" ?
> Choices (Up, Down, Both, Disabled). ?I have both.
>
> However, the issue is present, but far less pronounced in non-shared
lines.
> They have always said there was a connection delay at this site, but 
> with SCA it seems to be compounded.
>
> Maybe Group Call Pick will work better for them, they are all in the area.
>
> Sean
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> Anthony Minessale
> Sent: Friday, June 08, 2012 3:28 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Audio Connection Delay with SCA
>
> what type of phone? ?If it has a jitter buffer, you can try reducing 
> or disabling it.
> its probably a combo of jitter buffers charging up on the phone, and 
> the auto-ajust trying to fix the broken sdp for NAT.
>
>
> On Fri, Jun 8, 2012 at 12:12 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>> Hi,
>>
>>
>>
>> I finally got Shared Call Appearance working to 4 NAT connected
> phones!!!!!
>>
>>
>>
>> Now the users say when they answer that line, they must wait a second 
>> or so to start talking (no big deal), but they all claim it takes 
>> nearly 5 seconds to be able to hear the caller.? They answer the 
>> standard ?Thank You for calling XYZ, how can I help you today? which 
>> the caller DOES HEAR, but they cannot hear his response for the first
>> 5 seconds and ALWAYS have to ask them to repeat it.
>>
>>
>>
>> In the logs, I see what I believe is them picking up extension 299:
>>
>> ??? 2012-06-08 11:44:15.115249 [NOTICE] switch_channel.c:926 New 
>> Channel
>> sofia/external/sip:299 at 69.251.170.6:5063 

>>
>> And then the RPT finally connecting:
>>
>> ?? 2012-06-08 11:44:20.665094 [DEBUG] sofia_glue.c:3248 AUDIO RTP 
>> [sofia/external/sip:299 at 69.251.170.6:5064] 10.10.40.185 port 23500 ->
>> 69.251.170.6 port 16498 codec: 0 ms: 20
>>
>>
>>
>> Note the 5.5 second delay!? Granted there are 100+ lines between 
>> those if you want to see them, I will post them.
>>
>> After that audio is fine.
>>
>>
>>
>> Any thoughts would be appreciated.
>>
>> Sean
>>
>>
>> _____________________________________________________________________
>> _ ___ Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>  
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users 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
>> e
>> rs
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users 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
>
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users 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



--
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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




From sdevoy at bizfocused.com  Thu Jun 14 00:07:28 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 13 Jun 2012 16:07:28 -0400
Subject: [Freeswitch-users] Is SCA bridgeing Multithreaded?
Message-ID: <046801cd49a0$28819b70$7984d250$@bizfocused.com>

I see I can use Enterprise Originate :_: to connect calls to multiple phones
in threads to speed things up.  

 

Does Shared Call Appearance connect in threads or sequentially?

 

Thanks,

Sean

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From alex at thewinelake.com  Thu Jun 14 00:52:37 2012
From: alex at thewinelake.com (Alexander Lake)
Date: Wed, 13 Jun 2012 21:52:37 +0100
Subject: [Freeswitch-users] call_timeout and enterprise originate
In-Reply-To: <CALNZuw=17NZK6oR65Yg=hnPmdMjBQMX2XWbvHcYB8xNZtO4Rnw@mail.gmail.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
	<4FD86DB0.4030508@thewinelake.com>
	<4FD872B6.4020107@thewinelake.com>
	<CAPf+86XBW2gs2N2zidqMFN7bMV5VuraUk7QG=opYGNzrki+4nQ@mail.gmail.com>
	<CAPf+86VpHXqnnJpCOmq7TNu4g1WBXyO8PpsuStftWYiwLwhVBA@mail.gmail.com>
	<4FD895E6.7000101@thewinelake.com>
	<CALNZuwnDT23d44oR8-jPfpu4dKh-mh9GSQJzBOFtO9bWSUa4Rw@mail.gmail.com>
	<4FD8C5A1.8050905@thewinelake.com>
	<CALNZuw=17NZK6oR65Yg=hnPmdMjBQMX2XWbvHcYB8xNZtO4Rnw@mail.gmail.com>
Message-ID: <933886F1-C00C-4B73-9D76-049CB8D2E612@thewinelake.com>

export is deprecated? How is one supposed to do it instead?
Maybe I need to get some examples...

On 13 Jun 2012, at 18:34, Avi Marcus wrote:

> bridge seems to be fairly manual for assembling complicated bridge strings.
> 
> I think export works to all threads, but afaik, is deprecated.
> 
> I've used group_confirm on these threads, it's fine. Just make sure
> the group_confirm variable is in the right place...
> 
> -Avi
> 
> 
> On Wed, Jun 13, 2012 at 7:53 PM, Alex <alex at thewinelake.com> wrote:
>> Just 'cos it's a separate thread doesn't mean that it couldn't have been
>> initialised with channel variables from the spawning thread.
>> I presume that exports don't work either.
>> How about group_confirm?
>> This is looking awfully like writing our own everything!
>>> Hmm. re: enterprise originate remember that each is a separate thread,
>>> so {} vars in one thread won't work on the other. You need to do
>>> super-global<>  to make that happen.
>>> 
>>> I don't think we have a wiki page purely on building bridge strings yet...
>>> 
>>> -Avi Marcus
>>> 
>>> On Wed, Jun 13, 2012 at 4:30 PM, Alex<alex at thewinelake.com>  wrote:
>>>> Is there anything I should know about the combination of these two?
>>>> 
>>>> Alex
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> Join Us At ClueCon - Aug 7-9, 2012
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> -----
>>> No virus found in this message.
>>> Checked by AVG - www.avg.com
>>> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>>> 
>>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From avi at avimarcus.net  Thu Jun 14 00:55:56 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 13 Jun 2012 23:55:56 +0300
Subject: [Freeswitch-users] call_timeout and enterprise originate
In-Reply-To: <933886F1-C00C-4B73-9D76-049CB8D2E612@thewinelake.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
	<4FD86DB0.4030508@thewinelake.com> <4FD872B6.4020107@thewinelake.com>
	<CAPf+86XBW2gs2N2zidqMFN7bMV5VuraUk7QG=opYGNzrki+4nQ@mail.gmail.com>
	<CAPf+86VpHXqnnJpCOmq7TNu4g1WBXyO8PpsuStftWYiwLwhVBA@mail.gmail.com>
	<4FD895E6.7000101@thewinelake.com>
	<CALNZuwnDT23d44oR8-jPfpu4dKh-mh9GSQJzBOFtO9bWSUa4Rw@mail.gmail.com>
	<4FD8C5A1.8050905@thewinelake.com>
	<CALNZuw=17NZK6oR65Yg=hnPmdMjBQMX2XWbvHcYB8xNZtO4Rnw@mail.gmail.com>
	<933886F1-C00C-4B73-9D76-049CB8D2E612@thewinelake.com>
Message-ID: <CALNZuw=_SCGjNc_L=n-pyY0zwyynwOKYw101gKhSFGxZE75Lgw@mail.gmail.com>

Like you've been doing, mostly: setting them inside the bridge string.

-Avi


On Wed, Jun 13, 2012 at 11:52 PM, Alexander Lake <alex at thewinelake.com> wrote:
> export is deprecated? How is one supposed to do it instead?
> Maybe I need to get some examples...
>
> On 13 Jun 2012, at 18:34, Avi Marcus wrote:
>
>> bridge seems to be fairly manual for assembling complicated bridge strings.
>>
>> I think export works to all threads, but afaik, is deprecated.
>>
>> I've used group_confirm on these threads, it's fine. Just make sure
>> the group_confirm variable is in the right place...
>>
>> -Avi
>>
>>
>> On Wed, Jun 13, 2012 at 7:53 PM, Alex <alex at thewinelake.com> wrote:
>>> Just 'cos it's a separate thread doesn't mean that it couldn't have been
>>> initialised with channel variables from the spawning thread.
>>> I presume that exports don't work either.
>>> How about group_confirm?
>>> This is looking awfully like writing our own everything!
>>>> Hmm. re: enterprise originate remember that each is a separate thread,
>>>> so {} vars in one thread won't work on the other. You need to do
>>>> super-global<> ?to make that happen.
>>>>
>>>> I don't think we have a wiki page purely on building bridge strings yet...
>>>>
>>>> -Avi Marcus
>>>>
>>>> On Wed, Jun 13, 2012 at 4:30 PM, Alex<alex at thewinelake.com> ?wrote:
>>>>> Is there anything I should know about the combination of these two?
>>>>>
>>>>> Alex
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> -----
>>>> No virus found in this message.
>>>> Checked by AVG - www.avg.com
>>>> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>>>>
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From alex at thewinelake.com  Thu Jun 14 01:04:06 2012
From: alex at thewinelake.com (Alexander Lake)
Date: Wed, 13 Jun 2012 22:04:06 +0100
Subject: [Freeswitch-users] call_timeout and enterprise originate
In-Reply-To: <CALNZuw=_SCGjNc_L=n-pyY0zwyynwOKYw101gKhSFGxZE75Lgw@mail.gmail.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
	<4FD86DB0.4030508@thewinelake.com>
	<4FD872B6.4020107@thewinelake.com>
	<CAPf+86XBW2gs2N2zidqMFN7bMV5VuraUk7QG=opYGNzrki+4nQ@mail.gmail.com>
	<CAPf+86VpHXqnnJpCOmq7TNu4g1WBXyO8PpsuStftWYiwLwhVBA@mail.gmail.com>
	<4FD895E6.7000101@thewinelake.com>
	<CALNZuwnDT23d44oR8-jPfpu4dKh-mh9GSQJzBOFtO9bWSUa4Rw@mail.gmail.com>
	<4FD8C5A1.8050905@thewinelake.com>
	<CALNZuw=17NZK6oR65Yg=hnPmdMjBQMX2XWbvHcYB8xNZtO4Rnw@mail.gmail.com>
	<933886F1-C00C-4B73-9D76-049CB8D2E612@thewinelake.com>
	<CALNZuw=_SCGjNc_L=n-pyY0zwyynwOKYw101gKhSFGxZE75Lgw@mail.gmail.com>
Message-ID: <90AD4E68-0314-4CDB-83FA-BE5AEF27AE4C@thewinelake.com>

Ah. Is there a way to set common variables for the bridges?

On 13 Jun 2012, at 21:55, Avi Marcus wrote:

> Like you've been doing, mostly: setting them inside the bridge string.
> 
> -Avi
> 
> 
> On Wed, Jun 13, 2012 at 11:52 PM, Alexander Lake <alex at thewinelake.com> wrote:
>> export is deprecated? How is one supposed to do it instead?
>> Maybe I need to get some examples...
>> 
>> On 13 Jun 2012, at 18:34, Avi Marcus wrote:
>> 
>>> bridge seems to be fairly manual for assembling complicated bridge strings.
>>> 
>>> I think export works to all threads, but afaik, is deprecated.
>>> 
>>> I've used group_confirm on these threads, it's fine. Just make sure
>>> the group_confirm variable is in the right place...
>>> 
>>> -Avi
>>> 
>>> 
>>> On Wed, Jun 13, 2012 at 7:53 PM, Alex <alex at thewinelake.com> wrote:
>>>> Just 'cos it's a separate thread doesn't mean that it couldn't have been
>>>> initialised with channel variables from the spawning thread.
>>>> I presume that exports don't work either.
>>>> How about group_confirm?
>>>> This is looking awfully like writing our own everything!
>>>>> Hmm. re: enterprise originate remember that each is a separate thread,
>>>>> so {} vars in one thread won't work on the other. You need to do
>>>>> super-global<>  to make that happen.
>>>>> 
>>>>> I don't think we have a wiki page purely on building bridge strings yet...
>>>>> 
>>>>> -Avi Marcus
>>>>> 
>>>>> On Wed, Jun 13, 2012 at 4:30 PM, Alex<alex at thewinelake.com>  wrote:
>>>>>> Is there anything I should know about the combination of these two?
>>>>>> 
>>>>>> Alex
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> -----
>>>>> No virus found in this message.
>>>>> Checked by AVG - www.avg.com
>>>>> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>>>>> 
>>>>> 
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> Join Us At ClueCon - Aug 7-9, 2012
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mario_fs at mgtech.com  Thu Jun 14 01:07:31 2012
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 13 Jun 2012 14:07:31 -0700
Subject: [Freeswitch-users] NAT Router Recommendation For Large no of IP
	Phones Behind the Router
In-Reply-To: <CAGvuWYZnw05oKVcAArT9HaeqBagBanaeiG5ngdwNP4Mv79fXBQ@mail.gmail.com>
References: <CAGvuWYaqsqO6N0y+5+pJ2zyA0UqS0iAFjQiXuaSxssH52GcdVQ@mail.gmail.com>
	<729B9245-CBCA-4F7F-9B71-9142E0ADA448@mgtech.com>
	<4FD7766D.8040208@puzzled.xs4all.nl>
	<CAGvuWYZnw05oKVcAArT9HaeqBagBanaeiG5ngdwNP4Mv79fXBQ@mail.gmail.com>
Message-ID: <2993C055-06FD-4814-8BD1-7E85B72B6759@mgtech.com>

A little late but important to say: When I had the Linksys routers my FS had constant attempts from around the world, mostly China to break into it. I used WireShark to see the attempts, the Linksys could not keep the out.. Since the Zyxel USG100 it has stopped. It has an outstanding firewall and the object oriented rules really helps with SIP. Whatever you select, keep security in mind.
  
On Jun 12, 2012, at 12:23 PM, Lloyd Aloysius wrote:

> I try with the pfsense 2.0. Specially BLF causing lots of problems.
> 
> ====
> 
> In the past .. I try the following
> 
> 1. Linksys WRT54GL + Tomato ... Good for a small office < 5 Extension
> 2. Monowall with PC Engine Alix
> 
> 
> Does any one test with a Cisco ASA 5505? 
> 
> 
> 
> On Tue, Jun 12, 2012 at 1:03 PM, Patrick Lists <freeswitch-list at puzzled.xs4all.nl> wrote:
> On 12-06-12 18:20, Mario G wrote:
> > I looked for a year to find something with no NAT problems, dual WAN,
> > and auto fallback with load/balancing without causing FS any hiccups. I
> > highly recommend Zyxel, I use a USG100 with SPI ALG turned ON (I read
> > some people hate it) but it has been bullet proof. And the best part is
> > there is practically nothing to tell FS about! And when one WAN (DSL)
> > goes down the SIP traffic goes out the other one, even switches back
> > automatically! Pair this up with FS and you'll never look back. Hope
> > this helps.
> 
> Thanks for the info. Have you tried any Draytek Vigor routers like the
> 2920? I have no experience with them but was told they work fine in dual
> WAN setups.
> 
> Regards,
> Patrick
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From avi at avimarcus.net  Thu Jun 14 01:08:12 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 14 Jun 2012 00:08:12 +0300
Subject: [Freeswitch-users] call_timeout and enterprise originate
In-Reply-To: <90AD4E68-0314-4CDB-83FA-BE5AEF27AE4C@thewinelake.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
	<4FD86DB0.4030508@thewinelake.com> <4FD872B6.4020107@thewinelake.com>
	<CAPf+86XBW2gs2N2zidqMFN7bMV5VuraUk7QG=opYGNzrki+4nQ@mail.gmail.com>
	<CAPf+86VpHXqnnJpCOmq7TNu4g1WBXyO8PpsuStftWYiwLwhVBA@mail.gmail.com>
	<4FD895E6.7000101@thewinelake.com>
	<CALNZuwnDT23d44oR8-jPfpu4dKh-mh9GSQJzBOFtO9bWSUa4Rw@mail.gmail.com>
	<4FD8C5A1.8050905@thewinelake.com>
	<CALNZuw=17NZK6oR65Yg=hnPmdMjBQMX2XWbvHcYB8xNZtO4Rnw@mail.gmail.com>
	<933886F1-C00C-4B73-9D76-049CB8D2E612@thewinelake.com>
	<CALNZuw=_SCGjNc_L=n-pyY0zwyynwOKYw101gKhSFGxZE75Lgw@mail.gmail.com>
	<90AD4E68-0314-4CDB-83FA-BE5AEF27AE4C@thewinelake.com>
Message-ID: <CALNZuwnHb8=xrfPx3EQ3bEmLAE+=2Zi3wnWJu3PB_WvdDPCh7Q@mail.gmail.com>

[var1=myinfo] is per each bridge string. {var1=myinfo} applies to the
entire thread, meaning if you have a :_: it won't carry over to that.
<var1=myinfo> is a superglobal which is applied to ALL threads.

-Avi


On Thu, Jun 14, 2012 at 12:04 AM, Alexander Lake <alex at thewinelake.com> wrote:
> Ah. Is there a way to set common variables for the bridges?
>
> On 13 Jun 2012, at 21:55, Avi Marcus wrote:
>
>> Like you've been doing, mostly: setting them inside the bridge string.
>>
>> -Avi
>>
>>
>> On Wed, Jun 13, 2012 at 11:52 PM, Alexander Lake <alex at thewinelake.com> wrote:
>>> export is deprecated? How is one supposed to do it instead?
>>> Maybe I need to get some examples...
>>>
>>> On 13 Jun 2012, at 18:34, Avi Marcus wrote:
>>>
>>>> bridge seems to be fairly manual for assembling complicated bridge strings.
>>>>
>>>> I think export works to all threads, but afaik, is deprecated.
>>>>
>>>> I've used group_confirm on these threads, it's fine. Just make sure
>>>> the group_confirm variable is in the right place...
>>>>
>>>> -Avi
>>>>
>>>>
>>>> On Wed, Jun 13, 2012 at 7:53 PM, Alex <alex at thewinelake.com> wrote:
>>>>> Just 'cos it's a separate thread doesn't mean that it couldn't have been
>>>>> initialised with channel variables from the spawning thread.
>>>>> I presume that exports don't work either.
>>>>> How about group_confirm?
>>>>> This is looking awfully like writing our own everything!
>>>>>> Hmm. re: enterprise originate remember that each is a separate thread,
>>>>>> so {} vars in one thread won't work on the other. You need to do
>>>>>> super-global<> ?to make that happen.
>>>>>>
>>>>>> I don't think we have a wiki page purely on building bridge strings yet...
>>>>>>
>>>>>> -Avi Marcus
>>>>>>
>>>>>> On Wed, Jun 13, 2012 at 4:30 PM, Alex<alex at thewinelake.com> ?wrote:
>>>>>>> Is there anything I should know about the combination of these two?
>>>>>>>
>>>>>>> Alex
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> -----
>>>>>> No virus found in this message.
>>>>>> Checked by AVG - www.avg.com
>>>>>> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From alex at thewinelake.com  Thu Jun 14 01:27:44 2012
From: alex at thewinelake.com (Alexander Lake)
Date: Wed, 13 Jun 2012 22:27:44 +0100
Subject: [Freeswitch-users] call_timeout and enterprise originate
In-Reply-To: <CALNZuwnHb8=xrfPx3EQ3bEmLAE+=2Zi3wnWJu3PB_WvdDPCh7Q@mail.gmail.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
	<4FD86DB0.4030508@thewinelake.com>
	<4FD872B6.4020107@thewinelake.com>
	<CAPf+86XBW2gs2N2zidqMFN7bMV5VuraUk7QG=opYGNzrki+4nQ@mail.gmail.com>
	<CAPf+86VpHXqnnJpCOmq7TNu4g1WBXyO8PpsuStftWYiwLwhVBA@mail.gmail.com>
	<4FD895E6.7000101@thewinelake.com>
	<CALNZuwnDT23d44oR8-jPfpu4dKh-mh9GSQJzBOFtO9bWSUa4Rw@mail.gmail.com>
	<4FD8C5A1.8050905@thewinelake.com>
	<CALNZuw=17NZK6oR65Yg=hnPmdMjBQMX2XWbvHcYB8xNZtO4Rnw@mail.gmail.com>
	<933886F1-C00C-4B73-9D76-049CB8D2E612@thewinelake.com>
	<CALNZuw=_SCGjNc_L=n-pyY0zwyynwOKYw101gKhSFGxZE75Lgw@mail.gmail.com>
	<90AD4E68-0314-4CDB-83FA-BE5AEF27AE4C@thewinelake.com>
	<CALNZuwnHb8=xrfPx3EQ3bEmLAE+=2Zi3wnWJu3PB_WvdDPCh7Q@mail.gmail.com>
Message-ID: <8EB44FDA-1E5D-46B9-AA46-E2CD154EC725@thewinelake.com>

So an example of a current script would be:

<include>
 <extension name="0095301">
  <condition field="destination_number" expression="^(301|441223900482)$">
   <action application="set" data="sb_routing=${sb_routing}&441223900482_301_pers_Alex"/>
   <action application="set" data="tenant_id=0095"/>
   <action application="set" data="b_ext=301"/>
   <action application="export" data="alert_info=http://hzhnov11.dmclub.net/generic/tones/Personal.wav"/>
   <action application="set" data="call_timeout=60"/>
   <action application="set" data="hangup_after_bridge=true"/>
   <action application="set" data="continue_on_fail=true"/>
   <action application="set" data="bridge_generate_comfort_noise=true"/>
   <action application="export" data="sip_h_tenant_id=0095"/>
   <action application="export" data="sip_h_company_name=Test Account"/>
   <action application="export" data="sip_h_extension=301"/>
   <action application="export" data="sip_h_extension_type=PersonalExtension"/>
   <action application="export" data="sip_h_extension_name=Alex"/>
   <action application="export" data="sip_h_prev_sb_routing=${sb_routing}"/>
   <action application="set" data="group_confirm_key=exec"/>
   <action application="export" data="whisper_msg=say:d m switchboard Call for Test Account personal"/>
   <action application="export" data="accept_msg=say:press 1 to accept"/>
   <action application="set" data="ringback=$${uk-ring}"/>
   <action application="set" data="transfer_ringback_=$${uk-ring}"/>
   <action application="set" data="ignore_early_media=true"/>
   <action application="set" data="group_confirm_file=lua confirmcall.lua"/>
   <action application="bridge" data="[tenant_id=0095,b_ext=301,accept_mode=Direct,sip_h_sb_routing=${sb_routing}&441223900482_301_pers_Alex,call_timeout=20]user/0095301wp@${domain_name},[tenant_id=0095,b_ext=301,accept_mode=Direct,call_timeout=20,origination_caller_id_number=00953010${ani}]sofia/internal/898000000000207775911500 at 1.2.3.4,"/>
   <action application="set" data="comment=pers_inhours_VM" />
   <action application="answer" />
   <action application="sleep" data="1000"/>
   <action application="voicemail" data="default ${domain_name} 0095301"/>
  </condition>
 </extension>
</include>

Note that I'm using , as a separator (having briefly used :_:)

As I understand it, we could replace this with:

<include>
 <extension name="0095301">
  <condition field="destination_number" expression="^(301|441223900482)$">
   <action application="set" data="tenant_id=0095"/>
   <action application="set" data="b_ext=301"/>
   <action application="set" data="call_timeout=60"/>
   <action application="set" data="hangup_after_bridge=true"/>
   <action application="set" data="continue_on_fail=true"/>
   <action application="set" data="bridge_generate_comfort_noise=true"/>
   <action application="set" data="ringback=$${uk-ring}"/>
   <action application="set" data="transfer_ringback_=$${uk-ring}"/>
   <action application="set" data="ignore_early_media=true"/>
   <action application="bridge" data="<sb_routing=${sb_routing}&441223900482_301_pers_Alex,tenant_id=0095,b_ext=301,alert_info=http://hzhnov11.dmclub.net/generic/tones/Personal.wav,call_timeout=60,hangup_after_bridge=true,continue_on_fail=true,bridge_generate_comfort_noise=true,sip_h_tenant_id=0095,sip_h_company_name=Test Account,sip_h_extension=301,sip_h_extension_type=PersonalExtension,sip_h_extension_name=Alex,sip_h_prev_sb_routing=${sb_routing},group_confirm_key=exec,whisper_msg=say:d m switchboard Call for Test Account personal,accept_msg=say:press 1 to accept,ringback=$${uk-ring},transfer_ringback_=$${uk-ring},ignore_early_media=true,group_confirm_file=lua confirmcall.lua>[tenant_id=0095,b_ext=301,accept_mode=Direct,sip_h_sb_routing=${sb_routing}&441223900482_301_pers_Alex,call_timeout=20]user/0095301wp@${domain_name}:_:[tenant_id=0095,b_ext=301,accept_mode=Direct,call_timeout=20,origination_caller_id_number=00953010${ani}]sofia/internal/898000000000207775911500 at 1.2.3.4,"/>
   <action application="set" data="comment=pers_inhours_VM" />
   <action application="answer" />
   <action application="sleep" data="1000"/>
   <action application="voicemail" data="default ${domain_name} 0095301"/>
  </condition>
 </extension>
</include>

I'm not sure which of the channel variables is only for B-leg and which are for both (and which are essentially detritus!), but I guess there's no harm in putting in a few surplus ones.

Will give this a try tomorrow morning....

Cheers!
Alex

On 13 Jun 2012, at 22:08, Avi Marcus wrote:

> [var1=myinfo] is per each bridge string. {var1=myinfo} applies to the
> entire thread, meaning if you have a :_: it won't carry over to that.
> <var1=myinfo> is a superglobal which is applied to ALL threads.
> 
> -Avi
> 
> 
> On Thu, Jun 14, 2012 at 12:04 AM, Alexander Lake <alex at thewinelake.com> wrote:
>> Ah. Is there a way to set common variables for the bridges?
>> 
>> On 13 Jun 2012, at 21:55, Avi Marcus wrote:
>> 
>>> Like you've been doing, mostly: setting them inside the bridge string.
>>> 
>>> -Avi
>>> 
>>> 
>>> On Wed, Jun 13, 2012 at 11:52 PM, Alexander Lake <alex at thewinelake.com> wrote:
>>>> export is deprecated? How is one supposed to do it instead?
>>>> Maybe I need to get some examples...
>>>> 
>>>> On 13 Jun 2012, at 18:34, Avi Marcus wrote:
>>>> 
>>>>> bridge seems to be fairly manual for assembling complicated bridge strings.
>>>>> 
>>>>> I think export works to all threads, but afaik, is deprecated.
>>>>> 
>>>>> I've used group_confirm on these threads, it's fine. Just make sure
>>>>> the group_confirm variable is in the right place...
>>>>> 
>>>>> -Avi
>>>>> 
>>>>> 
>>>>> On Wed, Jun 13, 2012 at 7:53 PM, Alex <alex at thewinelake.com> wrote:
>>>>>> Just 'cos it's a separate thread doesn't mean that it couldn't have been
>>>>>> initialised with channel variables from the spawning thread.
>>>>>> I presume that exports don't work either.
>>>>>> How about group_confirm?
>>>>>> This is looking awfully like writing our own everything!
>>>>>>> Hmm. re: enterprise originate remember that each is a separate thread,
>>>>>>> so {} vars in one thread won't work on the other. You need to do
>>>>>>> super-global<>  to make that happen.
>>>>>>> 
>>>>>>> I don't think we have a wiki page purely on building bridge strings yet...
>>>>>>> 
>>>>>>> -Avi Marcus
>>>>>>> 
>>>>>>> On Wed, Jun 13, 2012 at 4:30 PM, Alex<alex at thewinelake.com>  wrote:
>>>>>>>> Is there anything I should know about the combination of these two?
>>>>>>>> 
>>>>>>>> Alex
>>>>>>>> 
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>> 
>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>> 
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>> 
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> -----
>>>>>>> No virus found in this message.
>>>>>>> Checked by AVG - www.avg.com
>>>>>>> Version: 2012.0.2177 / Virus Database: 2433/5065 - Release Date: 06/12/12
>>>>>>> 
>>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> Join Us At ClueCon - Aug 7-9, 2012
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From alex at thewinelake.com  Thu Jun 14 01:29:46 2012
From: alex at thewinelake.com (Alexander Lake)
Date: Wed, 13 Jun 2012 22:29:46 +0100
Subject: [Freeswitch-users] NAT Router Recommendation For Large no of IP
	Phones Behind the Router
In-Reply-To: <2993C055-06FD-4814-8BD1-7E85B72B6759@mgtech.com>
References: <CAGvuWYaqsqO6N0y+5+pJ2zyA0UqS0iAFjQiXuaSxssH52GcdVQ@mail.gmail.com>
	<729B9245-CBCA-4F7F-9B71-9142E0ADA448@mgtech.com>
	<4FD7766D.8040208@puzzled.xs4all.nl>
	<CAGvuWYZnw05oKVcAArT9HaeqBagBanaeiG5ngdwNP4Mv79fXBQ@mail.gmail.com>
	<2993C055-06FD-4814-8BD1-7E85B72B6759@mgtech.com>
Message-ID: <DFA4D58C-BE2E-49F1-908A-671AEE448EF8@thewinelake.com>

Interesting stuff. What about those of us who run fs on boxes "in the cloud" - we run fail2ban, but perhaps we should also be considering something more sophisticated. Are there any good software stateful firewalls that might suit?

On 13 Jun 2012, at 22:07, Mario G wrote:

> A little late but important to say: When I had the Linksys routers my FS had constant attempts from around the world, mostly China to break into it. I used WireShark to see the attempts, the Linksys could not keep the out.. Since the Zyxel USG100 it has stopped. It has an outstanding firewall and the object oriented rules really helps with SIP. Whatever you select, keep security in mind.
>   
> On Jun 12, 2012, at 12:23 PM, Lloyd Aloysius wrote:
> 
>> I try with the pfsense 2.0. Specially BLF causing lots of problems.
>> 
>> ====
>> 
>> In the past .. I try the following
>> 
>> 1. Linksys WRT54GL + Tomato ... Good for a small office < 5 Extension
>> 2. Monowall with PC Engine Alix
>> 
>> 
>> Does any one test with a Cisco ASA 5505? 
>> 
>> 
>> 
>> On Tue, Jun 12, 2012 at 1:03 PM, Patrick Lists <freeswitch-list at puzzled.xs4all.nl> wrote:
>> On 12-06-12 18:20, Mario G wrote:
>> > I looked for a year to find something with no NAT problems, dual WAN,
>> > and auto fallback with load/balancing without causing FS any hiccups. I
>> > highly recommend Zyxel, I use a USG100 with SPI ALG turned ON (I read
>> > some people hate it) but it has been bullet proof. And the best part is
>> > there is practically nothing to tell FS about! And when one WAN (DSL)
>> > goes down the SIP traffic goes out the other one, even switches back
>> > automatically! Pair this up with FS and you'll never look back. Hope
>> > this helps.
>> 
>> Thanks for the info. Have you tried any Draytek Vigor routers like the
>> 2920? I have no experience with them but was told they work fine in dual
>> WAN setups.
>> 
>> Regards,
>> Patrick
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing 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 chris at gonumina.com  Thu Jun 14 02:06:13 2012
From: chris at gonumina.com (Chris Ferreira)
Date: Wed, 13 Jun 2012 18:06:13 -0400
Subject: [Freeswitch-users] NAT Router Recommendation For Large no of IP
 Phones Behind the Router
In-Reply-To: <CAGvuWYZnw05oKVcAArT9HaeqBagBanaeiG5ngdwNP4Mv79fXBQ@mail.gmail.com>
References: <CAGvuWYaqsqO6N0y+5+pJ2zyA0UqS0iAFjQiXuaSxssH52GcdVQ@mail.gmail.com>
	<729B9245-CBCA-4F7F-9B71-9142E0ADA448@mgtech.com>
	<4FD7766D.8040208@puzzled.xs4all.nl>
	<CAGvuWYZnw05oKVcAArT9HaeqBagBanaeiG5ngdwNP4Mv79fXBQ@mail.gmail.com>
Message-ID: <-7772935820556311152@unknownmsgid>

You may want to try pfsense 2.0.1 as there were a lot of bug fixes.

A client of mine uses a Cosco ASA 5500 Series with my Hosted FreeSWITCH
implementation. I'm not sure if there are any BLF issues as they don't use
the feature.




___________________
Mobile Reply

On Jun 12, 2012, at 3:26 PM, Lloyd Aloysius <lloyd.aloysius at gmail.com>
wrote:

I try with the pfsense 2.0. Specially BLF causing lots of problems.

====

In the past .. I try the following

1. Linksys WRT54GL + Tomato ... Good for a small office < 5 Extension
2. Monowall with PC Engine Alix


Does any one test with a Cisco ASA 5505?



On Tue, Jun 12, 2012 at 1:03 PM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 12-06-12 18:20, Mario G wrote:
> > I looked for a year to find something with no NAT problems, dual WAN,
> > and auto fallback with load/balancing without causing FS any hiccups. I
> > highly recommend Zyxel, I use a USG100 with SPI ALG turned ON (I read
> > some people hate it) but it has been bullet proof. And the best part is
> > there is practically nothing to tell FS about! And when one WAN (DSL)
> > goes down the SIP traffic goes out the other one, even switches back
> > automatically! Pair this up with FS and you'll never look back. Hope
> > this helps.
>
> Thanks for the info. Have you tried any Draytek Vigor routers like the
> 2920? I have no experience with them but was told they work fine in dual
> WAN setups.
>
> Regards,
> Patrick
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing 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 jack at livecall.com  Thu Jun 14 03:55:35 2012
From: jack at livecall.com (Jack)
Date: Wed, 13 Jun 2012 16:55:35 -0700
Subject: [Freeswitch-users] FSClient UA error
In-Reply-To: <CALn233qfV_onHuCJCXtm-Lp7V+DtLd-Fy=Ji3q3DV9ndfCjVtg@mail.gmail.com>
References: <4FCF8F0F.8010004@livecall.com>
	<1339023006371-7579490.post@n2.nabble.com>
	<4FCFE653.7020600@livecall.com>
	<CALn233qfV_onHuCJCXtm-Lp7V+DtLd-Fy=Ji3q3DV9ndfCjVtg@mail.gmail.com>
Message-ID: <4FD92877.4060309@livecall.com>

I found the problem....   Apparently the folder _C:\Documents and 
Settings\SpecificUser\Local Settings\Application 
Data\Mitch_Capper\FSClient.exe_Url_4r2lzffycnauuydk5l21oei2aljmfy4b_  
was not deleted when I uninstalled the previous version.
I uninstalled the current version and then manually deleted the folder 
and then reinstalled and it works great!

Thanks for your efforts on this.
jack



On 6/6/2012 4:54 PM, Mitch Capper wrote:
> You can use fs_cli to connect up to see what error is happening but
> make sure you have TLS disabled and change the ports in the sofia
> settings and see if that fixes it.
>
> ~Mitch
>
> On Wed, Jun 6, 2012 at 4:22 PM, Jack<jack at livecall.com>  wrote:
>> 5060 should be available, I do run other clients that use 5060  but make
>> sure they are all the way closed out before trying FSClient.
>> FS is running on a different box.
>>
>> On 6/6/2012 3:50 PM, Jeff Lenk wrote:
>>> Is something already using the ports that FSClient is trying to use? Are you
>>> running fs on this box? Or another sip client?
>>>
>>> --
>>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-UA-error-tp7579476p7579490.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun 14 05:01:11 2012
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Wed, 13 Jun 2012 18:01:11 -0700 (PDT)
Subject: [Freeswitch-users] lcr sql variable
In-Reply-To: <1339545639.80187.YahooMailClassic@web160306.mail.bf1.yahoo.com>
Message-ID: <1339635671.7620.YahooMailClassic@web160301.mail.bf1.yahoo.com>

Hi,

Anyone could help me...

--- On Tue, 6/12/12, FERNANDO VILLARROEL <fvillarroel at yahoo.com> wrote:

> From: FERNANDO VILLARROEL <fvillarroel at yahoo.com>
> Subject: [Freeswitch-users] lcr sql variable
> To: freeswitch-users at lists.freeswitch.org
> Date: Tuesday, June 12, 2012, 9:00 PM
> Hi all.
> 
> I need setup some variable like foo to my cdr from lcr :
> 
> <param name="export_fields" value="tarifa"/>
> 
> 
> <param name="custom_sql" value="select a.tarifa as tarifa
> ......
> 
> 
> Returning Dialstring
> [lcr_carrier=carrier1,lcr_rate=0.15000,nibble_account=1004,nibble_rate=0.15000,lowbal_amt=0,nobal_amt=0,tarifa=1]user/1006
> 
> My dialplan:
> 
> <extension name="Outbound to PSTN 11 Digits">
> ? ? ? ? ? ? ? ?
> <condition field="destination_number"
> expression="^(10[01][0-9])$">
> ? ? ? ? ? ? ? ?
> ? ? ? ? <action application="set"
> data="hangup_after_bridge=false"/>
> ? ? ? ? ? ? ? ?
> ? ? ? ? <action application="set"
> data="continue_on_fail=true"/>
> ? ? ? ? ? ? ? ?
> ? ? ? ? <action application="lcr"
> data="$1 pg_prefix2"/>
> 
> ? ? ? ? ? ? ? ?
> ? ? ? ? <action
> application="set"? data="tarifa=${tarifa}"/>
> ? ? ? ? ? ? ? ?
> ? ? ? ? <action application="bridge"
> data="${lcr_auto_route}"/>
> ? ? ? ? ? ? ? ?
> </condition>
> ? ? ? ? </extension>
> 
> But the log show :
> 
> mod_dptools.c:1294 sofia/internal/1004 at 192.168.1.108 SET
> [tarifa]=[UNDEF]
> 
> 
> So how i can get the value of the variable "tarifa" and set
> for my CDR?
> 
> Regards.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From mitch.capper at gmail.com  Thu Jun 14 06:16:40 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 13 Jun 2012 19:16:40 -0700
Subject: [Freeswitch-users] FSClient UA error
In-Reply-To: <4FD92877.4060309@livecall.com>
References: <4FCF8F0F.8010004@livecall.com>
	<1339023006371-7579490.post@n2.nabble.com>
	<4FCFE653.7020600@livecall.com>
	<CALn233qfV_onHuCJCXtm-Lp7V+DtLd-Fy=Ji3q3DV9ndfCjVtg@mail.gmail.com>
	<4FD92877.4060309@livecall.com>
Message-ID: <CALn233pQ1Le-az0Qa_EmmL9T8QgFC4hSV2XR60a8m1Ek=HTxpQ@mail.gmail.com>

Happy you got it working, that is your settings file you should _not_
have to delete it between versions (infact .net should automatically
upgrade it).  It must have somehow got corrupted causing the issue.
Stay tuned new version of FSClient coming soon with some nice new
features thanks to todays conference.

~Mitch

On Wed, Jun 13, 2012 at 4:55 PM, Jack <jack at livecall.com> wrote:
> I found the problem....?? Apparently the folder? C:\Documents and
> Settings\SpecificUser\Local Settings\Application
> Data\Mitch_Capper\FSClient.exe_Url_4r2lzffycnauuydk5l21oei2aljmfy4b? was not
> deleted when I uninstalled the previous version.
> I uninstalled the current version and then manually deleted the folder and
> then reinstalled and it works great!
>
> Thanks for your efforts on this.
> jack
>
>
>
>
> On 6/6/2012 4:54 PM, Mitch Capper wrote:
>
> You can use fs_cli to connect up to see what error is happening but
> make sure you have TLS disabled and change the ports in the sofia
> settings and see if that fixes it.
>
> ~Mitch
>
> On Wed, Jun 6, 2012 at 4:22 PM, Jack <jack at livecall.com> wrote:
>
> 5060 should be available, I do run other clients that use 5060 ?but make
> sure they are all the way closed out before trying FSClient.
> FS is running on a different box.
>
> On 6/6/2012 3:50 PM, Jeff Lenk wrote:
>
> Is something already using the ports that FSClient is trying to use? Are you
> running fs on this box? Or another sip client?
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FSClient-UA-error-tp7579476p7579490.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From ocset at the800group.com  Thu Jun 14 06:27:50 2012
From: ocset at the800group.com (ocset)
Date: Thu, 14 Jun 2012 10:27:50 +0800
Subject: [Freeswitch-users] Brute-force attack
Message-ID: <4FD94C26.10800@the800group.com>

Hi

I have deployed Freeswiitch on windows 7 and since there is no fail2ban 
on windows, I was wondering what the real risk is with opening it up to 
the internet. If I was to ensure that all users and passwords were 
extremely difficult to guess (passwords like "2$53E_d7?^2!3s$"), what 
are the risks that I am exposing myself to? Is there a type of DoS for 
voip where hackers can just flood my system with requests simply to be 
malicious?

There are VB windows scripts available that emulate what fail2ban does 
on Linux but I was just wondering whether I really need to implement 
this level of security if I can control the password complexity in 
Freeswitch.

Thanks
O



From bobc at devassert.com  Thu Jun 14 06:41:03 2012
From: bobc at devassert.com (Bob Coleman)
Date: Thu, 14 Jun 2012 02:41:03 +0000
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <4FD94C26.10800@the800group.com>
References: <4FD94C26.10800@the800group.com>
Message-ID: <BAY170-W898DC3644D39A2FEAB331FCCF40@phx.gbl>


If you are only connecting to specified SIP providers, then you could specify those endpoints at the firewall level, which would help.
Regards
Bob> Date: Thu, 14 Jun 2012 10:27:50 +0800
> From: ocset at the800group.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Brute-force attack
> 
> Hi
> 
> I have deployed Freeswiitch on windows 7 and since there is no fail2ban 
> on windows, I was wondering what the real risk is with opening it up to 
> the internet. If I was to ensure that all users and passwords were 
> extremely difficult to guess (passwords like "2$53E_d7?^2!3s$"), what 
> are the risks that I am exposing myself to? Is there a type of DoS for 
> voip where hackers can just flood my system with requests simply to be 
> malicious?
> 
> There are VB windows scripts available that emulate what fail2ban does 
> on Linux but I was just wondering whether I really need to implement 
> this level of security if I can control the password complexity in 
> Freeswitch.
> 
> Thanks
> O
 		 	   		  
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From ocset at the800group.com  Thu Jun 14 07:04:41 2012
From: ocset at the800group.com (ocset)
Date: Thu, 14 Jun 2012 11:04:41 +0800
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <BAY170-W898DC3644D39A2FEAB331FCCF40@phx.gbl>
References: <4FD94C26.10800@the800group.com>
	<BAY170-W898DC3644D39A2FEAB331FCCF40@phx.gbl>
Message-ID: <4FD954C9.8030607@the800group.com>

Thanks Bob

The reason for opening the port would be to allow a user running a "Viop 
phone" app on their smartphone to register with Freeswitch, so that 
calls can be transferred to them when they are away from the office.

Regards
O

On 14/06/12 10:41, Bob Coleman wrote:
> If you are only connecting to specified SIP providers, then you 
> could specify those endpoints at the firewall level, which would help.
>
> Regards
>
> Bob
> > Date: Thu, 14 Jun 2012 10:27:50 +0800
> > From: ocset at the800group.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: [Freeswitch-users] Brute-force attack
> >
> > Hi
> >
> > I have deployed Freeswiitch on windows 7 and since there is no fail2ban
> > on windows, I was wondering what the real risk is with opening it up to
> > the internet. If I was to ensure that all users and passwords were
> > extremely difficult to guess (passwords like "2$53E_d7?^2!3s$"), what
> > are the risks that I am exposing myself to? Is there a type of DoS for
> > voip where hackers can just flood my system with requests simply to be
> > malicious?
> >
> > There are VB windows scripts available that emulate what fail2ban does
> > on Linux but I was just wondering whether I really need to implement
> > this level of security if I can control the password complexity in
> > Freeswitch.
> >
> > Thanks
> > O
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jaybinks at gmail.com  Thu Jun 14 07:23:23 2012
From: jaybinks at gmail.com (jay binks)
Date: Thu, 14 Jun 2012 13:23:23 +1000
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <4FD94C26.10800@the800group.com>
References: <4FD94C26.10800@the800group.com>
Message-ID: <CAO47Hpcuh7OFEgjyjAZx9bq9MrPPscqhx9159aaU03rv57=P9A@mail.gmail.com>

Strong passwords are a great start, but fail2ban does a little more than
this.

you could move off port 5060 to something un-conventional, meaning your
less likely to get scanned / brute forced.

Jay

On 14 June 2012 12:27, ocset <ocset at the800group.com> wrote:

> Hi
>
> I have deployed Freeswiitch on windows 7 and since there is no fail2ban
> on windows, I was wondering what the real risk is with opening it up to
> the internet. If I was to ensure that all users and passwords were
> extremely difficult to guess (passwords like "2$53E_d7?^2!3s$"), what
> are the risks that I am exposing myself to? Is there a type of DoS for
> voip where hackers can just flood my system with requests simply to be
> malicious?
>
> There are VB windows scripts available that emulate what fail2ban does
> on Linux but I was just wondering whether I really need to implement
> this level of security if I can control the password complexity in
> Freeswitch.
>
> Thanks
> O
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

Jay
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From tang.du at hotmail.com  Thu Jun 14 07:51:34 2012
From: tang.du at hotmail.com (tangdu)
Date: Wed, 13 Jun 2012 20:51:34 -0700 (PDT)
Subject: [Freeswitch-users] =?utf-8?q?Error_message_=22_=5BERR=5D_switch?=
 =?utf-8?q?=5Fcpp=2Ecpp=3A48_Cannot_queue_any_more_events=E2=80=9C?=
Message-ID: <1339645894596-7579817.post@n2.nabble.com>


   I use Mod event to fill channel cariables  in my mysql table?When
freeswitch runing some time?I got a lot of the following error messages:
[ERR] switch_cpp.cpp:48 Cannot queue any more events?
  This messages telling me that i have consumed too many events in my event
consumer object.   it has reached its max. Then the channel cariables can
not be filled in mysql?
    How can I solve this?  
Thanks?
    


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-message-ERR-switch-cpp-cpp-48-Cannot-queue-any-more-events-tp7579817.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Thu Jun 14 08:39:15 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 14 Jun 2012 07:39:15 +0300
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <CAO47Hpcuh7OFEgjyjAZx9bq9MrPPscqhx9159aaU03rv57=P9A@mail.gmail.com>
References: <4FD94C26.10800@the800group.com>
	<CAO47Hpcuh7OFEgjyjAZx9bq9MrPPscqhx9159aaU03rv57=P9A@mail.gmail.com>
Message-ID: <CALNZuwkF+Hw0HfpCONq0u8OJW29=tfTeAcudLYS69K04qAX2Ww@mail.gmail.com>

That's not necessarily the best kind of password... see http://xkcd.com/936/
and then http://tech.dropbox.com/?p=165

-Avi


On Thu, Jun 14, 2012 at 6:23 AM, jay binks <jaybinks at gmail.com> wrote:
> Strong passwords are a great start, but fail2ban does a little more than
> this.
>
> you could move off port 5060 to something un-conventional, meaning your
less
> likely to get scanned / brute forced.
>
> Jay
>
> On 14 June 2012 12:27, ocset <ocset at the800group.com> wrote:
>>
>> Hi
>>
>> I have deployed Freeswiitch on windows 7 and since there is no fail2ban
>> on windows, I was wondering what the real risk is with opening it up to
>> the internet. If I was to ensure that all users and passwords were
>> extremely difficult to guess (passwords like "2$53E_d7?^2!3s$"), what
>> are the risks that I am exposing myself to? Is there a type of DoS for
>> voip where hackers can just flood my system with requests simply to be
>> malicious?
>>
>> There are VB windows scripts available that emulate what fail2ban does
>> on Linux but I was just wondering whether I really need to implement
>> this level of security if I can control the password complexity in
>> Freeswitch.
>>
>> Thanks
>> O
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jack at livecall.com  Thu Jun 14 09:43:57 2012
From: jack at livecall.com (Jack)
Date: Wed, 13 Jun 2012 22:43:57 -0700
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <4FD94C26.10800@the800group.com>
References: <4FD94C26.10800@the800group.com>
Message-ID: <4FD97A1D.1050008@livecall.com>

I have run into this same problem with win2003.  You may need to check 
to see if windows 7 has IPSEC service.
If so, you can set up a blocked list in ipsec that you can add ip 
addresses to and windows won't let them into your machine.

Click on Start menu
choose Administrative Tools
choose Services
Find IPSEC Services - double click to open properties - make sure it is 
set to Automatic and started.

You can create the block list by issuing the following commands from a 
command window:

netsh ipsec static add filteraction name=Block action=block
netsh ipsec static add filter filterlist=BlockList 
srcaddr=192.168.192.100 dstaddr=me
netsh ipsec static add policy name=Block assign=yes activatedefaultrule=no
netsh ipsec static add rule name=BlockList policy=Block 
filterlist=BlockList filteraction=Block
netsh ipsec static delete filter filterlist=BlockList 
srcaddr=192.168.192.100 dstaddr=Me


create a directory called blockip

now in notepad  create blockip.bat with the following line in it:
netsh ipsec static add filter filterlist=BlockList srcaddr=%1 dstaddr=me

Now , in notepad, createunblockip.bat with the following line in it:
netsh ipsec static delete filter filterlist=BlockList srcaddr=%1 dstaddr=me


to block ip address 123.123.123.123  type blockip 123.123.123.123      
at a command prompt.

to unblock ip address 123.123.123.123  type unblockip 123.123.123.123 
      at a command prompt.

You can use xml_curl to keep track of hit frequency and do the blocking 
for you.


hope that helps....
jack


On 6/13/2012 7:27 PM, ocset wrote:
> Hi
>
> I have deployed Freeswiitch on windows 7 and since there is no fail2ban
> on windows, I was wondering what the real risk is with opening it up to
> the internet. If I was to ensure that all users and passwords were
> extremely difficult to guess (passwords like "2$53E_d7?^2!3s$"), what
> are the risks that I am exposing myself to? Is there a type of DoS for
> voip where hackers can just flood my system with requests simply to be
> malicious?
>
> There are VB windows scripts available that emulate what fail2ban does
> on Linux but I was just wondering whether I really need to implement
> this level of security if I can control the password complexity in
> Freeswitch.
>
> Thanks
> O
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shaheryarkh at googlemail.com  Thu Jun 14 09:46:16 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Thu, 14 Jun 2012 07:46:16 +0200
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <CALNZuwkF+Hw0HfpCONq0u8OJW29=tfTeAcudLYS69K04qAX2Ww@mail.gmail.com>
References: <4FD94C26.10800@the800group.com>
	<CAO47Hpcuh7OFEgjyjAZx9bq9MrPPscqhx9159aaU03rv57=P9A@mail.gmail.com>
	<CALNZuwkF+Hw0HfpCONq0u8OJW29=tfTeAcudLYS69K04qAX2Ww@mail.gmail.com>
Message-ID: <CAFZQphzkbg5_4tH0YmYLoJ4Ft_rN07nLHBWZMkcyWVbczAUe1A@mail.gmail.com>

I would strongly suggest to move your production system to Linux, which is
by far secure and controllable then Windows. Right now, if somebody does
not breaks into your voip setup using some bruteforce / DOS attack, s/he
can still exploit some hole in Windows to crack your security. Windows is
simply not secure enough to production grade performance.

Thank you.


On Thu, Jun 14, 2012 at 6:39 AM, Avi Marcus <avi at avimarcus.net> wrote:

> That's not necessarily the best kind of password... see
> http://xkcd.com/936/ and then http://tech.dropbox.com/?p=165
>
> -Avi
>
>
>
> On Thu, Jun 14, 2012 at 6:23 AM, jay binks <jaybinks at gmail.com> wrote:
> > Strong passwords are a great start, but fail2ban does a little more than
> > this.
> >
> > you could move off port 5060 to something un-conventional, meaning your
> less
> > likely to get scanned / brute forced.
> >
> > Jay
> >
> > On 14 June 2012 12:27, ocset <ocset at the800group.com> wrote:
> >>
> >> Hi
> >>
> >> I have deployed Freeswiitch on windows 7 and since there is no fail2ban
> >> on windows, I was wondering what the real risk is with opening it up to
> >> the internet. If I was to ensure that all users and passwords were
> >> extremely difficult to guess (passwords like "2$53E_d7?^2!3s$"), what
> >> are the risks that I am exposing myself to? Is there a type of DoS for
> >> voip where hackers can just flood my system with requests simply to be
> >> malicious?
> >>
> >> There are VB windows scripts available that emulate what fail2ban does
> >> on Linux but I was just wondering whether I really need to implement
> >> this level of security if I can control the password complexity in
> >> Freeswitch.
> >>
> >> Thanks
> >> O
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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
> >
> > Jay
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From ronmccar at gmail.com  Thu Jun 14 01:12:30 2012
From: ronmccar at gmail.com (Ron McCarthy)
Date: Wed, 13 Jun 2012 14:12:30 -0700
Subject: [Freeswitch-users] SCCP Question
Message-ID: <CAMQkVDKNJbQcENH-23+QQrAmTxM2=iMVgxnZtw9syEhdnfO7_g@mail.gmail.com>

Hi List,

Has anyone been running SCCP with a larger number of phones? Im looking to
deploy like 75+ phones and I want to keep SCCP so I don't have to upgrade
them and for the SLA, some phones also have no SIP software for them so im
forced to keep SCCP. Does anyone have any experience with this? From what
ive read the SCCP support works and works well, im just worried about
trying to run this many phones and if im missing any sort of issues that
could come up. I would like to use FreeSwitch instead of Asterisk for this,
just worried about functionality and the stability of it.

Thanks!
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From esander83 at hushmail.com  Thu Jun 14 08:31:51 2012
From: esander83 at hushmail.com (esander83 at hushmail.com)
Date: Wed, 13 Jun 2012 23:31:51 -0500
Subject: [Freeswitch-users] Does mod_fifo not take in
	effective_caller_id_name?
Message-ID: <20120614043151.7A69A14DBD8@smtp.hushmail.com>

Hi All,
I have this basic dialplan:
I can see the call get logged via the first extension but it's not
passing the info to modfifo.  I can remove the 2nd extension and put a
basic: 
and the caller id name info is displayed on the phones but not when
routing through mod_fifo.  I have tried both set/export and it still
doesn't seem to work.
Is this correct?
Thanks,eric_s
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From peter.olsson at visionutveckling.se  Thu Jun 14 10:16:04 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 14 Jun 2012 06:16:04 +0000
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <CAFZQphzkbg5_4tH0YmYLoJ4Ft_rN07nLHBWZMkcyWVbczAUe1A@mail.gmail.com>
References: <4FD94C26.10800@the800group.com>
	<CAO47Hpcuh7OFEgjyjAZx9bq9MrPPscqhx9159aaU03rv57=P9A@mail.gmail.com>
	<CALNZuwkF+Hw0HfpCONq0u8OJW29=tfTeAcudLYS69K04qAX2Ww@mail.gmail.com>,
	<CAFZQphzkbg5_4tH0YmYLoJ4Ft_rN07nLHBWZMkcyWVbczAUe1A@mail.gmail.com>
Message-ID: <D6D8040D-2350-4B59-9FC0-90B230537D25@visionutveckling.se>

I use both Windows and Linux systems. As long as you know how to manage both systems, there is not a big difference when it comes to exploits and general security (not anymore anyway, if you use current versions). I would say that the biggest issue here is the knowledge of the people managing the systems. And it's usually more secure to manage a system that you know, then a system you don't know much about, even though that system is considered to be more secure.

Lots has happened since Windows 95 :)

Anyway, for this kind of setup I would also prefer Linux, but mostly for the possibilites with fail2ban etc, which doesn't exist on Windows. I'm thinking of writiling something similar for Windows, hopefully I get som time for that soon...

/Peter

14 jun 2012 kl. 07:51 skrev "Muhammad Shahzad" <shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>>:

I would strongly suggest to move your production system to Linux, which is by far secure and controllable then Windows. Right now, if somebody does not breaks into your voip setup using some bruteforce / DOS attack, s/he can still exploit some hole in Windows to crack your security. Windows is simply not secure enough to production grade performance.

Thank you.


On Thu, Jun 14, 2012 at 6:39 AM, Avi Marcus <avi at avimarcus.net<mailto:avi at avimarcus.net>> wrote:
That's not necessarily the best kind of password... see http://xkcd.com/936/ and then http://tech.dropbox.com/?p=165

-Avi



On Thu, Jun 14, 2012 at 6:23 AM, jay binks <jaybinks at gmail.com<mailto:jaybinks at gmail.com>> wrote:
> Strong passwords are a great start, but fail2ban does a little more than
> this.
>
> you could move off port 5060 to something un-conventional, meaning your less
> likely to get scanned / brute forced.
>
> Jay
>
> On 14 June 2012 12:27, ocset <ocset at the800group.com<mailto:ocset at the800group.com>> wrote:
>>
>> Hi
>>
>> I have deployed Freeswiitch on windows 7 and since there is no fail2ban
>> on windows, I was wondering what the real risk is with opening it up to
>> the internet. If I was to ensure that all users and passwords were
>> extremely difficult to guess (passwords like "2$53E_d7?^2!3s$"), what
>> are the risks that I am exposing myself to? Is there a type of DoS for
>> voip where hackers can just flood my system with requests simply to be
>> malicious?
>>
>> There are VB windows scripts available that emulate what fail2ban does
>> on Linux but I was just wondering whether I really need to implement
>> this level of security if I can control the password complexity in
>> Freeswitch.
>>
>> Thanks
>> O
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> 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
>
>
>
>
> --
> Sincerely
>
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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




--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com<mailto:shari_786pk at hotmail.com>
Email: shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>
!DSPAM:4fd978c432761360223007!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


!DSPAM:4fd978c432761360223007!


From shaheryarkh at googlemail.com  Thu Jun 14 10:33:45 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Thu, 14 Jun 2012 08:33:45 +0200
Subject: [Freeswitch-users] SCCP Question
In-Reply-To: <CAMQkVDKNJbQcENH-23+QQrAmTxM2=iMVgxnZtw9syEhdnfO7_g@mail.gmail.com>
References: <CAMQkVDKNJbQcENH-23+QQrAmTxM2=iMVgxnZtw9syEhdnfO7_g@mail.gmail.com>
Message-ID: <CAFZQphxb7BH3DBP6LH+BeHM4W2GcHvteb9WVtUXBeREDG3_S0Q@mail.gmail.com>

Ah, i misread your email at first, thought you are setting up 75+K SCCP
phones, while its just 75+ phones. :-)

Anyways, yes FreeSWITCH works pretty good for SCCP, especially if you have
previously used asterisk with them, otherwise you may run into problem with
some phones which have very old firmware, you probably have to upgrade
their firmware.

I once setup a 60 SCCP phone setup with FreeSWITCH and it worked without
much problem. So go on and if you run into problem you can contact this
mailing list for help. ;-)

Thank you.


On Wed, Jun 13, 2012 at 11:12 PM, Ron McCarthy <ronmccar at gmail.com> wrote:

> Hi List,
>
> Has anyone been running SCCP with a larger number of phones? Im looking to
> deploy like 75+ phones and I want to keep SCCP so I don't have to upgrade
> them and for the SLA, some phones also have no SIP software for them so im
> forced to keep SCCP. Does anyone have any experience with this? From what
> ive read the SCCP support works and works well, im just worried about
> trying to run this many phones and if im missing any sort of issues that
> could come up. I would like to use FreeSwitch instead of Asterisk for this,
> just worried about functionality and the stability of it.
>
> Thanks!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From saami_mh at ymail.com  Thu Jun 14 11:09:45 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Thu, 14 Jun 2012 00:09:45 -0700 (PDT)
Subject: [Freeswitch-users] how to configuring nibblebill module to only
	billing every amount of heartbeat?
Message-ID: <1339657785.30594.YahooMailNeo@web120105.mail.ne1.yahoo.com>

hi,
in "nibblebill" module of freeswitch,?billing will only occur at the every 60 second by set like:(?<action application="nibblebill" data="heartbeat 60"/>)
but while making call and call is answered ?the minimmun amount of cash is reduced and also ?billing is done every 60 seconds,
(in addition of billing every 60 seconds the minimmum amount of cash also reduced at the first time the call is answered)
is it possible to set somethings so that don't reduce (billing) ?the minimum amount of cash at the same time call is answered and only billing every 60 second?
??
thanks so much
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From saami_mh at ymail.com  Thu Jun 14 11:49:53 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Thu, 14 Jun 2012 00:49:53 -0700 (PDT)
Subject: [Freeswitch-users] how to configuring nibblebill module to only
	billing every amount of heartbeat?
In-Reply-To: <1339657785.30594.YahooMailNeo@web120105.mail.ne1.yahoo.com>
References: <1339657785.30594.YahooMailNeo@web120105.mail.ne1.yahoo.com>
Message-ID: <1339660193.19106.YahooMailNeo@web120106.mail.ne1.yahoo.com>



add descriptions as:
when  Set the initial answer time the cash is reduced , and don't want the cash is reduced at the time of call answered,but 
want to billing cash (reduced it) until the time is reached to the amount of heartbeat.
-------------------------------------------------------------------------------------------------------------------------
From:Samira Mh <saami_mh at ymail.com>

To: Free SWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Thursday, June 14, 2012 11:39 AM
Subject: [Freeswitch-users] how to configuring nibblebill module to only billing every amount of heartbeat?
 

hi,
in "nibblebill" module of freeswitch,?billing will only occur at the every 60 second by set like:(?<action application="nibblebill" data="heartbeat 60"/>)
but while making call and call is answered ?the minimmun amount of cash is reduced and also ?billing is done every 60 seconds,
(in addition of billing every 60 seconds the minimmum amount of cash also reduced at the first time the call is answered)
is it possible to set somethings so that don't reduce (billing) ?the minimum amount of cash at the same time call is answered and only billing every 60 second?
??
thanks so much
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From mi.ke at null.net  Thu Jun 14 12:04:14 2012
From: mi.ke at null.net (Mi Ke)
Date: Thu, 14 Jun 2012 04:04:14 -0400
Subject: [Freeswitch-users] separate channel playback
Message-ID: <20120614080415.50590@gmx.com>

Dear FreeSwitchers,

 Is it possible to select a particular channel (left or right) when I play stereo audio file to the call leg or conference?

 Thanks / MiKe
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From vvenkatar at gmail.com  Thu Jun 14 12:07:13 2012
From: vvenkatar at gmail.com (Venkatesh)
Date: Thu, 14 Jun 2012 01:07:13 -0700
Subject: [Freeswitch-users] Collect DTMF via mod_event_socket
Message-ID: <CAP_69ms3WGg84zobjQ1P-+qqPnU_e6k+LqQDkq_WztoMi+mi2A@mail.gmail.com>

Hello !

I have been looking at the documentation but am unable to find it. I wanted
to know the correct syntax I need to issue via mod_event_socket if I want
to collect DTMF and report back the digits entered to my application. Great
if anybody can provide me with pointers.

Venkatesh
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From bobc at devassert.com  Thu Jun 14 12:18:10 2012
From: bobc at devassert.com (Bob Coleman)
Date: Thu, 14 Jun 2012 08:18:10 +0000
Subject: [Freeswitch-users] Collect DTMF via mod_event_socket
In-Reply-To: <CAP_69ms3WGg84zobjQ1P-+qqPnU_e6k+LqQDkq_WztoMi+mi2A@mail.gmail.com>
References: <CAP_69ms3WGg84zobjQ1P-+qqPnU_e6k+LqQDkq_WztoMi+mi2A@mail.gmail.com>
Message-ID: <BAY170-W13975A488C43B546F127DBECCF40@phx.gbl>


Hi Venkatesh, You could use this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits or this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read On the main page you go to the docs link at the top, then everything is linked off the left hand side Bob
 Date: Thu, 14 Jun 2012 01:07:13 -0700
From: vvenkatar at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Collect DTMF via mod_event_socket

Hello !
I have been looking at the documentation but am unable to find it. I wanted to know the correct syntax I need to issue via mod_event_socket if I want to collect DTMF and report back the digits entered to my application. Great if anybody can provide me with pointers.

Venkatesh
 		 	   		  
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From anton.jugatsu at gmail.com  Thu Jun 14 12:27:23 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Thu, 14 Jun 2012 12:27:23 +0400
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <D6D8040D-2350-4B59-9FC0-90B230537D25@visionutveckling.se>
References: <4FD94C26.10800@the800group.com>
	<CAO47Hpcuh7OFEgjyjAZx9bq9MrPPscqhx9159aaU03rv57=P9A@mail.gmail.com>
	<CALNZuwkF+Hw0HfpCONq0u8OJW29=tfTeAcudLYS69K04qAX2Ww@mail.gmail.com>
	<CAFZQphzkbg5_4tH0YmYLoJ4Ft_rN07nLHBWZMkcyWVbczAUe1A@mail.gmail.com>
	<D6D8040D-2350-4B59-9FC0-90B230537D25@visionutveckling.se>
Message-ID: <CANBhyPBPyGdwQhvZEbcK7iVFQEKwgC0uaD9qOb9udBEDiLR0eQ@mail.gmail.com>

I would suggest to create separate profile for remote workers. For example,
external-road-warrios. So you can play with ext-rtp-ip and ext-sip-ip.

2012/6/14 Peter Olsson <peter.olsson at visionutveckling.se>

> I use both Windows and Linux systems. As long as you know how to manage
> both systems, there is not a big difference when it comes to exploits and
> general security (not anymore anyway, if you use current versions). I would
> say that the biggest issue here is the knowledge of the people managing the
> systems. And it's usually more secure to manage a system that you know,
> then a system you don't know much about, even though that system is
> considered to be more secure.
>
> Lots has happened since Windows 95 :)
>
> Anyway, for this kind of setup I would also prefer Linux, but mostly for
> the possibilites with fail2ban etc, which doesn't exist on Windows. I'm
> thinking of writiling something similar for Windows, hopefully I get som
> time for that soon...
>
> /Peter
>
> 14 jun 2012 kl. 07:51 skrev "Muhammad Shahzad" <shaheryarkh at googlemail.com
> <mailto:shaheryarkh at googlemail.com>>:
>
> I would strongly suggest to move your production system to Linux, which is
> by far secure and controllable then Windows. Right now, if somebody does
> not breaks into your voip setup using some bruteforce / DOS attack, s/he
> can still exploit some hole in Windows to crack your security. Windows is
> simply not secure enough to production grade performance.
>
> Thank you.
>
>
> On Thu, Jun 14, 2012 at 6:39 AM, Avi Marcus <avi at avimarcus.net<mailto:
> avi at avimarcus.net>> wrote:
> That's not necessarily the best kind of password... see
> http://xkcd.com/936/ and then http://tech.dropbox.com/?p=165
>
> -Avi
>
>
>
> On Thu, Jun 14, 2012 at 6:23 AM, jay binks <jaybinks at gmail.com<mailto:
> jaybinks at gmail.com>> wrote:
> > Strong passwords are a great start, but fail2ban does a little more than
> > this.
> >
> > you could move off port 5060 to something un-conventional, meaning your
> less
> > likely to get scanned / brute forced.
> >
> > Jay
> >
> > On 14 June 2012 12:27, ocset <ocset at the800group.com<mailto:
> ocset at the800group.com>> wrote:
> >>
> >> Hi
> >>
> >> I have deployed Freeswiitch on windows 7 and since there is no fail2ban
> >> on windows, I was wondering what the real risk is with opening it up to
> >> the internet. If I was to ensure that all users and passwords were
> >> extremely difficult to guess (passwords like "2$53E_d7?^2!3s$"), what
> >> are the risks that I am exposing myself to? Is there a type of DoS for
> >> voip where hackers can just flood my system with requests simply to be
> >> malicious?
> >>
> >> There are VB windows scripts available that emulate what fail2ban does
> >> on Linux but I was just wondering whether I really need to implement
> >> this level of security if I can control the password complexity in
> >> Freeswitch.
> >>
> >> Thanks
> >> O
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> 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
> >
> >
> >
> >
> > --
> > Sincerely
> >
> > Jay
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > 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
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com<mailto:shari_786pk at hotmail.com>
> Email: shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>
> !DSPAM:4fd978c432761360223007!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>
> !DSPAM:4fd978c432761360223007!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.olsson at visionutveckling.se  Thu Jun 14 13:04:45 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 14 Jun 2012 09:04:45 +0000
Subject: [Freeswitch-users] Collect DTMF via mod_event_socket
Message-ID: <1FFF97C269757C458224B7C895F35F15107E3B@cantor.std.visionutv.se>

Or just listen for DTMF events, and act on those. I personally find that's the easiest approach.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Bob Coleman
Skickat: den 14 juni 2012 10:18
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Collect DTMF via mod_event_socket

Hi Venkatesh,

You could use this:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits

or this:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read

On the main page you go to the docs link at the top, then everything is linked off the left hand side

Bob

________________________________
Date: Thu, 14 Jun 2012 01:07:13 -0700
From: vvenkatar at gmail.com<mailto:vvenkatar at gmail.com>
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Collect DTMF via mod_event_socket

Hello !

I have been looking at the documentation but am unable to find it. I wanted to know the correct syntax I need to issue via mod_event_socket if I want to collect DTMF and report back the digits entered to my application. Great if anybody can provide me with pointers.

Venkatesh
!DSPAM:4fd99c9a32769547714299!
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From erkan at speedingtrade.com  Thu Jun 14 13:10:23 2012
From: erkan at speedingtrade.com (=?utf-8?Q?Erkan_=C3=9Cnl=C3=BC_=28simalube=29?=)
Date: Thu, 14 Jun 2012 12:10:23 +0300
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <CAFZQphzkbg5_4tH0YmYLoJ4Ft_rN07nLHBWZMkcyWVbczAUe1A@mail.gmail.com>
References: <4FD94C26.10800@the800group.com>	<CAO47Hpcuh7OFEgjyjAZx9bq9MrPPscqhx9159aaU03rv57=P9A@mail.gmail.com>	<CALNZuwkF+Hw0HfpCONq0u8OJW29=tfTeAcudLYS69K04qAX2Ww@mail.gmail.com>
	<CAFZQphzkbg5_4tH0YmYLoJ4Ft_rN07nLHBWZMkcyWVbczAUe1A@mail.gmail.com>
Message-ID: <009501cd4a0d$87f89d20$97e9d760$@speedingtrade.com>

Our System runs on Windows on a hosted dedicated server since 4 years without problems.

So I think the choice is not Linux or Windows, the right choice is where you have your knowledge. Windows is also absolute secure if you know which settings are important for a good security.

 

Kind regards

Erkan

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad
Sent: Thursday, June 14, 2012 8:46 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Brute-force attack

 

I would strongly suggest to move your production system to Linux, which is by far secure and controllable then Windows. Right now, if somebody does not breaks into your voip setup using some bruteforce / DOS attack, s/he can still exploit some hole in Windows to crack your security. Windows is simply not secure enough to production grade performance.

Thank you.



On Thu, Jun 14, 2012 at 6:39 AM, Avi Marcus <avi at avimarcus.net> wrote:

That's not necessarily the best kind of password... see http://xkcd.com/936/ and then http://tech.dropbox.com/?p=165

-Avi




On Thu, Jun 14, 2012 at 6:23 AM, jay binks <jaybinks at gmail.com> wrote:
> Strong passwords are a great start, but fail2ban does a little more than
> this.
>
> you could move off port 5060 to something un-conventional, meaning your less
> likely to get scanned / brute forced.
>
> Jay
>
> On 14 June 2012 12:27, ocset <ocset at the800group.com> wrote:
>>
>> Hi
>>
>> I have deployed Freeswiitch on windows 7 and since there is no fail2ban
>> on windows, I was wondering what the real risk is with opening it up to
>> the internet. If I was to ensure that all users and passwords were
>> extremely difficult to guess (passwords like "2$53E_d7?^2!3s$"), what
>> are the risks that I am exposing myself to? Is there a type of DoS for
>> voip where hackers can just flood my system with requests simply to be
>> malicious?
>>
>> There are VB windows scripts available that emulate what fail2ban does
>> on Linux but I was just wondering whether I really need to implement
>> this level of security if I can control the password complexity in
>> Freeswitch.
>>
>> Thanks
>> O
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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




-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com

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From alex at thewinelake.com  Thu Jun 14 13:22:01 2012
From: alex at thewinelake.com (Alex)
Date: Thu, 14 Jun 2012 10:22:01 +0100
Subject: [Freeswitch-users] call_timeout and enterprise originate
In-Reply-To: <4FD8C5A1.8050905@thewinelake.com>
References: <CALaOL+j5NtyUCinr98ymdCC7RyVVrtqT9S8OPTJhiq=xyvbd+Q@mail.gmail.com>
	<CAMwZ-t+owj1UggkVLE9k5LzrbiWh2D156b4A0dQOs3GpeN=a6Q@mail.gmail.com>
	<CALaOL+iGEd+_Vdvn-r0ubCmCACPNHZcoFmT3yNwCQVrgqnzSdg@mail.gmail.com>
	<4FD867F7.6090701@thewinelake.com>
	<CAPf+86V+Lq8JbC5zG1hN7GN1PN2p=dwPi2sdaSFWN-kD+ta3XQ@mail.gmail.com>
	<4FD86DB0.4030508@thewinelake.com>
	<4FD872B6.4020107@thewinelake.com>
	<CAPf+86XBW2gs2N2zidqMFN7bMV5VuraUk7QG=opYGNzrki+4nQ@mail.gmail.com>
	<CAPf+86VpHXqnnJpCOmq7TNu4g1WBXyO8PpsuStftWYiwLwhVBA@mail.gmail.com>
	<4FD895E6.7000101@thewinelake.com>
	<CALNZuwnDT23d44oR8-jPfpu4dKh-mh9GSQJzBOFtO9bWSUa4Rw@mail.gmail.com>
	<4FD8C5A1.8050905@thewinelake.com>
Message-ID: <4FD9AD39.1040306@thewinelake.com>

Looks like < > is the solution - many thanks!

...now I seem to have an issue with playAndGetDigits....


From alex at digitalmail.com  Thu Jun 14 13:26:52 2012
From: alex at digitalmail.com (Alex Lake)
Date: Thu, 14 Jun 2012 10:26:52 +0100
Subject: [Freeswitch-users] playAndGetDigits working as designed?
Message-ID: <4FD9AE5C.50008@digitalmail.com>

I have the following in a lua script:

session:streamFile(whisper)
dtmf = session:playAndGetDigits(1,1,1,5000,"#",accept,accept,"\\d+")

The idea is that the caller hears "Call for Alex of Wibble 
Corp"..."Press 1 to accept" and then he has 5s to press 1, otherwise 
it's assumed that the call has gone to mobile voicemail.

What seems to be happening is:

the caller hears "Call for Alex of Wibble Corp"..."Press 1 to accept" 
<pause 5s> "Press 1 to accept" <immediate hangup>

(I'm surprised that it says it twice, and ignores the <timeout> attribute)

Actually, I wouldn't mind something that went

"Call for Alex of Wibble Corp"..."Press 1 to accept" <pause 3s> "Press 1 
to accept" <pause 3s> <hangup>

But telling the user to press 1 and IMMEDIATELY hanging up seems wrong!


From jaybinks at gmail.com  Thu Jun 14 13:55:50 2012
From: jaybinks at gmail.com (jay binks)
Date: Thu, 14 Jun 2012 19:55:50 +1000
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <D6D8040D-2350-4B59-9FC0-90B230537D25@visionutveckling.se>
References: <4FD94C26.10800@the800group.com>
	<CAO47Hpcuh7OFEgjyjAZx9bq9MrPPscqhx9159aaU03rv57=P9A@mail.gmail.com>
	<CALNZuwkF+Hw0HfpCONq0u8OJW29=tfTeAcudLYS69K04qAX2Ww@mail.gmail.com>
	<CAFZQphzkbg5_4tH0YmYLoJ4Ft_rN07nLHBWZMkcyWVbczAUe1A@mail.gmail.com>
	<D6D8040D-2350-4B59-9FC0-90B230537D25@visionutveckling.se>
Message-ID: <CAO47HpcPqDm3gLCtMPoiooOnMAok+E-VhMAth6bG-3kJCjg9ew@mail.gmail.com>

Hey ..

On 14 June 2012 16:16, Peter Olsson <peter.olsson at visionutveckling.se>wrote:

> Anyway, for this kind of setup I would also prefer Linux, but mostly for
> the possibilites with fail2ban etc, which doesn't exist on Windows. I'm
> thinking of writiling something similar for Windows, hopefully I get som
> time for that soon...
>
> /Peter
>

if you do get round to writing something like this let me know.  ( not that
I even have a windows box )
but when I wrote the patches for FS to do the Fail2Ban compatible logging,
I created ESL Events also
with the intention that we could do a FS module "mod_security" or something
( no time given to the name )

it would be fairly simple to move some of the fail2ban functionality into
such a module that could either call out to simple scripts or
insert firewall rules its self.

I have some ideas here, and I would probably be interested in working on
this with others.

Jay
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From peter.olsson at visionutveckling.se  Thu Jun 14 14:07:17 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 14 Jun 2012 10:07:17 +0000
Subject: [Freeswitch-users] Brute-force attack
Message-ID: <1FFF97C269757C458224B7C895F35F1510821F@cantor.std.visionutv.se>

Jay,

Yes I've seen about those ideas, and I think it's a really good way to handle this. If I get the time to move these things further, I will let you know.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r jay binks
Skickat: den 14 juni 2012 11:56
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Brute-force attack

Hey ..

On 14 June 2012 16:16, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Anyway, for this kind of setup I would also prefer Linux, but mostly for the possibilites with fail2ban etc, which doesn't exist on Windows. I'm thinking of writiling something similar for Windows, hopefully I get som time for that soon...

/Peter

if you do get round to writing something like this let me know.  ( not that I even have a windows box )
but when I wrote the patches for FS to do the Fail2Ban compatible logging, I created ESL Events also
with the intention that we could do a FS module "mod_security" or something ( no time given to the name )

it would be fairly simple to move some of the fail2ban functionality into such a module that could either call out to simple scripts or
insert firewall rules its self.

I have some ideas here, and I would probably be interested in working on this with others.

Jay

!DSPAM:4fd9b38232761034425819!
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From ocset at the800group.com  Thu Jun 14 14:29:50 2012
From: ocset at the800group.com (ocset)
Date: Thu, 14 Jun 2012 18:29:50 +0800
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <CANBhyPBPyGdwQhvZEbcK7iVFQEKwgC0uaD9qOb9udBEDiLR0eQ@mail.gmail.com>
References: <4FD94C26.10800@the800group.com>
	<CAO47Hpcuh7OFEgjyjAZx9bq9MrPPscqhx9159aaU03rv57=P9A@mail.gmail.com>
	<CALNZuwkF+Hw0HfpCONq0u8OJW29=tfTeAcudLYS69K04qAX2Ww@mail.gmail.com>
	<CAFZQphzkbg5_4tH0YmYLoJ4Ft_rN07nLHBWZMkcyWVbczAUe1A@mail.gmail.com>
	<D6D8040D-2350-4B59-9FC0-90B230537D25@visionutveckling.se>
	<CANBhyPBPyGdwQhvZEbcK7iVFQEKwgC0uaD9qOb9udBEDiLR0eQ@mail.gmail.com>
Message-ID: <4FD9BD1E.5040705@the800group.com>

Anton

I know nothing about ext-rtp-ip and ext-sip-ip. Could you please explain 
how this will help in making the system more secure?

Thanks
O

On 14/06/12 16:27, Anton Kvashenkin wrote:
> I would suggest to create separate profile for remote workers. For 
> example, external-road-warrios. So you can play with ext-rtp-ip and 
> ext-sip-ip.
>
> 2012/6/14 Peter Olsson <peter.olsson at visionutveckling.se 
> <mailto:peter.olsson at visionutveckling.se>>
>
>     I use both Windows and Linux systems. As long as you know how to
>     manage both systems, there is not a big difference when it comes
>     to exploits and general security (not anymore anyway, if you use
>     current versions). I would say that the biggest issue here is the
>     knowledge of the people managing the systems. And it's usually
>     more secure to manage a system that you know, then a system you
>     don't know much about, even though that system is considered to be
>     more secure.
>
>     Lots has happened since Windows 95 :)
>
>     Anyway, for this kind of setup I would also prefer Linux, but
>     mostly for the possibilites with fail2ban etc, which doesn't exist
>     on Windows. I'm thinking of writiling something similar for
>     Windows, hopefully I get som time for that soon...
>
>     /Peter
>
>     14 jun 2012 kl. 07:51 skrev "Muhammad Shahzad"
>     <shaheryarkh at googlemail.com
>     <mailto:shaheryarkh at googlemail.com><mailto:shaheryarkh at googlemail.com
>     <mailto:shaheryarkh at googlemail.com>>>:
>
>     I would strongly suggest to move your production system to Linux,
>     which is by far secure and controllable then Windows. Right now,
>     if somebody does not breaks into your voip setup using some
>     bruteforce / DOS attack, s/he can still exploit some hole in
>     Windows to crack your security. Windows is simply not secure
>     enough to production grade performance.
>
>     Thank you.
>
>
>     On Thu, Jun 14, 2012 at 6:39 AM, Avi Marcus <avi at avimarcus.net
>     <mailto:avi at avimarcus.net><mailto:avi at avimarcus.net
>     <mailto:avi at avimarcus.net>>> wrote:
>     That's not necessarily the best kind of password... see
>     http://xkcd.com/936/ and then http://tech.dropbox.com/?p=165
>
>     -Avi
>
>
>
>     On Thu, Jun 14, 2012 at 6:23 AM, jay binks <jaybinks at gmail.com
>     <mailto:jaybinks at gmail.com><mailto:jaybinks at gmail.com
>     <mailto:jaybinks at gmail.com>>> wrote:
>     > Strong passwords are a great start, but fail2ban does a little
>     more than
>     > this.
>     >
>     > you could move off port 5060 to something un-conventional,
>     meaning your less
>     > likely to get scanned / brute forced.
>     >
>     > Jay
>     >
>     > On 14 June 2012 12:27, ocset <ocset at the800group.com
>     <mailto:ocset at the800group.com><mailto:ocset at the800group.com
>     <mailto:ocset at the800group.com>>> wrote:
>     >>
>     >> Hi
>     >>
>     >> I have deployed Freeswiitch on windows 7 and since there is no
>     fail2ban
>     >> on windows, I was wondering what the real risk is with opening
>     it up to
>     >> the internet. If I was to ensure that all users and passwords were
>     >> extremely difficult to guess (passwords like
>     "2$53E_d7?^2!3s$"), what
>     >> are the risks that I am exposing myself to? Is there a type of
>     DoS for
>     >> voip where hackers can just flood my system with requests
>     simply to be
>     >> malicious?
>     >>
>     >> There are VB windows scripts available that emulate what
>     fail2ban does
>     >> on Linux but I was just wondering whether I really need to
>     implement
>     >> this level of security if I can control the password complexity in
>     >> Freeswitch.
>     >>
>     >> Thanks
>     >> O
>     >>
>     >>
>     >>
>     _________________________________________________________________________
>     >> Professional FreeSWITCH Consulting Services:
>     >> consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     >> http://www.freeswitchsolutions.com
>     >>
>     >> 
>     >> 
>     >>
>     >> Official FreeSWITCH Sites
>     >> http://www.freeswitch.org
>     >> http://wiki.freeswitch.org
>     >> http://www.cluecon.com
>     >>
>     >> Join Us At ClueCon - Aug 7-9, 2012
>     >>
>     >> 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
>     >
>     >
>     >
>     >
>     > --
>     > Sincerely
>     >
>     > Jay
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     > http://www.freeswitchsolutions.com
>     >
>     > 
>     > 
>     >
>     > Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://wiki.freeswitch.org
>     > http://www.cluecon.com
>     >
>     > Join Us At ClueCon - Aug 7-9, 2012
>     >
>     > 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
>     >
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
>
>     --
>     Muhammad Shahzad
>     -----------------------------------
>     CISCO Rich Media Communication Specialist (CRMCS)
>     CISCO Certified Network Associate (CCNA)
>     Cell: +92 334 422 40 88
>     MSN: shari_786pk at hotmail.com
>     <mailto:shari_786pk at hotmail.com><mailto:shari_786pk at hotmail.com
>     <mailto:shari_786pk at hotmail.com>>
>     Email: shaheryarkh at googlemail.com
>     <mailto:shaheryarkh at googlemail.com><mailto:shaheryarkh at googlemail.com
>     <mailto:shaheryarkh at googlemail.com>>
>     !DSPAM:4fd978c432761360223007!
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>     !DSPAM:4fd978c432761360223007!
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From ocset at the800group.com  Thu Jun 14 14:35:24 2012
From: ocset at the800group.com (ocset)
Date: Thu, 14 Jun 2012 18:35:24 +0800
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <CAO47HpcPqDm3gLCtMPoiooOnMAok+E-VhMAth6bG-3kJCjg9ew@mail.gmail.com>
References: <4FD94C26.10800@the800group.com>
	<CAO47Hpcuh7OFEgjyjAZx9bq9MrPPscqhx9159aaU03rv57=P9A@mail.gmail.com>
	<CALNZuwkF+Hw0HfpCONq0u8OJW29=tfTeAcudLYS69K04qAX2Ww@mail.gmail.com>
	<CAFZQphzkbg5_4tH0YmYLoJ4Ft_rN07nLHBWZMkcyWVbczAUe1A@mail.gmail.com>
	<D6D8040D-2350-4B59-9FC0-90B230537D25@visionutveckling.se>
	<CAO47HpcPqDm3gLCtMPoiooOnMAok+E-VhMAth6bG-3kJCjg9ew@mail.gmail.com>
Message-ID: <4FD9BE6C.5040608@the800group.com>

If you are looking for a fail2ban equivalent on Windows, how about this 
script

http://serverfault.com/questions/43360/cygwin-sshd-autoblock-failed-logins/43900#43900

I have not tried it but it may be a good starting point for something to 
use with Freeswitch

Regards
O


On 14/06/12 17:55, jay binks wrote:
> Hey ..
>
> On 14 June 2012 16:16, Peter Olsson <peter.olsson at visionutveckling.se 
> <mailto:peter.olsson at visionutveckling.se>> wrote:
>
>     Anyway, for this kind of setup I would also prefer Linux, but
>     mostly for the possibilites with fail2ban etc, which doesn't exist
>     on Windows. I'm thinking of writiling something similar for
>     Windows, hopefully I get som time for that soon...
>
>     /Peter
>
>
> if you do get round to writing something like this let me know.  ( not 
> that I even have a windows box )
> but when I wrote the patches for FS to do the Fail2Ban compatible 
> logging, I created ESL Events also
> with the intention that we could do a FS module "mod_security" or 
> something ( no time given to the name )
>
> it would be fairly simple to move some of the fail2ban functionality 
> into such a module that could either call out to simple scripts or
> insert firewall rules its self.
>
> I have some ideas here, and I would probably be interested in working 
> on this with others.
>
> Jay
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From awais-nazeer at hotmail.com  Thu Jun 14 14:43:30 2012
From: awais-nazeer at hotmail.com (awais nazir)
Date: Thu, 14 Jun 2012 15:43:30 +0500
Subject: [Freeswitch-users] Source of hangup in cdr
Message-ID: <BLU162-W14DDD20C2B7AF6FFD77A5288F40@phx.gbl>


Hi folks

CSendan we write direction of call hangup in cdr , like source , destination or freeswitch (as it hangsup itself for example unauthenticated call).

--waisee
 		 	   		  
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From anton.jugatsu at gmail.com  Thu Jun 14 14:58:06 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Thu, 14 Jun 2012 14:58:06 +0400
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <4FD9BD1E.5040705@the800group.com>
References: <4FD94C26.10800@the800group.com>
	<CAO47Hpcuh7OFEgjyjAZx9bq9MrPPscqhx9159aaU03rv57=P9A@mail.gmail.com>
	<CALNZuwkF+Hw0HfpCONq0u8OJW29=tfTeAcudLYS69K04qAX2Ww@mail.gmail.com>
	<CAFZQphzkbg5_4tH0YmYLoJ4Ft_rN07nLHBWZMkcyWVbczAUe1A@mail.gmail.com>
	<D6D8040D-2350-4B59-9FC0-90B230537D25@visionutveckling.se>
	<CANBhyPBPyGdwQhvZEbcK7iVFQEKwgC0uaD9qOb9udBEDiLR0eQ@mail.gmail.com>
	<4FD9BD1E.5040705@the800group.com>
Message-ID: <CANBhyPBVHtpptOdfkgdWoWVS1kPtiU4TdLi-+qs8WQ3EayMv7Q@mail.gmail.com>

For example, you don't need to open 5060 port to a whole world, just 5090
(the port you use for connecting road warriors).
http://wiki.freeswitch.org/wiki/Nat
http://wiki.freeswitch.org/wiki/External_profile

2012/6/14 ocset <ocset at the800group.com>

>  Anton
>
> I know nothing about ext-rtp-ip and ext-sip-ip. Could you please explain
> how this will help in making the system more secure?
>
> Thanks
> O
>
> On 14/06/12 16:27, Anton Kvashenkin wrote:
>
> I would suggest to create separate profile for remote workers. For
> example, external-road-warrios. So you can play with ext-rtp-ip and
> ext-sip-ip.
>
> 2012/6/14 Peter Olsson <peter.olsson at visionutveckling.se>
>
>> I use both Windows and Linux systems. As long as you know how to manage
>> both systems, there is not a big difference when it comes to exploits and
>> general security (not anymore anyway, if you use current versions). I would
>> say that the biggest issue here is the knowledge of the people managing the
>> systems. And it's usually more secure to manage a system that you know,
>> then a system you don't know much about, even though that system is
>> considered to be more secure.
>>
>> Lots has happened since Windows 95 :)
>>
>> Anyway, for this kind of setup I would also prefer Linux, but mostly for
>> the possibilites with fail2ban etc, which doesn't exist on Windows. I'm
>> thinking of writiling something similar for Windows, hopefully I get som
>> time for that soon...
>>
>> /Peter
>>
>> 14 jun 2012 kl. 07:51 skrev "Muhammad Shahzad" <
>> shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>>:
>>
>> I would strongly suggest to move your production system to Linux, which
>> is by far secure and controllable then Windows. Right now, if somebody does
>> not breaks into your voip setup using some bruteforce / DOS attack, s/he
>> can still exploit some hole in Windows to crack your security. Windows is
>> simply not secure enough to production grade performance.
>>
>> Thank you.
>>
>>
>>  On Thu, Jun 14, 2012 at 6:39 AM, Avi Marcus <avi at avimarcus.net<mailto:
>> avi at avimarcus.net>> wrote:
>> That's not necessarily the best kind of password... see
>> http://xkcd.com/936/ and then http://tech.dropbox.com/?p=165
>>
>> -Avi
>>
>>
>>
>>  On Thu, Jun 14, 2012 at 6:23 AM, jay binks <jaybinks at gmail.com<mailto:
>> jaybinks at gmail.com>> wrote:
>> > Strong passwords are a great start, but fail2ban does a little more than
>> > this.
>> >
>> > you could move off port 5060 to something un-conventional, meaning your
>> less
>> > likely to get scanned / brute forced.
>> >
>> > Jay
>> >
>>  > On 14 June 2012 12:27, ocset <ocset at the800group.com<mailto:
>> ocset at the800group.com>> wrote:
>> >>
>> >> Hi
>> >>
>> >> I have deployed Freeswiitch on windows 7 and since there is no fail2ban
>> >> on windows, I was wondering what the real risk is with opening it up to
>> >> the internet. If I was to ensure that all users and passwords were
>> >> extremely difficult to guess (passwords like "2$53E_d7?^2!3s$"), what
>> >> are the risks that I am exposing myself to? Is there a type of DoS for
>> >> voip where hackers can just flood my system with requests simply to be
>> >> malicious?
>> >>
>> >> There are VB windows scripts available that emulate what fail2ban does
>> >> on Linux but I was just wondering whether I really need to implement
>> >> this level of security if I can control the password complexity in
>> >> Freeswitch.
>> >>
>> >> Thanks
>> >> O
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>>  >> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> Join Us At ClueCon - Aug 7-9, 2012
>> >>
>> >> 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
>> >
>> >
>> >
>> >
>> > --
>> > Sincerely
>> >
>> > Jay
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>>  > consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > Join Us At ClueCon - Aug 7-9, 2012
>> >
>> > 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
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>>  consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> 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
>>
>>
>>
>>
>> --
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>>  MSN: shari_786pk at hotmail.com<mailto:shari_786pk at hotmail.com>
>> Email: shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>
>> !DSPAM:4fd978c432761360223007!
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>>  consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> 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
>>
>>
>>  !DSPAM:4fd978c432761360223007!
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From th982a at googlemail.com  Thu Jun 14 15:24:10 2012
From: th982a at googlemail.com (Tamer Higazi)
Date: Thu, 14 Jun 2012 13:24:10 +0200
Subject: [Freeswitch-users] freeswitch and mod_mISDN in NT mode?!
Message-ID: <4FD9C9DA.5010409@googlemail.com>

Does anyone of you know, if it's possible right now to run FS with
mod_misdn in NT mode, to connect an ISDN phone to an HFC board?!



Tamer
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From andrew at cassidywebservices.co.uk  Thu Jun 14 15:29:23 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 14 Jun 2012 12:29:23 +0100
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <CANBhyPBVHtpptOdfkgdWoWVS1kPtiU4TdLi-+qs8WQ3EayMv7Q@mail.gmail.com>
References: <4FD94C26.10800@the800group.com>
	<CAO47Hpcuh7OFEgjyjAZx9bq9MrPPscqhx9159aaU03rv57=P9A@mail.gmail.com>
	<CALNZuwkF+Hw0HfpCONq0u8OJW29=tfTeAcudLYS69K04qAX2Ww@mail.gmail.com>
	<CAFZQphzkbg5_4tH0YmYLoJ4Ft_rN07nLHBWZMkcyWVbczAUe1A@mail.gmail.com>
	<D6D8040D-2350-4B59-9FC0-90B230537D25@visionutveckling.se>
	<CANBhyPBPyGdwQhvZEbcK7iVFQEKwgC0uaD9qOb9udBEDiLR0eQ@mail.gmail.com>
	<4FD9BD1E.5040705@the800group.com>
	<CANBhyPBVHtpptOdfkgdWoWVS1kPtiU4TdLi-+qs8WQ3EayMv7Q@mail.gmail.com>
Message-ID: <CAMbW9oLMys-qBwoHpBhsGXbAF+hDeU12=T2P4yjtW-ZfY7Q+3A@mail.gmail.com>

It IS possible to use windows 7 powershell scripts run on a schedule to
dynamically add and remove firewall rules, it's something a firend and I
did before for RDP. The RDP version is here:
http://www.jonsdocs.org.uk/wiki/index.php/PSLogonFailures

I'm sure with a little tweaking it can be made to work with freeswitch log
files.

On 14 June 2012 11:58, Anton Kvashenkin <anton.jugatsu at gmail.com> wrote:

> For example, you don't need to open 5060 port to a whole world, just 5090
> (the port you use for connecting road warriors).
> http://wiki.freeswitch.org/wiki/Nat
> http://wiki.freeswitch.org/wiki/External_profile
>
>
> 2012/6/14 ocset <ocset at the800group.com>
>
>>  Anton
>>
>> I know nothing about ext-rtp-ip and ext-sip-ip. Could you please explain
>> how this will help in making the system more secure?
>>
>> Thanks
>> O
>>
>> On 14/06/12 16:27, Anton Kvashenkin wrote:
>>
>> I would suggest to create separate profile for remote workers. For
>> example, external-road-warrios. So you can play with ext-rtp-ip and
>> ext-sip-ip.
>>
>> 2012/6/14 Peter Olsson <peter.olsson at visionutveckling.se>
>>
>>> I use both Windows and Linux systems. As long as you know how to manage
>>> both systems, there is not a big difference when it comes to exploits and
>>> general security (not anymore anyway, if you use current versions). I would
>>> say that the biggest issue here is the knowledge of the people managing the
>>> systems. And it's usually more secure to manage a system that you know,
>>> then a system you don't know much about, even though that system is
>>> considered to be more secure.
>>>
>>> Lots has happened since Windows 95 :)
>>>
>>> Anyway, for this kind of setup I would also prefer Linux, but mostly for
>>> the possibilites with fail2ban etc, which doesn't exist on Windows. I'm
>>> thinking of writiling something similar for Windows, hopefully I get som
>>> time for that soon...
>>>
>>> /Peter
>>>
>>> 14 jun 2012 kl. 07:51 skrev "Muhammad Shahzad" <
>>> shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>>:
>>>
>>> I would strongly suggest to move your production system to Linux, which
>>> is by far secure and controllable then Windows. Right now, if somebody does
>>> not breaks into your voip setup using some bruteforce / DOS attack, s/he
>>> can still exploit some hole in Windows to crack your security. Windows is
>>> simply not secure enough to production grade performance.
>>>
>>> Thank you.
>>>
>>>
>>>  On Thu, Jun 14, 2012 at 6:39 AM, Avi Marcus <avi at avimarcus.net<mailto:
>>> avi at avimarcus.net>> wrote:
>>> That's not necessarily the best kind of password... see
>>> http://xkcd.com/936/ and then http://tech.dropbox.com/?p=165
>>>
>>> -Avi
>>>
>>>
>>>
>>>  On Thu, Jun 14, 2012 at 6:23 AM, jay binks <jaybinks at gmail.com<mailto:
>>> jaybinks at gmail.com>> wrote:
>>> > Strong passwords are a great start, but fail2ban does a little more
>>> than
>>> > this.
>>> >
>>> > you could move off port 5060 to something un-conventional, meaning
>>> your less
>>> > likely to get scanned / brute forced.
>>> >
>>> > Jay
>>> >
>>>  > On 14 June 2012 12:27, ocset <ocset at the800group.com<mailto:
>>> ocset at the800group.com>> wrote:
>>> >>
>>> >> Hi
>>> >>
>>> >> I have deployed Freeswiitch on windows 7 and since there is no
>>> fail2ban
>>> >> on windows, I was wondering what the real risk is with opening it up
>>> to
>>> >> the internet. If I was to ensure that all users and passwords were
>>> >> extremely difficult to guess (passwords like "2$53E_d7?^2!3s$"), what
>>> >> are the risks that I am exposing myself to? Is there a type of DoS for
>>> >> voip where hackers can just flood my system with requests simply to be
>>> >> malicious?
>>> >>
>>> >> There are VB windows scripts available that emulate what fail2ban does
>>> >> on Linux but I was just wondering whether I really need to implement
>>> >> this level of security if I can control the password complexity in
>>> >> Freeswitch.
>>> >>
>>> >> Thanks
>>> >> O
>>> >>
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>>  >> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> 
>>> >> 
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://wiki.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> Join Us At ClueCon - Aug 7-9, 2012
>>> >>
>>> >> 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
>>> >
>>> >
>>> >
>>> >
>>> > --
>>> > Sincerely
>>> >
>>> > Jay
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>>  > consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > Join Us At ClueCon - Aug 7-9, 2012
>>> >
>>> > 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
>>> >
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>>  consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> 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
>>>
>>>
>>>
>>>
>>> --
>>> Muhammad Shahzad
>>> -----------------------------------
>>> CISCO Rich Media Communication Specialist (CRMCS)
>>> CISCO Certified Network Associate (CCNA)
>>> Cell: +92 334 422 40 88
>>>  MSN: shari_786pk at hotmail.com<mailto:shari_786pk at hotmail.com>
>>> Email: shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>
>>> !DSPAM:4fd978c432761360223007!
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>>  consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> 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
>>>
>>>
>>>  !DSPAM:4fd978c432761360223007!
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From andrew at cassidywebservices.co.uk  Thu Jun 14 15:34:44 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 14 Jun 2012 12:34:44 +0100
Subject: [Freeswitch-users] Does mod_fifo not take in
	effective_caller_id_name?
In-Reply-To: <20120614043151.7A69A14DBD8@smtp.hushmail.com>
References: <20120614043151.7A69A14DBD8@smtp.hushmail.com>
Message-ID: <CAMbW9oJ6ntsSGwbsyOpygtShRrkYGXuwJr428JZMmTFwb_LhLQ@mail.gmail.com>

mod_callcenter also ignores those variables unless you set one called
cc_export_vars. Perhaps there's a similar mechanism for mod_fifo?

On 14 June 2012 05:31, <esander83 at hushmail.com> wrote:

> Hi All,
>
> I have this basic dialplan:
>
> <extension name="setcallname" continue="true">
>       <condition field="caller_id_number" expression="^MYCELLPHONE$">
>         <action application="export" data="effective_caller_id_name=E
> Sander" inline="true"/>
>          <action application="log" data="INFO ES CELL"/>
>       </condition>
>     </extension>
>
>   <extension name="send caller to FIFO">
>     <condition field="destination_number" expression="^DIDNUMBER$">
>       <action application="answer"/>
>       <action application="set" data="fifo_music=$${hold_music}"/>
>       <action application="playback" data="ivr/ivr-hold_connect_call.wav"/>
>       <action application="fifo" data="FIFO0 in"/>
>     </condition>
>   </extension>
>
> I can see the call get logged via the first extension but it's not passing
> the info to modfifo.  I can remove the 2nd extension and put a basic:
> <action application="bridge" data="user/1052"/>
>
> and the caller id name info is displayed on the phones but not when
> routing through mod_fifo.  I have tried both set/export and it still
> doesn't seem to work.
>
> Is this correct?
>
> Thanks,
> eric_s
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From avi at avimarcus.net  Thu Jun 14 15:35:07 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 14 Jun 2012 14:35:07 +0300
Subject: [Freeswitch-users] NAT-ed phone only got some incoming calls?
Message-ID: <CALNZuwn8WA9Onndn_2eCYArth3yvQkmB1RQUCfam6qXgmveSnQ@mail.gmail.com>

I have a customer with a natted Yealink T20P...

Current reg: http://pastebin.freeswitch.org/19290
Two call traces, FS log side: http://pastebin.freeswitch.org/19286

One rang, one didn't. The local information for the phones is the same --
so NAT seems to be handled properly. Is this an issue with the remote
router dropping packets or something...?
I see 4 inbound calls to that device today, and 1 of them doesn't have a
progress stamp so I assume that's the same as above -- it never got a 180
back.

I think I can dig up the actual sip traces, if that matters, but it's not
in the most convenient of formats.

Some illumination would be appreciated.

Thanks,
-Avi Marcus
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From avi at avimarcus.net  Thu Jun 14 17:02:55 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 14 Jun 2012 16:02:55 +0300
Subject: [Freeswitch-users] playAndGetDigits working as designed?
In-Reply-To: <4FD9AE5C.50008@digitalmail.com>
References: <4FD9AE5C.50008@digitalmail.com>
Message-ID: <CALNZuwmGTuhH0edXG26-L-CF84MpuJvp+-Byod+PEis=UONOSA@mail.gmail.com>

it's playing twice because of the second accept -- that's what it plays
upon invalid entry. But you don't want any message on non-match.
You might want getdigits<http://wiki.freeswitch.org/wiki/Mod_lua#session:getDigits>,
a
simpler way to get input without retry options.

Wait, are you using this for confirm of pickup for hunting to mobiles?
Doesn't http://wiki.freeswitch.org/wiki/Variable_group_confirm_file work
for you? It seems to fit my use case.
-Avi


On Thu, Jun 14, 2012 at 12:26 PM, Alex Lake <alex at digitalmail.com> wrote:

> I have the following in a lua script:
>
> session:streamFile(whisper)
> dtmf = session:playAndGetDigits(1,1,1,5000,"#",accept,accept,"\\d+")
>
> The idea is that the caller hears "Call for Alex of Wibble
> Corp"..."Press 1 to accept" and then he has 5s to press 1, otherwise
> it's assumed that the call has gone to mobile voicemail.
>
> What seems to be happening is:
>
> the caller hears "Call for Alex of Wibble Corp"..."Press 1 to accept"
> <pause 5s> "Press 1 to accept" <immediate hangup>
>
> (I'm surprised that it says it twice, and ignores the <timeout> attribute)
>
> Actually, I wouldn't mind something that went
>
> "Call for Alex of Wibble Corp"..."Press 1 to accept" <pause 3s> "Press 1
> to accept" <pause 3s> <hangup>
>
> But telling the user to press 1 and IMMEDIATELY hanging up seems wrong!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tb at cool.de  Thu Jun 14 10:28:32 2012
From: tb at cool.de (Thomas Brettinger)
Date: Thu, 14 Jun 2012 08:28:32 +0200
Subject: [Freeswitch-users] Relatively new FreeSwitch user,
	calls from outside not terminated
Message-ID: <244FC5DA-2A80-4CA7-9CDD-8C7AF29B2109@cool.de>

Hi,

I started toying around with FreeSwitch on PFsense a while ago, and now did a fresh install on a machine inside my lan.

Setup is:

FS --- PFsense (Firewall, NAT) --- Internet --- sipgate.de

Routing inside and outside works like a charm. However, I have the issue that if a call is hung up on a local phone, FreeSwitch doesn't hang up on the sipgate.de leg.
It doesn't make a difference if the call was originated from inside or outside. The call is left open for an unknown time and freeswitch doesn't seem to notice that the inside has hung up (at least, nothing appears in the sip trace or log).

As soon as the remote party hangs up, siptrace and logfile look as they probably should.

Does anyone have an idea where I should start looking for this issue? It *could* have something to do with the firewall between, however I don't believe so, as the call is hung up on the local network and freeswitch doesn't seem to notice.

Oh, and I tried this with Cisco 69xx (SIP) and Siemens GIgaset 501V (SIP) phones.

Any hints appreciated.

Thanks in advance,

Thomas
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From netcentrica at gmail.com  Thu Jun 14 17:42:37 2012
From: netcentrica at gmail.com (Adam Raszynski)
Date: Thu, 14 Jun 2012 15:42:37 +0200
Subject: [Freeswitch-users] ESL originate - call events are sent too fast
Message-ID: <CAJWEF2bLeCPdx0UyvGmuTaTpPTYn1itrF8km-Xv1gN_7gods+g@mail.gmail.com>

Hi all

I have some little problem with events generated by ESL originated calls.
Maybe someone had similar problem and could post how to solve it.

I would like to originate call and receive events related only to that
call. At the same time my originate script is executed multiple times i.e.
100 times.

Current version of my script works like this:
- send api originate [params]target via ESL
- wait for originate response to get uuid
- add filter: filter Unique-ID <uuid of my call>
- subscribe for events: event plain CHANNEL_ANSWER CHANNEL_HANGUP CUSTOM
spandsp::txfaxresult

Unfortunately execution of all above statements sometimes take a while and
in meanwhile FreeSWITCH generates event and send it back to my script
BEFORE it has subscribed to receive events. It happens for example for very
fast call answer or hangup.

In that case I lost important answer or hangup event and all processing
logic is broken.

I've tried to change sequence of commands, but when I put event
subscription before adding filter then I get events from other channels. I
also can't add filter earlier because need to wait for originate response.

Any hints on this?
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From modesto at isimples.com.br  Thu Jun 14 18:40:47 2012
From: modesto at isimples.com.br (Antonio Modesto)
Date: Thu, 14 Jun 2012 11:40:47 -0300
Subject: [Freeswitch-users] How to configure ring back for calls through
	freetdm?
Message-ID: <1339684847.3957.12.camel@modesto.localdomain.net>

Hi,

	I've written a lua script to make calls using freetdm channels, the
problem is that there is not a ringback tone to the calling party, so
it's not possible to know what is going on until the caller party
answers the call. What do I have to do to enable this?


Thanks.



From peter.olsson at visionutveckling.se  Thu Jun 14 18:47:23 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 14 Jun 2012 14:47:23 +0000
Subject: [Freeswitch-users] ESL originate - call events are sent too fast
Message-ID: <1FFF97C269757C458224B7C895F35F151089B6@cantor.std.visionutv.se>

You will have to generate your own channel uuid and then set origination_uuid when you do the originate call in FS. Also, you will need to subscribe to this uuid before trying to originate the actual call.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Adam Raszynski
Skickat: den 14 juni 2012 15:43
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] ESL originate - call events are sent too fast

Hi all

I have some little problem with events generated by ESL originated calls.
Maybe someone had similar problem and could post how to solve it.

I would like to originate call and receive events related only to that call. At the same time my originate script is executed multiple times i.e. 100 times.
Current version of my script works like this:
- send api originate [params]target via ESL
- wait for originate response to get uuid
- add filter: filter Unique-ID <uuid of my call>
- subscribe for events: event plain CHANNEL_ANSWER CHANNEL_HANGUP CUSTOM spandsp::txfaxresult
Unfortunately execution of all above statements sometimes take a while and in meanwhile FreeSWITCH generates event and send it back to my script BEFORE it has subscribed to receive events. It happens for example for very fast call answer or hangup.
In that case I lost important answer or hangup event and all processing logic is broken.

I've tried to change sequence of commands, but when I put event subscription before adding filter then I get events from other channels. I also can't add filter earlier because need to wait for originate response.

Any hints on this?



!DSPAM:4fd9f54532761072142040!
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From peter.olsson at visionutveckling.se  Thu Jun 14 18:51:00 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 14 Jun 2012 14:51:00 +0000
Subject: [Freeswitch-users] How to configure ring back for calls
	through	freetdm?
Message-ID: <1FFF97C269757C458224B7C895F35F151089C5@cantor.std.visionutv.se>

Try to set the channel variable "ringback" to the tone you want to play.

/Peter



-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Antonio Modesto
Skickat: den 14 juni 2012 16:41
Till: Freeswitch Users
?mne: [Freeswitch-users] How to configure ring back for calls through freetdm?

Hi,

	I've written a lua script to make calls using freetdm channels, the problem is that there is not a ringback tone to the calling party, so it's not possible to know what is going on until the caller party answers the call. What do I have to do to enable this?


Thanks.


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

!DSPAM:4fd9f59932767773115490!



From dujinfang at gmail.com  Thu Jun 14 18:56:08 2012
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 14 Jun 2012 22:56:08 +0800
Subject: [Freeswitch-users] Developers' Salon,
	sched_broadcast from FreeSWITCH-CN
Message-ID: <E7C952BAB06744089BB455F648F8776E@gmail.com>

All, 

Sorry I didn't promote as it's mainly targeted to the Chinese Community and it's the first Salon in China and we want to keep a low profile.

I'm the creator and maintainer of FreeSWITCH-CN(http://www.freeswitch.org.cn ). We'll have a Salon at this Sunday (June 17th, UTC+8) afternoon in Beijing, China.

http://www.freeswitch.org.cn/blog/past/2012/4/16/freeswitchcnzhong-wen-she-qu-2012shou-jie-kai-fa-zhe-sha-long/ 

According to http://freeswitch-cn.eventbrite.com/ , we have 56 registers till now and we do accept blind reg.

It will be in a coffee bar and if anyone happens to drop by is welcome to have a cup of coffee.

The Chinese Community is growing and I will can have more news to ClueCon.

And I have one news now. I personally received the first donation a few days ago for the in-writing book: FreeSWITCH in Action (You won't use this name in the second edition of FS book, right? Michael :) ) , it's published on site using CC-BY-NC-ND.

http://www.freeswitch.org.cn/document

My aim of running the FreeSWITCH-CN community was to help Chinese speakers to learn FreeSWITCH. And I had put a Paypal donation button on and planed to donate back a percentage to the core developers but get nothing till now.

I just figured out that there's basically no way for us legally collecting money in my country publicly. So I replaced the paypal donation to a guide link directly to freeswitch.org and if anyone donate with a remark of FreeSWITCH-CN it's probably from our community, or probably never.

Anyway, I will use the first being-donated-money to buy a few things and will give out at the Salon, it's not that much, but it's something. And if there's future donations happens privately to me I will definitely happy to donate back to upstream.

Ok, back to the Salon. I will report more here after the Salon and probably we could run a tiny ClueCon in China next year?



Seven.

-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn

Sent with Sparrow (http://www.sparrowmailapp.com)

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From sdevoy at bizfocused.com  Thu Jun 14 19:08:34 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 14 Jun 2012 11:08:34 -0400
Subject: [Freeswitch-users] Is SCA bridgeing Multithreaded?
In-Reply-To: <046801cd49a0$28819b70$7984d250$@bizfocused.com>
References: <046801cd49a0$28819b70$7984d250$@bizfocused.com>
Message-ID: <026801cd4a3f$9151e1b0$b3f5a510$@bizfocused.com>

Bump.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Devoy
Sent: Wednesday, June 13, 2012 4:07 PM
To: FreeSWITCH-users at lists.freeswitch.org
Subject: [Freeswitch-users] Is SCA bridgeing Multithreaded?

 

I see I can use Enterprise Originate :_: to connect calls to multiple phones
in threads to speed things up.  

 

Does Shared Call Appearance connect in threads or sequentially?

 

Thanks,

Sean

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From vvenkatar at gmail.com  Thu Jun 14 19:18:48 2012
From: vvenkatar at gmail.com (Venkatesh)
Date: Thu, 14 Jun 2012 08:18:48 -0700
Subject: [Freeswitch-users] Collect DTMF via mod_event_socket
In-Reply-To: <1FFF97C269757C458224B7C895F35F15107E3B@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F15107E3B@cantor.std.visionutv.se>
Message-ID: <CAP_69mucgf-YLfg92FAAM8SkepOXzdJEV9K-RV5g5-vdc9QX5A@mail.gmail.com>

Peter:

I tried subscribing to DTMF events by sending the following command via
mod_event_socket:

event DTMF

I don't seem to be getting any events back when I press a bunch of digits.



On Thu, Jun 14, 2012 at 2:04 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  Or just listen for DTMF events, and act on those. I personally find
> that?s the easiest approach.****
>
> ** **
>
> /Peter****
>
> ** **
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Bob Coleman
> *Skickat:* den 14 juni 2012 10:18
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] Collect DTMF via mod_event_socket****
>
> ** **
>
> Hi Venkatesh,
>
> You could use this:
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
>
> or this:
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read
>
> On the main page you go to the docs link at the top, then everything is
> linked off the left hand side
>
> Bob
>  ****
>  ------------------------------
>
> Date: Thu, 14 Jun 2012 01:07:13 -0700
> From: vvenkatar at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Collect DTMF via mod_event_socket
>
> Hello !****
>
> ** **
>
> I have been looking at the documentation but am unable to find it. I
> wanted to know the correct syntax I need to issue via mod_event_socket if I
> want to collect DTMF and report back the digits entered to my application.
> Great if anybody can provide me with pointers.****
>
> ** **
>
> Venkatesh****
>
> !DSPAM:4fd99c9a32769547714299! ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jnankin at gmail.com  Thu Jun 14 19:22:03 2012
From: jnankin at gmail.com (Joshua Nankin)
Date: Thu, 14 Jun 2012 10:22:03 -0500
Subject: [Freeswitch-users] Spandsp events not being fired
In-Reply-To: <CAAYWZ8mUFBF1rcU_ErEURDgjCpmr8bGeZA8CCCfeQKPj2sOfBg@mail.gmail.com>
References: <CAAYWZ8mUFBF1rcU_ErEURDgjCpmr8bGeZA8CCCfeQKPj2sOfBg@mail.gmail.com>
Message-ID: <CAAYWZ8mNw=5=vak_j7XjnkHij+DisxQdVmUg5acZdCPuLTXELw@mail.gmail.com>

Anyone?

I'm using the ESL to listen to events, and I'm listening for ALL.  Not
seeing any spandsp events.

On Wed, Jun 13, 2012 at 12:50 PM, Joshua Nankin <jnankin at gmail.com> wrote:

> I'm sending a fax with freeswitch, but the events listed at
> http://wiki.freeswitch.org/wiki/Mod_spandsp#Events do not seem to be
> fired.  Do I need to enable them somehow?
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From peter.olsson at visionutveckling.se  Thu Jun 14 19:36:41 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 14 Jun 2012 15:36:41 +0000
Subject: [Freeswitch-users] Collect DTMF via mod_event_socket
In-Reply-To: <CAP_69mucgf-YLfg92FAAM8SkepOXzdJEV9K-RV5g5-vdc9QX5A@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F15107E3B@cantor.std.visionutv.se>,
	<CAP_69mucgf-YLfg92FAAM8SkepOXzdJEV9K-RV5g5-vdc9QX5A@mail.gmail.com>
Message-ID: <1DC1FCF7-1427-4917-A14A-5AC2AB16AAFD@visionutveckling.se>

If i remember correctly, that should be event plain DTMF or something?

/Peter

14 jun 2012 kl. 17:24 skrev "Venkatesh" <vvenkatar at gmail.com<mailto:vvenkatar at gmail.com>>:

Peter:

I tried subscribing to DTMF events by sending the following command via mod_event_socket:

event DTMF

I don't seem to be getting any events back when I press a bunch of digits.



On Thu, Jun 14, 2012 at 2:04 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Or just listen for DTMF events, and act on those. I personally find that?s the easiest approach.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Bob Coleman
Skickat: den 14 juni 2012 10:18
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] Collect DTMF via mod_event_socket

Hi Venkatesh,

You could use this:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits

or this:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read

On the main page you go to the docs link at the top, then everything is linked off the left hand side

Bob

________________________________
Date: Thu, 14 Jun 2012 01:07:13 -0700
From: vvenkatar at gmail.com<mailto:vvenkatar at gmail.com>
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Collect DTMF via mod_event_socket

Hello !

I have been looking at the documentation but am unable to find it. I wanted to know the correct syntax I need to issue via mod_event_socket if I want to collect DTMF and report back the digits entered to my application. Great if anybody can provide me with pointers.

Venkatesh

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


!DSPAM:4fd9ff0d32761977123140!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


!DSPAM:4fd9ff0d32761977123140!


From jeff at jefflenk.com  Thu Jun 14 19:41:58 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 14 Jun 2012 08:41:58 -0700 (PDT)
Subject: [Freeswitch-users] Spandsp events not being fired
In-Reply-To: <CAAYWZ8mNw=5=vak_j7XjnkHij+DisxQdVmUg5acZdCPuLTXELw@mail.gmail.com>
References: <CAAYWZ8mUFBF1rcU_ErEURDgjCpmr8bGeZA8CCCfeQKPj2sOfBg@mail.gmail.com>
	<CAAYWZ8mNw=5=vak_j7XjnkHij+DisxQdVmUg5acZdCPuLTXELw@mail.gmail.com>
Message-ID: <1339688518364-7579856.post@n2.nabble.com>

Do you see the data in the debug log? The events are generated in the same
place.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Spandsp-events-not-being-fired-tp7579797p7579856.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Thu Jun 14 19:50:58 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 14 Jun 2012 08:50:58 -0700 (PDT)
Subject: [Freeswitch-users] Is SCA bridgeing Multithreaded?
In-Reply-To: <046801cd49a0$28819b70$7984d250$@bizfocused.com>
References: <046801cd49a0$28819b70$7984d250$@bizfocused.com>
Message-ID: <1339689058375-7579857.post@n2.nabble.com>

I think anthm basically answered this question in your other thread. What is
the problem that you are trying to solve. Is it the same audio startup
issue?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Is-SCA-bridgeing-Multithreaded-tp7579801p7579857.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Thu Jun 14 19:53:37 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Jun 2012 08:53:37 -0700
Subject: [Freeswitch-users] Developers' Salon,
	sched_broadcast from FreeSWITCH-CN
In-Reply-To: <E7C952BAB06744089BB455F648F8776E@gmail.com>
References: <E7C952BAB06744089BB455F648F8776E@gmail.com>
Message-ID: <CAKzWOxVpZyOwy-FQsY=u3epeOrs61c2FTbwxuugARH5y48mCSQ@mail.gmail.com>

Seven,

Thanks for promoting FreeSWITCH in the Chinese community and in China
itself. I hope that everything goes well for you. Also, I believe the title
of the book you are writing is just fine.

-MC

On Thu, Jun 14, 2012 at 7:56 AM, Seven Du <dujinfang at gmail.com> wrote:

> All,
>
> Sorry I didn't promote as it's mainly targeted to the Chinese Community
> and it's the first Salon in China and we want to keep a low profile.
>
> I'm the creator and maintainer of FreeSWITCH-CN(
> http://www.freeswitch.org.cn ). We'll have a Salon at this Sunday (June
> 17th, UTC+8) afternoon in Beijing, China.
>
>
> http://www.freeswitch.org.cn/blog/past/2012/4/16/freeswitchcnzhong-wen-she-qu-2012shou-jie-kai-fa-zhe-sha-long/
>
>
> According to http://freeswitch-cn.eventbrite.com/ , we have 56 registers
> till now and we do accept blind reg.
>
> It will be in a coffee bar and if anyone happens to drop by is welcome to
> have a cup of coffee.
>
> The Chinese Community is growing and I will can have more news to ClueCon.
>
> And I have one news now. I personally received the first donation a few
> days ago for the in-writing book: FreeSWITCH in Action (You won't use this
> name in the second edition of FS book, right? Michael :) ) , it's published
> on site using CC-BY-NC-ND.
>
> http://www.freeswitch.org.cn/document
>
> My aim of running the FreeSWITCH-CN community was to help Chinese speakers
> to learn FreeSWITCH. And I had put a Paypal donation button on and planed
> to donate back a percentage to the core developers but get nothing till now.
>
> I just figured out that there's basically no way for us legally collecting
> money in my country publicly. So I replaced the paypal donation to a guide
> link directly to freeswitch.org and if anyone donate with a remark of
> FreeSWITCH-CN it's probably from our community, or probably never.
>
> Anyway, I will use the first being-donated-money to buy a few things and
> will give out at the Salon, it's not that much, but it's something. And if
> there's future donations happens privately to me I will definitely happy to
> donate back to upstream.
>
> Ok, back to the Salon. I will report more here after the Salon and
> probably we could run a tiny ClueCon in China next year?
>
>
>
> Seven.
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> Sent with Sparrow <http://www.sparrowmailapp.com>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sdevoy at bizfocused.com  Thu Jun 14 20:02:57 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 14 Jun 2012 12:02:57 -0400
Subject: [Freeswitch-users] Is SCA bridgeing Multithreaded?
In-Reply-To: <1339689058375-7579857.post@n2.nabble.com>
References: <046801cd49a0$28819b70$7984d250$@bizfocused.com>
	<1339689058375-7579857.post@n2.nabble.com>
Message-ID: <02d701cd4a47$2a4d60e0$7ee822a0$@bizfocused.com>

I am sorry, I don't recall an answer, not can I find one in search, but
DIGEST grouping prevents most search for me.

No the audio issue was just too slow (single) processor  for Freeswitch.

My customer is complaining the SCA phones are ringing in sequence and not
"simultaneously".  She says the first ring on the first phone completes
before the second phone starts and so on for the 4 phones.  Two seconds is a
very long delay for an SCA ring.  I could switch to Enterprise originate to
individual extensions if that will help.

Sean

 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Thursday, June 14, 2012 11:51 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Is SCA bridgeing Multithreaded?

I think anthm basically answered this question in your other thread. What is
the problem that you are trying to solve. Is it the same audio startup
issue?

--
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Is-SCA-bridgeing-Multithreaded
-tp7579801p7579857.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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 Jun 14 20:40:52 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jun 2012 11:40:52 -0500
Subject: [Freeswitch-users]
	=?windows-1252?q?Error_message_=22_=5BERR=5D_s?=
	=?windows-1252?q?witch=5Fcpp=2Ecpp=3A48_Cannot_queue_any_more_even?=
	=?windows-1252?q?ts=93?=
In-Reply-To: <1339645894596-7579817.post@n2.nabble.com>
References: <1339645894596-7579817.post@n2.nabble.com>
Message-ID: <CAKbxfG9vD5KjpiPC2tMhnhHA-jGc-TkVpG7CF-7wgw84rdwvRQ@mail.gmail.com>

consume some?
That means you have built up 5000 unserviced events.

I added a patch to latest git to add a param for the size of the queue
but 5000 is fairly generous already.

EventConsumer("event_name", "", 100000)

On Wed, Jun 13, 2012 at 10:51 PM, tangdu <tang.du at hotmail.com> wrote:
>
> ? I use Mod event to fill channel cariables ?in my mysql table?When
> freeswitch runing some time?I got a lot of the following error messages:
> [ERR] switch_cpp.cpp:48 Cannot queue any more events?
> ?This messages telling me that i have consumed too many events in my event
> consumer object. ? it has reached its max. Then the channel cariables can
> not be filled in mysql?
> ? ?How can I solve this?
> Thanks?
>
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-message-ERR-switch-cpp-cpp-48-Cannot-queue-any-more-events-tp7579817.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From jeff at jefflenk.com  Thu Jun 14 21:03:26 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 14 Jun 2012 10:03:26 -0700 (PDT)
Subject: [Freeswitch-users] Is SCA bridgeing Multithreaded?
In-Reply-To: <02d701cd4a47$2a4d60e0$7ee822a0$@bizfocused.com>
References: <046801cd49a0$28819b70$7984d250$@bizfocused.com>
	<1339689058375-7579857.post@n2.nabble.com>
	<02d701cd4a47$2a4d60e0$7ee822a0$@bizfocused.com>
Message-ID: <1339693406060-7579861.post@n2.nabble.com>

This sounds like a dialplan configuration issue. Maybe you could post the
relevant portion of your dialplan for review.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Is-SCA-bridgeing-Multithreaded-tp7579801p7579861.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From drk at drkngs.net  Thu Jun 14 21:03:14 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Thu, 14 Jun 2012 10:03:14 -0700
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <CAO47HpcPqDm3gLCtMPoiooOnMAok+E-VhMAth6bG-3kJCjg9ew@mail.gmail.com>
Message-ID: <20120614170314.2f57f40d@mail.tritonwest.net>

I already have implmented this a while ago, however it's burried in a product. Now that I see there is some interest, I can pull it out, and make it a standalone module.   
   
It's implmented in a C# module which runs under mod_managed. Has a single IApiPlugin class so that mod_managed will keep it loaded, and so you have an API command to shut it down, so you can replace it while it's running, and another static class that implments INotifyPluginLoaded to start a thread that sits in an EventConsumer loop.   
   
The tracking of failures is done by updating a Dictionary<String, myTrakingClass>() where the Key is the IP address as a string. Since it's very easy to call any WMI/CIMV2 queries/updates from managed code, it can just add and remove either IPSEC filter, or Advanced firewall rules directly.  
   
If you all would like me to do this, please respond. The more the response, the higher it will be on my priority list.  
   
--Dave
      _____  

  From: jay binks [mailto:jaybinks at gmail.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Thu, 14 Jun 2012 02:55:50 -0700
Subject: Re: [Freeswitch-users] Brute-force attack

Hey ..  

  
On 14 June 2012 16:16, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
  Anyway, for this kind of setup I would also prefer Linux, but mostly for the possibilites with fail2ban etc, which doesn't exist on Windows. I'm thinking of writiling something similar for Windows, hopefully I get som time for that soon...

/Peter
  

  
if you do get round to writing something like this let me know.  ( not that I even have a windows box )  
but when I wrote the patches for FS to do the Fail2Ban compatible logging, I created ESL Events also   
with the intention that we could do a FS module "mod_security" or something ( no time given to the name )  

  
it would be fairly simple to move some of the fail2ban functionality into such a module that could either call out to simple scripts or  
insert firewall rules its self.     

  
I have some ideas here, and I would probably be interested in working on this with others.  

  
Jay  
       
   
 
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From drk at drkngs.net  Thu Jun 14 21:22:47 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Thu, 14 Jun 2012 10:22:47 -0700
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <CAMbW9oLMys-qBwoHpBhsGXbAF+hDeU12=T2P4yjtW-ZfY7Q+3A@mail.gmail.com>
Message-ID: <20120614172247.a483449d@mail.tritonwest.net>

The problem with that, is it's parsing the windows event log. You could add scripts/code to FS to write failures to the application event log (or a custom one), but that's a lot more work then just putting all the code to do it all in a managed DLL that can be invoked via mod_managed. No external processes, no invoking PowerShelll or WScript/CScript.  
   
On another note, if you want to do this w/ external scripts or other external methods of interfacing w/ freeswitch, you may want to play with the RC (RTM is about 60 days away) of Windows 8/Server 2012. WMF 3.0 (Windows Managment Framework 3.0) has been extended so that the HTTP interfaces which were only SOAP WS-Managment in v2, have been extended to more endpoints, including REST/XML and OData (www.odata.org like the google or Netflix APIs). This makes it easy to do directly from any scripting language or external system.  
   
--Dave
      _____  

  From: Andrew Cassidy [mailto:andrew at cassidywebservices.co.uk]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Thu, 14 Jun 2012 04:29:23 -0700
Subject: Re: [Freeswitch-users] Brute-force attack

It IS possible to use windows 7 powershell scripts run on a schedule to dynamically add and remove firewall rules, it's something a firend and I did before for RDP. The RDP version is here:  http://www.jonsdocs.org.uk/wiki/index.php/PSLogonFailures   

  
I'm sure with a little tweaking it can be made to work with freeswitch log files.

  
On 14 June 2012 11:58, Anton Kvashenkin <anton.jugatsu at gmail.com> wrote:
  For example, you don't need to open 5060 port to a whole world, just 5090 (the port you use for connecting road warriors). http://wiki.freeswitch.org/wiki/Nat   http://wiki.freeswitch.org/wiki/External_profile   
  


  
2012/6/14 ocset <ocset at the800group.com>
    
Anton

I know nothing about ext-rtp-ip and ext-sip-ip. Could you please explain how this will help in making the system more secure?

Thanks
O 
  
  

On 14/06/12 16:27, Anton Kvashenkin wrote:   I would suggest to create separate profile for remote workers. For example, external-road-warrios. So you can play with ext-rtp-ip and ext-sip-ip.

  
2012/6/14 Peter Olsson <peter.olsson at visionutveckling.se>
  I use both Windows and Linux systems. As long as you know how to manage both systems, there is not a big difference when it comes to exploits and general security (not anymore anyway, if you use current versions). I would say that the biggest issue here is the knowledge of the people managing the systems. And it's usually more secure to manage a system that you know, then a system you don't know much about, even though that system is considered to be more secure.

Lots has happened since Windows 95 :)

Anyway, for this kind of setup I would also prefer Linux, but mostly for the possibilites with fail2ban etc, which doesn't exist on Windows. I'm thinking of writiling something similar for Windows, hopefully I get som time for that soon...

/Peter

14 jun 2012 kl. 07:51 skrev "Muhammad Shahzad" <shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>>:
  

I would strongly suggest to move your production system to Linux, which is by far secure and controllable then Windows. Right now, if somebody does not breaks into your voip setup using some bruteforce / DOS attack, s/he can still exploit some hole in Windows to crack your security. Windows is simply not secure enough to production grade performance.

Thank you.


  
On Thu, Jun 14, 2012 at 6:39 AM, Avi Marcus <avi at avimarcus.net<mailto:avi at avimarcus.net>> wrote:
That's not necessarily the best kind of password... see http://xkcd.com/936/ and then http://tech.dropbox.com/?p=165

-Avi



  
On Thu, Jun 14, 2012 at 6:23 AM, jay binks <jaybinks at gmail.com<mailto:jaybinks at gmail.com>> wrote:
> Strong passwords are a great start, but fail2ban does a little more than
> this.
>
> you could move off port 5060 to something un-conventional, meaning your less
> likely to get scanned / brute forced.
>
> Jay
>
  
> On 14 June 2012 12:27, ocset <ocset at the800group.com<mailto:ocset at the800group.com>> wrote:
>>
>> Hi
>>
>> I have deployed Freeswiitch on windows 7 and since there is no fail2ban
>> on windows, I was wondering what the real risk is with opening it up to
>> the internet. If I was to ensure that all users and passwords were
>> extremely difficult to guess (passwords like "2$53E_d7?^2!3s$"), what
>> are the risks that I am exposing myself to? Is there a type of DoS for
>> voip where hackers can just flood my system with requests simply to be
>> malicious?
>>
>> There are VB windows scripts available that emulate what fail2ban does
>> on Linux but I was just wondering whether I really need to implement
>> this level of security if I can control the password complexity in
>> Freeswitch.
>>
>> Thanks
>> O
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
  
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> 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
>
>
>
>
> --
> Sincerely
>
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
  
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
  
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
  
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http://www.freeswitch.org




--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com<mailto:shari_786pk at hotmail.com>
Email: shaheryarkh at googlemail.com<mailto:shaheryarkh at googlemail.com>
!DSPAM:4fd978c432761360223007!
  
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
  
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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
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!DSPAM:4fd978c432761360223007!
  
  

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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Join Us At ClueCon - Aug 7-9, 2012

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_________________________________________________________________________  Professional FreeSWITCH Consulting Services:  consulting at freeswitch.org  http://www.freeswitchsolutions.com          Official FreeSWITCH Sites  http://www.freeswitch.org  http://wiki.freeswitch.org  http://www.cluecon.com    Join Us At ClueCon - Aug 7-9, 2012    FreeSWITCH-users mailing list  FreeSWITCH-users at lists.freeswitch.org  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users  http://www.freeswitch.org  
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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Join Us At ClueCon - Aug 7-9, 2012

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http://www.freeswitch.org



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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-- 
Andrew Cassidy BSc (Hons) MBCS SSCA  
Managing Director  
  

  

  
T 03300 100 960  F 03300 100 961  
E andrew at cassidywebservices.co.uk  
W www.cassidywebservices.co.uk
      
   
 
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From jeff at jefflenk.com  Thu Jun 14 21:25:37 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 14 Jun 2012 10:25:37 -0700 (PDT)
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <20120614170314.2f57f40d@mail.tritonwest.net>
References: <4FD94C26.10800@the800group.com>
	<CAO47Hpcuh7OFEgjyjAZx9bq9MrPPscqhx9159aaU03rv57=P9A@mail.gmail.com>
	<CALNZuwkF+Hw0HfpCONq0u8OJW29=tfTeAcudLYS69K04qAX2Ww@mail.gmail.com>
	<CAFZQphzkbg5_4tH0YmYLoJ4Ft_rN07nLHBWZMkcyWVbczAUe1A@mail.gmail.com>
	<D6D8040D-2350-4B59-9FC0-90B230537D25@visionutveckling.se>
	<CAO47HpcPqDm3gLCtMPoiooOnMAok+E-VhMAth6bG-3kJCjg9ew@mail.gmail.com>
	<20120614170314.2f57f40d@mail.tritonwest.net>
Message-ID: <1339694737970-7579863.post@n2.nabble.com>

Thanks Dave,

This would be a great contribution for windows users. I would be happy to
help is any way that I can.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Brute-force-attack-tp7579813p7579863.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ronmccar at gmail.com  Thu Jun 14 21:22:36 2012
From: ronmccar at gmail.com (Ron McCarthy)
Date: Thu, 14 Jun 2012 10:22:36 -0700
Subject: [Freeswitch-users] SCCP Question
In-Reply-To: <CAFZQphxb7BH3DBP6LH+BeHM4W2GcHvteb9WVtUXBeREDG3_S0Q@mail.gmail.com>
References: <CAMQkVDKNJbQcENH-23+QQrAmTxM2=iMVgxnZtw9syEhdnfO7_g@mail.gmail.com>
	<CAFZQphxb7BH3DBP6LH+BeHM4W2GcHvteb9WVtUXBeREDG3_S0Q@mail.gmail.com>
Message-ID: <CAMQkVDJGBQ=WryDZhZU+-Q=vJeaxvzx5-K8gFT-0t7+37HCGUw@mail.gmail.com>

Yeah, I would not even think of trying 75k+ phones, besides that would cost
about $500 zillion in Cisco money!

Ill start some testing and see what I find out, hopefully all goes well as
you said.

Thanks!

On Wed, Jun 13, 2012 at 11:33 PM, Muhammad Shahzad <
shaheryarkh at googlemail.com> wrote:

> Ah, i misread your email at first, thought you are setting up 75+K SCCP
> phones, while its just 75+ phones. :-)
>
> Anyways, yes FreeSWITCH works pretty good for SCCP, especially if you have
> previously used asterisk with them, otherwise you may run into problem with
> some phones which have very old firmware, you probably have to upgrade
> their firmware.
>
> I once setup a 60 SCCP phone setup with FreeSWITCH and it worked without
> much problem. So go on and if you run into problem you can contact this
> mailing list for help. ;-)
>
> Thank you.
>
>
> On Wed, Jun 13, 2012 at 11:12 PM, Ron McCarthy <ronmccar at gmail.com> wrote:
>
>> Hi List,
>>
>> Has anyone been running SCCP with a larger number of phones? Im looking
>> to deploy like 75+ phones and I want to keep SCCP so I don't have to
>> upgrade them and for the SLA, some phones also have no SIP software for
>> them so im forced to keep SCCP. Does anyone have any experience with this?
>> From what ive read the SCCP support works and works well, im just worried
>> about trying to run this many phones and if im missing any sort of issues
>> that could come up. I would like to use FreeSwitch instead of Asterisk for
>> this, just worried about functionality and the stability of it.
>>
>> Thanks!
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vvenkatar at gmail.com  Thu Jun 14 21:46:18 2012
From: vvenkatar at gmail.com (Venkatesh)
Date: Thu, 14 Jun 2012 10:46:18 -0700
Subject: [Freeswitch-users] Collect DTMF via mod_event_socket
In-Reply-To: <1DC1FCF7-1427-4917-A14A-5AC2AB16AAFD@visionutveckling.se>
References: <1FFF97C269757C458224B7C895F35F15107E3B@cantor.std.visionutv.se>
	<CAP_69mucgf-YLfg92FAAM8SkepOXzdJEV9K-RV5g5-vdc9QX5A@mail.gmail.com>
	<1DC1FCF7-1427-4917-A14A-5AC2AB16AAFD@visionutveckling.se>
Message-ID: <CAP_69mtSq-V0+NysxDunbo0V5J-uEOotK_o+5cQ1ZDihVyP3Ug@mail.gmail.com>

Sorry, that is what I was sending, but there were no events triggered when
I entered digits.

Venkatesh

On Thu, Jun 14, 2012 at 8:36 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> If i remember correctly, that should be event plain DTMF or something?
>
> /Peter
>
> 14 jun 2012 kl. 17:24 skrev "Venkatesh" <vvenkatar at gmail.com<mailto:
> vvenkatar at gmail.com>>:
>
> Peter:
>
> I tried subscribing to DTMF events by sending the following command via
> mod_event_socket:
>
> event DTMF
>
> I don't seem to be getting any events back when I press a bunch of digits.
>
>
>
> On Thu, Jun 14, 2012 at 2:04 AM, Peter Olsson <
> peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
> wrote:
> Or just listen for DTMF events, and act on those. I personally find that?s
> the easiest approach.
>
> /Peter
>
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:
> freeswitch-users-bounces at lists.freeswitch.org> [mailto:
> freeswitch-users-bounces at lists.freeswitch.org<mailto:
> freeswitch-users-bounces at lists.freeswitch.org>] F?r Bob Coleman
> Skickat: den 14 juni 2012 10:18
> Till: freeswitch-users at lists.freeswitch.org<mailto:
> freeswitch-users at lists.freeswitch.org>
> ?mne: Re: [Freeswitch-users] Collect DTMF via mod_event_socket
>
> Hi Venkatesh,
>
> You could use this:
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
>
> or this:
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read
>
> On the main page you go to the docs link at the top, then everything is
> linked off the left hand side
>
> Bob
>
> ________________________________
> Date: Thu, 14 Jun 2012 01:07:13 -0700
> From: vvenkatar at gmail.com<mailto:vvenkatar at gmail.com>
> To: freeswitch-users at lists.freeswitch.org<mailto:
> freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Collect DTMF via mod_event_socket
>
> Hello !
>
> I have been looking at the documentation but am unable to find it. I
> wanted to know the correct syntax I need to issue via mod_event_socket if I
> want to collect DTMF and report back the digits entered to my application.
> Great if anybody can provide me with pointers.
>
> Venkatesh
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>
> !DSPAM:4fd9ff0d32761977123140!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>
> !DSPAM:4fd9ff0d32761977123140!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.olsson at visionutveckling.se  Thu Jun 14 21:57:44 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 14 Jun 2012 17:57:44 +0000
Subject: [Freeswitch-users] Collect DTMF via mod_event_socket
In-Reply-To: <CAP_69mtSq-V0+NysxDunbo0V5J-uEOotK_o+5cQ1ZDihVyP3Ug@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F15107E3B@cantor.std.visionutv.se>
	<CAP_69mucgf-YLfg92FAAM8SkepOXzdJEV9K-RV5g5-vdc9QX5A@mail.gmail.com>
	<1DC1FCF7-1427-4917-A14A-5AC2AB16AAFD@visionutveckling.se>,
	<CAP_69mtSq-V0+NysxDunbo0V5J-uEOotK_o+5cQ1ZDihVyP3Ug@mail.gmail.com>
Message-ID: <1FFF97C269757C458224B7C895F35F15108FE1@cantor.std.visionutv.se>

The only thing I might think of, is that FS doesn't actually detect any DTMF's. If you have DEBUG logs enabled, you should also see info about DTMF in the log.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Venkatesh [vvenkatar at gmail.com]
Skickat: den 14 juni 2012 19:46
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Collect DTMF via mod_event_socket

Sorry, that is what I was sending, but there were no events triggered when I entered digits.

Venkatesh

On Thu, Jun 14, 2012 at 8:36 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
If i remember correctly, that should be event plain DTMF or something?

/Peter

14 jun 2012 kl. 17:24 skrev "Venkatesh" <vvenkatar at gmail.com<mailto:vvenkatar at gmail.com><mailto:vvenkatar at gmail.com<mailto:vvenkatar at gmail.com>>>:

Peter:

I tried subscribing to DTMF events by sending the following command via mod_event_socket:

event DTMF

I don't seem to be getting any events back when I press a bunch of digits.



On Thu, Jun 14, 2012 at 2:04 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se><mailto:peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>> wrote:
Or just listen for DTMF events, and act on those. I personally find that?s the easiest approach.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org><mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org><mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>>] F?r Bob Coleman
Skickat: den 14 juni 2012 10:18
Till: 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>>
?mne: Re: [Freeswitch-users] Collect DTMF via mod_event_socket

Hi Venkatesh,

You could use this:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits

or this:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read

On the main page you go to the docs link at the top, then everything is linked off the left hand side

Bob

________________________________
Date: Thu, 14 Jun 2012 01:07:13 -0700
From: vvenkatar at gmail.com<mailto:vvenkatar at gmail.com><mailto:vvenkatar at gmail.com<mailto:vvenkatar at gmail.com>>
To: 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>>
Subject: [Freeswitch-users] Collect DTMF via mod_event_socket

Hello !

I have been looking at the documentation but am unable to find it. I wanted to know the correct syntax I need to issue via mod_event_socket if I want to collect DTMF and report back the digits entered to my application. Great if anybody can provide me with pointers.

Venkatesh

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


!DSPAM:4fd9ff0d32761977123140!

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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
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!DSPAM:4fda21ca32761441613306!


From gabe at gundy.org  Thu Jun 14 22:13:33 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 14 Jun 2012 12:13:33 -0600
Subject: [Freeswitch-users] Brute-force attack
In-Reply-To: <4FD94C26.10800@the800group.com>
References: <4FD94C26.10800@the800group.com>
Message-ID: <CAMwZ-t+sqS8vcpfbaaEqurPf-EKBo_=5Z_jvtd9TGJZzUE+mcw@mail.gmail.com>

On Wed, Jun 13, 2012 at 8:27 PM, ocset <ocset at the800group.com> wrote:
> If I was to ensure that all users and passwords were
> extremely difficult to guess (passwords like "2$53E_d7?^2!3s$"), what
> are the risks that I am exposing myself to? Is there a type of DoS for
> voip where hackers can just flood my system with requests simply to be
> malicious?

If you happen to be using mod_xml_curl for your directory look ups,
the dumb dictionary attack can really hurt. If you have strong
passwords, it might not be them getting access to the system that
you'll need to worry about.  It's more likely getting your web and db
servers hammered that you'll need to worry about :)


Gabe


From netcentrica at gmail.com  Thu Jun 14 22:19:55 2012
From: netcentrica at gmail.com (Adam Raszynski)
Date: Thu, 14 Jun 2012 20:19:55 +0200
Subject: [Freeswitch-users] Spandsp events not being fired
In-Reply-To: <CAAYWZ8mNw=5=vak_j7XjnkHij+DisxQdVmUg5acZdCPuLTXELw@mail.gmail.com>
References: <CAAYWZ8mUFBF1rcU_ErEURDgjCpmr8bGeZA8CCCfeQKPj2sOfBg@mail.gmail.com>
	<CAAYWZ8mNw=5=vak_j7XjnkHij+DisxQdVmUg5acZdCPuLTXELw@mail.gmail.com>
Message-ID: <CAJWEF2bRs=qS9Ldep2LN+ADnja53iqDtRX6C+YZ6y1qePc0RgA@mail.gmail.com>

I use this and it handles answer, hangup and fax result events:

event plain CHANNEL_ANSWER CHANNEL_HANGUP CUSTOM spandsp::txfaxresult


you can also add spandsp::rxfaxresult at the end

2012/6/14 Joshua Nankin <jnankin at gmail.com>

> Anyone?
>
> I'm using the ESL to listen to events, and I'm listening for ALL.  Not
> seeing any spandsp events.
>
> On Wed, Jun 13, 2012 at 12:50 PM, Joshua Nankin <jnankin at gmail.com> wrote:
>
>> I'm sending a fax with freeswitch, but the events listed at
>> http://wiki.freeswitch.org/wiki/Mod_spandsp#Events do not seem to be
>> fired.  Do I need to enable them somehow?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From patrick at sunsus.net  Thu Jun 14 23:29:52 2012
From: patrick at sunsus.net (sunsus)
Date: Thu, 14 Jun 2012 12:29:52 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch 304 Unauthorized
In-Reply-To: <4FD7B4AE.6030805@cupis.co.uk>
References: <1339334863709-7579670.post@n2.nabble.com>
	<1339524483645-7579753.post@n2.nabble.com>
	<CAKV-jFwWLygHJVozz41=5rsjXMVuTrDsek98Rty2DL4zF1AC1w@mail.gmail.com>
	<1339526433720-7579758.post@n2.nabble.com>
	<4FD7962E.9070401@cupis.co.uk>
	<1339531138888-7579764.post@n2.nabble.com>
	<4FD7B4AE.6030805@cupis.co.uk>
Message-ID: <1339702192936-7579870.post@n2.nabble.com>

Hello Paul 

The Problem with the Hangup, was a policy issue on the phone because it
rejected non tls calls.  Sorry was my fault.

But no I changes page everything to sip.netvoip.ch and changed the bridge to
the gateway as you told me. But I have still the issue with authorization: 
Log: 
http://pastebin.freeswitch.org/19292

Gateway: 
http://pastebin.freeswitch.org/19294

bridge:
<action application="bridge"
data="sofia/gateway/netvoip-41440000000/0041$2$3" />

Do you really sure this is not a bug? 

regards

Patrick 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-401-Unauthorized-tp7579670p7579870.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From patrick at sunsus.net  Thu Jun 14 23:45:54 2012
From: patrick at sunsus.net (sunsus)
Date: Thu, 14 Jun 2012 12:45:54 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch 304 Unauthorized
In-Reply-To: <4FD7B4AE.6030805@cupis.co.uk>
References: <1339334863709-7579670.post@n2.nabble.com>
	<1339524483645-7579753.post@n2.nabble.com>
	<CAKV-jFwWLygHJVozz41=5rsjXMVuTrDsek98Rty2DL4zF1AC1w@mail.gmail.com>
	<1339526433720-7579758.post@n2.nabble.com>
	<4FD7962E.9070401@cupis.co.uk>
	<1339531138888-7579764.post@n2.nabble.com>
	<4FD7B4AE.6030805@cupis.co.uk>
Message-ID: <1339703154568-7579871.post@n2.nabble.com>

Hello Paul 

And if I change to the first server sip-1.netvoip.ch: 
freeswitch at internal> sofia_dig sip.netvoip.ch
Preference	    Weight	 Transport	      Port	   Address
================================================================================
         1	     0.100	       udp	      5060	62.65.137.113
         1	     0.100	       udp	      5060	62.65.137.114


everything works, so it seams to be a bug with a DNS Loadbalancing!?
http://pastebin.freeswitch.org/19295

regards

Patrick 


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-401-Unauthorized-tp7579670p7579871.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kris at kriskinc.com  Fri Jun 15 00:06:52 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 14 Jun 2012 16:06:52 -0400
Subject: [Freeswitch-users] Blog post about SIP
Message-ID: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>

FreeSWITCHers,

  As some of you may have noticed I frequently comment on SIP-related
FreeSWITCH posts.  Long story short I decided to finally put some of
these thoughts and experiences down in writing.  What I ended up with
is a 21 page (and counting) document describing various SIP header
fields, SDP information, RTP issues, DTMF issues, NAT traversal
technologies, interop headaches, etc.

  I wrote a little intro and linked to the document if any of you
would like to check it out (link at the end of post):

http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html

  Of course I'd appreciate any feedback you may have.  What do you
disagree with?  Find a typo?  Have burning SIP/RTP/SDP/T.38/codec
questions you'd like to see me address?  Should I get into more
implementation (FreeSWITCH) specific issues?  Let me know.

Thanks!

-- 
Kristian Kielhofner


From jeff at jefflenk.com  Fri Jun 15 00:54:14 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 14 Jun 2012 13:54:14 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch 304 Unauthorized
In-Reply-To: <1339703154568-7579871.post@n2.nabble.com>
References: <1339334863709-7579670.post@n2.nabble.com>
	<1339524483645-7579753.post@n2.nabble.com>
	<CAKV-jFwWLygHJVozz41=5rsjXMVuTrDsek98Rty2DL4zF1AC1w@mail.gmail.com>
	<1339526433720-7579758.post@n2.nabble.com>
	<4FD7962E.9070401@cupis.co.uk>
	<1339531138888-7579764.post@n2.nabble.com>
	<4FD7B4AE.6030805@cupis.co.uk>
	<1339703154568-7579871.post@n2.nabble.com>
Message-ID: <1339707254878-7579873.post@n2.nabble.com>

Read this old thread.

http://freeswitch-users.2379917.n2.nabble.com/Authorizations-when-using-DNS-SRV-bug-td3480547.html#a3480944

This is a configuration problem with dns on their side.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-401-Unauthorized-tp7579670p7579873.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gcd at i.ph  Fri Jun 15 02:02:29 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 15 Jun 2012 06:02:29 +0800
Subject: [Freeswitch-users] Blog post about SIP
In-Reply-To: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
References: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
Message-ID: <CAHKqMb5QmGJsFCTOXku3QqHZxJM1+Bu6tnP1kyzqJxpNm_UdJg@mail.gmail.com>

hi Kristian,

what a relief for me to browse your stuff. the RFC docs are daunting for
me. this is a nice companion to the RFCs. now i can understand the SIP
messages in FS much better. keep it up. i'll give u feedback whatever i see
needs correction. and ... THANKS A LOT!

-nandy

On Fri, Jun 15, 2012 at 4:06 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> FreeSWITCHers,
>
>  As some of you may have noticed I frequently comment on SIP-related
> FreeSWITCH posts.  Long story short I decided to finally put some of
> these thoughts and experiences down in writing.  What I ended up with
> is a 21 page (and counting) document describing various SIP header
> fields, SDP information, RTP issues, DTMF issues, NAT traversal
> technologies, interop headaches, etc.
>
>  I wrote a little intro and linked to the document if any of you
> would like to check it out (link at the end of post):
>
> http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html
>
>  Of course I'd appreciate any feedback you may have.  What do you
> disagree with?  Find a typo?  Have burning SIP/RTP/SDP/T.38/codec
> questions you'd like to see me address?  Should I get into more
> implementation (FreeSWITCH) specific issues?  Let me know.
>
> Thanks!
>
> --
> Kristian Kielhofner
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From curriegrad2004 at gmail.com  Fri Jun 15 04:34:48 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 14 Jun 2012 17:34:48 -0700
Subject: [Freeswitch-users] Developers' Salon,
	sched_broadcast from FreeSWITCH-CN
In-Reply-To: <CAKzWOxVpZyOwy-FQsY=u3epeOrs61c2FTbwxuugARH5y48mCSQ@mail.gmail.com>
References: <E7C952BAB06744089BB455F648F8776E@gmail.com>
	<CAKzWOxVpZyOwy-FQsY=u3epeOrs61c2FTbwxuugARH5y48mCSQ@mail.gmail.com>
Message-ID: <CAMWiMy88BAivJ8R14kgx+QuxG_F9wdxNtF5Tmo1JBhqzQKqaUA@mail.gmail.com>

Only in Beijing right now? It would be nice if there was also an event in
Hong Kong.

A mini cluecon there is also a great idea there for people who are
experiencing difficulties on obtaining visas for the USA.
On Jun 14, 2012 8:54 AM, "Michael Collins" <msc at freeswitch.org> wrote:

> Seven,
>
> Thanks for promoting FreeSWITCH in the Chinese community and in China
> itself. I hope that everything goes well for you. Also, I believe the title
> of the book you are writing is just fine.
>
> -MC
>
> On Thu, Jun 14, 2012 at 7:56 AM, Seven Du <dujinfang at gmail.com> wrote:
>
>> All,
>>
>> Sorry I didn't promote as it's mainly targeted to the Chinese Community
>> and it's the first Salon in China and we want to keep a low profile.
>>
>> I'm the creator and maintainer of FreeSWITCH-CN(
>> http://www.freeswitch.org.cn ). We'll have a Salon at this Sunday (June
>> 17th, UTC+8) afternoon in Beijing, China.
>>
>>
>> http://www.freeswitch.org.cn/blog/past/2012/4/16/freeswitchcnzhong-wen-she-qu-2012shou-jie-kai-fa-zhe-sha-long/
>>
>>
>> According to http://freeswitch-cn.eventbrite.com/ , we have 56 registers
>> till now and we do accept blind reg.
>>
>> It will be in a coffee bar and if anyone happens to drop by is welcome to
>> have a cup of coffee.
>>
>> The Chinese Community is growing and I will can have more news to ClueCon.
>>
>> And I have one news now. I personally received the first donation a few
>> days ago for the in-writing book: FreeSWITCH in Action (You won't use this
>> name in the second edition of FS book, right? Michael :) ) , it's published
>> on site using CC-BY-NC-ND.
>>
>> http://www.freeswitch.org.cn/document
>>
>> My aim of running the FreeSWITCH-CN community was to help Chinese
>> speakers to learn FreeSWITCH. And I had put a Paypal donation button on and
>> planed to donate back a percentage to the core developers but get nothing
>> till now.
>>
>> I just figured out that there's basically no way for us legally
>> collecting money in my country publicly. So I replaced the paypal donation
>> to a guide link directly to freeswitch.org and if anyone donate with a
>> remark of FreeSWITCH-CN it's probably from our community, or probably never.
>>
>> Anyway, I will use the first being-donated-money to buy a few things and
>> will give out at the Salon, it's not that much, but it's something. And if
>> there's future donations happens privately to me I will definitely happy to
>> donate back to upstream.
>>
>> Ok, back to the Salon. I will report more here after the Salon and
>> probably we could run a tiny ClueCon in China next year?
>>
>>
>>
>> Seven.
>>
>> --
>> About: http://about.me/dujinfang
>> Blog: http://www.dujinfang.com
>> Proj:  http://www.freeswitch.org.cn
>>
>> Sent with Sparrow <http://www.sparrowmailapp.com>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Jun 15 04:59:16 2012
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 15 Jun 2012 08:59:16 +0800
Subject: [Freeswitch-users] Developers' Salon,
 sched_broadcast from FreeSWITCH-CN
In-Reply-To: <CAMWiMy88BAivJ8R14kgx+QuxG_F9wdxNtF5Tmo1JBhqzQKqaUA@mail.gmail.com>
References: <E7C952BAB06744089BB455F648F8776E@gmail.com>
	<CAKzWOxVpZyOwy-FQsY=u3epeOrs61c2FTbwxuugARH5y48mCSQ@mail.gmail.com>
	<CAMWiMy88BAivJ8R14kgx+QuxG_F9wdxNtF5Tmo1JBhqzQKqaUA@mail.gmail.com>
Message-ID: <D6C19787369944BF9A5391A6DA532A4A@gmail.com>

Only in Beijing right now and it's not targeted internationally right now. As it's our first try. We have some members in Shenzhen who are eager to come but can't due to long travel. Probable we could try Shenzhen next year. Hongkong is also a good idea but we need a host, seems the author of mod_say_zh is HK based so it is possible. Anyway, when we get more experience we could do sth. bigger and better.

Thanks.


On Friday, June 15, 2012 at 8:34 AM, curriegrad2004 wrote:

> Only in Beijing right now? It would be nice if there was also an event in Hong Kong.
> A mini cluecon there is also a great idea there for people who are experiencing difficulties on obtaining visas for the USA.
> On Jun 14, 2012 8:54 AM, "Michael Collins" <msc at freeswitch.org (mailto:msc at freeswitch.org)> wrote:
> > Seven,
> > 
> > Thanks for promoting FreeSWITCH in the Chinese community and in China itself. I hope that everything goes well for you. Also, I believe the title of the book you are writing is just fine.
> > 
> > -MC
> > 
> > On Thu, Jun 14, 2012 at 7:56 AM, Seven Du <dujinfang at gmail.com (mailto:dujinfang at gmail.com)> wrote:
> > > All, 
> > > 
> > > Sorry I didn't promote as it's mainly targeted to the Chinese Community and it's the first Salon in China and we want to keep a low profile.
> > > 
> > > I'm the creator and maintainer of FreeSWITCH-CN(http://www.freeswitch.org.cn ). We'll have a Salon at this Sunday (June 17th, UTC+8) afternoon in Beijing, China. 
> > > 
> > > http://www.freeswitch.org.cn/blog/past/2012/4/16/freeswitchcnzhong-wen-she-qu-2012shou-jie-kai-fa-zhe-sha-long/  
> > > 
> > > According to http://freeswitch-cn.eventbrite.com/ , we have 56 registers till now and we do accept blind reg.
> > > 
> > > It will be in a coffee bar and if anyone happens to drop by is welcome to have a cup of coffee.
> > > 
> > > The Chinese Community is growing and I will can have more news to ClueCon.
> > > 
> > > And I have one news now. I personally received the first donation a few days ago for the in-writing book: FreeSWITCH in Action (You won't use this name in the second edition of FS book, right? Michael :) ) , it's published on site using CC-BY-NC-ND. 
> > > 
> > > http://www.freeswitch.org.cn/document
> > > 
> > > My aim of running the FreeSWITCH-CN community was to help Chinese speakers to learn FreeSWITCH. And I had put a Paypal donation button on and planed to donate back a percentage to the core developers but get nothing till now. 
> > > 
> > > I just figured out that there's basically no way for us legally collecting money in my country publicly. So I replaced the paypal donation to a guide link directly to freeswitch.org (http://freeswitch.org) and if anyone donate with a remark of FreeSWITCH-CN it's probably from our community, or probably never. 
> > > 
> > > Anyway, I will use the first being-donated-money to buy a few things and will give out at the Salon, it's not that much, but it's something. And if there's future donations happens privately to me I will definitely happy to donate back to upstream. 
> > > 
> > > Ok, back to the Salon. I will report more here after the Salon and probably we could run a tiny ClueCon in China next year?
> > > 
> > > 
> > > 
> > > Seven.
> > > 
> > > -- 
> > > About: http://about.me/dujinfang
> > > Blog: http://www.dujinfang.com
> > > Proj:  http://www.freeswitch.org.cn
> > > 
> > > Sent with Sparrow (http://www.sparrowmailapp.com)
> > > 
> > > 
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > > http://www.freeswitchsolutions.com
> > > 
> > > 
> > > 
> > > 
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > > 
> > > Join Us At ClueCon - Aug 7-9, 2012
> > > 
> > > 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
> > > 
> > 
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > Join Us At ClueCon - Aug 7-9, 2012
> > 
> > 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
> > 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> 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
> 
> 


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From chris at opencsta.org  Fri Jun 15 05:05:06 2012
From: chris at opencsta.org (Chris Mylonas)
Date: Fri, 15 Jun 2012 11:05:06 +1000
Subject: [Freeswitch-users] Blog post about SIP
In-Reply-To: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
References: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
Message-ID: <67350785-202E-47BA-9D04-B41581F22F71@opencsta.org>

Hi Kristian,

A similar thing happened on the apache-tapestry mailing list to Igor [1]
He was writing a "Tapestry 5 In Action" book for Manning Publications and after the initial MEAP (early access program for enthusiastic readers!) it got cut because of the lack of numbers.

He didn't have any trouble from Manning to take his existing work and go down the self-publishing route with lulu [2] - he's a pretty busy but friendly fellow.

HTH,
Chris


[1] http://blog.tapestry5.de/index.php/news/
[2] http://www.lulu.com/



On 15/06/2012, at 6:06 AM, Kristian Kielhofner wrote:

> FreeSWITCHers,
> 
>  As some of you may have noticed I frequently comment on SIP-related
> FreeSWITCH posts.  Long story short I decided to finally put some of
> these thoughts and experiences down in writing.  What I ended up with
> is a 21 page (and counting) document describing various SIP header
> fields, SDP information, RTP issues, DTMF issues, NAT traversal
> technologies, interop headaches, etc.
> 
>  I wrote a little intro and linked to the document if any of you
> would like to check it out (link at the end of post):
> 
> http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html
> 
>  Of course I'd appreciate any feedback you may have.  What do you
> disagree with?  Find a typo?  Have burning SIP/RTP/SDP/T.38/codec
> questions you'd like to see me address?  Should I get into more
> implementation (FreeSWITCH) specific issues?  Let me know.
> 
> Thanks!
> 
> -- 
> Kristian Kielhofner
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sipmaillist at gmail.com  Fri Jun 15 07:53:07 2012
From: sipmaillist at gmail.com (Jakson Kalsson)
Date: Fri, 15 Jun 2012 11:53:07 +0800
Subject: [Freeswitch-users] AMR and AMR-WB support
Message-ID: <CAG8p5B1A-ewBY3RZy0pforfx2D0o6sp68O1PWTGqUEO3QDbTtg@mail.gmail.com>

Hi all, I have a project for the 3G related, AMR and AMR-WB support.

I'm using the client develop suite from the PortSIP(http://www.portsip.com),
as their said
support the AMR, AMR-WB with RFC4867.

Now I have to setup a SIP server/SIP PBX in our Lab for test, does the
Freeswitch
support these codecs and RFC4867 ?

Also, any other Server/PBX which support AMR, AMR-WB recommended are
welcome.


Best regards,
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From unau0 at gmx.de  Fri Jun 15 09:44:26 2012
From: unau0 at gmx.de (Mitja)
Date: Fri, 15 Jun 2012 07:44:26 +0200
Subject: [Freeswitch-users] Where are my freetdm sip_headers?
Message-ID: <4FDACBBA.5060302@gmx.de>

Hello all,

I'm have some problems with my freetdm module lately. To be more precise 
freetdm with ftmod_sangoma_isdn for ISDN handling.

One of which is, that I cant get early media to be played in pre_answer 
state (aka progress_media). But I think I saw a Jira on that one, where 
I will try to get the information/contribute if possible.

The other problem I have is that on an incoming call I cant reliably get 
the TON for displaying purpouses. Actually Display would be alright, but 
calling back the other side is just uncomfortable if your number is 
missing a leading 0.
So I thought I just have to activate the sip_headers in 
autoloads/freetdm.conf.xml as described here:
http://wiki.freeswitch.org/wiki/Mod_freetdm#Channel_Variables (actually 
right above)

But when I call my info Application, I cant see any of the headers.

freetdm.conf.xml: (I set the param sip_headers three times after several 
tries)
<configuration name="freetdm.conf" description="Freetdm Configuration">
<settings>
<param name="debug" value="0"/>
<param name="sip_headers" value="true" />
<!--<param name="hold-music" value="$${moh_uri}"/>-->
<!--<param name="enable-analog-option" value="call-swap"/>-->
<!--<param name="enable-analog-option" value="3-way"/>-->
</settings>
<config_profiles>
<profile name="my_pri_te_1">
<param name="switchtype" value="euroisdn" />
<param name="interface" value="pri_cpe"/>
<param name="sip_headers" value="true" />
</profile>
</config_profiles>
<sangoma_pri_spans>
<span name="wp1" cfgprofile="my_pri_te_1">
<param name="dialplan" value="XML"/>
<param name="context" value="from_pstn"/>
<param name="tei" value="0"/>
<param name="overlap" value="no"/>
<param name="min_digits" value="4"/>
<param name="setup arbitration" value="no"/>
<param name="facility" value="yes"/>
<xparam name="facility-timeout" value="0"/>
<param name="outbound-called-ton" value="national"/>
<param name="outbound-called-npi" value="ISDN"/>
<param name="outbound-calling-ton" value="national"/>
<param name="outbound-calling-npi" value="ISDN"/>
<param name="outbound-rdnis-ton" value="national"/>
<param name="outbound-rdnis-npi" value="ISDN"/>
<param name="outbound-bearer_cap" value="speech"/>
<param name="outbound-bearer_layer1" value="alaw"/>
<param name="sip_headers" value="true" />
<yparam name="early-media-override" value="on-alert" />
<xparam name="local-number" value=""/>
</span>
</sangoma_pri_spans>
<analog_spans>
</analog_spans>
</configuration>


Output of Info on incoming call:
2012-06-15 07:19:41.691799 [INFO] mod_dptools.c:1463 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-Call-State: [RINGING]
Channel-State-Number: [4]
Channel-Name: [FreeTDM/1:6/8001]
Unique-ID: [7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-HIT-Dialplan: [true]
Channel-Call-UUID: [7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
Answer-State: [ringing]
Channel-Read-Codec-Name: [PCMA]
Channel-Read-Codec-Rate: [8000]
Channel-Read-Codec-Bit-Rate: [64000]
Channel-Write-Codec-Name: [PCMA]
Channel-Write-Codec-Rate: [8000]
Channel-Write-Codec-Bit-Rate: [64000]
Caller-Direction: [inbound]
Caller-Username: [FreeTDM]
Caller-Dialplan: [XML]
Caller-Caller-ID-Number: [152532xxxxx]
Caller-ANI: [152532xxxxx]
Caller-Destination-Number: [8001]
Caller-Unique-ID: [7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
Caller-Source: [mod_freetdm]
Caller-Context: [from_pstn]
Caller-Channel-Name: [FreeTDM/1:6/8001]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1339737581650800]
Caller-Channel-Created-Time: [1339737581650800]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_direction: [inbound]
variable_uuid: [7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
variable_session_id: [18]
variable_read_codec: [PCMA]
variable_read_rate: [8000]
variable_write_codec: [PCMA]
variable_write_rate: [8000]
variable_channel_name: [FreeTDM/1:6/8001]
variable_freetdm_span_name: [wp1]
variable_freetdm_span_number: [1]
variable_freetdm_chan_number: [6]
variable_freetdm_bearer_capability: [0]
variable_freetdm_bearer_layer1: [3]
variable_freetdm_calling_party_category: [unknown]
variable_screening_ind: [network-provided]
variable_presentation_ind: [presentation-allowed]
variable_call_uuid: [7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
variable_process_cdr: [false]
variable_effective_caller_id_number: [0152532xxxxx]
variable_et_first_dialed_ext: [8001]
variable_absolute_codec_string: [PCMA]
variable_continue_on_fail: [true]
variable_hangup_after_bridge: [false]
variable_nolocal:execute_on_ring: [hash 
insert/last_caller/8001/7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
variable_nolocal:ignore_display_updates: [true]
variable_export_vars: 
[absolute_codec_string,nolocal:execute_on_ring,nolocal:ignore_display_updates]
variable_record_session: [015253xxxxx.8001.2012-6-15-7-19-41.wav]
variable_execute_on_answer: [lua pickup_group_cleanup.lua 
7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
variable_api_hangup_hook: [lua pickup_group_cleanup.lua 
7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
variable_current_application: [info]


Am I overlooking something, is my ISDN knowledge just terrible or do I 
have to configure something another way round?

Thanks in advance!

Regards
Mitja



From unau0 at gmx.de  Fri Jun 15 09:55:18 2012
From: unau0 at gmx.de (Mitja)
Date: Fri, 15 Jun 2012 07:55:18 +0200
Subject: [Freeswitch-users] Where are my freetdm sip_headers?
In-Reply-To: <4FDACBBA.5060302@gmx.de>
References: <4FDACBBA.5060302@gmx.de>
Message-ID: <4FDACE46.90905@gmx.de>

Two pieces of information I forgot:
freeswitch at xxx.xxx.xxx.xxx@internal> version
FreeSWITCH Version 1.2.0-rc2

cat .git/refs/heads/master
fc2bb00eb19edc245ae6e9bcdff814c07f1e603b from June 11th

libsng_isdn version 7.20.2

Greetz
Mitja

> Hello all,
>
> I'm have some problems with my freetdm module lately. To be more 
> precise freetdm with ftmod_sangoma_isdn for ISDN handling.
>
> One of which is, that I cant get early media to be played in 
> pre_answer state (aka progress_media). But I think I saw a Jira on 
> that one, where I will try to get the information/contribute if possible.
>
> The other problem I have is that on an incoming call I cant reliably 
> get the TON for displaying purpouses. Actually Display would be 
> alright, but calling back the other side is just uncomfortable if your 
> number is missing a leading 0.
> So I thought I just have to activate the sip_headers in 
> autoloads/freetdm.conf.xml as described here:
> http://wiki.freeswitch.org/wiki/Mod_freetdm#Channel_Variables 
> (actually right above)
>
> But when I call my info Application, I cant see any of the headers.
>
> freetdm.conf.xml: (I set the param sip_headers three times after 
> several tries)
> <configuration name="freetdm.conf" description="Freetdm Configuration">
> <settings>
> <param name="debug" value="0"/>
> <param name="sip_headers" value="true" />
> <!--<param name="hold-music" value="$${moh_uri}"/>-->
> <!--<param name="enable-analog-option" value="call-swap"/>-->
> <!--<param name="enable-analog-option" value="3-way"/>-->
> </settings>
> <config_profiles>
> <profile name="my_pri_te_1">
> <param name="switchtype" value="euroisdn" />
> <param name="interface" value="pri_cpe"/>
> <param name="sip_headers" value="true" />
> </profile>
> </config_profiles>
> <sangoma_pri_spans>
> <span name="wp1" cfgprofile="my_pri_te_1">
> <param name="dialplan" value="XML"/>
> <param name="context" value="from_pstn"/>
> <param name="tei" value="0"/>
> <param name="overlap" value="no"/>
> <param name="min_digits" value="4"/>
> <param name="setup arbitration" value="no"/>
> <param name="facility" value="yes"/>
> <xparam name="facility-timeout" value="0"/>
> <param name="outbound-called-ton" value="national"/>
> <param name="outbound-called-npi" value="ISDN"/>
> <param name="outbound-calling-ton" value="national"/>
> <param name="outbound-calling-npi" value="ISDN"/>
> <param name="outbound-rdnis-ton" value="national"/>
> <param name="outbound-rdnis-npi" value="ISDN"/>
> <param name="outbound-bearer_cap" value="speech"/>
> <param name="outbound-bearer_layer1" value="alaw"/>
> <param name="sip_headers" value="true" />
> <yparam name="early-media-override" value="on-alert" />
> <xparam name="local-number" value=""/>
> </span>
> </sangoma_pri_spans>
> <analog_spans>
> </analog_spans>
> </configuration>
>
>
> Output of Info on incoming call:
> 2012-06-15 07:19:41.691799 [INFO] mod_dptools.c:1463 CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-Call-State: [RINGING]
> Channel-State-Number: [4]
> Channel-Name: [FreeTDM/1:6/8001]
> Unique-ID: [7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
> Call-Direction: [inbound]
> Presence-Call-Direction: [inbound]
> Channel-HIT-Dialplan: [true]
> Channel-Call-UUID: [7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
> Answer-State: [ringing]
> Channel-Read-Codec-Name: [PCMA]
> Channel-Read-Codec-Rate: [8000]
> Channel-Read-Codec-Bit-Rate: [64000]
> Channel-Write-Codec-Name: [PCMA]
> Channel-Write-Codec-Rate: [8000]
> Channel-Write-Codec-Bit-Rate: [64000]
> Caller-Direction: [inbound]
> Caller-Username: [FreeTDM]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Number: [152532xxxxx]
> Caller-ANI: [152532xxxxx]
> Caller-Destination-Number: [8001]
> Caller-Unique-ID: [7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
> Caller-Source: [mod_freetdm]
> Caller-Context: [from_pstn]
> Caller-Channel-Name: [FreeTDM/1:6/8001]
> Caller-Profile-Index: [1]
> Caller-Profile-Created-Time: [1339737581650800]
> Caller-Channel-Created-Time: [1339737581650800]
> Caller-Channel-Answered-Time: [0]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_direction: [inbound]
> variable_uuid: [7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
> variable_session_id: [18]
> variable_read_codec: [PCMA]
> variable_read_rate: [8000]
> variable_write_codec: [PCMA]
> variable_write_rate: [8000]
> variable_channel_name: [FreeTDM/1:6/8001]
> variable_freetdm_span_name: [wp1]
> variable_freetdm_span_number: [1]
> variable_freetdm_chan_number: [6]
> variable_freetdm_bearer_capability: [0]
> variable_freetdm_bearer_layer1: [3]
> variable_freetdm_calling_party_category: [unknown]
> variable_screening_ind: [network-provided]
> variable_presentation_ind: [presentation-allowed]
> variable_call_uuid: [7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
> variable_process_cdr: [false]
> variable_effective_caller_id_number: [0152532xxxxx]
> variable_et_first_dialed_ext: [8001]
> variable_absolute_codec_string: [PCMA]
> variable_continue_on_fail: [true]
> variable_hangup_after_bridge: [false]
> variable_nolocal:execute_on_ring: [hash 
> insert/last_caller/8001/7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
> variable_nolocal:ignore_display_updates: [true]
> variable_export_vars: 
> [absolute_codec_string,nolocal:execute_on_ring,nolocal:ignore_display_updates]
> variable_record_session: [015253xxxxx.8001.2012-6-15-7-19-41.wav]
> variable_execute_on_answer: [lua pickup_group_cleanup.lua 
> 7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
> variable_api_hangup_hook: [lua pickup_group_cleanup.lua 
> 7e1f9f6c-f5e3-42ac-9311-341d09dcb7be]
> variable_current_application: [info]
>
>
> Am I overlooking something, is my ISDN knowledge just terrible or do I 
> have to configure something another way round?
>
> Thanks in advance!
>
> Regards
> Mitja
>



From jaasmailing at gmail.com  Fri Jun 15 14:09:24 2012
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Fri, 15 Jun 2012 12:09:24 +0200
Subject: [Freeswitch-users] BYE authentication problem
Message-ID: <4FDB09D4.4050703@gmail.com>

Hello,

my FS has a trunk with an openser:

FS -> openser (sip.test.com - 1.2.3.4) -> cisco gateway

I have configured a gateway with this sip proxy / realm and I have a 
problem with BYE.
When I call from PSTN (cisco gateway) to FS, the cisco device set a 
realm "1.2.3.4" and openser ask for authentication, while FS tell:

2012-06-14 14:23:44.018150 [ERR] sofia_reg.c:2160 Cannot locate any 
authentication credentials to complete an authentication request for 
realm '"1.2.3.4"'

I know that FS asks that realm configured in the gateway section 
(sip.test.com) should match the realm in the challenge packet from the 
other device (that instead is 1.2.3.4), but the Reverse DNS of 1.2.3.4 
is sip.test.com (and viceversa).

Should FS respond to the challenge packet? How can I solve this problem?


Best Regards,
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From awais-nazeer at hotmail.com  Fri Jun 15 17:17:04 2012
From: awais-nazeer at hotmail.com (awais nazir)
Date: Fri, 15 Jun 2012 18:17:04 +0500
Subject: [Freeswitch-users] freeswitch to write cdr after every bridge
 attempt (failed or successful
Message-ID: <BLU162-W1132ABF1876877BBBFAE4788FB0@phx.gbl>


Hi

I am trying to use following context to get failover gateways working

    <action application="set" data="hangup_after_bridge=true"/>
    <action application="bridge" data="sofia/local_profile/1111 at example1.company.com" />
    <action application="bridge" data="sofia/local_profile/1111 at example2.company.com" />

if call on first gateway fails , call connection is attempted on second gateway but FS is writing only one CDR.How can it write CDR on every bridge attempt.

BR
--waisee

 		 	   		  
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From ocset at the800group.com  Fri Jun 15 17:12:56 2012
From: ocset at the800group.com (ocset)
Date: Fri, 15 Jun 2012 21:12:56 +0800
Subject: [Freeswitch-users] Best place to set variables?
Message-ID: <4FDB34D8.1040208@the800group.com>

Hi

In the FreeSWITCH-Cookbook, there is the following example

<extension name="ring_sequentially">
<condition field="destination_number" expression="^(2001)$">
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge" 
data={ignore_early_media=true}sofia/internal/userA at local.pbx.com"/>
<action application="bridge" 
data={ignore_early_media=true}sofia/internal/userB at local.pbx.com"/>
</condition>
</extension>

Is there any reason why the "ignore_early_media=true" was not set before 
the bridge commands as is done with the "continue_on_fail"  and 
"hangup_after_bridge" variables?

Example:

<extension name="ring_sequentially">
<condition field="destination_number" expression="^(2001)$">
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="ignore_early_media=true"/>
<action application="bridge" data=sofia/internal/userA at local.pbx.com"/>
<action application="bridge" data=sofia/internal/userB at local.pbx.com"/>
</condition>
</extension>

Thanks
O


From ocset at the800group.com  Fri Jun 15 17:40:14 2012
From: ocset at the800group.com (ocset)
Date: Fri, 15 Jun 2012 21:40:14 +0800
Subject: [Freeswitch-users] No dial-tone on Siemens C610
Message-ID: <4FDB3B3E.6060300@the800group.com>

Hi

I have deployed 3 cordless Siemens C610 phones on FS and was very 
surprised that there is no dial-tone when a user makes a call. The users 
hears a sound similar to an engage signal while FS makes the connection, 
followed by the sound of the phone ringing once the connection is 
established. (hope that makes sense).

Siemens reckons that is the way their phones behave. Is there a way for 
FS to simulate a dial-tone while the user waits for the other parties 
phone to start ringing?

Thanks
O


From avi at avimarcus.net  Fri Jun 15 17:47:21 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 15 Jun 2012 16:47:21 +0300
Subject: [Freeswitch-users] Best place to set variables?
In-Reply-To: <4FDB34D8.1040208@the800group.com>
References: <4FDB34D8.1040208@the800group.com>
Message-ID: <CALNZuwnB0xMDASeKAG8qX20DYfpq-nbgr06PTO4HgaD6nr+Vqg@mail.gmail.com>

Short answer: some variables are for the A leg, and some are for the B leg.
The two in their own set tags are for the A leg's execution of the
bridge.. whether it should keep trying or not.
The one inside the bridge is specific to that leg and bridge attempt
of how it should behave.

Perhaps someone wants to give the longer answer, but I don't have time
right now.

-Avi


On Fri, Jun 15, 2012 at 4:12 PM, ocset <ocset at the800group.com> wrote:
> Hi
>
> In the FreeSWITCH-Cookbook, there is the following example
>
> <extension name="ring_sequentially">
> <condition field="destination_number" expression="^(2001)$">
> <action application="set" data="continue_on_fail=true"/>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="bridge"
> data={ignore_early_media=true}sofia/internal/userA at local.pbx.com"/>
> <action application="bridge"
> data={ignore_early_media=true}sofia/internal/userB at local.pbx.com"/>
> </condition>
> </extension>
>
> Is there any reason why the "ignore_early_media=true" was not set before
> the bridge commands as is done with the "continue_on_fail" ?and
> "hangup_after_bridge" variables?
>
> Example:
>
> <extension name="ring_sequentially">
> <condition field="destination_number" expression="^(2001)$">
> <action application="set" data="continue_on_fail=true"/>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set" data="ignore_early_media=true"/>
> <action application="bridge" data=sofia/internal/userA at local.pbx.com"/>
> <action application="bridge" data=sofia/internal/userB at local.pbx.com"/>
> </condition>
> </extension>
>
> Thanks
> O
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From avi at avimarcus.net  Fri Jun 15 17:51:09 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 15 Jun 2012 16:51:09 +0300
Subject: [Freeswitch-users] freeswitch to write cdr after every bridge
 attempt (failed or successful
In-Reply-To: <BLU162-W1132ABF1876877BBBFAE4788FB0@phx.gbl>
References: <BLU162-W1132ABF1876877BBBFAE4788FB0@phx.gbl>
Message-ID: <CALNZuwn_b2A8tBb270rAmEzX+wm48no8Stf7NjYPPqqAkJnonA@mail.gmail.com>

By default, the modules for logging only log the A leg of the call.
You can turn on b-leg logging in whichever module you are using for
call logs. Do note you'll have to reconstruct the call then, based on
which is the A-leg and which is B leg, and using bridge_uuid,
signal_bond or last_bridge_to of the B leg (whichever is set) to see
which A leg it came from.

(Also note on enterprise originates: they are reported as leg A in
xml_cdr and need to be tied using the variable:
ent_originate_aleg_uuid)

This kind of "advanced" CDR handling should probably be wikified...

-Avi


On Fri, Jun 15, 2012 at 4:17 PM, awais nazir <awais-nazeer at hotmail.com> wrote:
> Hi
>
> I am trying to use following context to get failover gateways working
>
> ??? <action application="set" data="hangup_after_bridge=true"/>
> ??? <action application="bridge"
> data="sofia/local_profile/1111 at example1.company.com" />
> ??? <action application="bridge"
> data="sofia/local_profile/1111 at example2.company.com" />
>
> if call on first gateway fails , call connection is attempted on second
> gateway but FS is writing only one CDR.How can it write CDR on every bridge
> attempt.
>
> BR
> --waisee
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From eduardonunesp at gmail.com  Fri Jun 15 18:00:24 2012
From: eduardonunesp at gmail.com (Eduardo Nunes Pereira)
Date: Fri, 15 Jun 2012 11:00:24 -0300
Subject: [Freeswitch-users] Multiple ring with mod_fifo
Message-ID: <61FED3F76A9C483D8D3DD63E73587029@gmail.com>

Hi, folks 

I have two members in my fifo, and i want to ring the two members when a call comes into fifo, so i had tried many ways to accomplish, but no success, my fifo file conf is:

<configuration name="fifo.conf" description="FIFO Configuration">
  <settings>
    <!--<param name="delete-all-outbound-member-on-startup" value="false"/> -->
  </settings>
  <fifos>
    <fifo name="cool_fifo@$${domain}" importance="0">
      <member timeout="60" simo="1" lag="20">{call_timeout=30,fifo_member_wait=wait}user/2001@$${domain}</member>
      <member timeout="60" simo="1" lag="20">{call_timeout=30,fifo_member_wait=wait}user/1000@$${domain}</member>
    </fifo>
  </fifos>
</configuration>


I have tried put simo="2", but nothing, sorry my limited idea of fifo, is possible to ring multiple phones registered in the fifo ?  

--  
Eduardo Nunes Pereira
skype: eduardonunesp
msn: eduardonunesp
http://about.me/eduardonunesp

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From fvillarroel at yahoo.com  Fri Jun 15 18:22:02 2012
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Fri, 15 Jun 2012 07:22:02 -0700 (PDT)
Subject: [Freeswitch-users] setup variable
Message-ID: <1339770122.54258.YahooMailClassic@web160301.mail.bf1.yahoo.com>

I need setup some variable like foo to my cdr from lcr :
 
My lcr.conf.xml
 
 <param name="export_fields" value="foo"/>
 
 
 <param name="custom_sql" value="select a.foo as foo
 ......
 
 
 Returning Dialstring
 [lcr_carrier=carrier1,lcr_rate=0.15000,nibble_account=1004,nibble_rate=0.15000,lowbal_amt=0,nobal_amt=0,foo=1]user/1006
 
 My dialplan:
 
 <extension name="Outbound to PSTN 11 Digits">
              
 <condition field="destination_number"  expression="^(10[01][0-9])$">
                
         <action application="set"  data="hangup_after_bridge=false"/>
                
         <action application="set"  data="continue_on_fail=true"/>
                
         <action application="lcr"  data="$1 pg_prefix2"/>
 
                
         <action application="set"  data="foo=${foo}"/>
                
         <action application="bridge" data="${lcr_auto_route}"/>
                
 </condition>
</extension>
 
 The log show :
 
 mod_dptools.c:1294 sofia/internal/1004 at 192.168.1.108 SET
 [foo]=[UNDEF]
 
 
 So how i can get the value of the variable "foo" and set
 for my CDR?
 
 Regards.


From ocset at the800group.com  Fri Jun 15 18:30:52 2012
From: ocset at the800group.com (ocset)
Date: Fri, 15 Jun 2012 22:30:52 +0800
Subject: [Freeswitch-users] Best place to set variables?
In-Reply-To: <CALNZuwnB0xMDASeKAG8qX20DYfpq-nbgr06PTO4HgaD6nr+Vqg@mail.gmail.com>
References: <4FDB34D8.1040208@the800group.com>
	<CALNZuwnB0xMDASeKAG8qX20DYfpq-nbgr06PTO4HgaD6nr+Vqg@mail.gmail.com>
Message-ID: <4FDB471C.8080306@the800group.com>

Thanks for your answer.

What you are saying is that the "continue_on_fail=true" and 
"hangup_after_bridge=true" variables are set back to their default 
values in the"B" leg of the dailplan. I thought the variable that are 
set would persist for the whole <condition> section.

Regards
O

On 15/06/12 21:47, Avi Marcus wrote:
> Short answer: some variables are for the A leg, and some are for the B leg.
> The two in their own set tags are for the A leg's execution of the
> bridge.. whether it should keep trying or not.
> The one inside the bridge is specific to that leg and bridge attempt
> of how it should behave.
>
> Perhaps someone wants to give the longer answer, but I don't have time
> right now.
>
> -Avi
>
>
> On Fri, Jun 15, 2012 at 4:12 PM, ocset<ocset at the800group.com>  wrote:
>> Hi
>>
>> In the FreeSWITCH-Cookbook, there is the following example
>>
>> <extension name="ring_sequentially">
>> <condition field="destination_number" expression="^(2001)$">
>> <action application="set" data="continue_on_fail=true"/>
>> <action application="set" data="hangup_after_bridge=true"/>
>> <action application="bridge"
>> data={ignore_early_media=true}sofia/internal/userA at local.pbx.com"/>
>> <action application="bridge"
>> data={ignore_early_media=true}sofia/internal/userB at local.pbx.com"/>
>> </condition>
>> </extension>
>>
>> Is there any reason why the "ignore_early_media=true" was not set before
>> the bridge commands as is done with the "continue_on_fail"  and
>> "hangup_after_bridge" variables?
>>
>> Example:
>>
>> <extension name="ring_sequentially">
>> <condition field="destination_number" expression="^(2001)$">
>> <action application="set" data="continue_on_fail=true"/>
>> <action application="set" data="hangup_after_bridge=true"/>
>> <action application="set" data="ignore_early_media=true"/>
>> <action application="bridge" data=sofia/internal/userA at local.pbx.com"/>
>> <action application="bridge" data=sofia/internal/userB at local.pbx.com"/>
>> </condition>
>> </extension>
>>
>> Thanks
>> O
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From avi at avimarcus.net  Fri Jun 15 18:40:49 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 15 Jun 2012 17:40:49 +0300
Subject: [Freeswitch-users] Best place to set variables?
In-Reply-To: <4FDB471C.8080306@the800group.com>
References: <4FDB34D8.1040208@the800group.com>
	<CALNZuwnB0xMDASeKAG8qX20DYfpq-nbgr06PTO4HgaD6nr+Vqg@mail.gmail.com>
	<4FDB471C.8080306@the800group.com>
Message-ID: <CALNZuw=bzuUqy1azvSxMNyZkKzaUnHw-RyeDt-QRUVvTMZSfwQ@mail.gmail.com>

No. They do persist for the whole <condition> just they aren't
necessarily checked by each bridge line.
-Avi


On Fri, Jun 15, 2012 at 5:30 PM, ocset <ocset at the800group.com> wrote:
> Thanks for your answer.
>
> What you are saying is that the "continue_on_fail=true" and
> "hangup_after_bridge=true" variables are set back to their default
> values in the"B" leg of the dailplan. I thought the variable that are
> set would persist for the whole <condition> section.
>
> Regards
> O
>
> On 15/06/12 21:47, Avi Marcus wrote:
>> Short answer: some variables are for the A leg, and some are for the B leg.
>> The two in their own set tags are for the A leg's execution of the
>> bridge.. whether it should keep trying or not.
>> The one inside the bridge is specific to that leg and bridge attempt
>> of how it should behave.
>>
>> Perhaps someone wants to give the longer answer, but I don't have time
>> right now.
>>
>> -Avi
>>
>>
>> On Fri, Jun 15, 2012 at 4:12 PM, ocset<ocset at the800group.com> ?wrote:
>>> Hi
>>>
>>> In the FreeSWITCH-Cookbook, there is the following example
>>>
>>> <extension name="ring_sequentially">
>>> <condition field="destination_number" expression="^(2001)$">
>>> <action application="set" data="continue_on_fail=true"/>
>>> <action application="set" data="hangup_after_bridge=true"/>
>>> <action application="bridge"
>>> data={ignore_early_media=true}sofia/internal/userA at local.pbx.com"/>
>>> <action application="bridge"
>>> data={ignore_early_media=true}sofia/internal/userB at local.pbx.com"/>
>>> </condition>
>>> </extension>
>>>
>>> Is there any reason why the "ignore_early_media=true" was not set before
>>> the bridge commands as is done with the "continue_on_fail" ?and
>>> "hangup_after_bridge" variables?
>>>
>>> Example:
>>>
>>> <extension name="ring_sequentially">
>>> <condition field="destination_number" expression="^(2001)$">
>>> <action application="set" data="continue_on_fail=true"/>
>>> <action application="set" data="hangup_after_bridge=true"/>
>>> <action application="set" data="ignore_early_media=true"/>
>>> <action application="bridge" data=sofia/internal/userA at local.pbx.com"/>
>>> <action application="bridge" data=sofia/internal/userB at local.pbx.com"/>
>>> </condition>
>>> </extension>
>>>
>>> Thanks
>>> O
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 gmail.com  Fri Jun 15 18:48:52 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 15 Jun 2012 16:48:52 +0200
Subject: [Freeswitch-users] Blog post about SIP
In-Reply-To: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
References: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
Message-ID: <CALXCt0p3J69qzZu=x8JsmLia89d+Ee9mrYN3N8z9hGZ_NocytQ@mail.gmail.com>

On Thu, Jun 14, 2012 at 10:06 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> FreeSWITCHers,
>
>
> ?I wrote a little intro and linked to the document if any of you
> would like to check it out (link at the end of post):
>
> http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html
>

Hi Kristian,

you're a gifted writer!

Lot of fun actually.

More, more !

Looking forward to "Son of SIP", "SIP Strikes Back", "Revenge of the SIP", ...

-giovanni


> --
> Kristian Kielhofner
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 curriegrad2004 at gmail.com  Fri Jun 15 19:43:41 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 15 Jun 2012 08:43:41 -0700
Subject: [Freeswitch-users] AMR and AMR-WB support
In-Reply-To: <CAG8p5B1A-ewBY3RZy0pforfx2D0o6sp68O1PWTGqUEO3QDbTtg@mail.gmail.com>
References: <CAG8p5B1A-ewBY3RZy0pforfx2D0o6sp68O1PWTGqUEO3QDbTtg@mail.gmail.com>
Message-ID: <CAMWiMy9iMvUfojBr1G7cjAy4nvZXUi=VY6qRU1bwLzmBQe8=Nw@mail.gmail.com>

FreeSWITCH supports amr in passthrough mode only. Not sure about the rfc
4867 part.
On Jun 14, 2012 8:54 PM, "Jakson Kalsson" <sipmaillist at gmail.com> wrote:

> Hi all, I have a project for the 3G related, AMR and AMR-WB support.
>
> I'm using the client develop suite from the PortSIP(http://www.portsip.com),
> as their said
> support the AMR, AMR-WB with RFC4867.
>
> Now I have to setup a SIP server/SIP PBX in our Lab for test, does the
> Freeswitch
> support these codecs and RFC4867 ?
>
> Also, any other Server/PBX which support AMR, AMR-WB recommended are
> welcome.
>
>
> Best regards,
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From curriegrad2004 at gmail.com  Fri Jun 15 19:47:12 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 15 Jun 2012 08:47:12 -0700
Subject: [Freeswitch-users] No dial-tone on Siemens C610
In-Reply-To: <4FDB3B3E.6060300@the800group.com>
References: <4FDB3B3E.6060300@the800group.com>
Message-ID: <CAMWiMy-qqiH+WHcO8s3Q0oEbPcwVtJt_Xjyey_EouBovQtuohg@mail.gmail.com>

You could solve this problem by making the Siemens phone behave like an
automatic ring down phone by making it dial an extension that simulates a
dial tone.

Look in the default dialplan, under the features context for such example.
On Jun 15, 2012 6:41 AM, "ocset" <ocset at the800group.com> wrote:

> Hi
>
> I have deployed 3 cordless Siemens C610 phones on FS and was very
> surprised that there is no dial-tone when a user makes a call. The users
> hears a sound similar to an engage signal while FS makes the connection,
> followed by the sound of the phone ringing once the connection is
> established. (hope that makes sense).
>
> Siemens reckons that is the way their phones behave. Is there a way for
> FS to simulate a dial-tone while the user waits for the other parties
> phone to start ringing?
>
> Thanks
> O
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Peter.Stevens at bbc.co.uk  Fri Jun 15 20:19:57 2012
From: Peter.Stevens at bbc.co.uk (Peter Stevens)
Date: Fri, 15 Jun 2012 17:19:57 +0100
Subject: [Freeswitch-users] Problem passing and playing audio files into a
	conference
Message-ID: <9B08C5DC53DDB7468007DE57C9FAF49DC89CB7@bbcxues30.national.core.bbc.co.uk>

Hi all,

I hope that someone can help with solving the following problem.

I have tried using the example code at the bottom of
http://wiki.freeswitch.org/wiki/Mod_conference page, but
stripped back to not record (so as to limit the number of possible
problems) but just to play back selected wav files.

Q: How to create a dialplan to have callers speak their name before
joining the conference and announce that to other participants when they
join?
This is the code I'm using...having added a couple of delays

<extension name="Record Name and schedule conf announce">
   <condition field="destination_number" expression="^55(3\d\d\d)$">
   <action application="answer"/>
     <action application="playback"
data="ivr/ivr-call_being_transferred.wav"/>
     <action application="sleep" data="1000"/>
     <action application="log" data="INFO Caller Id Name is:
${caller_id_name}"/> <!-- display for reference only -->
<!-- disable original line -->
<!--      <action application="set" data="res=${sched_api +1 none
conference $1-${domain} play
file_string://${namefile}!${sound_prefix}/conference/8000/conf-has_joine
d.wav}"/> -->
     <action application="set" data="res=${sched_api (+2 none conference
$1-${domain} play
file_string://${sound_prefix}/conference/8000/conf-welcome.wav!${sound_p
refix}/conference/8000/conf-has_joined.wav)}"/>
     <action application="sleep" data="1000"/>
     <action application="transfer" data="$1 XML default"/>
	 </condition>
</extension>

Using this line, instead of the original:

     <action application="set" data="res=${sched_api (+2 none conference
$1-${domain} play
file_string://${sound_prefix}/conference/8000/conf-welcome.wav!${sound_p
refix}/conference/8000/conf-has_joined.wav)}"/>
with the () brackets as shown at least set the res ok, i.e. I didn't get
this error message:

EXECUTE sofia/nat/1101 at DOMAIN set(res=)
2012-06-14 09:56:47.493255 [DEBUG] mod_dptools.c:748
sofia/nat/1101 at DOMAIN SET [res]=[UNDEF]

(note that I have removed the domain name and replaced it with DOMAIN in
these messages)

but got this instead:

2012-06-15 16:24:55.713235 [DEBUG] switch_scheduler.c:214 Added task 6
sched_api_function (none) to run at 1339773897
EXECUTE sofia/nat/1209 at DOMAIN set(res=+OK Added: 6)
2012-06-15 16:24:55.713235 [DEBUG] mod_dptools.c:748
sofia/nat/1209 at DOMAIN SET [res]=[+OK Added: 6]

which appears to be some improvement!
As before the call got transferred to the conference, but didn't run the
additional audio clips
specified in the action application set up line shown above and then ran
the hold music.

Instead I got the following invalid file format error message from
file_string:

2012-06-15 16:24:57.481975 [ERR] switch_core_file.c:71 Invalid file
format [file_string] for
[/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-welcome.wav!/o
pt/freeswitch/sounds/en/us/callie/conference/8000/conf-has_joined.wav]!
2012-06-15 16:24:57.481975 [DEBUG] mod_commands.c:2391 Command
conference(3000-132.185.142.18 play
file_string:///opt/freeswitch/sounds/en/us/callie/conference/8000/conf-w
elcome.wav!/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-has_
joined.wav):
(play) File:
file_string:///opt/freeswitch/sounds/en/us/callie/conference/8000/conf-w
elcome.wav!/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-has_
joined.wav not found.

2012-06-15 16:24:57.481975 [DEBUG] switch_scheduler.c:138 Deleting task
6 sched_api_function (none)
2012-06-15 16:24:59.473199 [DEBUG] mod_local_stream.c:346 Opening Stream
[moh/8000] 8000hz

The two particular audio files conf-welcome.wav and conf-has_joined.wav
do exist at the specified location.

>From what I have seen on various posts in the lists, the file_stream
syntax seems to be correct, so what might the problem be
parsing and playing these files once into the conference?

Thanks for any help

Peter


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From jeremyc at ssimicro.com  Fri Jun 15 20:39:27 2012
From: jeremyc at ssimicro.com (Jeremy Childs)
Date: Fri, 15 Jun 2012 10:39:27 -0600
Subject: [Freeswitch-users] Painted into a corner: outbound-codec-prefs,
 late negotiation, and inherit codec
Message-ID: <4FDB653F.6030000@ssimicro.com>

A quick introduction to what I'm trying to do:

I have two SIP profiles, "internal" (port 5060) and "offpremise" (port 
5070). Both of these profiles are recording registrations to the same 
registration db, so they can be reached by user/nnn. When a device 
leaves the local LAN and heads out into the field, firewall port 
translation maps port 5060 to 5070, so FS "knows" the extension is now 
off-premise.

All profiles on the box are set to inbound-late-negotiation and 
inherit-codec, to reduce transcoding.

What I would like to do is set a very restrictive list of codecs on the 
"offpremise" profile, so that an exension registered to "offpremise" 
only sees a set of low-bandwidth codecs. To facilitate this, I have set 
outbound-codec-prefs on the profile.

After much testing, I have determined that "outbound-codec-prefs" in 
this configuration ends up ADDING codecs to the offer, and will not 
remove any codecs offered by the A-leg. Interestingly, 
inbound-codec-prefs seems to remove codec from the A-leg offer.

So, my question: is there any way of restricting the list of codecs 
offered on the B-leg using the profile configuration (and not from 
within the dialplan)? If there is any answer at all, I'd suspect it's 
one of the following:

  * a global setting that changes the behavior of outbound-codec-prefs
    to be restrictive, instead of additive (the source doesn't seem to
    suggest this)
  * a way to set absolute_codec_string (or other variables) from within
    the profile itself
  * a way to set a variable (again, likely absolute_code_string) to the
    registration instance so the codec list can be explicitly specified
  * any other far-out ideas for accomplishing this kind of seamless
    "roaming"

Any ideas or hints greatly appreciated.

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From sdevoy at bizfocused.com  Fri Jun 15 20:39:59 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 15 Jun 2012 12:39:59 -0400
Subject: [Freeswitch-users] A request from a user seeking a Unique Dial Plan
	Solution
Message-ID: <02cb01cd4b15$815462c0$83fd2840$@bizfocused.com>

HI All,

 

I have a user that wants all DID calls to go to the receptionist . BUT if
she is unavailable or ON THE PHONE then go to a sequential or parallel hunt.
The full request is basically ring ext 300, if unanswered in 5 seconds
CONTINUE ringing 300, but add in the other extensions.  At first call
intercept seemed like a great choice, but if the receptionist in on a call
her phone does not "ring" so nobody knows to pick it up.

 

I was thinking <bridge> to receptionist with timeout of 5, then <bridge) to
all including receptionist.  However, I am concerned that she may try to
answer BETWEEN to the <bridge>s and get a dial tone.

 

Does anyone have any clever ideas?

 

Thanks,

Sean

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From patrick at sunsus.net  Fri Jun 15 21:23:02 2012
From: patrick at sunsus.net (sunsus)
Date: Fri, 15 Jun 2012 10:23:02 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH and netvoip.ch 304 Unauthorized
In-Reply-To: <1339707254878-7579873.post@n2.nabble.com>
References: <1339334863709-7579670.post@n2.nabble.com>
	<1339524483645-7579753.post@n2.nabble.com>
	<CAKV-jFwWLygHJVozz41=5rsjXMVuTrDsek98Rty2DL4zF1AC1w@mail.gmail.com>
	<1339526433720-7579758.post@n2.nabble.com>
	<4FD7962E.9070401@cupis.co.uk>
	<1339531138888-7579764.post@n2.nabble.com>
	<4FD7B4AE.6030805@cupis.co.uk>
	<1339703154568-7579871.post@n2.nabble.com>
	<1339707254878-7579873.post@n2.nabble.com>
Message-ID: <1339780982533-7579899.post@n2.nabble.com>

Hello Jeff 

Thanks for the information, thats what I searched for. Now I know it's not a
bug, but a provider problem.

regards

Patrick 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-401-Unauthorized-tp7579670p7579899.html
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From freeswitch at zoho.com  Fri Jun 15 21:24:35 2012
From: freeswitch at zoho.com (dingdong)
Date: Fri, 15 Jun 2012 10:24:35 -0700
Subject: [Freeswitch-users] Blog post about SIP
In-Reply-To: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
References: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
Message-ID: <137f12cd230.5442606051204082350.2378772053723064798@zoho.com>

very nice and helpful,and yes,it would really be more useful if somehow related to how does FS do SIP.

Thanks

---- On Thu, 14 Jun 2012 13:06:52 -0700 Kristian Kielhofner &lt;kris at kriskinc.com&gt; wrote ---- 


FreeSWITCHers, 
 
 As some of you may have noticed I frequently comment on SIP-related 
FreeSWITCH posts. Long story short I decided to finally put some of 
these thoughts and experiences down in writing. What I ended up with 
is a 21 page (and counting) document describing various SIP header 
fields, SDP information, RTP issues, DTMF issues, NAT traversal 
technologies, interop headaches, etc. 
 
 I wrote a little intro and linked to the document if any of you 
would like to check it out (link at the end of post): 
 
http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html 
 
 Of course I'd appreciate any feedback you may have. What do you 
disagree with? Find a typo? Have burning SIP/RTP/SDP/T.38/codec 
questions you'd like to see me address? Should I get into more 
implementation (FreeSWITCH) specific issues? Let me know. 
 
Thanks! 
 
-- 
Kristian Kielhofner 
 
_________________________________________________________________________ 
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org 
http://www.freeswitchsolutions.com 
 
 
 
 
Official FreeSWITCH Sites 
http://www.freeswitch.org 
http://wiki.freeswitch.org 
http://www.cluecon.com 
 
Join Us At ClueCon - Aug 7-9, 2012 
 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.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.meaney at bsb.ie  Fri Jun 15 21:08:00 2012
From: brian.meaney at bsb.ie (Bobthebeat)
Date: Fri, 15 Jun 2012 10:08:00 -0700 (PDT)
Subject: [Freeswitch-users] Conference Caller Announce
Message-ID: <1339780080974-7579898.post@n2.nabble.com>

Hi Folks, 

 brand new Freeswitch user here. 

I have modified the example shown on the mod_conference
(http://wiki.freeswitch.org/wiki/Mod_conference#FAQ) wiki for my own
deployment. 

I am attempting to have a user record his name, then have this recording
announced on joining the conference, but I am struggling. Here is my
dialplan entry:- 


<extension name="main_number_11" continue="true">
      <condition field="destination_number" expression="^xxxxxxxxxxx$">
        <action application="export" data="hold_music=silence"/>
        <action application="set" data="namefile=/tmp/${uuid}-name.wav"
inline="true"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="playback" data="conference/conf-welcome.wav"/>
        <action application="sleep" data="1000"/>
        <action application="playback"
data="voicemail/vm-record_name1.wav"/>
        <action application="playback" data="tone_stream://%(1000,0,500)"/>
        <action application="record" data="${namefile} 1"/>
        <action application="playback"
data="ivr/ivr-call_being_transferred.wav"/>
        <action application="set" data="res=${sched_api +1 none conference
$1-${domain_name}@conference_2 play
file_string://${namefile}!conference/conf-has_joined.wav}"/>
        <action application="transfer" data="$1 XML default"/>
      </condition>
    </extension>

The call connects and I am prompted to record, then I am notified of the
imminent transfer, but it bombs out here. Sophia output reads 

Command conference($1-192.168.3.178 at conference_2 play
file_string:///tmp/b34c9679-9fdb-4934-a1d9-dd4b646c88e2-name.wav!conference/conf-has_joined.wav): 
Conference $1-192.168.3.178 at conference_2 not found 

I can dial into the conference without issues when I disregard the 'Set'
expression and just dial the conference directly. 

Can anybody point me in the right direction? 
Thank you very much, 
B 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Conference-Caller-Announce-tp7579898.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Fri Jun 15 21:42:53 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Jun 2012 10:42:53 -0700
Subject: [Freeswitch-users] Conference Caller Announce
In-Reply-To: <1339780080974-7579898.post@n2.nabble.com>
References: <1339780080974-7579898.post@n2.nabble.com>
Message-ID: <CAKzWOxURj0VpLk-BPyXLWuOn-Q8eFxBfYJLnm6NiQSMDeHeGvw@mail.gmail.com>

As best I can tell it looks like this line is the problem:
<condition field="destination_number" expression="^xxxxxxxxxxx$">

You need to wrap the capture value in parens like this:
<condition field="destination_number" expression="^(xxxxxxxxxxx)$">

That should make $1 have something useful in it.

-MC

On Fri, Jun 15, 2012 at 10:08 AM, Bobthebeat <brian.meaney at bsb.ie> wrote:

> Hi Folks,
>
>  brand new Freeswitch user here.
>
> I have modified the example shown on the mod_conference
> (http://wiki.freeswitch.org/wiki/Mod_conference#FAQ) wiki for my own
> deployment.
>
> I am attempting to have a user record his name, then have this recording
> announced on joining the conference, but I am struggling. Here is my
> dialplan entry:-
>
>
> <extension name="main_number_11" continue="true">
>      <condition field="destination_number" expression="^xxxxxxxxxxx$">
>        <action application="export" data="hold_music=silence"/>
>        <action application="set" data="namefile=/tmp/${uuid}-name.wav"
> inline="true"/>
>        <action application="answer"/>
>        <action application="sleep" data="1000"/>
>        <action application="playback" data="conference/conf-welcome.wav"/>
>        <action application="sleep" data="1000"/>
>        <action application="playback"
> data="voicemail/vm-record_name1.wav"/>
>        <action application="playback" data="tone_stream://%(1000,0,500)"/>
>        <action application="record" data="${namefile} 1"/>
>        <action application="playback"
> data="ivr/ivr-call_being_transferred.wav"/>
>        <action application="set" data="res=${sched_api +1 none conference
> $1-${domain_name}@conference_2 play
> file_string://${namefile}!conference/conf-has_joined.wav}"/>
>        <action application="transfer" data="$1 XML default"/>
>      </condition>
>    </extension>
>
> The call connects and I am prompted to record, then I am notified of the
> imminent transfer, but it bombs out here. Sophia output reads
>
> Command conference($1-192.168.3.178 at conference_2 play
>
> file_string:///tmp/b34c9679-9fdb-4934-a1d9-dd4b646c88e2-name.wav!conference/conf-has_joined.wav):
> Conference $1-192.168.3.178 at conference_2 not found
>
> I can dial into the conference without issues when I disregard the 'Set'
> expression and just dial the conference directly.
>
> Can anybody point me in the right direction?
> Thank you very much,
> B
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Conference-Caller-Announce-tp7579898.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From drk at drkngs.net  Fri Jun 15 21:48:06 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Fri, 15 Jun 2012 10:48:06 -0700
Subject: [Freeswitch-users]
	=?iso-8859-1?q?Best_place_to_set_variables=3F?=
In-Reply-To: <CALNZuw=bzuUqy1azvSxMNyZkKzaUnHw-RyeDt-QRUVvTMZSfwQ@mail.gmail.com>
Message-ID: <20120615174806.216e9027@mail.tritonwest.net>

Once a variable is set on a session (one leg) it presists till it's either "unset" or changed. Variables really don't have "default" values, there are just default ways that things work if the variables are not present. With that said you should only have to set variables like "hangup_after_bridge" and so on, only once on the A leg.  
   
The confusing part is what leg to set them on. The easiest way to figure that out, is to see how FS uses it. For example "hangup_after_bridge" is used by the Bridge application (actually in switch_ivr_bridge), so because it's an parameter that tells the bridge how to work, it needs to be set on the A leg. The bridge function connects two legs togetther, and the A leg is the one calling it.  
   
Other variables, the ones used for call origination, for example "origination_uuuid" are used to control how an outbound channel origiates the call. They could be used in the originate API call, with NO other leg preseent. If you are using them in a Bridge, then you must either set them on the B-Leg, or "export" them so that all new B-Legs created from the current, get those variables "cloned" to them.   
   
So in the case that you have more then one "Bridge" inside a condition, any variables you set with "set" will be there for each following bridge, however the ones you set in the channel of the bridge, the ones in the "[]" will not be there for the next ones. Becuase if the bridge fails, the B-Leg for that bridge is gone.  
   
--Dave
      _____  

  From: Avi Marcus [mailto:avi at avimarcus.net]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Fri, 15 Jun 2012 07:40:49 -0700
Subject: Re: [Freeswitch-users] Best place to set variables?

No. They do persist for the whole <condition> just they aren't
necessarily checked by each bridge line.
-Avi


On Fri, Jun 15, 2012 at 5:30 PM, ocset <ocset at the800group.com> wrote:
> Thanks for your answer.
>
> What you are saying is that the "continue_on_fail=true" and
> "hangup_after_bridge=true" variables are set back to their default
> values in the"B" leg of the dailplan. I thought the variable that are
> set would persist for the whole <condition> section.
>
> Regards
> O
>
> On 15/06/12 21:47, Avi Marcus wrote:
>> Short answer: some variables are for the A leg, and some are for the B leg.
>> The two in their own set tags are for the A leg's execution of the
>> bridge.. whether it should keep trying or not.
>> The one inside the bridge is specific to that leg and bridge attempt
>> of how it should behave.
>>
>> Perhaps someone wants to give the longer answer, but I don't have time
>> right now.
>>
>> -Avi
>>
>>
>> On Fri, Jun 15, 2012 at 4:12 PM, ocset<ocset at the800group.com>  wrote:
>>> Hi
>>>
>>> In the FreeSWITCH-Cookbook, there is the following example
>>>
>>> <extension name="ring_sequentially">
>>> <condition field="destination_number" expression="^(2001)$">
>>> <action application="set" data="continue_on_fail=true"/>
>>> <action application="set" data="hangup_after_bridge=true"/>
>>> <action application="bridge"
>>> data={ignore_early_media=true}sofia/internal/userA at local.pbx.com"/>
>>> <action application="bridge"
>>> data={ignore_early_media=true}sofia/internal/userB at local.pbx.com"/>
>>> </condition>
>>> </extension>
>>>
>>> Is there any reason why the "ignore_early_media=true" was not set before
>>> the bridge commands as is done with the "continue_on_fail"  and
>>> "hangup_after_bridge" variables?
>>>
>>> Example:
>>>
>>> <extension name="ring_sequentially">
>>> <condition field="destination_number" expression="^(2001)$">
>>> <action application="set" data="continue_on_fail=true"/>
>>> <action application="set" data="hangup_after_bridge=true"/>
>>> <action application="set" data="ignore_early_media=true"/>
>>> <action application="bridge" data=sofia/internal/userA at local.pbx.com"/>
>>> <action application="bridge" data=sofia/internal/userB at local.pbx.com"/>
>>> </condition>
>>> </extension>
>>>
>>> Thanks
>>> O
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 Jun 15 21:53:45 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Jun 2012 10:53:45 -0700
Subject: [Freeswitch-users] A request from a user seeking a Unique Dial
 Plan Solution
In-Reply-To: <02cb01cd4b15$815462c0$83fd2840$@bizfocused.com>
References: <02cb01cd4b15$815462c0$83fd2840$@bizfocused.com>
Message-ID: <CAKzWOxWeZkT5Gc-PZgZnTCkmOca6K5p4r2prbW=O5OsxmKbYsg@mail.gmail.com>

I think you probably need this:
http://wiki.freeswitch.org/wiki/Channel_Variables#leg_delay_start

Just do a single bridge, and use the leg_delay_start variable to "delay"
the leg(s) that need to start ringing 5 seconds later.

-MC

On Fri, Jun 15, 2012 at 9:39 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> HI All,****
>
> ** **
>
> I have a user that wants all DID calls to go to the receptionist ? BUT if
> she is unavailable or ON THE PHONE then go to a sequential or parallel
> hunt.  The full request is basically ring ext 300, if unanswered in 5
> seconds CONTINUE ringing 300, but add in the other extensions.  At first
> call intercept seemed like a great choice, but if the receptionist in on a
> call her phone does not ?ring? so nobody knows to pick it up.****
>
> ** **
>
> I was thinking <bridge> to receptionist with timeout of 5, then <bridge)
> to all including receptionist.  However, I am concerned that she may try to
> answer BETWEEN to the <bridge>s and get a dial tone.****
>
> ** **
>
> Does anyone have any clever ideas?****
>
> ** **
>
> Thanks,****
>
> Sean****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.meaney at BSB.IE  Fri Jun 15 22:21:47 2012
From: brian.meaney at BSB.IE (Brian Meaney)
Date: Fri, 15 Jun 2012 19:21:47 +0100
Subject: [Freeswitch-users] Conference Caller Announce
In-Reply-To: <CAKzWOxURj0VpLk-BPyXLWuOn-Q8eFxBfYJLnm6NiQSMDeHeGvw@mail.gmail.com>
References: <1339780080974-7579898.post@n2.nabble.com>
	<CAKzWOxURj0VpLk-BPyXLWuOn-Q8eFxBfYJLnm6NiQSMDeHeGvw@mail.gmail.com>
Message-ID: <6D03E4DF94D7644A97A32498EE786BD3076DB58879@bsbmserver2>

Hi and thanks for your response. I altered the expression to include parentheses, and the number is successfully picked up. I also attempted to simplify the 'res' expression by using $1-{domain}
However, it still wont connect to the conference. Sophia output is now:-

2012-06-15 18:12:53.303084 [DEBUG] mod_commands.c:3092 Command conference(35319017881-192.168.3.178 play file_string:///tmp/799d7748-e75b-425c-bccd-1e6632eacc72-name.wav!conference/conf-has_joined.wav):
Conference 35319017881-192.168.3.178 not found

Its clear my issue is now with how the 'res' statement should be formed. I now have:-

<action application="set" data="res=${sched_api +1 none conference $1 at conference_2 play file_string://${namefile}!conference/conf-has_joined.wav}"/>

How should I construct the statement such that it connects the caller to the conference named conference_2 ? Should I even be referencing the conference name, seeing as I already have the expression matching in place earlier in the definition?





From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 15 June 2012 18:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference Caller Announce

As best I can tell it looks like this line is the problem:
<condition field="destination_number" expression="^xxxxxxxxxxx$">

You need to wrap the capture value in parens like this:
<condition field="destination_number" expression="^(xxxxxxxxxxx)$">

That should make $1 have something useful in it.

-MC
On Fri, Jun 15, 2012 at 10:08 AM, Bobthebeat <brian.meaney at bsb.ie<mailto:brian.meaney at bsb.ie>> wrote:
Hi Folks,

 brand new Freeswitch user here.

I have modified the example shown on the mod_conference
(http://wiki.freeswitch.org/wiki/Mod_conference#FAQ) wiki for my own
deployment.

I am attempting to have a user record his name, then have this recording
announced on joining the conference, but I am struggling. Here is my
dialplan entry:-


<extension name="main_number_11" continue="true">
     <condition field="destination_number" expression="^xxxxxxxxxxx$">
       <action application="export" data="hold_music=silence"/>
       <action application="set" data="namefile=/tmp/${uuid}-name.wav"
inline="true"/>
       <action application="answer"/>
       <action application="sleep" data="1000"/>
       <action application="playback" data="conference/conf-welcome.wav"/>
       <action application="sleep" data="1000"/>
       <action application="playback"
data="voicemail/vm-record_name1.wav"/>
       <action application="playback" data="tone_stream://%(1000,0,500)"/>
       <action application="record" data="${namefile} 1"/>
       <action application="playback"
data="ivr/ivr-call_being_transferred.wav"/>
       <action application="set" data="res=${sched_api +1 none conference
$1-${domain_name}@conference_2 play
file_string://${namefile}!conference/conf-has_joined.wav}"/>
       <action application="transfer" data="$1 XML default"/>
     </condition>
   </extension>

The call connects and I am prompted to record, then I am notified of the
imminent transfer, but it bombs out here. Sophia output reads

Command conference($1-192.168.3.178 at conference_2 play
file_string:///tmp/b34c9679-9fdb-4934-a1d9-dd4b646c88e2-name.wav!conference/conf-has_joined.wav):
Conference $1-192.168.3.178 at conference_2 not found

I can dial into the conference without issues when I disregard the 'Set'
expression and just dial the conference directly.

Can anybody point me in the right direction?
Thank you very much,
B

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Conference-Caller-Announce-tp7579898.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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
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From sdevoy at bizfocused.com  Fri Jun 15 22:56:27 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 15 Jun 2012 14:56:27 -0400
Subject: [Freeswitch-users] A request from a user seeking a Unique Dial
	Plan Solution
In-Reply-To: <CAKzWOxWeZkT5Gc-PZgZnTCkmOca6K5p4r2prbW=O5OsxmKbYsg@mail.gmail.com>
References: <02cb01cd4b15$815462c0$83fd2840$@bizfocused.com>
	<CAKzWOxWeZkT5Gc-PZgZnTCkmOca6K5p4r2prbW=O5OsxmKbYsg@mail.gmail.com>
Message-ID: <03cd01cd4b28$91eebe10$b5cc3a30$@bizfocused.com>

Perfect thank you.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Friday, June 15, 2012 1:54 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] A request from a user seeking a Unique Dial
Plan Solution

 

I think you probably need this:
http://wiki.freeswitch.org/wiki/Channel_Variables#leg_delay_start

Just do a single bridge, and use the leg_delay_start variable to "delay" the
leg(s) that need to start ringing 5 seconds later.

-MC

On Fri, Jun 15, 2012 at 9:39 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

HI All,

 

I have a user that wants all DID calls to go to the receptionist . BUT if
she is unavailable or ON THE PHONE then go to a sequential or parallel hunt.
The full request is basically ring ext 300, if unanswered in 5 seconds
CONTINUE ringing 300, but add in the other extensions.  At first call
intercept seemed like a great choice, but if the receptionist in on a call
her phone does not "ring" so nobody knows to pick it up.

 

I was thinking <bridge> to receptionist with timeout of 5, then <bridge) to
all including receptionist.  However, I am concerned that she may try to
answer BETWEEN to the <bridge>s and get a dial tone.

 

Does anyone have any clever ideas?

 

Thanks,

Sean


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From yungwei at resolvity.com  Sat Jun 16 00:55:50 2012
From: yungwei at resolvity.com (Yungwei Chen)
Date: Fri, 15 Jun 2012 16:55:50 -0400
Subject: [Freeswitch-users] Having trouble calling a 4-agent queue (the 4th
 agent never gets called)
Message-ID: <33095823FD21DF429B481B5163264B799E8D0DDC93@VMBX102.ihostexchange.net>

Hi,

I am trying to test routing calls to a queue with 4 agents using mod_callcenter.

Here're the settings.
* the contact of each agent is set to {leg_timeout=22}sofia/gateway/p1/12341234 (a phone number that will never be answered)
* the max-wait-time of the queue is set to 88 seconds (22 * 4)
* the strategy of the queue is set to sequentially-by-agent-order
* each agent is assigned a unique position
* the status of each agent is Available

The problem is that mod_callcenter doesn't try to reach the 4th agent for some reason. 
If the max-wait-time of the queue is set to 98 seconds, the 4th agent will still be ignored and the first agent will be called twice (it starts over).
Any ideas why?

Thanks.


From msc at freeswitch.org  Sat Jun 16 01:59:32 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Jun 2012 14:59:32 -0700
Subject: [Freeswitch-users] Conference Caller Announce
In-Reply-To: <6D03E4DF94D7644A97A32498EE786BD3076DB58879@bsbmserver2>
References: <1339780080974-7579898.post@n2.nabble.com>
	<CAKzWOxURj0VpLk-BPyXLWuOn-Q8eFxBfYJLnm6NiQSMDeHeGvw@mail.gmail.com>
	<6D03E4DF94D7644A97A32498EE786BD3076DB58879@bsbmserver2>
Message-ID: <CAKzWOxXztcXaCMWx2-1+QYfstmEb7tTvnFEbV3H1b18CXgCQ+g@mail.gmail.com>

Show us the dialplan code where you put the caller into the conference.
-MC

On Fri, Jun 15, 2012 at 11:21 AM, Brian Meaney <brian.meaney at bsb.ie> wrote:

>  Hi and thanks for your response. I altered the expression to include
> parentheses, and the number is successfully picked up. I also attempted to
> simplify the ?res? expression by using $1-{domain}
> However, it still wont connect to the conference. Sophia output is now:-**
> **
>
> ** **
>
> 2012-06-15 18:12:53.303084 [DEBUG] mod_commands.c:3092 Command
> conference(35319017881-192.168.3.178 play
> file_string:///tmp/799d7748-e75b-425c-bccd-1e6632eacc72-name.wav!conference/conf-has_joined.wav):
> ****
>
> Conference 35319017881-192.168.3.178 not found****
>
> ** **
>
> Its clear my issue is now with how the ?res? statement should be formed. I
> now have:-****
>
> ** **
>
> <action application="set" data="res=${sched_api +1 none conference
> $1 at conference_2 play
> file_string://${namefile}!conference/conf-has_joined.wav}"/>****
>
> ** **
>
> How should I construct the statement such that it connects the caller to
> the conference named conference_2 ? Should I even be referencing the
> conference name, seeing as I already have the expression matching in place
> earlier in the definition?****
>
> ** **
>
>  ****
>
> ** **
>
>  ****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 15 June 2012 18:43
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Conference Caller Announce****
>
> ** **
>
> As best I can tell it looks like this line is the problem:
> <condition field="destination_number" expression="^xxxxxxxxxxx$">
>
> You need to wrap the capture value in parens like this:
> <condition field="destination_number" expression="^(xxxxxxxxxxx)$">
>
> That should make $1 have something useful in it.
>
> -MC****
>
> On Fri, Jun 15, 2012 at 10:08 AM, Bobthebeat <brian.meaney at bsb.ie> wrote:*
> ***
>
> Hi Folks,
>
>  brand new Freeswitch user here.
>
> I have modified the example shown on the mod_conference
> (http://wiki.freeswitch.org/wiki/Mod_conference#FAQ) wiki for my own
> deployment.
>
> I am attempting to have a user record his name, then have this recording
> announced on joining the conference, but I am struggling. Here is my
> dialplan entry:-
>
>
> <extension name="main_number_11" continue="true">
>      <condition field="destination_number" expression="^xxxxxxxxxxx$">
>        <action application="export" data="hold_music=silence"/>
>        <action application="set" data="namefile=/tmp/${uuid}-name.wav"
> inline="true"/>
>        <action application="answer"/>
>        <action application="sleep" data="1000"/>
>        <action application="playback" data="conference/conf-welcome.wav"/>
>        <action application="sleep" data="1000"/>
>        <action application="playback"
> data="voicemail/vm-record_name1.wav"/>
>        <action application="playback" data="tone_stream://%(1000,0,500)"/>
>        <action application="record" data="${namefile} 1"/>
>        <action application="playback"
> data="ivr/ivr-call_being_transferred.wav"/>
>        <action application="set" data="res=${sched_api +1 none conference
> $1-${domain_name}@conference_2 play
> file_string://${namefile}!conference/conf-has_joined.wav}"/>
>        <action application="transfer" data="$1 XML default"/>
>      </condition>
>    </extension>
>
> The call connects and I am prompted to record, then I am notified of the
> imminent transfer, but it bombs out here. Sophia output reads
>
> Command conference($1-192.168.3.178 at conference_2 play
>
> file_string:///tmp/b34c9679-9fdb-4934-a1d9-dd4b646c88e2-name.wav!conference/conf-has_joined.wav):
> Conference $1-192.168.3.178 at conference_2 not found
>
> I can dial into the conference without issues when I disregard the 'Set'
> expression and just dial the conference directly.
>
> Can anybody point me in the right direction?
> Thank you very much,
> B
>
>
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From mario_fs at mgtech.com  Sat Jun 16 02:26:31 2012
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 15 Jun 2012 15:26:31 -0700
Subject: [Freeswitch-users] A request from a user seeking a Unique Dial
	Plan Solution
In-Reply-To: <03cd01cd4b28$91eebe10$b5cc3a30$@bizfocused.com>
References: <02cb01cd4b15$815462c0$83fd2840$@bizfocused.com>
	<CAKzWOxWeZkT5Gc-PZgZnTCkmOca6K5p4r2prbW=O5OsxmKbYsg@mail.gmail.com>
	<03cd01cd4b28$91eebe10$b5cc3a30$@bizfocused.com>
Message-ID: <52841C4F-9F75-4768-956D-8511A752391D@mgtech.com>

Wish I had know about this in 2010. But... I also have a need to change the phones ringtone after the leg_delay. I bridge/ring extensions for 20 seconds, the run the export below, then bridge to all extensions again and add 2 cell phones. The changing ringtone let's us know the cells are now being called. I DO have the problem where if a call is picked up at the instant between bridges they have to hang up to get ringing to answer.

So the leg_delay works, but would love to know if there is a way to change a phones ring without performing a second bridge. Thanks.
Mario G

<action application="export" data="alert_info=n=Simple-5;w=2;c=5"/>
        

On Jun 15, 2012, at 11:56 AM, Sean Devoy wrote:

> Perfect thank you.
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
> Sent: Friday, June 15, 2012 1:54 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] A request from a user seeking a Unique Dial Plan Solution
>  
> I think you probably need this:
> http://wiki.freeswitch.org/wiki/Channel_Variables#leg_delay_start
> 
> Just do a single bridge, and use the leg_delay_start variable to "delay" the leg(s) that need to start ringing 5 seconds later.
> 
> -MC
> 
> On Fri, Jun 15, 2012 at 9:39 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> HI All,
>  
> I have a user that wants all DID calls to go to the receptionist ? BUT if she is unavailable or ON THE PHONE then go to a sequential or parallel hunt.  The full request is basically ring ext 300, if unanswered in 5 seconds CONTINUE ringing 300, but add in the other extensions.  At first call intercept seemed like a great choice, but if the receptionist in on a call her phone does not ?ring? so nobody knows to pick it up.
>  
> I was thinking <bridge> to receptionist with timeout of 5, then <bridge) to all including receptionist.  However, I am concerned that she may try to answer BETWEEN to the <bridge>s and get a dial tone.
>  
> Does anyone have any clever ideas?
>  
> Thanks,
> Sean
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing 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 esander83 at hushmail.com  Sat Jun 16 04:06:45 2012
From: esander83 at hushmail.com (Eric Sander)
Date: Fri, 15 Jun 2012 19:06:45 -0500
Subject: [Freeswitch-users]
	=?utf-8?q?Does_mod=5Ffifo_not_take_in=09effect?=
	=?utf-8?q?ive=5Fcaller=5Fid=5Fname=3F?=
Message-ID: <20120616000645.DCB4014DBD8@smtp.hushmail.com>

Any body know if effective_caller_id_name can be passed on to modfifo. Read the wiki on it and there's no info.  This would be a helpful feature to include as other parts of FS use this to display info.

thanks
eric_s

On Thursday, June 14, 2012 at 6:35 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk> wrote:
>
>mod_callcenter also ignores those variables unless you set one 
>called
>cc_export_vars. Perhaps there&#39;s a similar mechanism for 
>mod_fifo?
>
>On 14 June 2012 05:31,   wrote:
> Hi All,
>I have this basic dialplan: 
>I can see the call get logged via the first extension but it&#39;s 
>not
>passing the info to modfifo.  I can remove the 2nd extension and 
>put a
>basic: 
>and the caller id name info is displayed on the phones but not when
>routing through mod_fifo.  I have tried both set/export and it 
>still
>doesn&#39;t seem to work.
>Is this correct? 
>Thanks,eric_s
>
>___________________________________________________________________
>______
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> Join Us At ClueCon - Aug 7-9, 2012
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch
>-users
> http://www.freeswitch.org
>-- 
>Andrew Cassidy BSc (Hons) MBCS SSCAManaging Director
>
> T 03300 100 960  F 03300 100 961 E 
>andrew at cassidywebservices.co.uk W
>www.cassidywebservices.co.uk



From ocset at the800group.com  Sat Jun 16 10:56:30 2012
From: ocset at the800group.com (ocset)
Date: Sat, 16 Jun 2012 14:56:30 +0800
Subject: [Freeswitch-users] No dial-tone on Siemens C610
In-Reply-To: <CAMWiMy-qqiH+WHcO8s3Q0oEbPcwVtJt_Xjyey_EouBovQtuohg@mail.gmail.com>
References: <4FDB3B3E.6060300@the800group.com>
	<CAMWiMy-qqiH+WHcO8s3Q0oEbPcwVtJt_Xjyey_EouBovQtuohg@mail.gmail.com>
Message-ID: <4FDC2E1E.2020304@the800group.com>

Thanks for you reply.

When you say default dialplan, are you referring to this file - 
/usr/local/freeswitch/conf/dialplan/default.xml

I am not sure which examples you are referring to but I found this, 
which also seems to be a solution (just need to change the "bridge" line):

<extension name="ringback_183_uk_ring">
<condition field="destination_number" expression="^9181$">
<action application="set" data="ringback=$${uk-ring}"/>
<action application="bridge" data="{ignore_early_media=true}loopback/wait"/>
</condition>
</extension>

Could I be so bold as to ask you to reply back with the example you are 
referring to?

Thanks in advance
O.


On 15/06/12 23:47, curriegrad2004 wrote:
>
> You could solve this problem by making the Siemens phone behave like 
> an automatic ring down phone by making it dial an extension that 
> simulates a dial tone.
>
> Look in the default dialplan, under the features context for such example.
>
> On Jun 15, 2012 6:41 AM, "ocset" <ocset at the800group.com 
> <mailto:ocset at the800group.com>> wrote:
>
>     Hi
>
>     I have deployed 3 cordless Siemens C610 phones on FS and was very
>     surprised that there is no dial-tone when a user makes a call. The
>     users
>     hears a sound similar to an engage signal while FS makes the
>     connection,
>     followed by the sound of the phone ringing once the connection is
>     established. (hope that makes sense).
>
>     Siemens reckons that is the way their phones behave. Is there a
>     way for
>     FS to simulate a dial-tone while the user waits for the other parties
>     phone to start ringing?
>
>     Thanks
>     O
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From saami_mh at ymail.com  Sat Jun 16 13:00:24 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Sat, 16 Jun 2012 02:00:24 -0700 (PDT)
Subject: [Freeswitch-users] what is the best tools for sip stress test on
	freeswitch?
Message-ID: <1339837224.94411.YahooMailNeo@web120101.mail.ne1.yahoo.com>

hi,
what is the best tools for sip stress test on freeswitch?
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From nbhatti at gmail.com  Sat Jun 16 13:02:39 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Sat, 16 Jun 2012 12:02:39 +0300
Subject: [Freeswitch-users] what is the best tools for sip stress test
	on freeswitch?
In-Reply-To: <1339837224.94411.YahooMailNeo@web120101.mail.ne1.yahoo.com>
References: <1339837224.94411.YahooMailNeo@web120101.mail.ne1.yahoo.com>
Message-ID: <CACak_yyeD+Xo6T1jUnU4uCE8jFjP48wwUcrYELi4O1o86KqFZA@mail.gmail.com>

http://sipp.sourceforge.net/

On Sat, Jun 16, 2012 at 12:00 PM, Samira Mh <saami_mh at ymail.com> wrote:
> hi,
> what is the best tools for sip stress test on freeswitch?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From alec.taylor6 at gmail.com  Sat Jun 16 13:03:23 2012
From: alec.taylor6 at gmail.com (Alec Taylor)
Date: Sat, 16 Jun 2012 19:03:23 +1000
Subject: [Freeswitch-users] what is the best tools for sip stress test
	on freeswitch?
In-Reply-To: <1339837224.94411.YahooMailNeo@web120101.mail.ne1.yahoo.com>
References: <1339837224.94411.YahooMailNeo@web120101.mail.ne1.yahoo.com>
Message-ID: <CAO+9iGfnvsfwEs_xeaa26_i_Hn-jK_GnJ4bw7hbWZ8h1Yx4c7Q@mail.gmail.com>

See: http://wiki.freeswitch.org/wiki/Load_testing

And: http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations

On Sat, Jun 16, 2012 at 7:00 PM, Samira Mh <saami_mh at ymail.com> wrote:
> hi,
> what is the best tools for sip stress test on freeswitch?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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.meaney at BSB.IE  Sat Jun 16 14:21:46 2012
From: brian.meaney at BSB.IE (Brian Meaney)
Date: Sat, 16 Jun 2012 11:21:46 +0100
Subject: [Freeswitch-users] Conference Caller Announce
In-Reply-To: <CAKzWOxXztcXaCMWx2-1+QYfstmEb7tTvnFEbV3H1b18CXgCQ+g@mail.gmail.com>
References: <1339780080974-7579898.post@n2.nabble.com>
	<CAKzWOxURj0VpLk-BPyXLWuOn-Q8eFxBfYJLnm6NiQSMDeHeGvw@mail.gmail.com>
	<6D03E4DF94D7644A97A32498EE786BD3076DB58879@bsbmserver2>
	<CAKzWOxXztcXaCMWx2-1+QYfstmEb7tTvnFEbV3H1b18CXgCQ+g@mail.gmail.com>
Message-ID: <6D03E4DF94D7644A97A32498EE786BD3076DB58923@bsbmserver2>

Thanks MC,

 Here it is:-

<extension name="main_number_11" continue="true">
      <condition field="destination_number" expression="^(xxxxxxxxxxx)$">
        <action application="export" data="hold_music=silence"/>
        <action application="set" data="namefile=/tmp/${uuid}-name.wav" inline="true"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="playback" data="conference/conf-welcome.wav"/>
        <action application="sleep" data="1000"/>
        <action application="playback" data="voicemail/vm-record_name1.wav"/>
        <action application="playback" data="tone_stream://%(1000,0,500)"/>
        <action application="record" data="${namefile} 1"/>
        <action application="playback" data="ivr/ivr-call_being_transferred.wav"/>
        <action application="set" data="res=${sched_api +1 none conference $1 at conference_2 play file_string://${namefile}!conference/conf-has_joined.wav}"/>
        <action application="transfer" data="$1 XML default"/>
      </condition>
    </extension>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 15 June 2012 23:00
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference Caller Announce

Show us the dialplan code where you put the caller into the conference.
-MC
On Fri, Jun 15, 2012 at 11:21 AM, Brian Meaney <brian.meaney at bsb.ie<mailto:brian.meaney at bsb.ie>> wrote:
Hi and thanks for your response. I altered the expression to include parentheses, and the number is successfully picked up. I also attempted to simplify the 'res' expression by using $1-{domain}
However, it still wont connect to the conference. Sophia output is now:-

2012-06-15 18:12:53.303084 [DEBUG] mod_commands.c:3092 Command conference(35319017881-192.168.3.178 play file_string:///tmp/799d7748-e75b-425c-bccd-1e6632eacc72-name.wav!conference/conf-has_joined.wav):
Conference 35319017881-192.168.3.178 not found

Its clear my issue is now with how the 'res' statement should be formed. I now have:-

<action application="set" data="res=${sched_api +1 none conference $1 at conference_2 play file_string://${namefile}!conference/conf-has_joined.wav}"/>

How should I construct the statement such that it connects the caller to the conference named conference_2 ? Should I even be referencing the conference name, seeing as I already have the expression matching in place earlier in the definition?





From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: 15 June 2012 18:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference Caller Announce

As best I can tell it looks like this line is the problem:
<condition field="destination_number" expression="^xxxxxxxxxxx$">

You need to wrap the capture value in parens like this:
<condition field="destination_number" expression="^(xxxxxxxxxxx)$">

That should make $1 have something useful in it.

-MC
On Fri, Jun 15, 2012 at 10:08 AM, Bobthebeat <brian.meaney at bsb.ie<mailto:brian.meaney at bsb.ie>> wrote:
Hi Folks,

 brand new Freeswitch user here.

I have modified the example shown on the mod_conference
(http://wiki.freeswitch.org/wiki/Mod_conference#FAQ) wiki for my own
deployment.

I am attempting to have a user record his name, then have this recording
announced on joining the conference, but I am struggling. Here is my
dialplan entry:-


<extension name="main_number_11" continue="true">
     <condition field="destination_number" expression="^xxxxxxxxxxx$">
       <action application="export" data="hold_music=silence"/>
       <action application="set" data="namefile=/tmp/${uuid}-name.wav"
inline="true"/>
       <action application="answer"/>
       <action application="sleep" data="1000"/>
       <action application="playback" data="conference/conf-welcome.wav"/>
       <action application="sleep" data="1000"/>
       <action application="playback"
data="voicemail/vm-record_name1.wav"/>
       <action application="playback" data="tone_stream://%(1000,0,500)"/>
       <action application="record" data="${namefile} 1"/>
       <action application="playback"
data="ivr/ivr-call_being_transferred.wav"/>
       <action application="set" data="res=${sched_api +1 none conference
$1-${domain_name}@conference_2 play
file_string://${namefile}!conference/conf-has_joined.wav}"/>
       <action application="transfer" data="$1 XML default"/>
     </condition>
   </extension>

The call connects and I am prompted to record, then I am notified of the
imminent transfer, but it bombs out here. Sophia output reads

Command conference($1-192.168.3.178 at conference_2 play
file_string:///tmp/b34c9679-9fdb-4934-a1d9-dd4b646c88e2-name.wav!conference/conf-has_joined.wav):
Conference $1-192.168.3.178 at conference_2 not found

I can dial into the conference without issues when I disregard the 'Set'
expression and just dial the conference directly.

Can anybody point me in the right direction?
Thank you very much,
B


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From awais-nazeer at hotmail.com  Sat Jun 16 22:07:14 2012
From: awais-nazeer at hotmail.com (awais nazir)
Date: Sat, 16 Jun 2012 23:07:14 +0500
Subject: [Freeswitch-users] freeswitch to write cdr after every bridge
 attempt (failed or successful
Message-ID: <BLU162-W175906B51EDEE234B9DD7D88FA0@phx.gbl>


Hello 

Thanks enabling leg B is certainly giving expected results but it's now showing up few variables including some user defined variables defined in the dialplan being executed. This problem did not exist in default enabled leg A.



On Fri, Jun 15, 2012 at 6:51 PM, Avi Marcus <avi at avimarcus.net> wrote:
By default, the modules for logging only log the A leg of the call.

You can turn on b-leg logging in whichever module you are using for

call logs. Do note you'll have to reconstruct the call then, based on

which is the A-leg and which is B leg, and using bridge_uuid,

signal_bond or last_bridge_to of the B leg (whichever is set) to see

which A leg it came from.



(Also note on enterprise originates: they are reported as leg A in

xml_cdr and need to be tied using the variable:

ent_originate_aleg_uuid)



This kind of "advanced" CDR handling should probably be wikified...



-Avi





On Fri, Jun 15, 2012 at 4:17 PM, awais nazir <awais-nazeer at hotmail.com> wrote:

> Hi

>

> I am trying to use following context to get failover gateways working

>

>     <action application="set" data="hangup_after_bridge=true"/>

>     <action application="bridge"

> data="sofia/local_profile/1111 at example1.company.com" />

>     <action application="bridge"

> data="sofia/local_profile/1111 at example2.company.com" />

>

> if call on first gateway fails , call connection is attempted on second

> gateway but FS is writing only one CDR.How can it write CDR on every bridge

> attempt.

>

> BR

> --waisee

 		 	   		  
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From vvenkatar at gmail.com  Sun Jun 17 05:04:02 2012
From: vvenkatar at gmail.com (Venkatesh)
Date: Sat, 16 Jun 2012 18:04:02 -0700
Subject: [Freeswitch-users] Problem with accessing HTTPs URL's via CURL with
	FreeSwitch.
Message-ID: <CAP_69mt+O=ow1nyLmBXUJ3VvbzbJ6H+qZ8_H=MgKZYSELz-weg@mail.gmail.com>

Hi !

I have a javascript application that needs to access a web service via
HTTPs. I tried using session.execute("curl", restURL) to access the same.
It looks like freeswitch/mod_curl is not even making an attempt to connect
to my web service when I use a HTTPs URL (verified by running tcpdump on
the host when running my application). However the application works if I
substitute HTTPs with plain HTTP. I wanted to know if this is a
configuration issue of some sort or if this is a bug? Would appreciate any
help.

Venkatesh
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From curriegrad2004 at gmail.com  Sun Jun 17 05:12:06 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 16 Jun 2012 18:12:06 -0700
Subject: [Freeswitch-users] No dial-tone on Siemens C610
In-Reply-To: <4FDC2E1E.2020304@the800group.com>
References: <4FDB3B3E.6060300@the800group.com>
	<CAMWiMy-qqiH+WHcO8s3Q0oEbPcwVtJt_Xjyey_EouBovQtuohg@mail.gmail.com>
	<4FDC2E1E.2020304@the800group.com>
Message-ID: <CAMWiMy-YFVxM964Qn4M9Kb7SdbxWHBX=4OknaxdrqSsUnodePA@mail.gmail.com>

Close, but the file I was referring to was conf/dialplan/features.xml

This is what you may want to have a look at:

    <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>

Replace the execute_extension action with transfer.

On Fri, Jun 15, 2012 at 11:56 PM, ocset <ocset at the800group.com> wrote:
> Thanks for you reply.
>
> When you say default dialplan, are you referring to this file -
> /usr/local/freeswitch/conf/dialplan/default.xml
>
> I am not sure which examples you are referring to but I found this, which
> also seems to be a solution (just need to change the "bridge" line):
>
> <extension name="ringback_183_uk_ring">
> ????? <condition field="destination_number" expression="^9181$">
> ??????? <action application="set" data="ringback=$${uk-ring}"/>
> ??????? <action application="bridge"
> data="{ignore_early_media=true}loopback/wait"/>
> ???? </condition>
> </extension>
>
> Could I be so bold as to ask you to reply back with the example you are
> referring to?
>
> Thanks in advance
> O.
>
>
>
> On 15/06/12 23:47, curriegrad2004 wrote:
>
> You could solve this problem by making the Siemens phone behave like an
> automatic ring down phone by making it dial an extension that simulates a
> dial tone.
>
> Look in the default dialplan, under the features context for such example.
>
> On Jun 15, 2012 6:41 AM, "ocset" <ocset at the800group.com> wrote:
>>
>> Hi
>>
>> I have deployed 3 cordless Siemens C610 phones on FS and was very
>> surprised that there is no dial-tone when a user makes a call. The users
>> hears a sound similar to an engage signal while FS makes the connection,
>> followed by the sound of the phone ringing once the connection is
>> established. (hope that makes sense).
>>
>> Siemens reckons that is the way their phones behave. Is there a way for
>> FS to simulate a dial-tone while the user waits for the other parties
>> phone to start ringing?
>>
>> Thanks
>> O
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From avi at avimarcus.net  Sun Jun 17 13:43:14 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 17 Jun 2012 12:43:14 +0300
Subject: [Freeswitch-users] Blog post about SIP
In-Reply-To: <137f12cd230.5442606051204082350.2378772053723064798@zoho.com>
References: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
	<137f12cd230.5442606051204082350.2378772053723064798@zoho.com>
Message-ID: <CALNZuwkZBvc=f0=ud8ZPXD56422vEXgP5LsMdqH_+pd0Rfb6Rg@mail.gmail.com>

Great doc!

Some short notes:

*This is broken. Cisco is the most well known vendor to make this mistake.
> These implementations should be fixed to use G729 as the codec name with
> payload type 18.*

Linksys / Sipura SPAs do this, too, I'm pretty sure even before they were
bough by cisco.

*Typically this NAT functionality is referred to as a SIP NAT helper, or
> SIP ALG. Of course there are other names.*

Many routers let you turn off NAT ALG, if you can find it in the menu.

*Latching has several issues:... scalability*

At what point does this seriously become an issue? 100 concurrent calls?
1000? If you're proxying media anyway the IP rewriting sounds pretty
minimal.

More info about NAT, one way media, and how sipsorcery is a SIP proxy that
doesn't ever proxy media but can still set up calls with 2 way media would
be much appreciated.

One other things I'm interested in is the ability to not handle media, for
latency & performance reasons:
What are the pitfalls with FreeSWITCH specifically that makes it not happy
with not handling media?
and.. if I still need FS to handle DTMF what are my options? Do many/most
carriers (on origination) support SIP INFO or something like that? (I run a
calling card with an option to hang up the call)


Thanks for the helpful info!
-Avi


On Fri, Jun 15, 2012 at 8:24 PM, dingdong <freeswitch at zoho.com> wrote:

> **
> very nice and helpful,and yes,it would really be more useful if somehow
> related to how does FS do SIP.
>
> Thanks
>
> ---- On Thu, 14 Jun 2012 13:06:52 -0700 *Kristian Kielhofner <
> kris at kriskinc.com>* wrote ----
>
> FreeSWITCHers,
>
> As some of you may have noticed I frequently comment on SIP-related
> FreeSWITCH posts. Long story short I decided to finally put some of
> these thoughts and experiences down in writing. What I ended up with
> is a 21 page (and counting) document describing various SIP header
> fields, SDP information, RTP issues, DTMF issues, NAT traversal
> technologies, interop headaches, etc.
>
> I wrote a little intro and linked to the document if any of you
> would like to check it out (link at the end of post):
>
> http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html
>
> Of course I'd appreciate any feedback you may have. What do you
> disagree with? Find a typo? Have burning SIP/RTP/SDP/T.38/codec
> questions you'd like to see me address? Should I get into more
> implementation (FreeSWITCH) specific issues? Let me know.
>
> Thanks!
>
> --
> Kristian Kielhofner
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveu at coppice.org  Sun Jun 17 14:44:02 2012
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 17 Jun 2012 18:44:02 +0800
Subject: [Freeswitch-users] Blog post about SIP
In-Reply-To: <CALNZuwkZBvc=f0=ud8ZPXD56422vEXgP5LsMdqH_+pd0Rfb6Rg@mail.gmail.com>
References: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
	<137f12cd230.5442606051204082350.2378772053723064798@zoho.com>
	<CALNZuwkZBvc=f0=ud8ZPXD56422vEXgP5LsMdqH_+pd0Rfb6Rg@mail.gmail.com>
Message-ID: <4FDDB4F2.5030707@coppice.org>

On 06/17/2012 05:43 PM, Avi Marcus wrote:
> Great doc!
>
> Some short notes:
>
>     *This is broken. Cisco is the most well known vendor to make this
>     mistake. These implementations should be fixed to use G729 as the
>     codec name with payload type 18.*
>
> Linksys / Sipura SPAs do this, too, I'm pretty sure even before they 
> were bough by cisco.
I thought the only Cisco products which did this were the ones which 
started as Linksys/Sipura products. The usual G.729 problem with real 
Cisco products is G.729 with the bits packed in the opposite order to 
the RFC. This isn't an error. Cisco implemented G.729 in RTP before 
there was a standard, and the packing order they chose just happened to 
be the opposite of the standard. Of course, modern Cisco kit implements 
the standard order, but they can still do the opposite order for 
backwards compatibility with their own kit. Sometimes you still meet 
systems configured to use that, and G.729 calls produce a horrible noise.
>
>     *Typically this NAT functionality is referred to as a SIP NAT
>     helper, or SIP ALG. Of course there are other names.*
>
> Many routers let you turn off NAT ALG, if you can find it in the menu.
>
>     *Latching has several issues:... scalability*
>
> At what point does this seriously become an issue? 100 concurrent 
> calls? 1000? If you're proxying media anyway the IP rewriting sounds 
> pretty minimal.
>
> More info about NAT, one way media, and how sipsorcery is a SIP proxy 
> that doesn't ever proxy media but can still set up calls with 2 way 
> media would be much appreciated.
>
> One other things I'm interested in is the ability to not handle media, 
> for latency & performance reasons:
> What are the pitfalls with FreeSWITCH specifically that makes it not 
> happy with not handling media?
> and.. if I still need FS to handle DTMF what are my options? Do 
> many/most carriers (on origination) support SIP INFO or something like 
> that? (I run a calling card with an option to hang up the call)
>

Steve



From vvenkatar at gmail.com  Sun Jun 17 19:25:56 2012
From: vvenkatar at gmail.com (Venkatesh)
Date: Sun, 17 Jun 2012 08:25:56 -0700
Subject: [Freeswitch-users] Support for NetEQ in FS.
Message-ID: <CAP_69mu95VLFtvV0nmDbR71xQ90=R=RzpOB773OADeG-x8a3KQ@mail.gmail.com>

Hi !

I wanted to know if there are plans to integrate NetEQ in FreeSwitch. I
read a few postings where there were some proposals indicating on these
lines, but wanted to see if anybody is working on the same?

Regards,
Venkatesh
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From Peter.Stevens at bbc.co.uk  Sun Jun 17 20:11:08 2012
From: Peter.Stevens at bbc.co.uk (Peter Stevens)
Date: Sun, 17 Jun 2012 17:11:08 +0100
Subject: [Freeswitch-users] Conference Caller Announce
References: <1339780080974-7579898.post@n2.nabble.com><CAKzWOxURj0VpLk-BPyXLWuOn-Q8eFxBfYJLnm6NiQSMDeHeGvw@mail.gmail.com><6D03E4DF94D7644A97A32498EE786BD3076DB58879@bsbmserver2><CAKzWOxXztcXaCMWx2-1+QYfstmEb7tTvnFEbV3H1b18CXgCQ+g@mail.gmail.com>
	<6D03E4DF94D7644A97A32498EE786BD3076DB58923@bsbmserver2>
Message-ID: <9B08C5DC53DDB7468007DE57C9FAF49D8ED7E5@bbcxues30.national.core.bbc.co.uk>

Hi Brian,
 
I've also been looking at this particular example this week and also posted a question! See [Freeswitch-users] Problem passing and playing audio files into a conference.
 
Mine will transfer into the default conference (using conference $1-${domain} in the res line below, but not play the files parsed into the res line.
 
My guess with yours is that you are specifying conference$1 at conference_2 in the res line. I think that it expects to find a conference_2 profile in the conference.conf.xml file, so if you don't have that it might well bomb out.
The other thing that you may also find is that with your existing res line in the debug log you may also be getting [res]=[UNDEF]

The only way I could get it not to produce this was to add a left brace after sched_api and a right brace after the last audio file to be parsed in the res line, as here:
 
<action application="set" data="res=${sched_api (+2 none conference $1-${domain} play file_string://${sound_prefix}/conference/8000/conf-welcome.wav!${sound_prefix}/conference/8000/conf-has_joined.wav)}"/>

I hope that this helps you and maybe you and/or someone else might be able to help me with the audio files being played once the conference starts.
 
Peter
 

________________________________

From: Brian Meaney [mailto:brian.meaney at BSB.IE]
Sent: Sat 16/06/2012 11:21
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference Caller Announce



Thanks MC,

 

 Here it is:-

 

<extension name="main_number_11" continue="true">

      <condition field="destination_number" expression="^(xxxxxxxxxxx)$">

        <action application="export" data="hold_music=silence"/>

        <action application="set" data="namefile=/tmp/${uuid}-name.wav" inline="true"/>

        <action application="answer"/>

        <action application="sleep" data="1000"/>

        <action application="playback" data="conference/conf-welcome.wav"/>

        <action application="sleep" data="1000"/>

        <action application="playback" data="voicemail/vm-record_name1.wav"/>

        <action application="playback" data="tone_stream://%(1000,0,500)"/>

        <action application="record" data="${namefile} 1"/>

        <action application="playback" data="ivr/ivr-call_being_transferred.wav"/>

        <action application="set" data="res=${sched_api +1 none conference $1 at conference_2 play file_string://${namefile}!conference/conf-has_joined.wav}"/>

        <action application="transfer" data="$1 XML default"/>

      </condition>

    </extension>

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 15 June 2012 23:00
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference Caller Announce

 

Show us the dialplan code where you put the caller into the conference.
-MC

On Fri, Jun 15, 2012 at 11:21 AM, Brian Meaney <brian.meaney at bsb.ie> wrote:

Hi and thanks for your response. I altered the expression to include parentheses, and the number is successfully picked up. I also attempted to simplify the 'res' expression by using $1-{domain}
However, it still wont connect to the conference. Sophia output is now:-

 

2012-06-15 18:12:53.303084 [DEBUG] mod_commands.c:3092 Command conference(35319017881-192.168.3.178 play file_string:///tmp/799d7748-e75b-425c-bccd-1e6632eacc72-name.wav!conference/conf-has_joined.wav):

Conference 35319017881-192.168.3.178 not found

 

Its clear my issue is now with how the 'res' statement should be formed. I now have:-

 

<action application="set" data="res=${sched_api +1 none conference $1 at conference_2 play file_string://${namefile}!conference/conf-has_joined.wav}"/>

 

How should I construct the statement such that it connects the caller to the conference named conference_2 ? Should I even be referencing the conference name, seeing as I already have the expression matching in place earlier in the definition?

 

 

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 15 June 2012 18:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference Caller Announce

 

As best I can tell it looks like this line is the problem:
<condition field="destination_number" expression="^xxxxxxxxxxx$">

You need to wrap the capture value in parens like this:
<condition field="destination_number" expression="^(xxxxxxxxxxx)$">

That should make $1 have something useful in it.

-MC

On Fri, Jun 15, 2012 at 10:08 AM, Bobthebeat <brian.meaney at bsb.ie> wrote:

Hi Folks,

 brand new Freeswitch user here.

I have modified the example shown on the mod_conference
(http://wiki.freeswitch.org/wiki/Mod_conference#FAQ) wiki for my own
deployment.

I am attempting to have a user record his name, then have this recording
announced on joining the conference, but I am struggling. Here is my
dialplan entry:-


<extension name="main_number_11" continue="true">
     <condition field="destination_number" expression="^xxxxxxxxxxx$">
       <action application="export" data="hold_music=silence"/>
       <action application="set" data="namefile=/tmp/${uuid}-name.wav"
inline="true"/>
       <action application="answer"/>
       <action application="sleep" data="1000"/>
       <action application="playback" data="conference/conf-welcome.wav"/>
       <action application="sleep" data="1000"/>
       <action application="playback"
data="voicemail/vm-record_name1.wav"/>
       <action application="playback" data="tone_stream://%(1000,0,500)"/>
       <action application="record" data="${namefile} 1"/>
       <action application="playback"
data="ivr/ivr-call_being_transferred.wav"/>
       <action application="set" data="res=${sched_api +1 none conference
$1-${domain_name}@conference_2 play
file_string://${namefile}!conference/conf-has_joined.wav}"/>
       <action application="transfer" data="$1 XML default"/>
     </condition>
   </extension>

The call connects and I am prompted to record, then I am notified of the
imminent transfer, but it bombs out here. Sophia output reads

Command conference($1-192.168.3.178 at conference_2 play
file_string:///tmp/b34c9679-9fdb-4934-a1d9-dd4b646c88e2-name.wav!conference/conf-has_joined.wav):
Conference $1-192.168.3.178 at conference_2 not found

I can dial into the conference without issues when I disregard the 'Set'
expression and just dial the conference directly.

Can anybody point me in the right direction?
Thank you very much,
B

P Please consider the environment before printing this email 

http://www.bbc.co.uk/
This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated.
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From awais-nazeer at hotmail.com  Sun Jun 17 20:27:02 2012
From: awais-nazeer at hotmail.com (awais nazir)
Date: Sun, 17 Jun 2012 21:27:02 +0500
Subject: [Freeswitch-users] freeswitch to write cdr after every bridge
 attempt (failed or successful
In-Reply-To: <BLU162-W175906B51EDEE234B9DD7D88FA0@phx.gbl>
References: <BLU162-W175906B51EDEE234B9DD7D88FA0@phx.gbl>
Message-ID: <BLU162-W18B59846DBB58245F9C1AF88F90@phx.gbl>



I will attempt get reply again, the channel variable being set in dialplan are seen in A leg cdr but not in B leg cdr, can somebody help ?


From: awais-nazeer at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] freeswitch to write cdr after every bridge attempt (failed or successful
Date: Sat, 16 Jun 2012 23:07:14 +0500





Hello 

Thanks enabling leg B is certainly giving expected results but it's now showing up few variables including some user defined variables defined in the dialplan being executed. This problem did not exist in default enabled leg A.



On Fri, Jun 15, 2012 at 6:51 PM, Avi Marcus <avi at avimarcus.net> wrote:
By default, the modules for logging only log the A leg of the call.

You can turn on b-leg logging in whichever module you are using for

call logs. Do note you'll have to reconstruct the call then, based on

which is the A-leg and which is B leg, and using bridge_uuid,

signal_bond or last_bridge_to of the B leg (whichever is set) to see

which A leg it came from.



(Also note on enterprise originates: they are reported as leg A in

xml_cdr and need to be tied using the variable:

ent_originate_aleg_uuid)



This kind of "advanced" CDR handling should probably be wikified...



-Avi





On Fri, Jun 15, 2012 at 4:17 PM, awais nazir <awais-nazeer at hotmail.com> wrote:

> Hi

>

> I am trying to use following context to get failover gateways working

>

>     <action application="set" data="hangup_after_bridge=true"/>

>     <action application="bridge"

> data="sofia/local_profile/1111 at example1.company.com" />

>     <action application="bridge"

> data="sofia/local_profile/1111 at example2.company.com" />

>

> if call on first gateway fails , call connection is attempted on second

> gateway but FS is writing only one CDR.How can it write CDR on every bridge

> attempt.

>

> BR

> --waisee

 		 	   		   		 	   		  
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From brian.meaney at BSB.IE  Sun Jun 17 20:32:12 2012
From: brian.meaney at BSB.IE (Brian Meaney)
Date: Sun, 17 Jun 2012 17:32:12 +0100
Subject: [Freeswitch-users] Conference Caller Announce
In-Reply-To: <9B08C5DC53DDB7468007DE57C9FAF49D8ED7E5@bbcxues30.national.core.bbc.co.uk>
References: <1339780080974-7579898.post@n2.nabble.com>
	<CAKzWOxURj0VpLk-BPyXLWuOn-Q8eFxBfYJLnm6NiQSMDeHeGvw@mail.gmail.com>
	<6D03E4DF94D7644A97A32498EE786BD3076DB58879@bsbmserver2>
	<CAKzWOxXztcXaCMWx2-1+QYfstmEb7tTvnFEbV3H1b18CXgCQ+g@mail.gmail.com>
	<6D03E4DF94D7644A97A32498EE786BD3076DB58923@bsbmserver2>
	<9B08C5DC53DDB7468007DE57C9FAF49D8ED7E5@bbcxues30.national.core.bbc.co.uk>
Message-ID: <811CC363-2B82-42D9-9AEA-E2BD77BB61AC@BSB.IE>

Hi Peter and thank you for your input. Glad to hear somebody else is searching for answers to this. Hopefully we can help one another with resolving the issue!

I have already defined the conference profile within the conference XML file, so I assume that's all that's required to put the caller into the conference.

If I discover anything I'll be sure to update this thread.

On 17 Jun 2012, at 17:16, "Peter Stevens" <Peter.Stevens at bbc.co.uk<mailto:Peter.Stevens at bbc.co.uk>> wrote:

Hi Brian,

I've also been looking at this particular example this week and also posted a question! See [Freeswitch-users] Problem passing and playing audio files into a conference.

Mine will transfer into the default conference (using conference $1-${domain} in the res line below, but not play the files parsed into the res line.

My guess with yours is that you are specifying conference$1 at conference_2<mailto:conference$1 at conference_2> in the res line. I think that it expects to find a conference_2 profile in the conference.conf.xml file, so if you don't have that it might well bomb out.
The other thing that you may also find is that with your existing res line in the debug log you may also be getting [res]=[UNDEF]
The only way I could get it not to produce this was to add a left brace after sched_api and a right brace after the last audio file to be parsed in the res line, as here:

<action application="set" data="res=${sched_api (+2 none conference $1-${domain} play file_string://${sound_prefix}/conference/8000/conf-welcome.wav!${sound_prefix}/conference/8000/conf-has_joined.wav)}"/>
I hope that this helps you and maybe you and/or someone else might be able to help me with the audio files being played once the conference starts.

Peter


________________________________
From: Brian Meaney [mailto:brian.meaney at BSB.IE]
Sent: Sat 16/06/2012 11:21
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference Caller Announce

Thanks MC,

 Here it is:-

<extension name="main_number_11" continue="true">
      <condition field="destination_number" expression="^(xxxxxxxxxxx)$">
        <action application="export" data="hold_music=silence"/>
        <action application="set" data="namefile=/tmp/${uuid}-name.wav" inline="true"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="playback" data="conference/conf-welcome.wav"/>
        <action application="sleep" data="1000"/>
        <action application="playback" data="voicemail/vm-record_name1.wav"/>
        <action application="playback" data="tone_stream://%(1000,0,500)"/>
        <action application="record" data="${namefile} 1"/>
        <action application="playback" data="ivr/ivr-call_being_transferred.wav"/>
        <action application="set" data="res=${sched_api +1 none conference $1 at conference_2 play file_string://${namefile}!conference/conf-has_joined.wav}"/>
        <action application="transfer" data="$1 XML default"/>
      </condition>
    </extension>

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 15 June 2012 23:00
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference Caller Announce

Show us the dialplan code where you put the caller into the conference.
-MC
On Fri, Jun 15, 2012 at 11:21 AM, Brian Meaney <brian.meaney at bsb.ie<mailto:brian.meaney at bsb.ie>> wrote:
Hi and thanks for your response. I altered the expression to include parentheses, and the number is successfully picked up. I also attempted to simplify the ?res? expression by using $1-{domain}
However, it still wont connect to the conference. Sophia output is now:-

2012-06-15 18:12:53.303084 [DEBUG] mod_commands.c:3092 Command conference(35319017881-192.168.3.178 play file_string:///tmp/799d7748-e75b-425c-bccd-1e6632eacc72-name.wav!conference/conf-has_joined.wav):
Conference 35319017881-192.168.3.178 not found

Its clear my issue is now with how the ?res? statement should be formed. I now have:-

<action application="set" data="res=${sched_api +1 none conference $1 at conference_2 play file_string://${namefile}!conference/conf-has_joined.wav}"/>

How should I construct the statement such that it connects the caller to the conference named conference_2 ? Should I even be referencing the conference name, seeing as I already have the expression matching in place earlier in the definition?





From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: 15 June 2012 18:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference Caller Announce

As best I can tell it looks like this line is the problem:
<condition field="destination_number" expression="^xxxxxxxxxxx$">

You need to wrap the capture value in parens like this:
<condition field="destination_number" expression="^(xxxxxxxxxxx)$">

That should make $1 have something useful in it.

-MC
On Fri, Jun 15, 2012 at 10:08 AM, Bobthebeat <brian.meaney at bsb.ie<mailto:brian.meaney at bsb.ie>> wrote:
Hi Folks,

 brand new Freeswitch user here.

I have modified the example shown on the mod_conference
(http://wiki.freeswitch.org/wiki/Mod_conference#FAQ) wiki for my own
deployment.

I am attempting to have a user record his name, then have this recording
announced on joining the conference, but I am struggling. Here is my
dialplan entry:-


<extension name="main_number_11" continue="true">
     <condition field="destination_number" expression="^xxxxxxxxxxx$">
       <action application="export" data="hold_music=silence"/>
       <action application="set" data="namefile=/tmp/${uuid}-name.wav"
inline="true"/>
       <action application="answer"/>
       <action application="sleep" data="1000"/>
       <action application="playback" data="conference/conf-welcome.wav"/>
       <action application="sleep" data="1000"/>
       <action application="playback"
data="voicemail/vm-record_name1.wav"/>
       <action application="playback" data="tone_stream://%(1000,0,500)"/>
       <action application="record" data="${namefile} 1"/>
       <action application="playback"
data="ivr/ivr-call_being_transferred.wav"/>
       <action application="set" data="res=${sched_api +1 none conference
$1-${domain_name}@conference_2 play
file_string://${namefile}!conference/conf-has_joined.wav}"/>
       <action application="transfer" data="$1 XML default"/>
     </condition>
   </extension>

The call connects and I am prompted to record, then I am notified of the
imminent transfer, but it bombs out here. Sophia output reads

Command conference($1-192.168.3.178 at conference_2 play
file_string:///tmp/b34c9679-9fdb-4934-a1d9-dd4b646c88e2-name.wav!conference/conf-has_joined.wav):
Conference $1-192.168.3.178 at conference_2 not found

I can dial into the conference without issues when I disregard the 'Set'
expression and just dial the conference directly.

Can anybody point me in the right direction?
Thank you very much,
B
P Please consider the environment before printing this email


http://www.bbc.co.uk
This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated.
If you have received it in error, please delete it from your system.
Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately.
Please note that the BBC monitors e-mails sent or received.
Further communication will signify your consent to this.
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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
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From avi at avimarcus.net  Sun Jun 17 21:15:19 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 17 Jun 2012 20:15:19 +0300
Subject: [Freeswitch-users] freeswitch to write cdr after every bridge
 attempt (failed or successful
In-Reply-To: <BLU162-W18B59846DBB58245F9C1AF88F90@phx.gbl>
References: <BLU162-W175906B51EDEE234B9DD7D88FA0@phx.gbl>
	<BLU162-W18B59846DBB58245F9C1AF88F90@phx.gbl>
Message-ID: <CALNZuwm=T5YhhyLzcBPQoUfrO3-sp8L+FpL1+4Vb44y8hDJXVg@mail.gmail.com>

Oh that's what you meant.
If it's in leg A, why do you expect it to be in Leg B CDR?

That said, there's an option to let you do that:
http://wiki.freeswitch.org/wiki/Variable_copy_xml_cdr but I've never tried
it.

-Avi


On Sun, Jun 17, 2012 at 7:27 PM, awais nazir <awais-nazeer at hotmail.com>wrote:

>
> I will attempt get reply again, the channel variable being set in dialplan
> are seen in A leg cdr but *not *in B leg cdr, can somebody help ?
>
>
> ------------------------------
> From: awais-nazeer at hotmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] freeswitch to write cdr after every bridge
> attempt (failed or successful
> Date: Sat, 16 Jun 2012 23:07:14 +0500
>
>
>  Hello
>
> Thanks enabling leg B is certainly giving expected results but it's now
> showing up few variables including some user defined variables defined in
> the dialplan being executed. This problem did not exist in default enabled
> leg A.
>
>
>
> On Fri, Jun 15, 2012 at 6:51 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
> By default, the modules for logging only log the A leg of the call.
> You can turn on b-leg logging in whichever module you are using for
> call logs. Do note you'll have to reconstruct the call then, based on
> which is the A-leg and which is B leg, and using bridge_uuid,
> signal_bond or last_bridge_to of the B leg (whichever is set) to see
> which A leg it came from.
>
> (Also note on enterprise originates: they are reported as leg A in
> xml_cdr and need to be tied using the variable:
> ent_originate_aleg_uuid)
>
> This kind of "advanced" CDR handling should probably be wikified...
>
> -Avi
>
>
> On Fri, Jun 15, 2012 at 4:17 PM, awais nazir <awais-nazeer at hotmail.com>
> wrote:
> > Hi
> >
> > I am trying to use following context to get failover gateways working
> >
> >     <action application="set" data="hangup_after_bridge=**true"/>
> >     <action application="bridge"
> > data="sofia/local_profile/1111**@example1.company.com<1111 at example1.company.com>"
> />
> >     <action application="bridge"
> > data="sofia/local_profile/1111**@example2.company.com<1111 at example2.company.com>"
> />
> >
> > if call on first gateway fails , call connection is attempted on second
> > gateway but FS is writing only one CDR.How can it write CDR on every
> bridge
> > attempt.
> >
> > BR
> > --waisee
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From awais-nazeer at hotmail.com  Sun Jun 17 21:25:35 2012
From: awais-nazeer at hotmail.com (awais nazir)
Date: Sun, 17 Jun 2012 22:25:35 +0500
Subject: [Freeswitch-users] (no subject)
Message-ID: <BLU162-W3540B9F1478EF8A3A9A35788F90@phx.gbl>


Actually , some user defined variables like huting attempt ,customer and vendor names I define in dialplan. The solution to make join with uuid is bit complicated and seem to require bit efforts like pushing in to mysql and then make join. Even then the variables only appears according to successful or last bridge attempt. 

On Sun, Jun 17, 2012 at 10:15 PM, Avi Marcus <avi at avimarcus.net> wrote:
Oh that's what you meant.If it's in leg A, why do you expect it to be in Leg B CDR?
That said, there's an option to let you do that: http://wiki.freeswitch.org/wiki/Variable_copy_xml_cdr but I've never tried it.

-Avi


On Sun, Jun 17, 2012 at 7:27 PM, awais nazir <awais-nazeer at hotmail.com> wrote:







I will attempt get reply again, the channel variable being set in dialplan are seen in A leg cdr but not in B leg cdr, can somebody help ?




From: awais-nazeer at hotmail.com
To: freeswitch-users at lists.freeswitch.org


Subject: Re: [Freeswitch-users] freeswitch to write cdr after every bridge attempt (failed or successful
Date: Sat, 16 Jun 2012 23:07:14 +0500





Hello 

Thanks enabling leg B is certainly giving expected results
 but it's now showing up few variables including some user defined 
variables defined in the dialplan being executed. This problem did not 
exist in default enabled leg A.





On Fri, Jun 15, 2012 at 6:51 PM, Avi Marcus <avi at avimarcus.net> wrote:


By default, the modules for logging only log the A leg of the call.

You can turn on b-leg logging in whichever module you are using for

call logs. Do note you'll have to reconstruct the call then, based on

which is the A-leg and which is B leg, and using bridge_uuid,

signal_bond or last_bridge_to of the B leg (whichever is set) to see

which A leg it came from.



(Also note on enterprise originates: they are reported as leg A in

xml_cdr and need to be tied using the variable:

ent_originate_aleg_uuid)



This kind of "advanced" CDR handling should probably be wikified...



-Avi





On Fri, Jun 15, 2012 at 4:17 PM, awais nazir <awais-nazeer at hotmail.com> wrote:

> Hi

>

> I am trying to use following context to get failover gateways working

>

>     <action application="set" data="hangup_after_bridge=true"/>

>     <action application="bridge"

> data="sofia/local_profile/1111 at example1.company.com" />

>     <action application="bridge"

> data="sofia/local_profile/1111 at example2.company.com" />

>

> if call on first gateway fails , call connection is attempted on second

> gateway but FS is writing only one CDR.How can it write CDR on every bridge

> attempt.

>

> BR

> --waisee

 		 	   		   		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com



Join Us At ClueCon - Aug 7-9, 2012



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com



Join Us At ClueCon - Aug 7-9, 2012



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

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From awais-nazeer at hotmail.com  Sun Jun 17 21:29:20 2012
From: awais-nazeer at hotmail.com (awais nazir)
Date: Sun, 17 Jun 2012 22:29:20 +0500
Subject: [Freeswitch-users] freeswitch to write cdr after every bridge
 attempt (failed or successful
Message-ID: <BLU162-W54A969DEF079760BD2CD3A88F90@phx.gbl>


Actually , some user defined variables like huting attempt ,customer and
 vendor names I define in dialplan. The solution to make join with uuid 
is bit complicated and seem to require bit efforts like pushing in to 
mysql and then make join. Even then the variables only appears according
 to successful or last bridge attempt. 

On Sun, Jun 17, 2012 at 10:15 PM, Avi Marcus <avi at avimarcus.net> wrote:
Oh that's what you meant.If it's in leg A, why do you expect it to be in Leg B CDR?
That said, there's an option to let you do that: http://wiki.freeswitch.org/wiki/Variable_copy_xml_cdr but I've never tried it.

-Avi


On Sun, Jun 17, 2012 at 7:27 PM, awais nazir <awais-nazeer at hotmail.com> wrote:







I will attempt get reply again, the channel variable being set in dialplan are seen in A leg cdr but not in B leg cdr, can somebody help ?




From: awais-nazeer at hotmail.com
To: freeswitch-users at lists.freeswitch.org


Subject: Re: [Freeswitch-users] freeswitch to write cdr after every bridge attempt (failed or successful
Date: Sat, 16 Jun 2012 23:07:14 +0500





Hello 

Thanks enabling leg B is certainly giving expected results
 but it's now showing up few variables including some user defined 
variables defined in the dialplan being executed. This problem did not 
exist in default enabled leg A.





On Fri, Jun 15, 2012 at 6:51 PM, Avi Marcus <avi at avimarcus.net> wrote:


By default, the modules for logging only log the A leg of the call.

You can turn on b-leg logging in whichever module you are using for

call logs. Do note you'll have to reconstruct the call then, based on

which is the A-leg and which is B leg, and using bridge_uuid,

signal_bond or last_bridge_to of the B leg (whichever is set) to see

which A leg it came from.



(Also note on enterprise originates: they are reported as leg A in

xml_cdr and need to be tied using the variable:

ent_originate_aleg_uuid)



This kind of "advanced" CDR handling should probably be wikified...



-Avi





On Fri, Jun 15, 2012 at 4:17 PM, awais nazir <awais-nazeer at hotmail.com> wrote:

> Hi

>

> I am trying to use following context to get failover gateways working

>

>     <action application="set" data="hangup_after_bridge=true"/>

>     <action application="bridge"

> data="sofia/local_profile/1111 at example1.company.com" />

>     <action application="bridge"

> data="sofia/local_profile/1111 at example2.company.com" />

>

> if call on first gateway fails , call connection is attempted on second

> gateway but FS is writing only one CDR.How can it write CDR on every bridge

> attempt.

>

> BR

> --waisee

 		 	   		   		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com

 		 	   		  
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From awais-nazeer at hotmail.com  Sun Jun 17 21:46:02 2012
From: awais-nazeer at hotmail.com (awais nazir)
Date: Sun, 17 Jun 2012 22:46:02 +0500
Subject: [Freeswitch-users] freeswitch to write cdr after every bridge
 attempt (failed or successful
In-Reply-To: <BLU162-W54A969DEF079760BD2CD3A88F90@phx.gbl>
References: <BLU162-W54A969DEF079760BD2CD3A88F90@phx.gbl>
Message-ID: <BLU162-W3041ABFA5CA4C1CCAA26C588F90@phx.gbl>




Thanks Avi, this link really helped, I am using export instead of set for variables.

Is there anyway we can put A and B legs logs separate?


From: awais-nazeer at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] freeswitch to write cdr after every bridge attempt (failed or successful
Date: Sun, 17 Jun 2012 22:29:20 +0500





Actually , some user defined variables like huting attempt ,customer and
 vendor names I define in dialplan. The solution to make join with uuid 
is bit complicated and seem to require bit efforts like pushing in to 
mysql and then make join. Even then the variables only appears according
 to successful or last bridge attempt. 

On Sun, Jun 17, 2012 at 10:15 PM, Avi Marcus <avi at avimarcus.net> wrote:
Oh that's what you meant.If it's in leg A, why do you expect it to be in Leg B CDR?
That said, there's an option to let you do that: http://wiki.freeswitch.org/wiki/Variable_copy_xml_cdr but I've never tried it.

-Avi


On Sun, Jun 17, 2012 at 7:27 PM, awais nazir <awais-nazeer at hotmail.com> wrote:







I will attempt get reply again, the channel variable being set in dialplan are seen in A leg cdr but not in B leg cdr, can somebody help ?




From: awais-nazeer at hotmail.com
To: freeswitch-users at lists.freeswitch.org


Subject: Re: [Freeswitch-users] freeswitch to write cdr after every bridge attempt (failed or successful
Date: Sat, 16 Jun 2012 23:07:14 +0500





Hello 

Thanks enabling leg B is certainly giving expected results
 but it's now showing up few variables including some user defined 
variables defined in the dialplan being executed. This problem did not 
exist in default enabled leg A.





On Fri, Jun 15, 2012 at 6:51 PM, Avi Marcus <avi at avimarcus.net> wrote:


By default, the modules for logging only log the A leg of the call.

You can turn on b-leg logging in whichever module you are using for

call logs. Do note you'll have to reconstruct the call then, based on

which is the A-leg and which is B leg, and using bridge_uuid,

signal_bond or last_bridge_to of the B leg (whichever is set) to see

which A leg it came from.



(Also note on enterprise originates: they are reported as leg A in

xml_cdr and need to be tied using the variable:

ent_originate_aleg_uuid)



This kind of "advanced" CDR handling should probably be wikified...



-Avi





On Fri, Jun 15, 2012 at 4:17 PM, awais nazir <awais-nazeer at hotmail.com> wrote:

> Hi

>

> I am trying to use following context to get failover gateways working

>

>     <action application="set" data="hangup_after_bridge=true"/>

>     <action application="bridge"

> data="sofia/local_profile/1111 at example1.company.com" />

>     <action application="bridge"

> data="sofia/local_profile/1111 at example2.company.com" />

>

> if call on first gateway fails , call connection is attempted on second

> gateway but FS is writing only one CDR.How can it write CDR on every bridge

> attempt.

>

> BR

> --waisee

 		 	   		   		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com

 		 	   		   		 	   		  
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From Peter.Stevens at bbc.co.uk  Mon Jun 18 17:38:39 2012
From: Peter.Stevens at bbc.co.uk (Peter Stevens)
Date: Mon, 18 Jun 2012 14:38:39 +0100
Subject: [Freeswitch-users] Conference Caller Announce
References: <1339780080974-7579898.post@n2.nabble.com><CAKzWOxURj0VpLk-BPyXLWuOn-Q8eFxBfYJLnm6NiQSMDeHeGvw@mail.gmail.com><6D03E4DF94D7644A97A32498EE786BD3076DB58879@bsbmserver2><CAKzWOxXztcXaCMWx2-1+QYfstmEb7tTvnFEbV3H1b18CXgCQ+g@mail.gmail.com><6D03E4DF94D7644A97A32498EE786BD3076DB58923@bsbmserver2><9B08C5DC53DDB7468007DE57C9FAF49D8ED7E5@bbcxues30.national.core.bbc.co.uk>
	<811CC363-2B82-42D9-9AEA-E2BD77BB61AC@BSB.IE>
Message-ID: <9B08C5DC53DDB7468007DE57C9FAF49D8ED7E9@bbcxues30.national.core.bbc.co.uk>

Hi Brian,
 
I was able to replicate your conference not found problem (purely by accident) and after a bit of investigation I think that I have probably found an answer to your problem.

In my case the last 4 digits of the extension number were 3000 (<condition field="destination_number" expression="^55(3\d\d\d)$">)
The last 4 digits become the value of $1 in this part of the res line ($1-${domain}) and also in the transfer line. So when the transfer occurs, the dialplan gets searched again but this time for a conference extension of 3000, which just happens to be one of the default conference numbers in the default.xml file (first 2 lines only):
<extension name="nb_conferences">
 <condition field="destination_number" expression="^(30\d{2})$">
 
and so it finds this extension (which I have in my dialplan) when the dialplan gets searched again.
 
I checked that this was the case, by changing the last 4 digits of my extension number to 3100 - which is the number for the wb_conferences in the same default.xml (which I have in my dialplan) and sure enough, it switches to that.
One more test confirmed this.
I don't have a conference number of 32xx in my dialplan. But by dialling 553200, the number gets picked up, it says the conference is being transferred, but is not found, so it bombs out. I think that if there is no
extension number that matches $1 in your dial plan, then it will fail to transfer
 
This might be worth trying: ensure that you have a conference number that matches your value of $1 in the destination number for your extension, but see if it works for you.

Maybe pop this INFO line into your dialplan, just to check the value of $1:

     <action application="log" data="INFO the dollar 1 value is: $1"/>
 
Not sure if there is a mechanism to reset the value of $1 (destination_number) after the extension has been picked up, but this time to point to the conference extension number required if different from the original destination number?
 
Let us know if this works and also if your audio gets played. My audio is still not playing as yet....
 
Peter

________________________________

From: Brian Meaney [mailto:brian.meaney at BSB.IE]
Sent: Sun 17/06/2012 17:32
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference Caller Announce


Hi Peter and thank you for your input. Glad to hear somebody else is searching for answers to this. Hopefully we can help one another with resolving the issue!

I have already defined the conference profile within the conference XML file, so I assume that's all that's required to put the caller into the conference.

If I discover anything I'll be sure to update this thread.

On 17 Jun 2012, at 17:16, "Peter Stevens" <Peter.Stevens at bbc.co.uk> wrote:



	Hi Brian,
	 
	I've also been looking at this particular example this week and also posted a question! See [Freeswitch-users] Problem passing and playing audio files into a conference.
	 
	Mine will transfer into the default conference (using conference $1-${domain} in the res line below, but not play the files parsed into the res line.
	 
	My guess with yours is that you are specifying conference$1 at conference_2 in the res line. I think that it expects to find a conference_2 profile in the conference.conf.xml file, so if you don't have that it might well bomb out.
	
	The other thing that you may also find is that with your existing res line in the debug log you may also be getting [res]=[UNDEF]
	
	The only way I could get it not to produce this was to add a left brace after sched_api and a right brace after the last audio file to be parsed in the res line, as here:
	 
	<action application="set" data="res=${sched_api (+2 none conference $1-${domain} play file_string://${sound_prefix}/conference/8000/conf-welcome.wav!${sound_prefix}/conference/8000/conf-has_joined.wav)}"/>
	
	I hope that this helps you and maybe you and/or someone else might be able to help me with the audio files being played once the conference starts.
	 
	Peter
	 

________________________________

	From: Brian Meaney [mailto:brian.meaney at BSB.IE]
	Sent: Sat 16/06/2012 11:21
	To: FreeSWITCH Users Help
	Subject: Re: [Freeswitch-users] Conference Caller Announce
	
	

	Thanks MC,

	 

	 Here it is:-

	 

	<extension name="main_number_11" continue="true">

	      <condition field="destination_number" expression="^(xxxxxxxxxxx)$">

	        <action application="export" data="hold_music=silence"/>

	        <action application="set" data="namefile=/tmp/${uuid}-name.wav" inline="true"/>

	        <action application="answer"/>

	        <action application="sleep" data="1000"/>

	        <action application="playback" data="conference/conf-welcome.wav"/>

	        <action application="sleep" data="1000"/>

	        <action application="playback" data="voicemail/vm-record_name1.wav"/>

	        <action application="playback" data="tone_stream://%(1000,0,500)"/>

	        <action application="record" data="${namefile} 1"/>

	        <action application="playback" data="ivr/ivr-call_being_transferred.wav"/>

	        <action application="set" data="res=${sched_api +1 none conference $1 at conference_2 play file_string://${namefile}!conference/conf-has_joined.wav}"/>

	        <action application="transfer" data="$1 XML default"/>

	      </condition>

	    </extension>

	 

	From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
	Sent: 15 June 2012 23:00
	To: FreeSWITCH Users Help
	Subject: Re: [Freeswitch-users] Conference Caller Announce

	 

	Show us the dialplan code where you put the caller into the conference.
	-MC

	On Fri, Jun 15, 2012 at 11:21 AM, Brian Meaney <brian.meaney at bsb.ie> wrote:

	Hi and thanks for your response. I altered the expression to include parentheses, and the number is successfully picked up. I also attempted to simplify the 'res' expression by using $1-{domain}
	However, it still wont connect to the conference. Sophia output is now:-

	 

	2012-06-15 18:12:53.303084 [DEBUG] mod_commands.c:3092 Command conference(35319017881-192.168.3.178 play file_string:///tmp/799d7748-e75b-425c-bccd-1e6632eacc72-name.wav!conference/conf-has_joined.wav):

	Conference 35319017881-192.168.3.178 not found

	 

	Its clear my issue is now with how the 'res' statement should be formed. I now have:-

	 

	<action application="set" data="res=${sched_api +1 none conference $1 at conference_2 play file_string://${namefile}!conference/conf-has_joined.wav}"/>

	 

	How should I construct the statement such that it connects the caller to the conference named conference_2 ? Should I even be referencing the conference name, seeing as I already have the expression matching in place earlier in the definition?

	 

	 

	 

	 

	 

	From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
	Sent: 15 June 2012 18:43
	To: FreeSWITCH Users Help
	Subject: Re: [Freeswitch-users] Conference Caller Announce

	 

	As best I can tell it looks like this line is the problem:
	<condition field="destination_number" expression="^xxxxxxxxxxx$">
	
	You need to wrap the capture value in parens like this:
	<condition field="destination_number" expression="^(xxxxxxxxxxx)$">
	
	That should make $1 have something useful in it.
	
	-MC

	On Fri, Jun 15, 2012 at 10:08 AM, Bobthebeat <brian.meaney at bsb.ie> wrote:

	Hi Folks,
	
	 brand new Freeswitch user here.
	
	I have modified the example shown on the mod_conference
	(http://wiki.freeswitch.org/wiki/Mod_conference#FAQ) wiki for my own
	deployment.
	
	I am attempting to have a user record his name, then have this recording
	announced on joining the conference, but I am struggling. Here is my
	dialplan entry:-
	
	
	<extension name="main_number_11" continue="true">
	     <condition field="destination_number" expression="^xxxxxxxxxxx$">
	       <action application="export" data="hold_music=silence"/>
	       <action application="set" data="namefile=/tmp/${uuid}-name.wav"
	inline="true"/>
	       <action application="answer"/>
	       <action application="sleep" data="1000"/>
	       <action application="playback" data="conference/conf-welcome.wav"/>
	       <action application="sleep" data="1000"/>
	       <action application="playback"
	data="voicemail/vm-record_name1.wav"/>
	       <action application="playback" data="tone_stream://%(1000,0,500)"/>
	       <action application="record" data="${namefile} 1"/>
	       <action application="playback"
	data="ivr/ivr-call_being_transferred.wav"/>
	       <action application="set" data="res=${sched_api +1 none conference
	$1-${domain_name}@conference_2 play
	file_string://${namefile}!conference/conf-has_joined.wav}"/>
	       <action application="transfer" data="$1 XML default"/>
	     </condition>
	   </extension>
	
	The call connects and I am prompted to record, then I am notified of the
	imminent transfer, but it bombs out here. Sophia output reads
	
	Command conference($1-192.168.3.178 at conference_2 play
	file_string:///tmp/b34c9679-9fdb-4934-a1d9-dd4b646c88e2-name.wav!conference/conf-has_joined.wav):
	Conference $1-192.168.3.178 at conference_2 not found
	
	I can dial into the conference without issues when I disregard the 'Set'
	expression and just dial the conference directly.
	
	Can anybody point me in the right direction?
	Thank you very much,
	B

	P Please consider the environment before printing this email 
	 

	http://www.bbc.co.uk <http://www.bbc.co.uk/> 
	This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated.
	If you have received it in error, please delete it from your system.
	Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately.
	Please note that the BBC monitors e-mails sent or received.
	Further communication will signify your consent to this. 

	_________________________________________________________________________
	Professional FreeSWITCH Consulting Services:
	consulting at freeswitch.org
	http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> 
	
	
	 </> 
	
	Official FreeSWITCH Sites
	http://www.freeswitch.org <http://www.freeswitch.org/> 
	http://wiki.freeswitch.org <http://wiki.freeswitch.org/> 
	http://www.cluecon.com <http://www.cluecon.com/> 
	
	Join Us At ClueCon - Aug 7-9, 2012
	
	FreeSWITCH-users mailing list
	FreeSWITCH-users at 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/> 
	

P Please consider the environment before printing this email 

http://www.bbc.co.uk/
This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated.
If you have received it in error, please delete it from your system.
Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately.
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From jbr at consiglia.dk  Mon Jun 18 17:38:55 2012
From: jbr at consiglia.dk (Jon Bruel)
Date: Mon, 18 Jun 2012 15:38:55 +0200
Subject: [Freeswitch-users] sendevent NOTIFY
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE014608F5348C@SBS2008SERVER.consiglia.local>

After some time working with BroadSoft, I have returned to the FreeSWITCH, and I'm impressed with the overall improvements in consistency and documentation. Great work!

I have tested the Event Socket sendevent NOTIFY command, and I have tried to check if I could change the SIP From and SIP To headers to suit my requirements. It looks as if there is no way.

So is there a way to specify the From- and To-headers, including the associated tags?

I need this to be able to control the phone sets (set it off-hook).

Thanks /Jon

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From alex at thewinelake.com  Mon Jun 18 18:03:45 2012
From: alex at thewinelake.com (Alex)
Date: Mon, 18 Jun 2012 15:03:45 +0100
Subject: [Freeswitch-users] lua trouble
Message-ID: <4FDF3541.4010808@thewinelake.com>

Trying to do a lua-based bit of IVR and inexplicably struggling with 
this bit. Maybe something to do with use of variables in 
session:playAndGetDigits or possibly the regexp?

                 destnum = getDestNumber()
                 session:execute("log", "INFO User dialled "..destnum)
                 confPrompt = "say:'Now calling telephone number 
"..destnum.."'"
                 session:execute("log", "INFO Confirming with "..confPrompt)
                 confirmkey = session:playAndGetDigits(0, 1, 1, 1000, 
"#", confPrompt, "", "\\d|\\*")
                 session:execute("log", "INFO mcm.lua Number confirm 
(not) interrupted with "..confirmkey.." mcmoption="..mcmoption)
                 if confirmkey == "*" then
                         -- return to call
                         break
                 end
                 session:execute("transfer","-bleg "..destnum.." XML 
dp"..tenant_id)

The problem is with
                 confirmkey = session:playAndGetDigits(0, 1, 1, 1000, 
"#", confPrompt, "", "\\d|\\*")

It complains of "Invalid Args" (and doesn't say anything). All it has to 
do is read out the number that it's intending to call, allowing the user 
to change their mind by hitting *.

Logfile reports:

2012-06-18 14:00:22.975225 [INFO] mod_dptools.c:1444 mcm2.lua getDestNumber
2012-06-18 14:00:22.975225 [DEBUG] switch_ivr_play_say.c:1309 Codec 
Activated L16 at 8000hz 1 channels 20ms
2012-06-18 14:00:26.035180 [DEBUG] switch_rtp.c:3457 RTP RECV DTMF 1:1040
2012-06-18 14:00:26.035180 [DEBUG] switch_ivr_play_say.c:1682 done 
playing file /home/oev/fs-pabx/prompts/dmsb12_mcm3.wav
2012-06-18 14:00:26.205278 [DEBUG] switch_rtp.c:3457 RTP RECV DTMF 2:720
2012-06-18 14:00:26.605240 [DEBUG] switch_rtp.c:3457 RTP RECV DTMF 3:1040
2012-06-18 14:00:27.415292 [DEBUG] switch_rtp.c:3457 RTP RECV DTMF #:1120
2012-06-18 14:00:27.415292 [DEBUG] switch_ivr_play_say.c:2026 Test Regex 
[123][\d+]
EXECUTE sofia/internal/898000000000202074904605 at 193.105.54.10 log(INFO 
User dialled 123)
2012-06-18 14:00:27.415292 [INFO] mod_dptools.c:1444 User dialled 123
EXECUTE sofia/internal/898000000000202074904605 at 193.105.54.10 log(INFO 
Confirming with say:'Now calling telephone number 123')
2012-06-18 14:00:27.415292 [INFO] mod_dptools.c:1444 Confirming with 
say:'Now calling telephone number 123'
2012-06-18 14:00:27.415292 [ERR] switch_ivr_play_say.c:1174 Invalid Args
EXECUTE sofia/internal/898000000000202074904605 at 193.105.54.10 log(INFO 
mcm.lua Number confirm (not) interrupted with  mcmoption=8)
2012-06-18 14:00:28.425260 [INFO] mod_dptools.c:1444 mcm.lua Number 
confirm (not) interrupted with  mcmoption=8
EXECUTE sofia/internal/898000000000202074904605 at 193.105.54.10 
transfer(-bleg 123 XML dp0)



From ocset at the800group.com  Mon Jun 18 18:06:09 2012
From: ocset at the800group.com (ocset)
Date: Mon, 18 Jun 2012 22:06:09 +0800
Subject: [Freeswitch-users] External profile & NAT
Message-ID: <4FDF35D1.306@the800group.com>

Hi

I would like to allow a person with a softphone app to register with 
Freeswitch (extension 1111) when they are outside the office network.

I would like to to implement this using an external profile. According 
to the wiki page (http://wiki.freeswitch.org/wiki/External_profile) I 
just need to create a copy of the "conf/sip_profiles/external.xml" file 
and change the port number to 5090 (and also open the port on the 
modem). When I do a "sofia status", this new gateway does not show up 
and to be honest, I don't quite understand how that works and what else 
to read to make sense of what to do.

What are the ways in which other Freeswitch admins are granting access 
to external devices?

ps. just to confirm, do I set external_rtp_ip and external_sip_ip to the 
fix IP address that my ISP has given me - the one that is shown by 
www.whatsmyip.org?

Please help
Thanks
O.


From mariya at cs.ucsb.edu  Mon Jun 18 01:48:36 2012
From: mariya at cs.ucsb.edu (Mariya Zheleva)
Date: Sun, 17 Jun 2012 14:48:36 -0700
Subject: [Freeswitch-users] Freeswitch [DEBUG] libdingaling.c:1242 TLS NOT
	SUPPORTED IN THIS BUILD!
Message-ID: <CANr3YS3PWU5B5LPyqNHUUkVDs0q=Mu_+=qFc0NJM6Vx3h-dNEA@mail.gmail.com>

Hello,

I am trying to configure freeswitch for outbound calls and SMS through
google voice using mod_dingaling.
I have succeeded with the SMS configuration but I can't get call to work.
It seems the problem is that TLS can't be successfully built
upon freeswitch installation. I followed the instructions here:
http://wiki.freeswitch.org/wiki/Mod_dingaling
I added the flag "-lgnutls" to gcc in relink_command in the file file
src/mod/endpoints/mod_dingaling/mod_dingaling.la, however I still get
*[DEBUG] libdingaling.c:1242 TLS NOT SUPPORTED IN THIS BUILD!*
in the freeswitch.log

This is the output when I make mod_dingaling-install.
*
*
*openbts at ubuntu:/usr/local/src/freeswitch$ sudo make mod_dingaling-install*
* /bin/bash /usr/local/src/freeswitch/quiet_libtool   --mode=install
/usr/bin/install -c   libfreeswitch.la '/usr/local/freeswitch/lib'*
*quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.so.1.0.0
/usr/local/freeswitch/lib/libfreeswitch.so.1.0.0*
*quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f
libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 &&
ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; })*
*quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f
libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so && ln
-s libfreeswitch.so.1.0.0 libfreeswitch.so; }; })*
*quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.lai
/usr/local/freeswitch/lib/libfreeswitch.la*
*quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.a
/usr/local/freeswitch/lib/libfreeswitch.a*
*quiet_libtool: install: chmod 644 /usr/local/freeswitch/lib/libfreeswitch.a
*
*quiet_libtool: install: ranlib /usr/local/freeswitch/lib/libfreeswitch.a*
*quiet_libtool: finish:
PATH="/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin:/usr/X11R6/bin:/sbin"
ldconfig -n /usr/local/freeswitch/lib*
*----------------------------------------------------------------------*
*Libraries have been installed in:*
*   /usr/local/freeswitch/lib*
*
*
*If you ever happen to want to link against installed libraries*
*in a given directory, LIBDIR, you must either use libtool, and*
*specify the full pathname of the library, or use the `-LLIBDIR'*
*flag during linking and do at least one of the following:*
*   - add LIBDIR to the `LD_LIBRARY_PATH' environment variable*
*     during execution*
*   - add LIBDIR to the `LD_RUN_PATH' environment variable*
*     during linking*
*   - use the `-Wl,-rpath -Wl,LIBDIR' linker flag*
*   - have your system administrator add LIBDIR to `/etc/ld.so.conf'*
*
*
*See any operating system documentation about shared libraries for*
*more information, such as the ld(1) and ld.so(8) manual pages.*
*----------------------------------------------------------------------*
*
*
*making install mod_dingaling*

Any suggestions as to how to build TLS are highly appreciated.

Thanks,
mariya~
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From adam.koleszar at virtual-call-center.eu  Mon Jun 18 15:59:22 2012
From: adam.koleszar at virtual-call-center.eu (=?ISO-8859-2?Q?=C1d=E1m_Kolesz=E1r?=)
Date: Mon, 18 Jun 2012 13:59:22 +0200
Subject: [Freeswitch-users] mod_syslog
Message-ID: <4FDF181A.6030701@virtual-call-center.eu>

Hi,

A have a little problem with the mod_syslog module. Freeswitch generates 
multiline log messages (for example: SDP or register messages) which 
syslog-ng OSE can't handle. The firs part of the message is sent 
correctly but missing the facility and ident from the other part of the 
message. Because of this the messages can not be filtered by ident (or 
facility) on server side and goes to wrong destination.

My question is, have you ever noticed this kind of problem?
Thank you for your help in advance.

Br,
Adam


From avi at avimarcus.net  Mon Jun 18 18:53:29 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 18 Jun 2012 17:53:29 +0300
Subject: [Freeswitch-users] Freeswitch [DEBUG] libdingaling.c:1242 TLS
 NOT SUPPORTED IN THIS BUILD!
In-Reply-To: <CANr3YS3PWU5B5LPyqNHUUkVDs0q=Mu_+=qFc0NJM6Vx3h-dNEA@mail.gmail.com>
References: <CANr3YS3PWU5B5LPyqNHUUkVDs0q=Mu_+=qFc0NJM6Vx3h-dNEA@mail.gmail.com>
Message-ID: <CALNZuwmYOWGX9NB8Rx7JAD8XA-bMLbXRwVC=A7QSLW7_=vKNhw@mail.gmail.com>

Ah, that's outdated.
"Since October 17,2011 this is no longer needed now OpenSSL is used."

Try using the package openssl-devel (centos?) or libssl-dev
(debian/ubuntu?) and probably re-bootstrapping / reconfig and try building
again.

Then please update the wiki.

I'm surprised that one didn't get fixed...


-Avi



On Mon, Jun 18, 2012 at 12:48 AM, Mariya Zheleva <mariya at cs.ucsb.edu> wrote:

> Hello,
>
> I am trying to configure freeswitch for outbound calls and SMS through
> google voice using mod_dingaling.
> I have succeeded with the SMS configuration but I can't get call to work.
> It seems the problem is that TLS can't be successfully built
> upon freeswitch installation. I followed the instructions here:
> http://wiki.freeswitch.org/wiki/Mod_dingaling
> I added the flag "-lgnutls" to gcc in relink_command in the file file
> src/mod/endpoints/mod_dingaling/mod_dingaling.la, however I still get
> *[DEBUG] libdingaling.c:1242 TLS NOT SUPPORTED IN THIS BUILD!*
> in the freeswitch.log
>
> This is the output when I make mod_dingaling-install.
> *
> *
> *openbts at ubuntu:/usr/local/src/freeswitch$ sudo make mod_dingaling-install
> *
> * /bin/bash /usr/local/src/freeswitch/quiet_libtool   --mode=install
> /usr/bin/install -c   libfreeswitch.la '/usr/local/freeswitch/lib'*
> *quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.so.1.0.0
> /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0*
> *quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f
> libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 &&
> ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; })*
> *quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f
> libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so && ln
> -s libfreeswitch.so.1.0.0 libfreeswitch.so; }; })*
> *quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.lai
> /usr/local/freeswitch/lib/libfreeswitch.la*
> *quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.a
> /usr/local/freeswitch/lib/libfreeswitch.a*
> *quiet_libtool: install: chmod 644
> /usr/local/freeswitch/lib/libfreeswitch.a*
> *quiet_libtool: install: ranlib /usr/local/freeswitch/lib/libfreeswitch.a*
> *quiet_libtool: finish:
> PATH="/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin:/usr/X11R6/bin:/sbin"
> ldconfig -n /usr/local/freeswitch/lib*
> *----------------------------------------------------------------------*
> *Libraries have been installed in:*
> *   /usr/local/freeswitch/lib*
> *
> *
> *If you ever happen to want to link against installed libraries*
> *in a given directory, LIBDIR, you must either use libtool, and*
> *specify the full pathname of the library, or use the `-LLIBDIR'*
> *flag during linking and do at least one of the following:*
> *   - add LIBDIR to the `LD_LIBRARY_PATH' environment variable*
> *     during execution*
> *   - add LIBDIR to the `LD_RUN_PATH' environment variable*
> *     during linking*
> *   - use the `-Wl,-rpath -Wl,LIBDIR' linker flag*
> *   - have your system administrator add LIBDIR to `/etc/ld.so.conf'*
> *
> *
> *See any operating system documentation about shared libraries for*
> *more information, such as the ld(1) and ld.so(8) manual pages.*
> *----------------------------------------------------------------------*
> *
> *
> *making install mod_dingaling*
>
> Any suggestions as to how to build TLS are highly appreciated.
>
> Thanks,
> mariya~
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun 18 18:55:32 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jun 2012 07:55:32 -0700
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <4FDF3541.4010808@thewinelake.com>
References: <4FDF3541.4010808@thewinelake.com>
Message-ID: <CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>

>
> The problem is with
>                 confirmkey = session:playAndGetDigits(0, 1, 1, 1000,
> "#", confPrompt, "", "\\d|\\*")
>

It complains of "Invalid Args" (and doesn't say anything). All it has to
> do is read out the number that it's intending to call, allowing the user
> to change their mind by hitting *.
>
>
You can't have an argument of "0" for the minimum number of digits. See
this page for details on play_and_get_digits args:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits

-MC
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From msc at freeswitch.org  Mon Jun 18 18:56:59 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jun 2012 07:56:59 -0700
Subject: [Freeswitch-users] External profile & NAT
In-Reply-To: <4FDF35D1.306@the800group.com>
References: <4FDF35D1.306@the800group.com>
Message-ID: <CAKzWOxUW5ehDU+5cc=FCLnXX6=_iA0Gr0REwa0=wz9huztAExw@mail.gmail.com>

After you add the external profile you need to reloadxml and then you need
to start the profile - assuming you didn't just restart FreeSWITCH or
reload mod_sofia.

-MC

On Mon, Jun 18, 2012 at 7:06 AM, ocset <ocset at the800group.com> wrote:

> Hi
>
> I would like to allow a person with a softphone app to register with
> Freeswitch (extension 1111) when they are outside the office network.
>
> I would like to to implement this using an external profile. According
> to the wiki page (http://wiki.freeswitch.org/wiki/External_profile) I
> just need to create a copy of the "conf/sip_profiles/external.xml" file
> and change the port number to 5090 (and also open the port on the
> modem). When I do a "sofia status", this new gateway does not show up
> and to be honest, I don't quite understand how that works and what else
> to read to make sense of what to do.
>
> What are the ways in which other Freeswitch admins are granting access
> to external devices?
>
> ps. just to confirm, do I set external_rtp_ip and external_sip_ip to the
> fix IP address that my ISP has given me - the one that is shown by
> www.whatsmyip.org?
>
> Please help
> Thanks
> O.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From avi at avimarcus.net  Mon Jun 18 19:00:34 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 18 Jun 2012 18:00:34 +0300
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
Message-ID: <CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>

Uhm I thought I just wrote code that did that last week in lua. Seemed to
work GREAT.

if(phone_required==true) then
...
 else
phone_min=0 --no minimum on the match length
end
phone=session:playAndGetDigits(phone_min, phone_max, 3, 20000, "#",
"phrase:cc_phone_prompt:"..phone_skip,
"phrase:cc_invalid_entry","\\d{"..phone_min..","..phone_max.."}");

-Avi


On Mon, Jun 18, 2012 at 5:55 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
>> The problem is with
>>                 confirmkey = session:playAndGetDigits(0, 1, 1, 1000,
>> "#", confPrompt, "", "\\d|\\*")
>>
>
> It complains of "Invalid Args" (and doesn't say anything). All it has to
>> do is read out the number that it's intending to call, allowing the user
>> to change their mind by hitting *.
>>
>>
> You can't have an argument of "0" for the minimum number of digits. See
> this page for details on play_and_get_digits args:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
>
> -MC
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun 18 19:09:46 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jun 2012 08:09:46 -0700
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
Message-ID: <CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>

But but but the wiki says! The wiki just CAN'T be wrong! ;) Actually,
according to switch_ivr_play_say.c it looks like you can have a min_digits
of "0". Still, I'd make the min digits "1" unless there's a compelling
reason to want zero digits.
-MC

On Mon, Jun 18, 2012 at 8:00 AM, Avi Marcus <avi at avimarcus.net> wrote:

> Uhm I thought I just wrote code that did that last week in lua. Seemed to
> work GREAT.
>
> if(phone_required==true) then
> ...
>  else
> phone_min=0 --no minimum on the match length
> end
> phone=session:playAndGetDigits(phone_min, phone_max, 3, 20000, "#",
> "phrase:cc_phone_prompt:"..phone_skip,
> "phrase:cc_invalid_entry","\\d{"..phone_min..","..phone_max.."}");
>
> -Avi
>
>
> On Mon, Jun 18, 2012 at 5:55 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>>> The problem is with
>>>                 confirmkey = session:playAndGetDigits(0, 1, 1, 1000,
>>> "#", confPrompt, "", "\\d|\\*")
>>>
>>
>> It complains of "Invalid Args" (and doesn't say anything). All it has to
>>> do is read out the number that it's intending to call, allowing the user
>>> to change their mind by hitting *.
>>>
>>>
>> You can't have an argument of "0" for the minimum number of digits. See
>> this page for details on play_and_get_digits args:
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
>>
>> -MC
>>
>>
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From avi at avimarcus.net  Mon Jun 18 19:15:11 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 18 Jun 2012 18:15:11 +0300
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
Message-ID: <CALNZuw=8Q=7cy30i3UFDJH77-tZ6HEQJiiRwRvzhVo2FpYLMKA@mail.gmail.com>

For me there was. "enter the value or press # to skip" because it's
optional... seems a very possible use-case. It's an IVR for gathering
information, not navigating a menu.
-Avi


On Mon, Jun 18, 2012 at 6:09 PM, Michael Collins <msc at freeswitch.org> wrote:

> But but but the wiki says! The wiki just CAN'T be wrong! ;) Actually,
> according to switch_ivr_play_say.c it looks like you can have a min_digits
> of "0". Still, I'd make the min digits "1" unless there's a compelling
> reason to want zero digits.
> -MC
>
>
> On Mon, Jun 18, 2012 at 8:00 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> Uhm I thought I just wrote code that did that last week in lua. Seemed to
>> work GREAT.
>>
>> if(phone_required==true) then
>> ...
>>  else
>> phone_min=0 --no minimum on the match length
>> end
>> phone=session:playAndGetDigits(phone_min, phone_max, 3, 20000, "#",
>> "phrase:cc_phone_prompt:"..phone_skip,
>> "phrase:cc_invalid_entry","\\d{"..phone_min..","..phone_max.."}");
>>
>> -Avi
>>
>>
>> On Mon, Jun 18, 2012 at 5:55 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>>> The problem is with
>>>>                 confirmkey = session:playAndGetDigits(0, 1, 1, 1000,
>>>> "#", confPrompt, "", "\\d|\\*")
>>>>
>>>
>>> It complains of "Invalid Args" (and doesn't say anything). All it has to
>>>> do is read out the number that it's intending to call, allowing the user
>>>> to change their mind by hitting *.
>>>>
>>>>
>>> You can't have an argument of "0" for the minimum number of digits. See
>>> this page for details on play_and_get_digits args:
>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
>>>
>>> -MC
>>>
>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gshepard at star2star.com  Mon Jun 18 19:19:08 2012
From: gshepard at star2star.com (Gabe Shepard)
Date: Mon, 18 Jun 2012 11:19:08 -0400
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
Message-ID: <CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>

I think in mod_dptools.c in the play_and_get_digits_function there's some
code that does:

    if (min_digits <= 1) {
        min_digits = 1;
    }

   My C is rusty and this may be irrelevant though!

-Gabe

On Mon, Jun 18, 2012 at 11:09 AM, Michael Collins <msc at freeswitch.org>wrote:

> But but but the wiki says! The wiki just CAN'T be wrong! ;) Actually,
> according to switch_ivr_play_say.c it looks like you can have a min_digits
> of "0". Still, I'd make the min digits "1" unless there's a compelling
> reason to want zero digits.
> -MC
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From alex at thewinelake.com  Mon Jun 18 20:02:58 2012
From: alex at thewinelake.com (Alex)
Date: Mon, 18 Jun 2012 17:02:58 +0100
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
	<CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
Message-ID: <4FDF5132.5030303@thewinelake.com>

I've changed the 0 to a 1 and it didn't make any difference.
I'm pretty sure it's all to do with trying to use "say" within the lua 
incarnation of playAndGetDigits.

> I think in mod_dptools.c in the play_and_get_digits_function there's 
> some code that does:
>
>     if (min_digits <= 1) {
>         min_digits = 1;
>     }
>
>    My C is rusty and this may be irrelevant though!
>
> -Gabe
>
> On Mon, Jun 18, 2012 at 11:09 AM, Michael Collins <msc at freeswitch.org 
> <mailto:msc at freeswitch.org>> wrote:
>
>     But but but the wiki says! The wiki just CAN'T be wrong! ;)
>     Actually, according to switch_ivr_play_say.c it looks like you can
>     have a min_digits of "0". Still, I'd make the min digits "1"
>     unless there's a compelling reason to want zero digits.
>     -MC
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2433/5076 - Release Date: 06/17/12
>


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From alex at thewinelake.com  Mon Jun 18 20:03:49 2012
From: alex at thewinelake.com (Alex)
Date: Mon, 18 Jun 2012 17:03:49 +0100
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
	<CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
Message-ID: <4FDF5165.7020907@thewinelake.com>

Maybe there's a simpler way to do a "speak" that's interruptable by DTMF 
and then finding out what the DTMF was?
> I think in mod_dptools.c in the play_and_get_digits_function there's 
> some code that does:
>
>     if (min_digits <= 1) {
>         min_digits = 1;
>     }
>
>    My C is rusty and this may be irrelevant though!
>
> -Gabe
>
> On Mon, Jun 18, 2012 at 11:09 AM, Michael Collins <msc at freeswitch.org 
> <mailto:msc at freeswitch.org>> wrote:
>
>     But but but the wiki says! The wiki just CAN'T be wrong! ;)
>     Actually, according to switch_ivr_play_say.c it looks like you can
>     have a min_digits of "0". Still, I'd make the min digits "1"
>     unless there's a compelling reason to want zero digits.
>     -MC
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2433/5076 - Release Date: 06/17/12
>


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From ocset at the800group.com  Mon Jun 18 20:06:31 2012
From: ocset at the800group.com (ocset)
Date: Tue, 19 Jun 2012 00:06:31 +0800
Subject: [Freeswitch-users] External profile & NAT
In-Reply-To: <CAKzWOxUW5ehDU+5cc=FCLnXX6=_iA0Gr0REwa0=wz9huztAExw@mail.gmail.com>
References: <4FDF35D1.306@the800group.com>
	<CAKzWOxUW5ehDU+5cc=FCLnXX6=_iA0Gr0REwa0=wz9huztAExw@mail.gmail.com>
Message-ID: <4FDF5207.7000204@the800group.com>

Thanks Michael

I found my mistake. I had copied the new external5090.xml file into the 
external directory and not into the same directory as external.xml. 
After restating FreeSWITCH, I can now see the new profile:

external5090    profile               sip:mod_sofia at 192.168.0.16:5090    
RUNNING (0)

I have removed the <gateway> section as I presume I don't need these 
gateways to be loaded for my new profile?

<gateways>
<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
</gateways>

Does the external softphone just connect using an existing  user/ext or 
do I now somehow create a gateway that limits which users are exposed to 
the internet.

Thanks again for your guidance.
O

On 18/06/12 22:56, Michael Collins wrote:
> After you add the external profile you need to reloadxml and then you 
> need to start the profile - assuming you didn't just restart 
> FreeSWITCH or reload mod_sofia.
>
> -MC
>
> On Mon, Jun 18, 2012 at 7:06 AM, ocset <ocset at the800group.com 
> <mailto:ocset at the800group.com>> wrote:
>
>     Hi
>
>     I would like to allow a person with a softphone app to register with
>     Freeswitch (extension 1111) when they are outside the office network.
>
>     I would like to to implement this using an external profile. According
>     to the wiki page (http://wiki.freeswitch.org/wiki/External_profile) I
>     just need to create a copy of the "conf/sip_profiles/external.xml"
>     file
>     and change the port number to 5090 (and also open the port on the
>     modem). When I do a "sofia status", this new gateway does not show up
>     and to be honest, I don't quite understand how that works and what
>     else
>     to read to make sense of what to do.
>
>     What are the ways in which other Freeswitch admins are granting access
>     to external devices?
>
>     ps. just to confirm, do I set external_rtp_ip and external_sip_ip
>     to the
>     fix IP address that my ISP has given me - the one that is shown by
>     www.whatsmyip.org <http://www.whatsmyip.org>?
>
>     Please help
>     Thanks
>     O.
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun 18 20:08:05 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jun 2012 09:08:05 -0700
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <4FDF5165.7020907@thewinelake.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
	<CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
	<4FDF5165.7020907@thewinelake.com>
Message-ID: <CAKzWOxWsAFSwdHgqE7Ehmrs6d9G3VKh2MhiAB3UdHCv0pYpbng@mail.gmail.com>

If you're just trying to do "say" (using the existing pre-recorded prompts)
or "speak" (for TTS) then just use a phrase macro. Look in conf/lang/en for
examples of different types of phrase macros. See chapter 6 of the original
FS book for a gentle tutorial on using phrase macros. (It mentions them in
the context of using XML IVRs but phrase macros can be used anywhere in
FreeSWITCH where a sound file would normally be specified.)

-MC

On Mon, Jun 18, 2012 at 9:03 AM, Alex <alex at thewinelake.com> wrote:

>  Maybe there's a simpler way to do a "speak" that's interruptable by DTMF
> and then finding out what the DTMF was?
>
> I think in mod_dptools.c in the play_and_get_digits_function there's some
> code that does:
>
>      if (min_digits <= 1) {
>         min_digits = 1;
>     }
>
>     My C is rusty and this may be irrelevant though!
>
>  -Gabe
>
> On Mon, Jun 18, 2012 at 11:09 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> But but but the wiki says! The wiki just CAN'T be wrong! ;) Actually,
>> according to switch_ivr_play_say.c it looks like you can have a min_digits
>> of "0". Still, I'd make the min digits "1" unless there's a compelling
>> reason to want zero digits.
>> -MC
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2177 / Virus Database: 2433/5076 - Release Date: 06/17/12
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From alex at thewinelake.com  Mon Jun 18 20:10:34 2012
From: alex at thewinelake.com (Alex)
Date: Mon, 18 Jun 2012 17:10:34 +0100
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
	<CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
Message-ID: <4FDF52FA.2070007@thewinelake.com>

Ah, just seen this: 
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073877.html

I thought I wanted say, but maybe could be persuaded to use speak if it 
came to it....
> I think in mod_dptools.c in the play_and_get_digits_function there's 
> some code that does:
>
>     if (min_digits <= 1) {
>         min_digits = 1;
>     }
>
>    My C is rusty and this may be irrelevant though!
>
> -Gabe
>
> On Mon, Jun 18, 2012 at 11:09 AM, Michael Collins <msc at freeswitch.org 
> <mailto:msc at freeswitch.org>> wrote:
>
>     But but but the wiki says! The wiki just CAN'T be wrong! ;)
>     Actually, according to switch_ivr_play_say.c it looks like you can
>     have a min_digits of "0". Still, I'd make the min digits "1"
>     unless there's a compelling reason to want zero digits.
>     -MC
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2433/5076 - Release Date: 06/17/12
>


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From avi at avimarcus.net  Mon Jun 18 20:10:24 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 18 Jun 2012 19:10:24 +0300
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <4FDF5132.5030303@thewinelake.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
	<CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
	<4FDF5132.5030303@thewinelake.com>
Message-ID: <CALNZuwkMn4MhmedCOt2RUKxhbORcCjhVozgwbAB1cNaHZZNhNQ@mail.gmail.com>

maybe.. try turning your "confPrompt" into a "phrase:confPrompt" and
see what happens.

<macro name="confPrompt" pause="100">
	<input pattern=".*"><match>
		<action function="speak-text" data="My spoken prompt."/>
	</match></input>
	</macro>

-Avi


From msc at freeswitch.org  Mon Jun 18 20:11:34 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jun 2012 09:11:34 -0700
Subject: [Freeswitch-users] External profile & NAT
In-Reply-To: <4FDF5207.7000204@the800group.com>
References: <4FDF35D1.306@the800group.com>
	<CAKzWOxUW5ehDU+5cc=FCLnXX6=_iA0Gr0REwa0=wz9huztAExw@mail.gmail.com>
	<4FDF5207.7000204@the800group.com>
Message-ID: <CAKzWOxWTz49csQmCupEXzCOepKR9+gVQDBvgrDUaODRbPUxv4Q@mail.gmail.com>

Correct, the gateways  section is only if you wish to create gateways to
connect to other servers. Since you plan to use this profile only for
inbound registrations you're fine. This profile can utilize existing users
from your user directory.
-MC

On Mon, Jun 18, 2012 at 9:06 AM, ocset <ocset at the800group.com> wrote:

>  Thanks Michael
>
> I found my mistake. I had copied the new external5090.xml file into the
> external directory and not into the same directory as external.xml. After
> restating FreeSWITCH, I can now see the new profile:
>
> external5090    profile               sip:mod_sofia at 192.168.0.16:5090
> RUNNING (0)
>
> I have removed the <gateway> section as I presume I don't need these
> gateways to be loaded for my new profile?
>
> <gateways>
>     <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
> </gateways>
>
> Does the external softphone just connect using an existing  user/ext or do
> I now somehow create a gateway that limits which users are exposed to the
> internet.
>
> Thanks again for your guidance.
> O
>
> On 18/06/12 22:56, Michael Collins wrote:
>
> After you add the external profile you need to reloadxml and then you need
> to start the profile - assuming you didn't just restart FreeSWITCH or
> reload mod_sofia.
>
> -MC
>
> On Mon, Jun 18, 2012 at 7:06 AM, ocset <ocset at the800group.com> wrote:
>
>> Hi
>>
>> I would like to allow a person with a softphone app to register with
>> Freeswitch (extension 1111) when they are outside the office network.
>>
>> I would like to to implement this using an external profile. According
>> to the wiki page (http://wiki.freeswitch.org/wiki/External_profile) I
>> just need to create a copy of the "conf/sip_profiles/external.xml" file
>> and change the port number to 5090 (and also open the port on the
>> modem). When I do a "sofia status", this new gateway does not show up
>> and to be honest, I don't quite understand how that works and what else
>> to read to make sense of what to do.
>>
>> What are the ways in which other Freeswitch admins are granting access
>> to external devices?
>>
>> ps. just to confirm, do I set external_rtp_ip and external_sip_ip to the
>> fix IP address that my ISP has given me - the one that is shown by
>> www.whatsmyip.org?
>>
>> Please help
>> Thanks
>> O.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Mon Jun 18 20:14:29 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 18 Jun 2012 19:14:29 +0300
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <4FDF52FA.2070007@thewinelake.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
	<CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
	<4FDF52FA.2070007@thewinelake.com>
Message-ID: <CALNZuwmhmW7Yp1owGtLK+z8EhG=URxn7c0YpZtkyZvFU24SNMA@mail.gmail.com>

Phrases let you mix and match speak, say, play-file, silence, other macros,
and even passing in options.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_phrase
http://wiki.freeswitch.org/wiki/Speech_Phrase_Management

-Avi
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From alex at thewinelake.com  Mon Jun 18 20:20:16 2012
From: alex at thewinelake.com (Alex)
Date: Mon, 18 Jun 2012 17:20:16 +0100
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <CALNZuwmhmW7Yp1owGtLK+z8EhG=URxn7c0YpZtkyZvFU24SNMA@mail.gmail.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
	<CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
	<4FDF52FA.2070007@thewinelake.com>
	<CALNZuwmhmW7Yp1owGtLK+z8EhG=URxn7c0YpZtkyZvFU24SNMA@mail.gmail.com>
Message-ID: <4FDF5540.5050708@thewinelake.com>

OK - something for a rainy day. No confirmations for now ;-)
> Phrases let you mix and match speak, say, play-file, silence, other 
> macros, and even passing in options.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_phrase
> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
>
> -Avi
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2433/5076 - Release Date: 06/17/12
>


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From alex at digitalmail.com  Mon Jun 18 20:26:07 2012
From: alex at digitalmail.com (Alex Lake)
Date: Mon, 18 Jun 2012 17:26:07 +0100
Subject: [Freeswitch-users] Using * as "terminate and cancel"
Message-ID: <4FDF569F.9020409@digitalmail.com>

Is there a way to get playAndGetDigits to allow the user to enter a 
(telephone) number, but if they decide part way through that they messed 
up and they want to abort, they can  press * - and then (without 
checking validation) the lua script knows that they pressed * rather 
than the more normal # (which means "I've finished - now go do it!").


From jaasmailing at gmail.com  Mon Jun 18 20:28:07 2012
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Mon, 18 Jun 2012 18:28:07 +0200
Subject: [Freeswitch-users] BYE authentication problem
In-Reply-To: <4FDB09D4.4050703@gmail.com>
References: <4FDB09D4.4050703@gmail.com>
Message-ID: <4FDF5717.70000@gmail.com>

Does anyone have the same problem?

Regards,


Il 15/06/12 12.09, Carlo Dimaggio ha scritto:
> Hello,
>
> my FS has a trunk with an openser:
>
> FS -> openser (sip.test.com - 1.2.3.4) -> cisco gateway
>
> I have configured a gateway with this sip proxy / realm and I have a 
> problem with BYE.
> When I call from PSTN (cisco gateway) to FS, the cisco device set a 
> realm "1.2.3.4" and openser ask for authentication, while FS tell:
>
> 2012-06-14 14:23:44.018150 [ERR] sofia_reg.c:2160 Cannot locate any 
> authentication credentials to complete an authentication request for 
> realm '"1.2.3.4"'
>
> I know that FS asks that realm configured in the gateway section 
> (sip.test.com) should match the realm in the challenge packet from the 
> other device (that instead is 1.2.3.4), but the Reverse DNS of 1.2.3.4 
> is sip.test.com (and viceversa).
>
> Should FS respond to the challenge packet? How can I solve this problem?
>
>
> Best Regards,

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From marketing at cluecon.com  Mon Jun 18 20:40:04 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 18 Jun 2012 09:40:04 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxX0pBC5J_TX+qe-KZN+Lr=zPtOY5ceAxu-CL4hXbsqtvg@mail.gmail.com>

We hope you're having a nice Monday today.

The past week was a busy one both for the
FreeSWITCH<http://www.freeswitch.org?cc12-0618>project and for
ClueCon <http://www.cluecon.com?cc12-0618>. Last week's community
conference call was a nice discussion about some of the things that will be
happening at ClueCon this year. We also had a visit from Diana Cionoiu of
the YATE <http://yate.null.ro/pmwiki?cc12-0618> project. This coming
Wednesday we will have Darren Schreiber, co-author of both FreeSWITCH books
and co-founder of the 2600hz project <http://www.2600hz.org?cc12-0618>, on
our call to give his world-famous "SIP 101" presentation. We definitely
look forward to that discussion.

We also wanted to highlight an interesting blog
post<http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html?cc12-0618>by
our friend and community member Kristian Kielhofner. Its provocative
title is: "Everything you wish you didn't need to know about VoIP." Over
the years Kristian has collected a lot of knowledge about interoperability
- and lack thereof - between various VoIP devices and servers. Whether
you're a VoIP novice or veteran we think you will appreciate seeing this
knowledge written down for the benefit of all.

In ClueCon news are happy to announce two training session and a birds of a
feather (BOF) meetup for this year's event. Paid attendees of ClueCon will
have their choice of FreeSWITCH or OpenSIPS training. The FreeSWITCH
training will be conducted by the aforementioned Darren Schreiber. The
OpenSIPS training will be conducted by none other than Bogdan-Andrei Iancu,
lead developer of the OpenSIPS project. The training sessions will take
place on Monday, August 6. On the evening of Wednesday, August 8 we will be
having a VoIP security ("VoIPSec") meetup for those interested in
discussing this subject with others who share an interest. Among those who
will be present are Phil Zimmermann of PGP and
ZRTP<http://zfone.com?cc12-0618>fame.

Be sure to register right away so that you can get your extra chances to
win. Anyone who registers before July 4th of this year will receive 10
entries in the great ClueCon giveaway. We have a lot of interesting items
to give away this year, so stay tuned. And while you are at the ClueCon
site to register <http://www.cluecon.com/register?cc12-0618>, but sure to
review the schedule <http://www.cluecon.com/schedule?cc12-0618> as we've
added a number of new speakers in the past week.

See you in August!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
cc12-0618
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From covici at ccs.covici.com  Mon Jun 18 20:52:18 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 18 Jun 2012 12:52:18 -0400
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKzWOxX0pBC5J_TX+qe-KZN+Lr=zPtOY5ceAxu-CL4hXbsqtvg@mail.gmail.com>
References: <CAKzWOxX0pBC5J_TX+qe-KZN+Lr=zPtOY5ceAxu-CL4hXbsqtvg@mail.gmail.com>
Message-ID: <15212.1340038338@ccs.covici.com>

I just wish the blog was not a google doc!  Can't read those.

Michael Collins <marketing at cluecon.com> wrote:

> We hope you're having a nice Monday today.
> 
> The past week was a busy one both for the
> FreeSWITCH<http://www.freeswitch.org?cc12-0618>project and for
> ClueCon <http://www.cluecon.com?cc12-0618>. Last week's community
> conference call was a nice discussion about some of the things that will be
> happening at ClueCon this year. We also had a visit from Diana Cionoiu of
> the YATE <http://yate.null.ro/pmwiki?cc12-0618> project. This coming
> Wednesday we will have Darren Schreiber, co-author of both FreeSWITCH books
> and co-founder of the 2600hz project <http://www.2600hz.org?cc12-0618>, on
> our call to give his world-famous "SIP 101" presentation. We definitely
> look forward to that discussion.
> 
> We also wanted to highlight an interesting blog
> post<http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html?cc12-0618>by
> our friend and community member Kristian Kielhofner. Its provocative
> title is: "Everything you wish you didn't need to know about VoIP." Over
> the years Kristian has collected a lot of knowledge about interoperability
> - and lack thereof - between various VoIP devices and servers. Whether
> you're a VoIP novice or veteran we think you will appreciate seeing this
> knowledge written down for the benefit of all.
> 
> In ClueCon news are happy to announce two training session and a birds of a
> feather (BOF) meetup for this year's event. Paid attendees of ClueCon will
> have their choice of FreeSWITCH or OpenSIPS training. The FreeSWITCH
> training will be conducted by the aforementioned Darren Schreiber. The
> OpenSIPS training will be conducted by none other than Bogdan-Andrei Iancu,
> lead developer of the OpenSIPS project. The training sessions will take
> place on Monday, August 6. On the evening of Wednesday, August 8 we will be
> having a VoIP security ("VoIPSec") meetup for those interested in
> discussing this subject with others who share an interest. Among those who
> will be present are Phil Zimmermann of PGP and
> ZRTP<http://zfone.com?cc12-0618>fame.
> 
> Be sure to register right away so that you can get your extra chances to
> win. Anyone who registers before July 4th of this year will receive 10
> entries in the great ClueCon giveaway. We have a lot of interesting items
> to give away this year, so stay tuned. And while you are at the ClueCon
> site to register <http://www.cluecon.com/register?cc12-0618>, but sure to
> review the schedule <http://www.cluecon.com/schedule?cc12-0618> as we've
> added a number of new speakers in the past week.
> 
> See you in August!
> 
> -- 
> Michael S Collins
> ClueCon Team
> http://www.cluecon.com
> 877-7-4ACLUE
> cc12-0618
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From alex at thewinelake.com  Mon Jun 18 20:55:57 2012
From: alex at thewinelake.com (Alex)
Date: Mon, 18 Jun 2012 17:55:57 +0100
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <CALNZuwmhmW7Yp1owGtLK+z8EhG=URxn7c0YpZtkyZvFU24SNMA@mail.gmail.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
	<CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
	<4FDF52FA.2070007@thewinelake.com>
	<CALNZuwmhmW7Yp1owGtLK+z8EhG=URxn7c0YpZtkyZvFU24SNMA@mail.gmail.com>
Message-ID: <4FDF5D9D.4060707@thewinelake.com>

More lua trouble - real beginner stuff, I'm afraid.

Why does this not work:

         destnum = session:playAndGetDigits(3, 20, 3, 5000, "#", 
"/home/oev/fs-pabx/prompts/dmsb12_mcm3.wav", "", "\\d+\\*")
         l = strlen(destnum)

The objection is


2012-06-18 16:52:30.415261 [ERR] mod_lua.cpp:198 
/usr/local/freeswitch/scripts/mcm2.lua:13: attempt to call global 
'strlen' (a nil value)
stack traceback:
         /usr/local/freeswitch/scripts/mcm2.lua:13: in function 
'getDestNumber'
         /usr/local/freeswitch/scripts/mcm2.lua:49: in function 
'unattendedTransfer'
         /usr/local/freeswitch/scripts/mcm2.lua:24: in function 
'processMcmOption'
         /usr/local/freeswitch/scripts/mcm2.lua:153: in main chunk

I was just assuming that lua is lua and that anything in the lua wiki 
(eg. http://www.lua.org/manual/2.4/node32.html#exstring) would work.

> Phrases let you mix and match speak, say, play-file, silence, other 
> macros, and even passing in options.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_phrase
> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
>
> -Avi
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2433/5076 - Release Date: 06/17/12
>


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From avi at avimarcus.net  Mon Jun 18 21:05:21 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 18 Jun 2012 20:05:21 +0300
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <4FDF5D9D.4060707@thewinelake.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
	<CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
	<4FDF52FA.2070007@thewinelake.com>
	<CALNZuwmhmW7Yp1owGtLK+z8EhG=URxn7c0YpZtkyZvFU24SNMA@mail.gmail.com>
	<4FDF5D9D.4060707@thewinelake.com>
Message-ID: <CALNZuw=SbKSu-kORw-5M3J-qniKBU6HVv9rCa_261Ce4hrxWeg@mail.gmail.com>

lua uses: myString:len() or tostring(myNumber):len()

-Avi


On Mon, Jun 18, 2012 at 7:55 PM, Alex <alex at thewinelake.com> wrote:

>  More lua trouble - real beginner stuff, I'm afraid.
>
> Why does this not work:
>
>         destnum = session:playAndGetDigits(3, 20, 3, 5000, "#",
> "/home/oev/fs-pabx/prompts/dmsb12_mcm3.wav", "", "\\d+\\*")
>         l = strlen(destnum)
>
> The objection is
>
>
> 2012-06-18 16:52:30.415261 [ERR] mod_lua.cpp:198
> /usr/local/freeswitch/scripts/mcm2.lua:13: attempt to call global 'strlen'
> (a nil value)
> stack traceback:
>         /usr/local/freeswitch/scripts/mcm2.lua:13: in function
> 'getDestNumber'
>         /usr/local/freeswitch/scripts/mcm2.lua:49: in function
> 'unattendedTransfer'
>         /usr/local/freeswitch/scripts/mcm2.lua:24: in function
> 'processMcmOption'
>         /usr/local/freeswitch/scripts/mcm2.lua:153: in main chunk
>
> I was just assuming that lua is lua and that anything in the lua wiki (eg.
> http://www.lua.org/manual/2.4/node32.html#exstring) would work.
>
>  Phrases let you mix and match speak, say, play-file, silence, other
> macros, and even passing in options.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_phrase
> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
>
> -Avi
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2177 / Virus Database: 2433/5076 - Release Date: 06/17/12
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From alex at thewinelake.com  Mon Jun 18 21:07:04 2012
From: alex at thewinelake.com (Alex)
Date: Mon, 18 Jun 2012 18:07:04 +0100
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <4FDF5D9D.4060707@thewinelake.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
	<CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
	<4FDF52FA.2070007@thewinelake.com>
	<CALNZuwmhmW7Yp1owGtLK+z8EhG=URxn7c0YpZtkyZvFU24SNMA@mail.gmail.com>
	<4FDF5D9D.4060707@thewinelake.com>
Message-ID: <4FDF6038.4020606@thewinelake.com>

OK, so replying to myself, I think I need to include the string 
library... I can see in lua.conf.xml that one can have a 
"module-directory" but I'm not sure where I should get these modules 
from. Any recommended quick reading on this?
> More lua trouble - real beginner stuff, I'm afraid.
>
> Why does this not work:
>
>         destnum = session:playAndGetDigits(3, 20, 3, 5000, "#", 
> "/home/oev/fs-pabx/prompts/dmsb12_mcm3.wav", "", "\\d+\\*")
>         l = strlen(destnum)
>
> The objection is
>
>
> 2012-06-18 16:52:30.415261 [ERR] mod_lua.cpp:198 
> /usr/local/freeswitch/scripts/mcm2.lua:13: attempt to call global 
> 'strlen' (a nil value)
> stack traceback:
>         /usr/local/freeswitch/scripts/mcm2.lua:13: in function 
> 'getDestNumber'
>         /usr/local/freeswitch/scripts/mcm2.lua:49: in function 
> 'unattendedTransfer'
>         /usr/local/freeswitch/scripts/mcm2.lua:24: in function 
> 'processMcmOption'
>         /usr/local/freeswitch/scripts/mcm2.lua:153: in main chunk
>
> I was just assuming that lua is lua and that anything in the lua wiki 
> (eg. http://www.lua.org/manual/2.4/node32.html#exstring) would work.
>
>> Phrases let you mix and match speak, say, play-file, silence, other 
>> macros, and even passing in options.
>>
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_phrase
>> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
>>
>> -Avi
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> No virus found in this message.
>> Checked by AVG - www.avg.com <http://www.avg.com>
>> Version: 2012.0.2177 / Virus Database: 2433/5076 - Release Date: 06/17/12
>>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2437/5077 - Release Date: 06/18/12
>


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From avi at avimarcus.net  Mon Jun 18 21:10:24 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 18 Jun 2012 20:10:24 +0300
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <4FDF6038.4020606@thewinelake.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
	<CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
	<4FDF52FA.2070007@thewinelake.com>
	<CALNZuwmhmW7Yp1owGtLK+z8EhG=URxn7c0YpZtkyZvFU24SNMA@mail.gmail.com>
	<4FDF5D9D.4060707@thewinelake.com> <4FDF6038.4020606@thewinelake.com>
Message-ID: <CALNZuwkBOwpBjuKW5kwm7tHfH5VOtGsC5Tj3pHAB8UuwjAjE3A@mail.gmail.com>

I didn't need to do any such thing... Did you get an error using
myString:len()?
-Avi


On Mon, Jun 18, 2012 at 8:07 PM, Alex <alex at thewinelake.com> wrote:

>  OK, so replying to myself, I think I need to include the string
> library... I can see in lua.conf.xml that one can have a "module-directory"
> but I'm not sure where I should get these modules from. Any recommended
> quick reading on this?
>
> More lua trouble - real beginner stuff, I'm afraid.
>
> Why does this not work:
>
>         destnum = session:playAndGetDigits(3, 20, 3, 5000, "#",
> "/home/oev/fs-pabx/prompts/dmsb12_mcm3.wav", "", "\\d+\\*")
>         l = strlen(destnum)
>
> The objection is
>
>
> 2012-06-18 16:52:30.415261 [ERR] mod_lua.cpp:198
> /usr/local/freeswitch/scripts/mcm2.lua:13: attempt to call global 'strlen'
> (a nil value)
> stack traceback:
>         /usr/local/freeswitch/scripts/mcm2.lua:13: in function
> 'getDestNumber'
>         /usr/local/freeswitch/scripts/mcm2.lua:49: in function
> 'unattendedTransfer'
>         /usr/local/freeswitch/scripts/mcm2.lua:24: in function
> 'processMcmOption'
>         /usr/local/freeswitch/scripts/mcm2.lua:153: in main chunk
>
> I was just assuming that lua is lua and that anything in the lua wiki (eg.
> http://www.lua.org/manual/2.4/node32.html#exstring) would work.
>
>  Phrases let you mix and match speak, say, play-file, silence, other
> macros, and even passing in options.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_phrase
> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
>
> -Avi
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2177 / Virus Database: 2433/5076 - Release Date: 06/17/12
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2177 / Virus Database: 2437/5077 - Release Date: 06/18/12
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Mon Jun 18 21:14:40 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 18 Jun 2012 18:14:40 +0100
Subject: [Freeswitch-users] mod_syslog
In-Reply-To: <4FDF181A.6030701@virtual-call-center.eu>
References: <4FDF181A.6030701@virtual-call-center.eu>
Message-ID: <161250C0-1AA2-4708-89AA-8EEB3212C544@gmail.com>

It appears to be a limitation in syslog-ng ose - a quick google turned this up which sounds similar to your problem:

https://lists.balabit.hu/pipermail/syslog-ng/2011-September/017212.html

Steve on iPhone




On 18 Jun 2012, at 12:59, ?d?m Kolesz?r<adam.koleszar at virtual-call-center.eu> wrote:

> Hi,
> 
> A have a little problem with the mod_syslog module. Freeswitch generates 
> multiline log messages (for example: SDP or register messages) which 
> syslog-ng OSE can't handle. The firs part of the message is sent 
> correctly but missing the facility and ident from the other part of the 
> message. Because of this the messages can not be filtered by ident (or 
> facility) on server side and goes to wrong destination.
> 
> My question is, have you ever noticed this kind of problem?
> Thank you for your help in advance.
> 
> Br,
> Adam
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Jun 18 21:29:41 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 18 Jun 2012 13:29:41 -0400
Subject: [Freeswitch-users] mod_syslog
In-Reply-To: <161250C0-1AA2-4708-89AA-8EEB3212C544@gmail.com>
References: <4FDF181A.6030701@virtual-call-center.eu>
	<161250C0-1AA2-4708-89AA-8EEB3212C544@gmail.com>
Message-ID: <1744390.8By8XOcLhx@sos>

 Syslog-ng ose handles multiline logs beginning with version 3.2. Here is an 
example:

Jun 18 13:26:02 east1 /usr/local/sbin/kamailio[20824]: INFO: <script>: 
incoming reply from udp:217.174.249.226:5062
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.85.163.5;branch=z9hG4bK5aed.4b7d8687.0
From: sip:dispatcher at localhost;tag=119eb1f393742816e68426b045d5a8e7-1177
To: <sip:217.174.249.226:5062>;tag=1FacrNHveN0ZH
Call-ID: 30f7ccd477b092dc-20828 at 64.85.163.5
CSeq: 10 OPTIONS
Contact: <sip:217.174.249.226:5062>
User-Agent: CWU SIP GW
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Length: 0


On Monday 18 June 2012 18:14:40 Steven Ayre wrote:
> It appears to be a limitation in syslog-ng ose - a quick google turned this
> up which sounds similar to your problem:
> 
> https://lists.balabit.hu/pipermail/syslog-ng/2011-September/017212.html
> 
> Steve on iPhone
> 
> On 18 Jun 2012, at 12:59, ?d?m Kolesz?r<adam.koleszar at virtual-call-
center.eu> wrote:
> > Hi,
> > 
> > A have a little problem with the mod_syslog module. Freeswitch generates
> > multiline log messages (for example: SDP or register messages) which
> > syslog-ng OSE can't handle. The firs part of the message is sent
> > correctly but missing the facility and ident from the other part of the
> > message. Because of this the messages can not be filtered by ident (or
> > facility) on server side and goes to wrong destination.
> > 
> > My question is, have you ever noticed this kind of problem?
> > Thank you for your help in advance.
> > 
> > Br,
> > Adam
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > Join Us At ClueCon - Aug 7-9, 2012
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From alex at thewinelake.com  Mon Jun 18 21:34:21 2012
From: alex at thewinelake.com (Alex Lake)
Date: Mon, 18 Jun 2012 18:34:21 +0100
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <CALNZuwkBOwpBjuKW5kwm7tHfH5VOtGsC5Tj3pHAB8UuwjAjE3A@mail.gmail.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
	<CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
	<4FDF52FA.2070007@thewinelake.com>
	<CALNZuwmhmW7Yp1owGtLK+z8EhG=URxn7c0YpZtkyZvFU24SNMA@mail.gmail.com>
	<4FDF5D9D.4060707@thewinelake.com>
	<4FDF6038.4020606@thewinelake.com>
	<CALNZuwkBOwpBjuKW5kwm7tHfH5VOtGsC5Tj3pHAB8UuwjAjE3A@mail.gmail.com>
Message-ID: <1C310345-714C-4B49-85EA-D4155B599A5D@digitalmail.com>

Dunno - haven't tried it, but will do now!



On 18 Jun 2012, at 18:10, Avi Marcus <avi at avimarcus.net> wrote:

> I didn't need to do any such thing... Did you get an error using myString:len()?
> -Avi
> 
> 
> On Mon, Jun 18, 2012 at 8:07 PM, Alex <alex at thewinelake.com> wrote:
> OK, so replying to myself, I think I need to include the string library... I can see in lua.conf.xml that one can have a "module-directory" but I'm not sure where I should get these modules from. Any recommended quick reading on this?
>> More lua trouble - real beginner stuff, I'm afraid.
>> 
>> Why does this not work:
>> 
>>         destnum = session:playAndGetDigits(3, 20, 3, 5000, "#", "/home/oev/fs-pabx/prompts/dmsb12_mcm3.wav", "", "\\d+\\*")
>>         l = strlen(destnum)
>> 
>> The objection is
>> 
>> 
>> 2012-06-18 16:52:30.415261 [ERR] mod_lua.cpp:198 /usr/local/freeswitch/scripts/mcm2.lua:13: attempt to call global 'strlen' (a nil value)
>> stack traceback:
>>         /usr/local/freeswitch/scripts/mcm2.lua:13: in function 'getDestNumber'
>>         /usr/local/freeswitch/scripts/mcm2.lua:49: in function 'unattendedTransfer'
>>         /usr/local/freeswitch/scripts/mcm2.lua:24: in function 'processMcmOption'
>>         /usr/local/freeswitch/scripts/mcm2.lua:153: in main chunk
>> 
>> I was just assuming that lua is lua and that anything in the lua wiki (eg. http://www.lua.org/manual/2.4/node32.html#exstring) would work. 
>> 
>>> Phrases let you mix and match speak, say, play-file, silence, other macros, and even passing in options.
>>> 
>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_phrase
>>> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
>>> 
>>> -Avi
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> Join Us At ClueCon - Aug 7-9, 2012
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> No virus found in this message.
>>> Checked by AVG - www.avg.com
>>> Version: 2012.0.2177 / Virus Database: 2433/5076 - Release Date: 06/17/12
>>> 
>> 
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> No virus found in this message.
>> Checked by AVG - www.avg.com
>> Version: 2012.0.2177 / Virus Database: 2437/5077 - Release Date: 06/18/12
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mariya at cs.ucsb.edu  Mon Jun 18 21:37:40 2012
From: mariya at cs.ucsb.edu (Mariya Zheleva)
Date: Mon, 18 Jun 2012 10:37:40 -0700
Subject: [Freeswitch-users] Freeswitch [DEBUG] libdingaling.c:1242 TLS
 NOT SUPPORTED IN THIS BUILD!
In-Reply-To: <CALNZuwmYOWGX9NB8Rx7JAD8XA-bMLbXRwVC=A7QSLW7_=vKNhw@mail.gmail.com>
References: <CANr3YS3PWU5B5LPyqNHUUkVDs0q=Mu_+=qFc0NJM6Vx3h-dNEA@mail.gmail.com>
	<CALNZuwmYOWGX9NB8Rx7JAD8XA-bMLbXRwVC=A7QSLW7_=vKNhw@mail.gmail.com>
Message-ID: <CANr3YS3U6e+DsrnHZruahESb9gAuTDFotLTHtKcXHnGU46LS8Q@mail.gmail.com>

Hi Avi,

Thanks a lot! I am using Ubuntu10.04 and installing
libssl-dev, re-bootstrapping, re-configuring and re-building helped solve
the problem.
Now I can do outbound calls using googlevoice.

I will update the wiki as soon as I get an account for it.

Thanks again.
mariya~

On Mon, Jun 18, 2012 at 7:53 AM, Avi Marcus <avi at avimarcus.net> wrote:

> Ah, that's outdated.
> "Since October 17,2011 this is no longer needed now OpenSSL is used."
>
> Try using the package openssl-devel (centos?) or libssl-dev
> (debian/ubuntu?) and probably re-bootstrapping / reconfig and try building
> again.
>
> Then please update the wiki.
>
> I'm surprised that one didn't get fixed...
>
>
> -Avi
>
>
>
> On Mon, Jun 18, 2012 at 12:48 AM, Mariya Zheleva <mariya at cs.ucsb.edu>wrote:
>
>> Hello,
>>
>> I am trying to configure freeswitch for outbound calls and SMS through
>> google voice using mod_dingaling.
>> I have succeeded with the SMS configuration but I can't get call to work.
>> It seems the problem is that TLS can't be successfully built
>> upon freeswitch installation. I followed the instructions here:
>> http://wiki.freeswitch.org/wiki/Mod_dingaling
>> I added the flag "-lgnutls" to gcc in relink_command in the file file
>> src/mod/endpoints/mod_dingaling/mod_dingaling.la, however I still get
>> *[DEBUG] libdingaling.c:1242 TLS NOT SUPPORTED IN THIS BUILD!*
>> in the freeswitch.log
>>
>> This is the output when I make mod_dingaling-install.
>> *
>> *
>> *openbts at ubuntu:/usr/local/src/freeswitch$ sudo make
>> mod_dingaling-install*
>> * /bin/bash /usr/local/src/freeswitch/quiet_libtool   --mode=install
>> /usr/bin/install -c   libfreeswitch.la '/usr/local/freeswitch/lib'*
>> *quiet_libtool: install: /usr/bin/install -c
>> .libs/libfreeswitch.so.1.0.0
>> /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0*
>> *quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f
>> libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 &&
>> ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; })*
>> *quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f
>> libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so && ln
>> -s libfreeswitch.so.1.0.0 libfreeswitch.so; }; })*
>> *quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.lai
>> /usr/local/freeswitch/lib/libfreeswitch.la*
>> *quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.a
>> /usr/local/freeswitch/lib/libfreeswitch.a*
>> *quiet_libtool: install: chmod 644
>> /usr/local/freeswitch/lib/libfreeswitch.a*
>> *quiet_libtool: install: ranlib /usr/local/freeswitch/lib/libfreeswitch.a
>> *
>> *quiet_libtool: finish:
>> PATH="/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin:/usr/X11R6/bin:/sbin"
>> ldconfig -n /usr/local/freeswitch/lib*
>> *----------------------------------------------------------------------*
>> *Libraries have been installed in:*
>> *   /usr/local/freeswitch/lib*
>> *
>> *
>> *If you ever happen to want to link against installed libraries*
>> *in a given directory, LIBDIR, you must either use libtool, and*
>> *specify the full pathname of the library, or use the `-LLIBDIR'*
>> *flag during linking and do at least one of the following:*
>> *   - add LIBDIR to the `LD_LIBRARY_PATH' environment variable*
>> *     during execution*
>> *   - add LIBDIR to the `LD_RUN_PATH' environment variable*
>> *     during linking*
>> *   - use the `-Wl,-rpath -Wl,LIBDIR' linker flag*
>> *   - have your system administrator add LIBDIR to `/etc/ld.so.conf'*
>> *
>> *
>> *See any operating system documentation about shared libraries for*
>> *more information, such as the ld(1) and ld.so(8) manual pages.*
>> *----------------------------------------------------------------------*
>> *
>> *
>> *making install mod_dingaling*
>>
>> Any suggestions as to how to build TLS are highly appreciated.
>>
>> Thanks,
>> mariya~
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Mariya Zheleva
Department of Computer Science
University of California
Santa Barbara, CA 93106-5110
http://cs.ucsb.edu/~mariya/
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From kris at kriskinc.com  Mon Jun 18 21:55:35 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 18 Jun 2012 13:55:35 -0400
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <15212.1340038338@ccs.covici.com>
References: <CAKzWOxX0pBC5J_TX+qe-KZN+Lr=zPtOY5ceAxu-CL4hXbsqtvg@mail.gmail.com>
	<15212.1340038338@ccs.covici.com>
Message-ID: <CAKDfjgeO1gaUsbpUjrVoAaBQHxAgvM-71+x7O4F27DZFOuPV8Q@mail.gmail.com>

I'm looking for a better way to publish it in the future.

Any suggestions?

On Mon, Jun 18, 2012 at 12:52 PM,  <covici at ccs.covici.com> wrote:
> I just wish the blog was not a google doc! ?Can't read those.
>
> Michael Collins <marketing at cluecon.com> wrote:
>
>> We hope you're having a nice Monday today.
>>
>> The past week was a busy one both for the
>> FreeSWITCH<http://www.freeswitch.org?cc12-0618>project and for
>> ClueCon <http://www.cluecon.com?cc12-0618>. Last week's community
>> conference call was a nice discussion about some of the things that will be
>> happening at ClueCon this year. We also had a visit from Diana Cionoiu of
>> the YATE <http://yate.null.ro/pmwiki?cc12-0618> project. This coming
>> Wednesday we will have Darren Schreiber, co-author of both FreeSWITCH books
>> and co-founder of the 2600hz project <http://www.2600hz.org?cc12-0618>, on
>> our call to give his world-famous "SIP 101" presentation. We definitely
>> look forward to that discussion.
>>
>> We also wanted to highlight an interesting blog
>> post<http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html?cc12-0618>by
>> our friend and community member Kristian Kielhofner. Its provocative
>> title is: "Everything you wish you didn't need to know about VoIP." Over
>> the years Kristian has collected a lot of knowledge about interoperability
>> - and lack thereof - between various VoIP devices and servers. Whether
>> you're a VoIP novice or veteran we think you will appreciate seeing this
>> knowledge written down for the benefit of all.
>>
>> In ClueCon news are happy to announce two training session and a birds of a
>> feather (BOF) meetup for this year's event. Paid attendees of ClueCon will
>> have their choice of FreeSWITCH or OpenSIPS training. The FreeSWITCH
>> training will be conducted by the aforementioned Darren Schreiber. The
>> OpenSIPS training will be conducted by none other than Bogdan-Andrei Iancu,
>> lead developer of the OpenSIPS project. The training sessions will take
>> place on Monday, August 6. On the evening of Wednesday, August 8 we will be
>> having a VoIP security ("VoIPSec") meetup for those interested in
>> discussing this subject with others who share an interest. Among those who
>> will be present are Phil Zimmermann of PGP and
>> ZRTP<http://zfone.com?cc12-0618>fame.
>>
>> Be sure to register right away so that you can get your extra chances to
>> win. Anyone who registers before July 4th of this year will receive 10
>> entries in the great ClueCon giveaway. We have a lot of interesting items
>> to give away this year, so stay tuned. And while you are at the ClueCon
>> site to register <http://www.cluecon.com/register?cc12-0618>, but sure to
>> review the schedule <http://www.cluecon.com/schedule?cc12-0618> as we've
>> added a number of new speakers in the past week.
>>
>> See you in August!
>>
>> --
>> Michael S Collins
>> ClueCon Team
>> http://www.cluecon.com
>> 877-7-4ACLUE
>> cc12-0618
>>
>> ----------------------------------------------------
>> Alternatives:
>>
>> ----------------------------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Kristian Kielhofner


From avi at avimarcus.net  Mon Jun 18 22:09:20 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 18 Jun 2012 21:09:20 +0300
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKDfjgeO1gaUsbpUjrVoAaBQHxAgvM-71+x7O4F27DZFOuPV8Q@mail.gmail.com>
References: <CAKzWOxX0pBC5J_TX+qe-KZN+Lr=zPtOY5ceAxu-CL4hXbsqtvg@mail.gmail.com>
	<15212.1340038338@ccs.covici.com>
	<CAKDfjgeO1gaUsbpUjrVoAaBQHxAgvM-71+x7O4F27DZFOuPV8Q@mail.gmail.com>
Message-ID: <CALNZuw=aXGL7DmFpOOq63gUaOjdF0fd+AsyLxWxVZusHAv5Csg@mail.gmail.com>

Github public repo, maybe. Then others can clone and make changes and you
can do a pull to merge if you approve.
-Avi


On Mon, Jun 18, 2012 at 8:55 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> I'm looking for a better way to publish it in the future.
>
> Any suggestions?
>
> On Mon, Jun 18, 2012 at 12:52 PM,  <covici at ccs.covici.com> wrote:
> > I just wish the blog was not a google doc!  Can't read those.
> >
> > Michael Collins <marketing at cluecon.com> wrote:
> >
> >> We hope you're having a nice Monday today.
> >>
> >> The past week was a busy one both for the
> >> FreeSWITCH<http://www.freeswitch.org?cc12-0618>project and for
> >> ClueCon <http://www.cluecon.com?cc12-0618>. Last week's community
> >> conference call was a nice discussion about some of the things that
> will be
> >> happening at ClueCon this year. We also had a visit from Diana Cionoiu
> of
> >> the YATE <http://yate.null.ro/pmwiki?cc12-0618> project. This coming
> >> Wednesday we will have Darren Schreiber, co-author of both FreeSWITCH
> books
> >> and co-founder of the 2600hz project <http://www.2600hz.org?cc12-0618>,
> on
> >> our call to give his world-famous "SIP 101" presentation. We definitely
> >> look forward to that discussion.
> >>
> >> We also wanted to highlight an interesting blog
> >> post<
> http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html?cc12-0618
> >by
> >> our friend and community member Kristian Kielhofner. Its provocative
> >> title is: "Everything you wish you didn't need to know about VoIP." Over
> >> the years Kristian has collected a lot of knowledge about
> interoperability
> >> - and lack thereof - between various VoIP devices and servers. Whether
> >> you're a VoIP novice or veteran we think you will appreciate seeing this
> >> knowledge written down for the benefit of all.
> >>
> >> In ClueCon news are happy to announce two training session and a birds
> of a
> >> feather (BOF) meetup for this year's event. Paid attendees of ClueCon
> will
> >> have their choice of FreeSWITCH or OpenSIPS training. The FreeSWITCH
> >> training will be conducted by the aforementioned Darren Schreiber. The
> >> OpenSIPS training will be conducted by none other than Bogdan-Andrei
> Iancu,
> >> lead developer of the OpenSIPS project. The training sessions will take
> >> place on Monday, August 6. On the evening of Wednesday, August 8 we
> will be
> >> having a VoIP security ("VoIPSec") meetup for those interested in
> >> discussing this subject with others who share an interest. Among those
> who
> >> will be present are Phil Zimmermann of PGP and
> >> ZRTP<http://zfone.com?cc12-0618>fame.
> >>
> >> Be sure to register right away so that you can get your extra chances to
> >> win. Anyone who registers before July 4th of this year will receive 10
> >> entries in the great ClueCon giveaway. We have a lot of interesting
> items
> >> to give away this year, so stay tuned. And while you are at the ClueCon
> >> site to register <http://www.cluecon.com/register?cc12-0618>, but sure
> to
> >> review the schedule <http://www.cluecon.com/schedule?cc12-0618> as
> we've
> >> added a number of new speakers in the past week.
> >>
> >> See you in August!
> >>
> >> --
> >> Michael S Collins
> >> ClueCon Team
> >> http://www.cluecon.com
> >> 877-7-4ACLUE
> >> cc12-0618
> >>
> >> ----------------------------------------------------
> >> Alternatives:
> >>
> >> ----------------------------------------------------
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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 life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> --
> Kristian Kielhofner
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From awais-nazeer at hotmail.com  Mon Jun 18 22:15:07 2012
From: awais-nazeer at hotmail.com (awais nazir)
Date: Mon, 18 Jun 2012 23:15:07 +0500
Subject: [Freeswitch-users] Actions before CDR writing (b leg only)
Message-ID: <BLU162-W235003A2E9EA0381B1212288F80@phx.gbl>


Hello Guys

I am using following scenario with series of attempts on gateways on failover

<action application="set" data="hangup_after_bridge=true"/>
    <action application="bridge" data="sofia/local_profile/1111 at example1.company.com" />
    <action application="bridge" data="sofia/local_profile/1111 at example2.company.com" />

I want to do some actions like setting some variables before CDR (leg B) is written like


<action application="set" data="hangup_after_bridge=true"/>
    <action application="bridge" data="sofia/local_profile/1111 at example1.company.com" />
<!-- do some actions-->
<!-- write CDR-->
    <action application="bridge" data="sofia/local_profile/1111 at example2.company.com" /><!-- do some actions-->

<!-- write CDR-->
.
.
.
.

Can anybody help me here.

Waisee----
 		 	   		  
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From megahohol at gmail.com  Mon Jun 18 22:21:38 2012
From: megahohol at gmail.com (Grygoriy Dobrovolskyy)
Date: Mon, 18 Jun 2012 20:21:38 +0200
Subject: [Freeswitch-users] HA setup completed but nothing in the database.
Message-ID: <CAAfngiHGi=oYeTLRP4OFWRV7R8T8=Le1RphmD=iarG1uN6UGOA@mail.gmail.com>

Hello list, i've completed HA setup with two FS boxes, connected them to
the ODBC(mysql) and replicated sip_recovery table between servers, the
problem i face is simple, i have nothing in there when i have calls.

External profile has this option set:
  <param name="track-calls" value="true"/>
  <param name="odbc-dsn" value="fsw-cnf:fs-usr:2>sdfjeslkjfd4<gH(^s"/>


switch.conf also the required options:
    <param name="core-db-dsn" value="fsw-cnf:fs-usr:sdfjeslkjfd4<gH(^s"/>



FS connects fine and have no problems with that, calls, channels are in the
database, so i am thinking that FS do not "track calls" because of my
dialplan which is the following:


    <condition field="${sip_to_user}" expression="(1234567789).*$"
break="never">
      <param name="track-calls" value="true"/>
      <action inline="true" application="eval" data="URI=${sip_to_user}"/>
      <action application="set" data="domain_name=$${domain}"/>
      <action application="bridge"
data="{absolute_codec_string=PCMA}sofia/internal/1234569787 at 83.164.189.13"/>
    </condition>

So what i do is simple, match the call, and bridge to external URI. I even
added <param name="track-calls" value="true"/> to test just before bridge.

Dear all please help me to figure this out, how to make FS track this calls
?
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From drk at drkngs.net  Mon Jun 18 22:30:18 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Mon, 18 Jun 2012 11:30:18 -0700
Subject: [Freeswitch-users] Conference Caller Announce
In-Reply-To: <9B08C5DC53DDB7468007DE57C9FAF49D8ED7E9@bbcxues30.national.core.bbc.co.uk>
Message-ID: <20120618183018.dac684c9@mail.tritonwest.net>

This is not about this exact problem, but you all do know that rather then using the scheduler, you ucan override the enter sound on a per call basis by seting the variable "conference_enter_sound".  
   
--Dave
      _____  

  From: Peter Stevens [mailto:Peter.Stevens at bbc.co.uk]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Mon, 18 Jun 2012 06:38:39 -0700
Subject: Re: [Freeswitch-users] Conference Caller Announce

  
  
Hi Brian,  
   
I was able to replicate your conference not found problem (purely by accident) and after a bit of investigation I think that I have probably found an answer to your problem.  

In my case the last 4 digits of the extension number were 3000 (<condition field="destination_number" expression="^55(3\d\d\d)$">)
The last 4 digits become the value of $1 in this part of the res line ($1-${domain}) and also in the transfer line. So when the transfer occurs, the dialplan gets searched again but this time for a conference extension of 3000, which just happens to be one of the default conference numbers in the default.xml file (first 2 lines only):  
<extension name="nb_conferences">
 <condition field="destination_number" expression="^(30\d{2})$">  
   
and so it finds this extension (which I have in my dialplan) when the dialplan gets searched again.  
   
I checked that this was the case, by changing the last 4 digits of my extension number to 3100 - which is the number for the wb_conferences in the same default.xml (which I have in my dialplan) and sure enough, it switches to that.  
One more test confirmed this.  
I don't have a conference number of 32xx in my dialplan. But by dialling 553200, the number gets picked up, it says the conference is being transferred, but is not found, so it bombs out. I think that if there is no  
extension number that matches $1 in your dial plan, then it will fail to transfer  
   
This might be worth trying: ensure that you have a conference number that matches your value of $1 in the destination number for your extension, but see if it works for you.
  
Maybe pop this INFO line into your dialplan, just to check the value of $1:  

     <action application="log" data="INFO the dollar 1 value is: $1"/>  
   
Not sure if there is a mechanism to reset the value of $1 (destination_number) after the extension has been picked up, but this time to point to the conference extension number required if different from the original destination number?  
   
Let us know if this works and also if your audio gets played. My audio is still not playing as yet....  
   
Peter  

    _____  

  From: Brian Meaney [mailto:brian.meaney at BSB.IE]
Sent: Sun 17/06/2012 17:32
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference Caller Announce

  
  
Hi Peter and thank you for your input. Glad to hear somebody else is searching for answers to this. Hopefully we can help one another with resolving the issue!  

  
I have already defined the conference profile within the conference XML file, so I assume that's all that's required to put the caller into the conference.  

  
If I discover anything I'll be sure to update this thread.

On 17 Jun 2012, at 17:16, "Peter Stevens" <Peter.Stevens at bbc.co.uk> wrote:

  
    
  
  
Hi Brian,  
   
I've also been looking at this particular example this week and also posted a question! See [Freeswitch-users] Problem passing and playing audio files into a conference.  
   
Mine will transfer into the default conference (using conference $1-${domain} in the res line below, but not play the files parsed into the res line.  
   
My guess with yours is that you are specifying conference$1 at conference_2 in the res line. I think that it expects to find a conference_2 profile in the conference.conf.xml file, so if you don't have that it might well bomb out.  
  
The other thing that you may also find is that with your existing res line in the debug log you may also be getting [res]=[UNDEF]
  
The only way I could get it not to produce this was to add a left brace after sched_api and a right brace after the last audio file to be parsed in the res line, as here:  
   
<action application="set" data="res=${sched_api (+2 none conference $1-${domain} play file_string://${sound_prefix}/conference/8000/conf-welcome.wav!${sound_prefix}/conference/8000/conf-has_joined.wav)}"/>
  
I hope that this helps you and maybe you and/or someone else might be able to help me with the audio files being played once the conference starts.  
   
Peter  
   

    _____  

  From: Brian Meaney [mailto:brian.meaney at BSB.IE]
Sent: Sat 16/06/2012 11:21
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference Caller Announce

  
  
  
Thanks MC,  
   
 Here it is:-  
   
<extension name="main_number_11" continue="true">  
      <condition field="destination_number" expression="^(xxxxxxxxxxx)$">  
        <action application="export" data="hold_music=silence"/>  
        <action application="set" data="namefile=/tmp/${uuid}-name.wav" inline="true"/>  
        <action application="answer"/>  
        <action application="sleep" data="1000"/>  
        <action application="playback" data="conference/conf-welcome.wav"/>  
        <action application="sleep" data="1000"/>  
        <action application="playback" data="voicemail/vm-record_name1.wav"/>  
        <action application="playback" data="tone_stream://%(1000,0,500)"/>  
        <action application="record" data="${namefile} 1"/>  
        <action application="playback" data="ivr/ivr-call_being_transferred.wav"/>  
        <action application="set" data="res=${sched_api +1 none conference $1 at conference_2 play file_string://${namefile}!conference/conf-has_joined.wav}"/>  
        <action application="transfer" data="$1 XML default"/>  
      </condition>  
    </extension>  
   
  
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 15 June 2012 23:00
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference Caller Announce  
   

Show us the dialplan code where you put the caller into the conference.
-MC  
  
On Fri, Jun 15, 2012 at 11:21 AM, Brian Meaney <brian.meaney at bsb.ie> wrote:  
  
  
Hi and thanks for your response. I altered the expression to include parentheses, and the number is successfully picked up. I also attempted to simplify the ?res? expression by using $1-{domain}
However, it still wont connect to the conference. Sophia output is now:-  
   
2012-06-15 18:12:53.303084 [DEBUG] mod_commands.c:3092 Command conference(35319017881-192.168.3.178 play file_string:///tmp/799d7748-e75b-425c-bccd-1e6632eacc72-name.wav!conference/conf-has_joined.wav):  
Conference 35319017881-192.168.3.178 not found  
   
Its clear my issue is now with how the ?res? statement should be formed. I now have:-  
   
<action application="set" data="res=${sched_api +1 none conference $1 at conference_2 play file_string://${namefile}!conference/conf-has_joined.wav}"/>  
   
How should I construct the statement such that it connects the caller to the conference named conference_2 ? Should I even be referencing the conference name, seeing as I already have the expression matching in place earlier in the definition?  
   
   
   
   
   
  
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 15 June 2012 18:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference Caller Announce  
   

As best I can tell it looks like this line is the problem:
<condition field="destination_number" expression="^xxxxxxxxxxx$">

You need to wrap the capture value in parens like this:
<condition field="destination_number" expression="^(xxxxxxxxxxx)$">

That should make $1 have something useful in it.

-MC  
  
On Fri, Jun 15, 2012 at 10:08 AM, Bobthebeat <brian.meaney at bsb.ie> wrote:  

Hi Folks,

 brand new Freeswitch user here.

I have modified the example shown on the mod_conference
(http://wiki.freeswitch.org/wiki/Mod_conference#FAQ) wiki for my own
deployment.

I am attempting to have a user record his name, then have this recording
announced on joining the conference, but I am struggling. Here is my
dialplan entry:-


<extension name="main_number_11" continue="true">
     <condition field="destination_number" expression="^xxxxxxxxxxx$">
       <action application="export" data="hold_music=silence"/>
       <action application="set" data="namefile=/tmp/${uuid}-name.wav"
inline="true"/>
       <action application="answer"/>
       <action application="sleep" data="1000"/>
       <action application="playback" data="conference/conf-welcome.wav"/>
       <action application="sleep" data="1000"/>
       <action application="playback"
data="voicemail/vm-record_name1.wav"/>
       <action application="playback" data="tone_stream://%(1000,0,500)"/>
       <action application="record" data="${namefile} 1"/>
       <action application="playback"
data="ivr/ivr-call_being_transferred.wav"/>
       <action application="set" data="res=${sched_api +1 none conference
$1-${domain_name}@conference_2 play
file_string://${namefile}!conference/conf-has_joined.wav}"/>
       <action application="transfer" data="$1 XML default"/>
     </condition>
   </extension>

The call connects and I am prompted to record, then I am notified of the
imminent transfer, but it bombs out here. Sophia output reads

Command conference($1-192.168.3.178 at conference_2 play
file_string:///tmp/b34c9679-9fdb-4934-a1d9-dd4b646c88e2-name.wav!conference/conf-has_joined.wav):
Conference $1-192.168.3.178 at conference_2 not found

I can dial into the conference without issues when I disregard the 'Set'
expression and just dial the conference directly.

Can anybody point me in the right direction?
Thank you very much,
BP Please consider the environment before printing this email   
 
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This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated.
If you have received it in error, please delete it from your system.
Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately.
Please note that the BBC monitors e-mails sent or received.
Further communication will signify your consent to this.     
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
P Please consider the environment before printing this email   
 
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This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated.
If you have received it in error, please delete it from your system.
Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately.
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From jeremyc at ssimicro.com  Mon Jun 18 22:50:29 2012
From: jeremyc at ssimicro.com (Jeremy Childs)
Date: Mon, 18 Jun 2012 12:50:29 -0600
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKDfjgeO1gaUsbpUjrVoAaBQHxAgvM-71+x7O4F27DZFOuPV8Q@mail.gmail.com>
References: <CAKzWOxX0pBC5J_TX+qe-KZN+Lr=zPtOY5ceAxu-CL4hXbsqtvg@mail.gmail.com>
	<15212.1340038338@ccs.covici.com>
	<CAKDfjgeO1gaUsbpUjrVoAaBQHxAgvM-71+x7O4F27DZFOuPV8Q@mail.gmail.com>
Message-ID: <4FDF7875.2070007@ssimicro.com>

Kristian, the document as it stands is amazingly good already - please 
continue authoring, regardless of where you publish it.

On 12-06-18 11:55 AM, Kristian Kielhofner wrote:
> I'm looking for a better way to publish it in the future.
>
> Any suggestions?
>
> On Mon, Jun 18, 2012 at 12:52 PM,<covici at ccs.covici.com>  wrote:
>> I just wish the blog was not a google doc!  Can't read those.
>>
>> Michael Collins<marketing at cluecon.com>  wrote:
>>
>>> We hope you're having a nice Monday today.
>>>
>>> The past week was a busy one both for the
>>> FreeSWITCH<http://www.freeswitch.org?cc12-0618>project and for
>>> ClueCon<http://www.cluecon.com?cc12-0618>. Last week's community
>>> conference call was a nice discussion about some of the things that will be
>>> happening at ClueCon this year. We also had a visit from Diana Cionoiu of
>>> the YATE<http://yate.null.ro/pmwiki?cc12-0618>  project. This coming
>>> Wednesday we will have Darren Schreiber, co-author of both FreeSWITCH books
>>> and co-founder of the 2600hz project<http://www.2600hz.org?cc12-0618>, on
>>> our call to give his world-famous "SIP 101" presentation. We definitely
>>> look forward to that discussion.
>>>
>>> We also wanted to highlight an interesting blog
>>> post<http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html?cc12-0618>by
>>> our friend and community member Kristian Kielhofner. Its provocative
>>> title is: "Everything you wish you didn't need to know about VoIP." Over
>>> the years Kristian has collected a lot of knowledge about interoperability
>>> - and lack thereof - between various VoIP devices and servers. Whether
>>> you're a VoIP novice or veteran we think you will appreciate seeing this
>>> knowledge written down for the benefit of all.
>>>
>>> In ClueCon news are happy to announce two training session and a birds of a
>>> feather (BOF) meetup for this year's event. Paid attendees of ClueCon will
>>> have their choice of FreeSWITCH or OpenSIPS training. The FreeSWITCH
>>> training will be conducted by the aforementioned Darren Schreiber. The
>>> OpenSIPS training will be conducted by none other than Bogdan-Andrei Iancu,
>>> lead developer of the OpenSIPS project. The training sessions will take
>>> place on Monday, August 6. On the evening of Wednesday, August 8 we will be
>>> having a VoIP security ("VoIPSec") meetup for those interested in
>>> discussing this subject with others who share an interest. Among those who
>>> will be present are Phil Zimmermann of PGP and
>>> ZRTP<http://zfone.com?cc12-0618>fame.
>>>
>>> Be sure to register right away so that you can get your extra chances to
>>> win. Anyone who registers before July 4th of this year will receive 10
>>> entries in the great ClueCon giveaway. We have a lot of interesting items
>>> to give away this year, so stay tuned. And while you are at the ClueCon
>>> site to register<http://www.cluecon.com/register?cc12-0618>, but sure to
>>> review the schedule<http://www.cluecon.com/schedule?cc12-0618>  as we've
>>> added a number of new speakers in the past week.
>>>
>>> See you in August!
>>>
>>> --
>>> Michael S Collins
>>> ClueCon Team
>>> http://www.cluecon.com
>>> 877-7-4ACLUE
>>> cc12-0618
>>>
>>> ----------------------------------------------------
>>> Alternatives:
>>>
>>> ----------------------------------------------------
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 life is like a penny.  You're going to lose it.  The question is:
>> How do
>> you spend it?
>>
>>          John Covici
>>          covici at ccs.covici.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>



From nestor at tiendalinux.com  Tue Jun 19 00:03:44 2012
From: nestor at tiendalinux.com (Nestor A. Diaz)
Date: Mon, 18 Jun 2012 15:03:44 -0500
Subject: [Freeswitch-users] Help with video call ( h.264 codec )
Message-ID: <bbda0e29f3b1d099674e11dec9e62f9c@asterix.tiendalinux.com>

 

Hi guys, i am configuring a video call between two grandstream video
phones (GXV-3000) which i don't recommend for being a crappy set of
devices.

 I am using the latest freeswitch FreeSWITCH version
1.2.0rc2+git~20120618T111915Z~11a745cf76+unclean~20120618T121534Z .


But no video call between those devices. 

However video call between
a Grandstream and an Android Bria Phone works ( i don't recomment bria
too, but it helps me with some tests).

 Here is the log of two
grandstream that fails:

 http://pastebin.com/tXZ4DhKM [1] 

it keeps
stuck on the line that says: 

2012-06-18 11:58:35.144424 [DEBUG]
sofia.c:5838 Channel sofia/internal/215 entering state [calling][0]


and then continues. 

And here is the log of a grandstream and a bria
that works: 

http://pastebin.com/nRzSr7WR [2] 

The call is started
from the Bria, starting the call from Grandstream will not work. 

If i
configure the Grandstream to not send video they can place calls and it
will work for audio, if i configure them to start video automatically
they will not work, in fact they can't place any call but they can
receive calls from other phones. 

But if i put: 

 <param
name="inbound-bypass-media" value="true"/>

at the sip profile they are
attached to they will work, however i don't like this solution, cause i
want to deploy them behind a firewall at different places.

Tried
before:

 <param name="inbound-proxy-media" value="true"/>

but it
didn't work, the only way to have this working is setting
inbound_bypass_media.

Any help on this issue ?

Thanks.

-- 
Nestor A.
Diaz
Ingeniero de Sistemas
Tel. +57.1-485-3020 x 211
Cel.
+57.316-227-3595
nestor at tiendalinux.com
www.tiendalinux.com



Links:
------
[1] http://pastebin.com/tXZ4DhKM
[2]
http://pastebin.com/nRzSr7WR
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From terry at digital-outpost.com  Mon Jun 18 23:50:03 2012
From: terry at digital-outpost.com (Terry Barnum)
Date: Mon, 18 Jun 2012 12:50:03 -0700
Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
Message-ID: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>

I recently started investigating FS and am curious about what the specific problem is with Macports and FS. I've been using Macports for a couple years and find it to be a great, actively maintained package manager that uses /opt/local, making it a very reliable way to keep production software working through Apple system updates. I've found the Macports team to be very responsive so if there are specific things that I can pass along, it might get fixed. Is there a link or thread that documents what's incompatible with FS?

Thanks,
-Terry



From mgg at giagnocavo.net  Tue Jun 19 00:32:48 2012
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Mon, 18 Jun 2012 20:32:48 +0000
Subject: [Freeswitch-users] Blog post about SIP
In-Reply-To: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
References: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B6BACD40@BY2PRD0710MB390.namprd07.prod.outlook.com>

Excellent work. It's great to see a practical guide to the complicated (and sometimes downright moronic) mess that encumbers VoIP.

Some notes:

- From viewing a medium sized production network, I see about 1% of the SIP packets received and processed are fragmented (MTU of 576 bytes somewhere along the way). Are you seeing implementations actually failing due to IP fragmentation of UDP packets? I tested sending 2KB length UDP messages to my proxies over the Internet, and it seems to work. Certainly sending frag'd SIP messages is more likely to work than the crazy idea of protocol transition, right? Who's really experiencing MTU issues?

- Watch out with ICE and VPNs. While communicating to a server on a public IP that's also available internally via VPN, an aggressive anti-NAT solution like ICE might determine there's a "direct" path via the private VPN IPs. If the VPN is not a good path (say, its implemented over SSL), connectivity will happen over that path, and audio quality suffers. Microsoft Lync does this (Google lots of complaints related to using Lync with a VPN), and the only workaround is to configure firewalls to block such traffic.
	- Subnote: I wonder how well things would work if SIP clients implemented UPnP, which is supported by tons of home routers?

- Re-INVITE messages *should* be distinguished by a to tag. But, there are many networks out there using software written by folks that didn't bother to read the SIP spec, and they will happily ignore an existing to tag and treat it like a new request. This has implications for the standard OpenSIPS config, for instance, which allows relaying of INVITEs if there's a to-tag.

-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner
Sent: Thursday, June 14, 2012 2:07 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Blog post about SIP

FreeSWITCHers,

  As some of you may have noticed I frequently comment on SIP-related FreeSWITCH posts.  Long story short I decided to finally put some of these thoughts and experiences down in writing.  What I ended up with is a 21 page (and counting) document describing various SIP header fields, SDP information, RTP issues, DTMF issues, NAT traversal technologies, interop headaches, etc.

  I wrote a little intro and linked to the document if any of you would like to check it out (link at the end of post):

http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html

  Of course I'd appreciate any feedback you may have.  What do you disagree with?  Find a typo?  Have burning SIP/RTP/SDP/T.38/codec questions you'd like to see me address?  Should I get into more implementation (FreeSWITCH) specific issues?  Let me know.



Thanks!

--
Kristian Kielhofner

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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




From kris at kriskinc.com  Tue Jun 19 00:41:34 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 18 Jun 2012 16:41:34 -0400
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <4FDF7875.2070007@ssimicro.com>
References: <CAKzWOxX0pBC5J_TX+qe-KZN+Lr=zPtOY5ceAxu-CL4hXbsqtvg@mail.gmail.com>
	<15212.1340038338@ccs.covici.com>
	<CAKDfjgeO1gaUsbpUjrVoAaBQHxAgvM-71+x7O4F27DZFOuPV8Q@mail.gmail.com>
	<4FDF7875.2070007@ssimicro.com>
Message-ID: <CAKDfjgc9hkvAdD2CDB5oNak5HmS-3=waPc8Ytsf1uD_wffvJKA@mail.gmail.com>

Thanks!  I'm looking forward to updating the document in the near
future with some of the feedback I've received so far.

On Mon, Jun 18, 2012 at 2:50 PM, Jeremy Childs <jeremyc at ssimicro.com> wrote:
> Kristian, the document as it stands is amazingly good already - please
> continue authoring, regardless of where you publish it.
>

-- 
Kristian Kielhofner


From kris at kriskinc.com  Tue Jun 19 00:44:50 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 18 Jun 2012 16:44:50 -0400
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <CALNZuw=aXGL7DmFpOOq63gUaOjdF0fd+AsyLxWxVZusHAv5Csg@mail.gmail.com>
References: <CAKzWOxX0pBC5J_TX+qe-KZN+Lr=zPtOY5ceAxu-CL4hXbsqtvg@mail.gmail.com>
	<15212.1340038338@ccs.covici.com>
	<CAKDfjgeO1gaUsbpUjrVoAaBQHxAgvM-71+x7O4F27DZFOuPV8Q@mail.gmail.com>
	<CALNZuw=aXGL7DmFpOOq63gUaOjdF0fd+AsyLxWxVZusHAv5Csg@mail.gmail.com>
Message-ID: <CAKDfjgf4O90Y6-JXk06dFpY-yMLpmhteRL9iEVW9C-PTosR7Vw@mail.gmail.com>

I'm not opposed to this, necessarily, but my first priority is making
updates as easy as possible.

I'm busy and lazy.  If updating the document means doing a git pull,
running some scripts, editing HTML/Latex/whatever, checking in, etc,
etc I'm not going to update the document as much.

If it's as simple as opening a new tab and clicking a bookmark I'm
much more likely to keep the document up to date ;).

On Mon, Jun 18, 2012 at 2:09 PM, Avi Marcus <avi at avimarcus.net> wrote:
> Github public repo, maybe. Then others can clone and make changes and you
> can do a pull to merge if you approve.
> -Avi
>

-- 
Kristian Kielhofner


From abaci64 at gmail.com  Tue Jun 19 00:50:11 2012
From: abaci64 at gmail.com (Abaci)
Date: Mon, 18 Jun 2012 16:50:11 -0400
Subject: [Freeswitch-users] get info from DB into channel variable in xml
	dialplan
Message-ID: <4FDF9483.7040701@gmail.com>

We are looking for a way to set a few variables in the XML dialplan with 
information stored in a MySQL DB, the simplest seems to be to use 
mod_odbc_query which will set channel variables (see 
http://wiki.freeswitch.org/wiki/Mod_odbc_query#APP), problem is that we 
prefer not to access the DB directly from xml.
The other option is to use mod_curl which gets the result into a channel 
variable named 'curl_response_data', problem is that we have multiple 
values that we need in separate channel variables, and I can't find any 
documentation on how to do that, is there any way to do this using json etc?



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From avi at avimarcus.net  Tue Jun 19 00:51:40 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 18 Jun 2012 23:51:40 +0300
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKDfjgf4O90Y6-JXk06dFpY-yMLpmhteRL9iEVW9C-PTosR7Vw@mail.gmail.com>
References: <CAKzWOxX0pBC5J_TX+qe-KZN+Lr=zPtOY5ceAxu-CL4hXbsqtvg@mail.gmail.com>
	<15212.1340038338@ccs.covici.com>
	<CAKDfjgeO1gaUsbpUjrVoAaBQHxAgvM-71+x7O4F27DZFOuPV8Q@mail.gmail.com>
	<CALNZuw=aXGL7DmFpOOq63gUaOjdF0fd+AsyLxWxVZusHAv5Csg@mail.gmail.com>
	<CAKDfjgf4O90Y6-JXk06dFpY-yMLpmhteRL9iEVW9C-PTosR7Vw@mail.gmail.com>
Message-ID: <CALNZuwmF0Z3bpK0RGr0eSdTRx_pBsHjOXGm2oV7um3XHjpC9Fw@mail.gmail.com>

On your own github repo's there's an "edit this file" link and it gives you
an editor in the browser.

I was mostly referring that others could submit "patches" that you could
approve easily.

-Avi


On Mon, Jun 18, 2012 at 11:44 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> I'm not opposed to this, necessarily, but my first priority is making
> updates as easy as possible.
>
> I'm busy and lazy.  If updating the document means doing a git pull,
> running some scripts, editing HTML/Latex/whatever, checking in, etc,
> etc I'm not going to update the document as much.
>
> If it's as simple as opening a new tab and clicking a bookmark I'm
> much more likely to keep the document up to date ;).
>
> On Mon, Jun 18, 2012 at 2:09 PM, Avi Marcus <avi at avimarcus.net> wrote:
> > Github public repo, maybe. Then others can clone and make changes and you
> > can do a pull to merge if you approve.
> > -Avi
> >
>
> --
> Kristian Kielhofner
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kris at kriskinc.com  Tue Jun 19 00:59:04 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 18 Jun 2012 16:59:04 -0400
Subject: [Freeswitch-users] Blog post about SIP
In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B6BACD40@BY2PRD0710MB390.namprd07.prod.outlook.com>
References: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
	<63B00DD1DA6A364E9F64A3A0BD2FE7B6BACD40@BY2PRD0710MB390.namprd07.prod.outlook.com>
Message-ID: <CAKDfjge7n9f6ojkBgJbA4L_-65_QGdWSGwwaC-GLaNAngcURGw@mail.gmail.com>

Thank you for your comments!

While this is off-topic for FreeSWITCH I'm not sure where else this
conversation could take place.  Apologies in advance to the list
admins - if you want me to take it elsewhere just say so :).

You could probably tell I wasn't exactly sure how to address SIP/UDP
and fragmentation issues.  More than anything it came out of the
various efforts deployed by some implementations to reduce SIP packet
size (omitting rtpmap lines, specifically).  I wanted to explain why
some implementations (FreeSWITCH being one) make attempts to minimize
SIP packet size, even at the expense of overall interoperability
(depending on how you want to look at it).  One could certainly argue
that the UDP -> TCP transition specified in RFC 3261 would lead to
more interop problems than "missing" rtpmap lines.

To answer the question I don't see a lot of UDP fragmentation issues
but then again the majority of my production UDP SIP traffic has
pretty small packet sizes.

Your comment on actively blocking undesired ICE candidate paths with a
firewall is well taken.  I'll add it to the doc!

Re-INVITEs.  Yep.  Remember one of the first rules of the document -
someone, somewhere has broken just about anything and everything.

On Mon, Jun 18, 2012 at 4:32 PM, Michael Giagnocavo <mgg at giagnocavo.net> wrote:
> Excellent work. It's great to see a practical guide to the complicated (and sometimes downright moronic) mess that encumbers VoIP.
>
> Some notes:
>
> - From viewing a medium sized production network, I see about 1% of the SIP packets received and processed are fragmented (MTU of 576 bytes somewhere along the way). Are you seeing implementations actually failing due to IP fragmentation of UDP packets? I tested sending 2KB length UDP messages to my proxies over the Internet, and it seems to work. Certainly sending frag'd SIP messages is more likely to work than the crazy idea of protocol transition, right? Who's really experiencing MTU issues?
>
> - Watch out with ICE and VPNs. While communicating to a server on a public IP that's also available internally via VPN, an aggressive anti-NAT solution like ICE might determine there's a "direct" path via the private VPN IPs. If the VPN is not a good path (say, its implemented over SSL), connectivity will happen over that path, and audio quality suffers. Microsoft Lync does this (Google lots of complaints related to using Lync with a VPN), and the only workaround is to configure firewalls to block such traffic.
> ? ? ? ?- Subnote: I wonder how well things would work if SIP clients implemented UPnP, which is supported by tons of home routers?
>
> - Re-INVITE messages *should* be distinguished by a to tag. But, there are many networks out there using software written by folks that didn't bother to read the SIP spec, and they will happily ignore an existing to tag and treat it like a new request. This has implications for the standard OpenSIPS config, for instance, which allows relaying of INVITEs if there's a to-tag.
>
> -Michael
>

-- 
Kristian Kielhofner


From avi at avimarcus.net  Tue Jun 19 01:00:53 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 19 Jun 2012 00:00:53 +0300
Subject: [Freeswitch-users] get info from DB into channel variable in
	xml dialplan
In-Reply-To: <4FDF9483.7040701@gmail.com>
References: <4FDF9483.7040701@gmail.com>
Message-ID: <CALNZuwm=72wpjkBaDqQt821vKX5y9MTP27y41rz=jJZgpGe2fw@mail.gmail.com>

lua is a lightweight option that can do sql or curl.

-Avi


On Mon, Jun 18, 2012 at 11:50 PM, Abaci <abaci64 at gmail.com> wrote:

>  We are looking for a way to set a few variables in the XML dialplan with
> information stored in a MySQL DB, the simplest seems to be to use
> mod_odbc_query which will set channel variables (see
> http://wiki.freeswitch.org/wiki/Mod_odbc_query#APP), problem is that we
> prefer not to access the DB directly from xml.
> The other option is to use mod_curl which gets the result into a channel
> variable named 'curl_response_data', problem is that we have multiple
> values that we need in separate channel variables, and I can't find any
> documentation on how to do that, is there any way to do this using json etc?
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kris at kriskinc.com  Tue Jun 19 01:01:47 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 18 Jun 2012 17:01:47 -0400
Subject: [Freeswitch-users] Blog post about SIP
In-Reply-To: <4FDDB4F2.5030707@coppice.org>
References: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
	<137f12cd230.5442606051204082350.2378772053723064798@zoho.com>
	<CALNZuwkZBvc=f0=ud8ZPXD56422vEXgP5LsMdqH_+pd0Rfb6Rg@mail.gmail.com>
	<4FDDB4F2.5030707@coppice.org>
Message-ID: <CAKDfjgdn91+9v7rcZj23A1nTqR-bO_jhN=SC=1DVyELLFiB-NA@mail.gmail.com>

On Sun, Jun 17, 2012 at 6:44 AM, Steve Underwood <steveu at coppice.org> wrote:
> I thought the only Cisco products which did this were the ones which
> started as Linksys/Sipura products. The usual G.729 problem with real
> Cisco products is G.729 with the bits packed in the opposite order to
> the RFC. This isn't an error. Cisco implemented G.729 in RTP before
> there was a standard, and the packing order they chose just happened to
> be the opposite of the standard. Of course, modern Cisco kit implements
> the standard order, but they can still do the opposite order for
> backwards compatibility with their own kit. Sometimes you still meet
> systems configured to use that, and G.729 calls produce a horrible noise.

You are generally correct, however, some of the rtmpap G729a craziness
seems to have moved to "Cisco proper".  I've seen IOS-powered media
gateways and CallManager express systems put G729a in the rtpmap. With
how much that hardware interacts with the current line of rebadged
Sipura/Linksys phones I can't say I'm completely surprised.

-- 
Kristian Kielhofner


From abaci64 at gmail.com  Tue Jun 19 01:08:42 2012
From: abaci64 at gmail.com (Abaci)
Date: Mon, 18 Jun 2012 17:08:42 -0400
Subject: [Freeswitch-users] get info from DB into channel variable in
 xml dialplan
In-Reply-To: <CALNZuwm=72wpjkBaDqQt821vKX5y9MTP27y41rz=jJZgpGe2fw@mail.gmail.com>
References: <4FDF9483.7040701@gmail.com>
	<CALNZuwm=72wpjkBaDqQt821vKX5y9MTP27y41rz=jJZgpGe2fw@mail.gmail.com>
Message-ID: <4FDF98DA.8050207@gmail.com>

I was wondering if there is a way to do it using curl without lua

On 6/18/2012 5:00 PM, Avi Marcus wrote:
> lua is a lightweight option that can do sql or curl.
>
> -Avi
>
>
> On Mon, Jun 18, 2012 at 11:50 PM, Abaci <abaci64 at gmail.com 
> <mailto:abaci64 at gmail.com>> wrote:
>
>     We are looking for a way to set a few variables in the XML
>     dialplan with information stored in a MySQL DB, the simplest seems
>     to be to use mod_odbc_query which will set channel variables (see
>     http://wiki.freeswitch.org/wiki/Mod_odbc_query#APP), problem is
>     that we prefer not to access the DB directly from xml.
>     The other option is to use mod_curl which gets the result into a
>     channel variable named 'curl_response_data', problem is that we
>     have multiple values that we need in separate channel variables,
>     and I can't find any documentation on how to do that, is there any
>     way to do this using json etc?
>
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From getj2k at yahoo.com  Tue Jun 19 00:48:34 2012
From: getj2k at yahoo.com (mtle)
Date: Mon, 18 Jun 2012 13:48:34 -0700 (PDT)
Subject: [Freeswitch-users] Gtalk does not ring local phones when making
	outbound calls
Message-ID: <1340052514.70031.YahooMailClassic@web161501.mail.bf1.yahoo.com>

Hi,I have successfully compiled and installed FS from the latest git as at today. ?Everything seems OK. ?The only issue is that I do not get any rings at all with gtalk. I wonder if anyone have had this problem lately and how to fix that? Here's my dialplan.
<extension name="gv_out">? ? <condition field="destination_number" expression="^1?(\d{10})$">? ? ? ? <action application="set" data="hangup_after_bridge=true"/>? ? ? ? <action application="ring_ready"/>? ? ? ?<!--<action application="set" data="bridge_generate_comfort_noise=true"/>-->? ? ? ?<!--<action application="set" data="execute_on_answer=start_dtmf_generate"/>-->? ? ? ? <action application="bridge" data="dingaling/gtalk/+1$1 at voice.google.com"/>? ? </condition></extension>
Thanks.
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From msc at freeswitch.org  Tue Jun 19 01:13:39 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jun 2012 14:13:39 -0700
Subject: [Freeswitch-users] Blog post about SIP
In-Reply-To: <CAKDfjge7n9f6ojkBgJbA4L_-65_QGdWSGwwaC-GLaNAngcURGw@mail.gmail.com>
References: <CAKDfjgenryyn2huw6wXkr5vH+HKPTRP_WT9wj251pWHvqR_Buw@mail.gmail.com>
	<63B00DD1DA6A364E9F64A3A0BD2FE7B6BACD40@BY2PRD0710MB390.namprd07.prod.outlook.com>
	<CAKDfjge7n9f6ojkBgJbA4L_-65_QGdWSGwwaC-GLaNAngcURGw@mail.gmail.com>
Message-ID: <CAKzWOxXsy57C4LGdppQ8dUnT=kug6Nkqz4nes-YkuSoe761p8A@mail.gmail.com>

On Mon, Jun 18, 2012 at 1:59 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Thank you for your comments!
>
> While this is off-topic for FreeSWITCH I'm not sure where else this
> conversation could take place.  Apologies in advance to the list
> admins - if you want me to take it elsewhere just say so :).
>

Naturally, the *best* place for this conversation to take place is the
Hyatt Chicago Magnificent Mile, August 7-9, 2012! ;)
-MC
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From msc at freeswitch.org  Tue Jun 19 01:15:44 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jun 2012 14:15:44 -0700
Subject: [Freeswitch-users] get info from DB into channel variable in
	xml dialplan
In-Reply-To: <4FDF9483.7040701@gmail.com>
References: <4FDF9483.7040701@gmail.com>
Message-ID: <CAKzWOxUhukjcVUK3gxdhEVPOa97vxjkDYL--f+zj4EZZBuRsEA@mail.gmail.com>

My guess is that if you want/need to skip Lua then you'll need to do some
regex magic to parse out the response data. However, I'm wondering what the
aversion to Lua is. It's ridiculously light and it's pretty easy to use.
What's the downside?
-MC

On Mon, Jun 18, 2012 at 1:50 PM, Abaci <abaci64 at gmail.com> wrote:

>  We are looking for a way to set a few variables in the XML dialplan with
> information stored in a MySQL DB, the simplest seems to be to use
> mod_odbc_query which will set channel variables (see
> http://wiki.freeswitch.org/wiki/Mod_odbc_query#APP), problem is that we
> prefer not to access the DB directly from xml.
> The other option is to use mod_curl which gets the result into a channel
> variable named 'curl_response_data', problem is that we have multiple
> values that we need in separate channel variables, and I can't find any
> documentation on how to do that, is there any way to do this using json etc?
>
>
>
>
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From abaci64 at gmail.com  Tue Jun 19 01:26:41 2012
From: abaci64 at gmail.com (Abaci)
Date: Mon, 18 Jun 2012 17:26:41 -0400
Subject: [Freeswitch-users] get info from DB into channel variable in
 xml dialplan
In-Reply-To: <CAKzWOxUhukjcVUK3gxdhEVPOa97vxjkDYL--f+zj4EZZBuRsEA@mail.gmail.com>
References: <4FDF9483.7040701@gmail.com>
	<CAKzWOxUhukjcVUK3gxdhEVPOa97vxjkDYL--f+zj4EZZBuRsEA@mail.gmail.com>
Message-ID: <4FDF9D11.9050200@gmail.com>

to avoid the overhead of XML+curl+lua+JSON4Lua and use just XML+curl. I 
know Lua is light but it adds up especially if you do more than one per 
call.

On 6/18/2012 5:15 PM, Michael Collins wrote:
> My guess is that if you want/need to skip Lua then you'll need to do 
> some regex magic to parse out the response data. However, I'm 
> wondering what the aversion to Lua is. It's ridiculously light and 
> it's pretty easy to use. What's the downside?
> -MC
>
> On Mon, Jun 18, 2012 at 1:50 PM, Abaci <abaci64 at gmail.com 
> <mailto:abaci64 at gmail.com>> wrote:
>
>     We are looking for a way to set a few variables in the XML
>     dialplan with information stored in a MySQL DB, the simplest seems
>     to be to use mod_odbc_query which will set channel variables (see
>     http://wiki.freeswitch.org/wiki/Mod_odbc_query#APP), problem is
>     that we prefer not to access the DB directly from xml.
>     The other option is to use mod_curl which gets the result into a
>     channel variable named 'curl_response_data', problem is that we
>     have multiple values that we need in separate channel variables,
>     and I can't find any documentation on how to do that, is there any
>     way to do this using json etc?
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun 19 01:52:44 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jun 2012 14:52:44 -0700
Subject: [Freeswitch-users] get info from DB into channel variable in
	xml dialplan
In-Reply-To: <4FDF9D11.9050200@gmail.com>
References: <4FDF9483.7040701@gmail.com>
	<CAKzWOxUhukjcVUK3gxdhEVPOa97vxjkDYL--f+zj4EZZBuRsEA@mail.gmail.com>
	<4FDF9D11.9050200@gmail.com>
Message-ID: <CAKzWOxXqPh-zP27hOu_xMkwbEyheaOUE5OzTRCyw4dCXPLTK5Q@mail.gmail.com>

If it's really a concern then I would just use a regex API and parse out
the stuff you need from the response data.
-MC

On Mon, Jun 18, 2012 at 2:26 PM, Abaci <abaci64 at gmail.com> wrote:

>  to avoid the overhead of XML+curl+lua+JSON4Lua and use just XML+curl. I
> know Lua is light but it adds up especially if you do more than one per
> call.
>
> On 6/18/2012 5:15 PM, Michael Collins wrote:
>
> My guess is that if you want/need to skip Lua then you'll need to do some
> regex magic to parse out the response data. However, I'm wondering what the
> aversion to Lua is. It's ridiculously light and it's pretty easy to use.
> What's the downside?
> -MC
>
> On Mon, Jun 18, 2012 at 1:50 PM, Abaci <abaci64 at gmail.com> wrote:
>
>>  We are looking for a way to set a few variables in the XML dialplan with
>> information stored in a MySQL DB, the simplest seems to be to use
>> mod_odbc_query which will set channel variables (see
>> http://wiki.freeswitch.org/wiki/Mod_odbc_query#APP), problem is that we
>> prefer not to access the DB directly from xml.
>> The other option is to use mod_curl which gets the result into a channel
>> variable named 'curl_response_data', problem is that we have multiple
>> values that we need in separate channel variables, and I can't find any
>> documentation on how to do that, is there any way to do this using json etc?
>>
>>
>>
>>
>
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From shadowym at hotmail.com  Tue Jun 19 02:01:46 2012
From: shadowym at hotmail.com (shadowym)
Date: Mon, 18 Jun 2012 15:01:46 -0700
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
Message-ID: <COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>

Who's idea was it to move the freeswitch directories in the RPM and why?

 

Is it too much to ask that directories are standardized and don't change?!

 

Do you know how distruptive it is when you move all the config files from
/opt to /etc.!  Not only that but the hierarchy has been changed.  The /conf
directory has been removed even though it's in all the documentation on the
Freeswitch wiki.  Wtf?  I am used to config files being in /etc for
Asterisk.  Despite all the disadvantages with Asterisk at least they are
sticking with that decision!  Of course if I compile from scratch I suppose
I can choose  the directory myself but for various reasons I used the RPM so
I don't have much choice in the matter unless I want to make my own RPM
which isn't very good use of my time imho.  But wait, it gets worse!
Whoever made this decision decided just moving everything and getting rid of
/conf directory was not disruptive enough.  So they decided to split it up
and leave the sound files where they are in /opt/freeswitch.  Again, wtf?!

 

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From msc at freeswitch.org  Tue Jun 19 02:12:56 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jun 2012 15:12:56 -0700
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
Message-ID: <CAKzWOxVAPEESKaW4-Ax2TLRuy03DeTRzkGV0RP1Jm=m76viQJw@mail.gmail.com>

Normally I would have slapped this email down for being unnecessarily
trollish, however if we s/emotions//g and get to the core of the issue then
I think we can have a productive discussion. Since I personally was not
involved in the decisions on the RPMs I will simply moderate the discussion.

Please everyone, let's keep the discussion civil and on point.

-MC

On Mon, Jun 18, 2012 at 3:01 PM, shadowym <shadowym at hotmail.com> wrote:

> Who?s idea was it to move the freeswitch directories in the RPM and why?**
> **
>
> ** **
>
> Is it too much to ask that directories are standardized and don?t change?!
> ****
>
> ** **
>
> Do you know how distruptive it is when you move all the config files from
> /opt to /etc.!  Not only that but the hierarchy has been changed.  The
> /conf directory has been removed even though it?s in all the documentation
> on the Freeswitch wiki.  Wtf?  I am used to config files being in /etc
> for Asterisk.  Despite all the disadvantages with Asterisk at least they
> are sticking with that decision!  Of course if I compile from scratch I
> suppose I can choose  the directory myself but for various reasons I used
> the RPM so I don?t have much choice in the matter unless I want to make
> my own RPM which isn?t very good use of my time imho.  But wait, it gets
> worse!  Whoever made this decision decided just moving everything and
> getting rid of /conf directory was not disruptive enough.  So they
> decided to split it up and leave the sound files where they are in
> /opt/freeswitch.  Again, wtf?!****
>
> ** **
>
>
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From moises.silva at gmail.com  Tue Jun 19 02:14:48 2012
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 18 Jun 2012 18:14:48 -0400
Subject: [Freeswitch-users] Where are my freetdm sip_headers?
In-Reply-To: <4FDACBBA.5060302@gmx.de>
References: <4FDACBBA.5060302@gmx.de>
Message-ID: <CAA4nhyA44SRLMtsB+HUu-oFWWszzzY7Jh_H0HdW=smDFkforkg@mail.gmail.com>

On Fri, Jun 15, 2012 at 1:44 AM, Mitja <unau0 at gmx.de> wrote:

> Hello all,
>
> I'm have some problems with my freetdm module lately. To be more precise
> freetdm with ftmod_sangoma_isdn for ISDN handling.
>
> One of which is, that I cant get early media to be played in pre_answer
> state (aka progress_media). But I think I saw a Jira on that one, where
> I will try to get the information/contribute if possible.
>
> The other problem I have is that on an incoming call I cant reliably get
> the TON for displaying purpouses. Actually Display would be alright, but
> calling back the other side is just uncomfortable if your number is
> missing a leading 0.
> So I thought I just have to activate the sip_headers in
> autoloads/freetdm.conf.xml as described here:
> http://wiki.freeswitch.org/wiki/Mod_freetdm#Channel_Variables (actually
> right above)
>
> But when I call my info Application, I cant see any of the headers.
>
> freetdm.conf.xml: (I set the param sip_headers three times after several
> tries)
> <configuration name="freetdm.conf" description="Freetdm Configuration">
> <settings>
> <param name="debug" value="0"/>
> <param name="sip_headers" value="true" />
>


This should be <param name="sip-headers" value="true" />, note that there
is a dash instead of an underscore.

*Moises Silva
**Manager, Software Engineering***

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



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From moises.silva at gmail.com  Tue Jun 19 02:34:10 2012
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 18 Jun 2012 18:34:10 -0400
Subject: [Freeswitch-users] Openzap bridge and DTMF issue
In-Reply-To: <ABCC4BED9408E84E91D9348B2DA819C6061BE6@tslsrv.TSL.LOCAL>
References: <Ac1IpMOubioO5YO7RIq1YQa/DFj4/A==>
	<ABCC4BED9408E84E91D9348B2DA819C6061BE6@tslsrv.TSL.LOCAL>
Message-ID: <CAA4nhyDiSguJrEGnTkSxZJdBKi==4kwbqmH-p8uwV6jrdgBHwQ@mail.gmail.com>

On Tue, Jun 12, 2012 at 10:07 AM, George D'mithrov <
george.d at telesoftlabs.com> wrote:

> **
>
> Hi,
>
> I wants to bridge calls with particular DID, which landing on one PRI span
> (Span_1) to another PRI span (Span_2).
>
>
> I'm able to bridge the calls between PRI span and able to talk to each
> other.
>
>
> My problem is that sometimes DTMF which received on Span_1 is not sending
> to Span2 and vice-verse. DTMF tones are hearing partially on receving end.
>
>
> Sometimes after a some delay (30-60 seconds) am getting the DTMF, but
> that's too some dtmf are missing.
>
>
> If I hangup the line while the dtmf is in queue then the dtmf are
> generating on the next call on the same line.
>
>
> But when I bridge the call on PRI with sip channel then am not getting any
> problem.
>
> I think the following log in freeswitch is usefull
>
> 2012-06-12 18:58:38.379506 [DEBUG] mod_openzap.c:725 queue DTMF [1]
> 2012-06-12 18:58:38.019513 [DEBUG] switch_ivr_bridge.c:391 Send signal
> OpenZAP/1:1/8040327325 [BREAK]
> after this  it takes a while to generate dtmf
>


Any reason you're using a deprecated module? you should be using
mod_freetdm and not mod_openzap. I think it's been 2 years since we renamed
openzap to freetdm and did all new development on freetdm. I don't
particularly remember if the issue you mention is fixed, but before I would
even look at the logs to see if there is an issue indeed you need to
upgrade to FreeTDM.


*Moises Silva
**Manager, Software Engineering***

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



**<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>

Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
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 | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`|
| YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
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From lazyvirus at gmx.com  Tue Jun 19 02:53:57 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Tue, 19 Jun 2012 00:53:57 +0200
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
Message-ID: <20120619005357.0cd10ecb@anubis.defcon1>

On Mon, 18 Jun 2012 15:01:46 -0700
shadowym <shadowym at hotmail.com> wrote:

> Is it too much to ask that directories are standardized and don't
> change?!

From what you say below, it is a case of standardization indeed,
as Linux policy is to _have_ all conf files into /etc/.

> Do you know how distruptive it is when you move all the config
> files from /opt to /etc.!

OMFG, what a PITA to do a: mv /opt/whatever /etc/freeswitch, indeed?.

>  Not only that but the hierarchy has
> been changed.  The /conf directory has been removed even though
> it's in all the documentation on the Freeswitch wiki.  Wtf?  I am
> used to config files being in /etc for Asterisk.  Despite all the
> disadvantages with Asterisk at least they are sticking with that
> decision!  Of course if I compile from scratch I suppose I can
> choose  the directory myself but for various reasons I used the
> RPM so I don't have much choice in the matter unless I want to
> make my own RPM which isn't very good use of my time imho.  But
> wait, it gets worse! Whoever made this decision decided just
> moving everything and getting rid of /conf directory was not
> disruptive enough.  So they decided to split it up and leave the
> sound files where they are in /opt/freeswitch.  Again, wtf?!

I guess your typing has been done under hot conditions - and this
can happen to anybody one day or another.

However, consider all the devs people work, what they're doing for
YOU (the user, that just have to modify a bit a complete native
conf to get a fully operative switch), the sweat they put in fixing
some quirks sometimes in minutes and you'll begin to see how
offensive your words are: "somebody" reversed the order of your 
FREE lunch, fork, knive & spoon and you're going off the deep end
for THAT?
In france, we have a swell stock phrase for that: 
"faire un caca nerveux" (literally: to make a nervous caca-poo).

So, please tell them it just a hot and epidermic reaction, no?

Jean-Yves
-- 
BOFH excuse #12:
dry joints on cable plug


From steveayre at gmail.com  Tue Jun 19 03:27:39 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 19 Jun 2012 00:27:39 +0100
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
Message-ID: <31E28D57-821F-4C2C-9558-B17B2DF2DC35@gmail.com>

The project is currently in a feature freeze working towards a version 1.2 release.

As part of this the Redhat and Debian packages are being improved and standardised to be compliant with the various FHS requirements. The move to /etc is precisely part of the standardisation that you are so desperate for.

Sound files are data and don't belong in /etc, personally I'd say they belong under /usr along with the test of the FreeSWITCH binaries - if they're still in /opt then (without following the work the rpm maintainer us doing) its probably a transition that's still in progress.

If this caught you by surprise, then you should always be testing a setup before putting it into production. Once 1.2 is released it'll become the stable branch and offer far fewer surprises.


Steve on iPhone



On 18 Jun 2012, at 23:01, shadowym <shadowym at hotmail.com> wrote:

> Who?s idea was it to move the freeswitch directories in the RPM and why?
>  
> Is it too much to ask that directories are standardized and don?t change?!
>  
> Do you know how distruptive it is when you move all the config files from /opt to /etc.!  Not only that but the hierarchy has been changed.  The /conf directory has been removed even though it?s in all the documentation on the Freeswitch wiki.  Wtf?  I am used to config files being in /etc for Asterisk.  Despite all the disadvantages with Asterisk at least they are sticking with that decision!  Of course if I compile from scratch I suppose I can choose  the directory myself but for various reasons I used the RPM so I don?t have much choice in the matter unless I want to make my own RPM which isn?t very good use of my time imho.  But wait, it gets worse!  Whoever made this decision decided just moving everything and getting rid of /conf directory was not disruptive enough.  So they decided to split it up and leave the sound files where they are in /opt/freeswitch.  Again, wtf?!
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From kris at kriskinc.com  Tue Jun 19 03:37:01 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 18 Jun 2012 19:37:01 -0400
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
Message-ID: <CAKDfjgcU-jRZF+iX++txudwVaUDm51khDbLwVhRRszFMGS29Lg@mail.gmail.com>

Write a script to make symlinks.  You won't know the difference.

On Mon, Jun 18, 2012 at 6:01 PM, shadowym <shadowym at hotmail.com> wrote:
> Who?s idea was it to move the freeswitch directories in the RPM and why?
>
>
>
> Is it too much to ask that directories are standardized and don?t change?!
>
>
>
> Do you know how distruptive it is when you move all the config files from
> /opt to /etc.!? Not only that but the hierarchy has been changed.? The /conf
> directory has been removed even though it?s in all the documentation on the
> Freeswitch wiki.? Wtf?? I am used to config files being in /etc for
> Asterisk.? Despite all the disadvantages with Asterisk at least they are
> sticking with that decision!? Of course if I compile from scratch I suppose
> I can choose? the directory myself but for various reasons I used the RPM so
> I don?t have much choice in the matter unless I want to make my own RPM
> which isn?t very good use of my time imho.? But wait, it gets worse!
> Whoever made this decision decided just moving everything and getting rid of
> /conf directory was not disruptive enough.? So they decided to split it up
> and leave the sound files where they are in /opt/freeswitch.? Again, wtf?!
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From william.king at quentustech.com  Tue Jun 19 03:41:28 2012
From: william.king at quentustech.com (William King)
Date: Mon, 18 Jun 2012 16:41:28 -0700
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
Message-ID: <4FDFBCA8.1050700@quentustech.com>

I believe there was an announcement on this mailing list alerting that 
the new rpms would be undergoing major changes to include moving the 
directory locations. Search this list for new threads started by SwK in 
the past 6 months.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 06/18/2012 03:01 PM, shadowym wrote:
>
> Who's idea was it to move the freeswitch directories in the RPMand why?
>
> Is it too much to ask that directories are standardized and don't change?!
>
> Do you know how distruptive it is when you move all the config files 
> from /opt to /etc.! Not only that but the hierarchy has been changed.  
> The /conf directory has been removed even though it's in all the 
> documentation on the Freeswitch wiki.  Wtf? I am used to config files 
> being in /etc for Asterisk.  Despite all the disadvantages with 
> Asterisk at least they are sticking with that decision!  Of course if 
> I compile from scratch I suppose I can choose  the directory myself 
> butfor various reasons I used the RPM so I don't have much choice in 
> the matterunless I want to make my own RPM which isn't very good use 
> of my time imho.  But wait, it gets worse!  Whoever made this decision 
> decided just moving everything and getting rid of /conf directory was 
> not disruptive enough. So they decided to split it up and leave the 
> sound files where they are in /opt/freeswitch. Again, wtf?!
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From chris at opencsta.org  Tue Jun 19 03:50:49 2012
From: chris at opencsta.org (Chris Mylonas)
Date: Tue, 19 Jun 2012 09:50:49 +1000
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <CALNZuwmF0Z3bpK0RGr0eSdTRx_pBsHjOXGm2oV7um3XHjpC9Fw@mail.gmail.com>
References: <CAKzWOxX0pBC5J_TX+qe-KZN+Lr=zPtOY5ceAxu-CL4hXbsqtvg@mail.gmail.com>
	<15212.1340038338@ccs.covici.com>
	<CAKDfjgeO1gaUsbpUjrVoAaBQHxAgvM-71+x7O4F27DZFOuPV8Q@mail.gmail.com>
	<CALNZuw=aXGL7DmFpOOq63gUaOjdF0fd+AsyLxWxVZusHAv5Csg@mail.gmail.com>
	<CAKDfjgf4O90Y6-JXk06dFpY-yMLpmhteRL9iEVW9C-PTosR7Vw@mail.gmail.com>
	<CALNZuwmF0Z3bpK0RGr0eSdTRx_pBsHjOXGm2oV7um3XHjpC9Fw@mail.gmail.com>
Message-ID: <2EFCD4D9-E427-4716-9885-6F659506410C@opencsta.org>

+1 github repo
attach your (c)opyright and whatever license the doco is so people know how/what to contribute

you never know what will happen.  a few network diagrams + format-conversion scripts (txt -> pdf -> html -> epub) from others and you are far from bound to your one document :)





On 19/06/2012, at 6:51 AM, Avi Marcus wrote:

> On your own github repo's there's an "edit this file" link and it gives you an editor in the browser.
> 
> I was mostly referring that others could submit "patches" that you could approve easily.
> 
> -Avi
> 
> 
> On Mon, Jun 18, 2012 at 11:44 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> I'm not opposed to this, necessarily, but my first priority is making
> updates as easy as possible.
> 
> I'm busy and lazy.  If updating the document means doing a git pull,
> running some scripts, editing HTML/Latex/whatever, checking in, etc,
> etc I'm not going to update the document as much.
> 
> If it's as simple as opening a new tab and clicking a bookmark I'm
> much more likely to keep the document up to date ;).
> 
> On Mon, Jun 18, 2012 at 2:09 PM, Avi Marcus <avi at avimarcus.net> wrote:
> > Github public repo, maybe. Then others can clone and make changes and you
> > can do a pull to merge if you approve.
> > -Avi
> >
> 
> --
> Kristian Kielhofner
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 zoho.com  Tue Jun 19 04:18:16 2012
From: freeswitch at zoho.com (dingdong)
Date: Mon, 18 Jun 2012 17:18:16 -0700
Subject: [Freeswitch-users] PRI + FreeTDM problems
Message-ID: <138021aa2d1.-4585004023753801168.3063868342642460335@zoho.com>

hello all,   i had this problem months ago and didn't really have the solution,just thinking that maybe someone from here can help.




our primary link for calling outside is a PRI.
so here's what we currently have, when we try to call 18778025246 and  1866-642-5246 - calls are not going through.


but when we call 18775682495 - calls go just fine.




attaching my freetdm.conf.xml too as the provider made me change some fields in there.






pasting the logs provided by our PRI provider telepacific.com,with comparisons for both numbers dialed and labelled as good and bad calls,
----------------------------------------------------------------------------------------------------------------------------------


GOOD ONE S2C4-1091895439 12-06-09 13:49:46 253961 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 3 Call Reference Value Orig 18 M 05 Message Type SETUP I 04 Bearer Capability Length = 3 80 Coding Standard CCITT Info Transfer Cap. Speech 90 Transfer Mode Circuit Transfer Rate 64 kbit/s A2 Layer 1 Protocol G.711 u-law I 18 Channel Identification Length = 3 A9 Interface ID Implicitly Defined Interface Type Primary S2C4-1091895439 12-06-09 13:49:46 253962 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 4 Preferred/Exclusive Exclusive D Channel Indicator Not D Channel Channel Selection As Indicated 83 Coding Standard CCITT Coding Type Number Channel Type B Channel 81 Channel Number 1 I 6C Calling Party Number Length = 13 00 Type of Number Unknown Type Numbering Plan Unknown Numbering Plan 81 Presentation Indicator Allowed S2C4-1091895439 12-06-09 13:49:46 253963 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 5 Screening Indicator User Provided - Passed Number 13231234567 I 70 Called Party Number Length = 12 80 Type of Number Unknown Type Numbering Plan Unknown Numbering Plan Number 18775682495 Raw Hex Data 08 02 00 12 05 04 03 80 90 A2 18 03 A9 83 81 6C 0D 00 81 31 33 32 33 34 35 34 34 30 36 30 70 0C 80 31 38 37 37 35 36 38 32 34 39 35 S2C4-1091895439 12-06-09 13:50:16 253972 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 6 ------------------------------------------------ Event 2 12/06/09 13:47:20 TEI 0 08 Q.931 Call Control Message Call Reference Value Dest 18 M 02 Message Type CALL PROCEEDING I 18 Channel Identification Length = 3 A9 Interface ID Implicitly Defined Interface Type Primary Preferred/Exclusive Exclusive D Channel Indicator Not D Channel Channel Selection As Indicated 83 Coding Standard CCITT Coding Type Number Channel Type B Channel S2C4-1091895439 12-06-09 13:50:17 253973 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 7 81 Channel Number 1 Raw Hex Data 08 02 80 12 02 18 03 A9 83 81 ------------------------------------------------ Event 3 12/06/09 13:47:21 TEI 0 08 Q.931 Call Control Message Call Reference Value Dest 18 M 03 Message Type PROGRESS I 1E Progress Indicator Length = 2 84 Coding Standard CCITT Location Remote Public Network 88 Value In-band Information/Pattern Avail S2C4-1091895439 12-06-09 13:50:17 253974 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 8 Raw Hex Data 08 02 80 12 03 1E 02 84 88 ------------------------------------------------ Event 4 12/06/09 13:47:21 TEI 0 08 Q.931 Call Control Message Call Reference Value Dest 18 M 07 Message Type CONNECT I 1E Progress Indicator Length = 2 84 Coding Standard CCITT Location Remote Public Network 82 Value Destination is not ISDN Raw Hex Data 08 02 80 12 07 1E 02 84 82 S2C4-1091895439 12-06-09 13:50:17 253975 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 9 ------------------------------------------------ Event 5 12/06/09 13:47:33 TEI 0 08 Q.931 Call Control Message Call Reference Value Orig 18 M 45 Message Type DISCONNECT I 08 Cause Length = 2 81 Coding Standard CCITT Location Local Private Network 90 Class Normal Event Value #16, Normal Call Clearing ------------------------------------------------ ------------------------------------------------ --------------------- BAD ONE S2C4-1091895439 12-06-09 13:50:48 253980 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 13 83 Coding Standard CCITT Coding Type Number Channel Type B Channel 81 Channel Number 1 I 6C Calling Party Number Length = 13 00 Type of Number Unknown Type Numbering Plan Unknown Numbering Plan 81 Presentation Indicator Allowed Screening Indicator User Provided - Passed Number 13231234567
 I 70 Called Party Number Length = 12 S2C4-1091895439 12-06-09 13:50:48 253981 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 14 80 Type of Number Unknown Type Numbering Plan Unknown Numbering Plan Number 18778025246 Raw Hex Data 08 02 00 13 05 04 03 80 90 A2 18 03 A9 83 81 6C 0D 00 81 31 33 32 33 34 35 34 34 30 36 30 70 0C 80 31 38 37 37 38 30 32 35 32 34 36 ------------------------------------------------ Event 9 12/06/09 13:48:05 TEI 0 08 Q.931 Call Control Message S2C4-1091895439 12-06-09 13:50:48 253982 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 15 Call Reference Value Dest 19 M 02 Message Type CALL PROCEEDING I 18 Channel Identification Length = 3 A9 Interface ID Implicitly Defined Interface Type Primary Preferred/Exclusive Exclusive D Channel Indicator Not D Channel Channel Selection As Indicated 83 Coding Standard CCITT Coding Type Number Channel Type B Channel 81 Channel Number 1 Raw Hex Data S2C4-1091895439 12-06-09 13:51:18 253993 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 16 08 02 80 13 02 18 03 A9 83 81 ------------------------------------------------ Event 10 12/06/09 13:48:05 TEI 0 08 Q.931 Call Control Message Call Reference Value Dest 19 M 03 Message Type PROGRESS I 1E Progress Indicator Length = 2 8A Coding Standard CCITT Location Beyond Interworking Network 81 Value Call is not End-to-End ISDN Raw Hex Data 08 02 80 13 03 1E 02 8A 81 ------------------------------------------------ S2C4-1091895439 12-06-09 13:51:19 253994 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 17 Event 11 12/06/09 13:48:19 TEI 0 08 Q.931 Call Control Message Call Reference Value Orig 19 M 45 Message Type DISCONNECT I 08 Cause Length = 2 81 Coding Standard CCITT Location Local Private Network 90 Class Normal Event Value #16, Normal Call Clearing Raw Hex Data S2C4-1091895439 12-06-09 13:51:19 253995 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 18 08 02 00 13 45 08 02 81 90 ------------------------------------------------ 






please take note that 18778025246/1866-642-5246 goes thru our sip provider just fine. no problems at all. not with the PRI.
i pressume both numbers are toll free.



i had help from sangoma support no response yet with the 'called number' issue.


wanrouter version


WANPIPE Release: 3.5.27
 i have updated the libsng_isdn to the newest driver as suggested,




pcap sent to sangoma support is saying that when calls go through, i am NOT specifying a 'called number'
but bad calls are having that 'called number' .


now the problem is i am not really setting anything specific (e.g adding that 'called number' field to specific numbers when called)









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From darius at dons.net.au  Tue Jun 19 03:23:45 2012
From: darius at dons.net.au (Daniel O'Connor)
Date: Tue, 19 Jun 2012 08:53:45 +0930
Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
In-Reply-To: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
References: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
Message-ID: <0AEB7B3A-5591-4694-83B5-E4208D5DA4A0@dons.net.au>


On 19/06/2012, at 5:20, Terry Barnum wrote:
> I recently started investigating FS and am curious about what the specific problem is with Macports and FS. I've been using Macports for a couple years and find it to be a great, actively maintained package manager that uses /opt/local, making it a very reliable way to keep production software working through Apple system updates. I've found the Macports team to be very responsive so if there are specific things that I can pass along, it might get fixed. Is there a link or thread that documents what's incompatible with FS?

There is no incompatibility per se - I use Macports and have installed FreeSwitch, although the only thing that FS actually uses from MP is libjpeg as it bundles everything else.

That said, I had trouble getting the bundled PortAudio working so I mangled the mod_portaudio Makefile.am to use the MP PortAudio and it worked fine.

--
Daniel O'Connor software and network engineer
for Genesis Software - http://www.gsoft.com.au
"The nice thing about standards is that there
are so many of them to choose from."
  -- Andrew Tanenbaum
GPG Fingerprint - 5596 B766 97C0 0E94 4347 295E E593 DC20 7B3F CE8C








From skent at noblesys.com  Tue Jun 19 04:22:22 2012
From: skent at noblesys.com (Steve Kent)
Date: Mon, 18 Jun 2012 20:22:22 -0400 (EDT)
Subject: [Freeswitch-users] PRI + FreeTDM problems
In-Reply-To: <138021aa2d1.-4585004023753801168.3063868342642460335@zoho.com>
Message-ID: <13378049.7977776.1340065342986.JavaMail.root@noblesys.com>

Larry just switched them over to tnci......



----- Original Message -----
From: "dingdong" <freeswitch at zoho.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Monday, June 18, 2012 8:18:16 PM
Subject: [Freeswitch-users] PRI + FreeTDM problems



hello all, 
i had this problem months ago and didn't really have the solution,just thinking that maybe someone from here can help. 




our primary link for calling outside is a PRI. 
so here's what we currently have, when we try to call 18778025246 and 1866-642-5246 - calls are not going through. 



but when we call 18775682495 - calls go just fine. 




attaching my freetdm.conf.xml too as the provider made me change some fields in there. 






pasting the logs provided by our PRI provider telepacific.com,with comparisons for both numbers dialed and labelled as good and bad calls, 
---------------------------------------------------------------------------------------------------------------------------------- 


GOOD ONE            S2C4-1091895439 12-06-09 13:49:46 253961 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 3                                 Call Reference Value              Orig        18                          M 05 Message Type                               SETUP                          I 04 Bearer Capability                   Length =   3                            80   Coding Standard                          CCITT                                 Info Transfer Cap.                      Speech                            90   Transfer Mode                          Circuit                                 Transfer Rate                        64 kbit/s                            A2   Layer 1 Protocol                   G.711 u-law                          I 18 Channel Identification              Length =   3                            A9   Interface ID                Implicitly Defined                                 Interface Type                         Primary        S2C4-1091895439 12-06-09 13:49:46 253962 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 4                                   Preferred/Exclusive                  Exclusive                                 D Channel Indicator              Not D Channel                                 Channel Selection                 As Indicated                            83   Coding Standard                          CCITT                                 Coding Type                             Number                                 Channel Type                         B Channel                            81   Channel Number                               1                          I 6C Calling Party Number                Length =  13                            00   Type of Number                    Unknown Type                                 Numbering Plan          Unknown Numbering Plan                            81   Presentation Indicator                 Allowed        S2C4-1091895439 12-06-09 13:49:46 253963 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 5                                   Screening Indicator     User Provided - Passed                                 Number                             13231234567                          I 70 Called Party Number                 Length =  12                            80   Type of Number                    Unknown Type                                 Numbering Plan          Unknown Numbering Plan                                 Number                             18775682495                                 Raw Hex Data                               08 02 00 12 05 04 03 80 90 A2 18 03 A9 83 81 6C                               0D 00 81 31 33 32 33 34 35 34 34 30 36 30 70 0C                               80 31 38 37 37 35 36 38 32 34 39 35        S2C4-1091895439 12-06-09 13:50:16 253972 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 6                                 ------------------------------------------------          Event 2  12/06/09 13:47:20  TEI 0       08 Q.931 Call Control Message          Call Reference Value              Dest        18     M 02 Message Type                     CALL PROCEEDING     I 18 Channel Identification              Length =   3       A9   Interface ID                Implicitly Defined            Interface Type                         Primary            Preferred/Exclusive                  Exclusive            D Channel Indicator              Not D Channel            Channel Selection                 As Indicated       83   Coding Standard                          CCITT            Coding Type                             Number            Channel Type                         B Channel        S2C4-1091895439 12-06-09 13:50:17 253973 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 7         81   Channel Number                               1            Raw Hex Data          08 02 80 12 02 18 03 A9 83 81          ------------------------------------------------          Event 3  12/06/09 13:47:21  TEI 0       08 Q.931 Call Control Message          Call Reference Value              Dest        18     M 03 Message Type                            PROGRESS     I 1E Progress Indicator                  Length =   2       84   Coding Standard                          CCITT            Location                 Remote Public Network       88   Value        In-band Information/Pattern Avail          S2C4-1091895439 12-06-09 13:50:17 253974 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 8            Raw Hex Data          08 02 80 12 03 1E 02 84 88          ------------------------------------------------          Event 4  12/06/09 13:47:21  TEI 0       08 Q.931 Call Control Message          Call Reference Value              Dest        18     M 07 Message Type                             CONNECT     I 1E Progress Indicator                  Length =   2       84   Coding Standard                          CCITT            Location                 Remote Public Network       82   Value                  Destination is not ISDN            Raw Hex Data          08 02 80 12 07 1E 02 84 82        S2C4-1091895439 12-06-09 13:50:17 253975 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 9            ------------------------------------------------                               Event 5  12/06/09 13:47:33  TEI 0                            08 Q.931 Call Control Message                               Call Reference Value              Orig        18                          M 45 Message Type                          DISCONNECT                          I 08 Cause                               Length =   2                            81   Coding Standard                          CCITT                                 Location                 Local Private Network                            90   Class                             Normal Event                                 Value                #16, Normal Call Clearing         ------------------------------------------------ ------------------------------------------------ ---------------------   BAD ONE              S2C4-1091895439 12-06-09 13:50:48 253980 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 13                              83   Coding Standard                          CCITT                                 Coding Type                             Number                                 Channel Type                         B Channel                            81   Channel Number                               1                          I 6C Calling Party Number                Length =  13                            00   Type of Number                    Unknown Type                                 Numbering Plan          Unknown Numbering Plan                            81   Presentation Indicator                 Allowed                                 Screening Indicator     User Provided - Passed                                 Number                             13231234567 
I 70 Called Party Number                 Length =  12        S2C4-1091895439 12-06-09 13:50:48 253981 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 14                              80   Type of Number                    Unknown Type                                 Numbering Plan          Unknown Numbering Plan                                 Number                             18778025246                                 Raw Hex Data                               08 02 00 13 05 04 03 80 90 A2 18 03 A9 83 81 6C                               0D 00 81 31 33 32 33 34 35 34 34 30 36 30 70 0C                               80 31 38 37 37 38 30 32 35 32 34 36                               ------------------------------------------------          Event 9  12/06/09 13:48:05  TEI 0       08 Q.931 Call Control Message        S2C4-1091895439 12-06-09 13:50:48 253982 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 15            Call Reference Value              Dest        19     M 02 Message Type                     CALL PROCEEDING     I 18 Channel Identification              Length =   3       A9   Interface ID                Implicitly Defined            Interface Type                         Primary            Preferred/Exclusive                  Exclusive            D Channel Indicator              Not D Channel            Channel Selection                 As Indicated       83   Coding Standard                          CCITT            Coding Type                             Number            Channel Type                         B Channel       81   Channel Number                               1            Raw Hex Data        S2C4-1091895439 12-06-09 13:51:18 253993 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 16            08 02 80 13 02 18 03 A9 83 81          ------------------------------------------------          Event 10  12/06/09 13:48:05  TEI 0       08 Q.931 Call Control Message          Call Reference Value              Dest        19     M 03 Message Type                            PROGRESS     I 1E Progress Indicator                  Length =   2       8A   Coding Standard                          CCITT            Location           Beyond Interworking Network       81   Value              Call is not End-to-End ISDN            Raw Hex Data          08 02 80 13 03 1E 02 8A 81          ------------------------------------------------        S2C4-1091895439 12-06-09 13:51:19 253994 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 17                                 Event 11  12/06/09 13:48:19  TEI 0                            08 Q.931 Call Control Message                               Call Reference Value              Orig        19                          M 45 Message Type                          DISCONNECT                          I 08 Cause                               Length =   2                            81   Coding Standard                          CCITT                                 Location                 Local Private Network                            90   Class                             Normal Event                                 Value                #16, Normal Call Clearing                                 Raw Hex Data        S2C4-1091895439 12-06-09 13:51:19 253995 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 18                                 08 02 00 13 45 08 02 81 90                               ------------------------------------------------ 






please take note that 18778025246/1866-642-5246 goes thru our sip provider just fine. no problems at all. not with the PRI. 
i pressume both numbers are toll free. 


i had help from sangoma support no response yet with the 'called number' issue. 


wanrouter version 


WANPIPE Release: 3.5.27 
i have updated the libsng_isdn to the newest driver as suggested, 




pcap sent to sangoma support is saying that when calls go through, i am NOT specifying a 'called number' 
but bad calls are having that 'called number' . 


now the problem is i am not really setting anything specific (e.g adding that 'called number' field to specific numbers when called) 








_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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please immediately notify the sender by email at the address 
shown above. This email may contain confidential or legally 
privileged information that is intended only for the use of the 
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From msc at freeswitch.org  Tue Jun 19 04:43:02 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jun 2012 17:43:02 -0700
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <4FDFBCA8.1050700@quentustech.com>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
	<4FDFBCA8.1050700@quentustech.com>
Message-ID: <CAKzWOxWR6PJqPb6NqTLRQa5WMGP6H4zVte1w0i2UZSDN9+fQMg@mail.gmail.com>

On Mon, Jun 18, 2012 at 4:41 PM, William King
<william.king at quentustech.com>wrote:

>  I believe there was an announcement on this mailing list alerting that
> the new rpms would be undergoing major changes to include moving the
> directory locations. Search this list for new threads started by SwK in the
> past 6 months.
>

There was indeed an announcement by SwK himself. It was as close to FNLoIP
(Flashing Neon Lights over IP) as you can get.
-MC
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From skent at noblesys.com  Tue Jun 19 04:44:37 2012
From: skent at noblesys.com (Steve Kent)
Date: Mon, 18 Jun 2012 20:44:37 -0400 (EDT)
Subject: [Freeswitch-users] PRI + FreeTDM problems
In-Reply-To: <13378049.7977776.1340065342986.JavaMail.root@noblesys.com>
Message-ID: <1761577.7979332.1340066677695.JavaMail.root@noblesys.com>

So sorry, wrong post.

----- Original Message -----
From: "Steve Kent" <skent at noblesys.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, June 18, 2012 8:22:22 PM
Subject: Re: [Freeswitch-users] PRI + FreeTDM problems

Larry just switched them over to tnci......



----- Original Message -----
From: "dingdong" <freeswitch at zoho.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Monday, June 18, 2012 8:18:16 PM
Subject: [Freeswitch-users] PRI + FreeTDM problems



hello all, 
i had this problem months ago and didn't really have the solution,just thinking that maybe someone from here can help. 




our primary link for calling outside is a PRI. 
so here's what we currently have, when we try to call 18778025246 and 1866-642-5246 - calls are not going through. 



but when we call 18775682495 - calls go just fine. 




attaching my freetdm.conf.xml too as the provider made me change some fields in there. 






pasting the logs provided by our PRI provider telepacific.com,with comparisons for both numbers dialed and labelled as good and bad calls, 
---------------------------------------------------------------------------------------------------------------------------------- 


GOOD ONE            S2C4-1091895439 12-06-09 13:49:46 253961 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 3                                 Call Reference Value              Orig        18                          M 05 Message Type                               SETUP                          I 04 Bearer Capability                   Length =   3                            80   Coding Standard                          CCITT                                 Info Transfer Cap.                      Speech                            90   Transfer Mode                          Circuit                                 Transfer Rate                        64 kbit/s                            A2   Layer 1 Protocol                   G.711 u-law                          I 18 Channel Identification              Length =   3                            A9   Interface ID                Implicitly Defined                                 Interface Type                         Primary 
        S2C4-1091895439 12-06-09 13:49:46 253962 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 4                                   Preferred/Exclusive                  Exclusive                                 D Channel Indicator              Not D Channel                                 Channel Selection                 As Indicated                            83   Coding Standard                          CCITT                                 Coding Type                             Number                                 Channel Type                         B Channel                            81   Channel Number                               1                          I 6C Calling Party Number                Length =  13                            00   Type of Number                    Unknown Type                                 Numbering Plan          Unknown Numbering Plan                            81   Presentation Indicator                 Allowed        S2C4-
 1091895439 12-06-09 13:49:46 253963 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 5                                   Screening Indicator     User Provided - Passed                                 Number                             13231234567                          I 70 Called Party Number                 Length =  12                            80   Type of Number                    Unknown Type                                 Numbering Plan          Unknown Numbering Plan                                 Number                             18775682495                                 Raw Hex Data                               08 02 00 12 05 04 03 80 90 A2 18 03 A9 83 81 6C                               0D 00 81 31 33 32 33 34 35 34 34 30 36 30 70 0C                               80 31 38 37 37 35 36 38 32 34 39 35        S2C4-1091895439 12-06-09 13:50:16 253972 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 6                                 --------------
 ----------------------------------          Event 2  12/06/09 13:47:20  TEI 0       08 Q.931 Call Control Message          Call Reference Value              Dest        18     M 02 Message Type                     CALL PROCEEDING     I 18 Channel Identification              Length =   3       A9   Interface ID                Implicitly Defined            Interface Type                         Primary            Preferred/Exclusive                  Exclusive            D Channel Indicator              Not D Channel            Channel Selection                 As Indicated       83   Coding Standard                          CCITT            Coding Type                             Number            Channel Type                         B Channel        S2C4-1091895439 12-06-09 13:50:17 253973 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 7         81   Channel Number                               1            Raw Hex Data          08 02 80 12 02 18 03 A9 83 81         
  ------------------------------------------------          Event 3  12/06/09 13:47:21  TEI 0       08 Q.931 Call Control Message          Call Reference Value              Dest        18     M 03 Message Type                            PROGRESS     I 1E Progress Indicator                  Length =   2       84   Coding Standard                          CCITT            Location                 Remote Public Network       88   Value        In-band Information/Pattern Avail          S2C4-1091895439 12-06-09 13:50:17 253974 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 8            Raw Hex Data          08 02 80 12 03 1E 02 84 88          ------------------------------------------------          Event 4  12/06/09 13:47:21  TEI 0       08 Q.931 Call Control Message          Call Reference Value              Dest        18     M 07 Message Type                             CONNECT     I 1E Progress Indicator                  Length =   2       84   Coding Standard       
                    CCITT            Location                 Remote Public Network       82   Value                  Destination is not ISDN            Raw Hex Data          08 02 80 12 07 1E 02 84 82        S2C4-1091895439 12-06-09 13:50:17 253975 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 9            ------------------------------------------------                               Event 5  12/06/09 13:47:33  TEI 0                            08 Q.931 Call Control Message                               Call Reference Value              Orig        18                          M 45 Message Type                          DISCONNECT                          I 08 Cause                               Length =   2                            81   Coding Standard                          CCITT                                 Location                 Local Private Network                            90   Class                             Normal Event                            
      Value                #16, Normal Call Clearing         ------------------------------------------------ ------------------------------------------------ ---------------------   BAD ONE              S2C4-1091895439 12-06-09 13:50:48 253980 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 13                              83   Coding Standard                          CCITT                                 Coding Type                             Number                                 Channel Type                         B Channel                            81   Channel Number                               1                          I 6C Calling Party Number                Length =  13                            00   Type of Number                    Unknown Type                                 Numbering Plan          Unknown Numbering Plan                            81   Presentation Indicator                 Allowed                                 Screening Indicator 
     User Provided - Passed                                 Number                             13231234567 
I 70 Called Party Number                 Length =  12        S2C4-1091895439 12-06-09 13:50:48 253981 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 14                              80   Type of Number                    Unknown Type                                 Numbering Plan          Unknown Numbering Plan                                 Number                             18778025246                                 Raw Hex Data                               08 02 00 13 05 04 03 80 90 A2 18 03 A9 83 81 6C                               0D 00 81 31 33 32 33 34 35 34 34 30 36 30 70 0C                               80 31 38 37 37 38 30 32 35 32 34 36                               ------------------------------------------------          Event 9  12/06/09 13:48:05  TEI 0       08 Q.931 Call Control Message        S2C4-1091895439 12-06-09 13:50:48 253982 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 15            Call Reference Value              Dest        1
 9     M 02 Message Type                     CALL PROCEEDING     I 18 Channel Identification              Length =   3       A9   Interface ID                Implicitly Defined            Interface Type                         Primary            Preferred/Exclusive                  Exclusive            D Channel Indicator              Not D Channel            Channel Selection                 As Indicated       83   Coding Standard                          CCITT            Coding Type                             Number            Channel Type                         B Channel       81   Channel Number                               1            Raw Hex Data        S2C4-1091895439 12-06-09 13:51:18 253993 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 16            08 02 80 13 02 18 03 A9 83 81          ------------------------------------------------          Event 10  12/06/09 13:48:05  TEI 0       08 Q.931 Call Control Message          Call Reference Value          
     Dest        19     M 03 Message Type                            PROGRESS     I 1E Progress Indicator                  Length =   2       8A   Coding Standard                          CCITT            Location           Beyond Interworking Network       81   Value              Call is not End-to-End ISDN            Raw Hex Data          08 02 80 13 03 1E 02 8A 81          ------------------------------------------------        S2C4-1091895439 12-06-09 13:51:19 253994 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 17                                 Event 11  12/06/09 13:48:19  TEI 0                            08 Q.931 Call Control Message                               Call Reference Value              Orig        19                          M 45 Message Type                          DISCONNECT                          I 08 Cause                               Length =   2                            81   Coding Standard                          CCITT                
                  Location                 Local Private Network                            90   Class                             Normal Event                                 Value                #16, Normal Call Clearing                                 Raw Hex Data        S2C4-1091895439 12-06-09 13:51:19 253995 MAINT LSANCAGJDS4 M  DUMP FILE ALL IN PROGRESS  SEGMENT 18                                 08 02 00 13 45 08 02 81 90                               ------------------------------------------------ 






please take note that 18778025246/1866-642-5246 goes thru our sip provider just fine. no problems at all. not with the PRI. 
i pressume both numbers are toll free. 


i had help from sangoma support no response yet with the 'called number' issue. 


wanrouter version 


WANPIPE Release: 3.5.27 
i have updated the libsng_isdn to the newest driver as suggested, 




pcap sent to sangoma support is saying that when calls go through, i am NOT specifying a 'called number' 
but bad calls are having that 'called number' . 


now the problem is i am not really setting anything specific (e.g adding that 'called number' field to specific numbers when called) 








_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

CONFIDENTIAL NOTICE : If you have received this email in error, 
please immediately notify the sender by email at the address 
shown above. This email may contain confidential or legally 
privileged information that is intended only for the use of the 
individual or entity named in this email. If you are not the 
intended recipient, you are hereby notified that any 
disclosure, copying, distribution or reliance upon the contents 
of this email is strictly prohibited. Please delete from your 
files if you are not the intended recipient. Thank you for your 
compliance.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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disclosure, copying, distribution or reliance upon the contents 
of this email is strictly prohibited. Please delete from your 
files if you are not the intended recipient. Thank you for your 
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From nestor at tiendalinux.com  Tue Jun 19 04:46:41 2012
From: nestor at tiendalinux.com (Nestor A. Diaz)
Date: Mon, 18 Jun 2012 19:46:41 -0500
Subject: [Freeswitch-users] Help with video call ( h.264 codec )
In-Reply-To: <bbda0e29f3b1d099674e11dec9e62f9c@asterix.tiendalinux.com>
References: <bbda0e29f3b1d099674e11dec9e62f9c@asterix.tiendalinux.com>
Message-ID: <f147744c119dc8ea2b6fc65018ca4083@asterix.tiendalinux.com>

Just to add the freeswitch keep sending udp messages to this phone, but 
the phone never process those messages, i have configured the phone to 
log at the debug level and it logs every sip message, but none of the 
messages sent from freeswitch, i use tcpdump and confirmed the packages 
went out from FS and sent to the phone, but the phone never processed 
those messages ... well anyway those are grandstream phones, a total 
piece of junk.

This is the message FS keep sending .

send 1490 bytes to udp/[172.24.103.44]:5060 at 22:30:23.032784:
    
------------------------------------------------------------------------
    INVITE sip:215 at 172.24.103.44:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 172.24.107.1:5090;rport;branch=z9hG4bK8tZND2F7QccBQ
    Max-Forwards: 69
    From: "Ext. 216" <sip:216 at 172.24.107.1>;tag=cyQ3Fagm3FBvr
    To: <sip:215 at 172.24.103.44:5060;transport=udp>
    Call-ID: fff8c62a-3437-1230-1388-00224d56ac0c
    CSeq: 29693240 INVITE
    Contact: <sip:mod_sofia at 172.24.107.1:5090>
    User-Agent: 
FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120618T111915Z~11a745cf76+unclean~20120618T121534Z
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
    Supported: timer, precondition, path, replaces
    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, 
sla, include-session-description, presence.winfo, message-summary, refer
    Content-Type: application/sdp
    Content-Disposition: session
    Content-Length: 511
    X-FS-Support: update_display,send_info
    Remote-Party-ID: "Ext. 216" 
<sip:216 at 172.24.107.1>;party=calling;screen=yes;privacy=off

    v=0
    o=FreeSWITCH 1569664329 1569664330 IN IP4 172.24.107.1
    s=FreeSWITCH
    c=IN IP4 172.24.107.1
    t=0 0
    m=audio 12058 RTP/AVP 0 8 4 18 3 2 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-11
    a=ptime:20
    m=video 12044 RTP/AVP 99
    a=rtpmap:99 H264/90000
    a=fmtp:99 profile-level-id=428014; packetization-mode=0; 
sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg==
    a=framerate:15
    
------------------------------------------------------------------------

-- 
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57.1-485-3020 x 211
Cel. +57.316-227-3595
nestor at tiendalinux.com
www.tiendalinux.com


From chris at opencsta.org  Tue Jun 19 04:59:21 2012
From: chris at opencsta.org (Chris Mylonas)
Date: Tue, 19 Jun 2012 10:59:21 +1000
Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
In-Reply-To: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
References: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
Message-ID: <70B8EC54-1F06-47C3-959E-D4E6C909337B@opencsta.org>

Hi Terry,

I was having a hissy fit with a short while ago.  It was about libtool being a coit (au slang)

http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-OSX-Libtool-Macports-td7548764.html

Excuse the exhaustive gibber, the error messages will mean the most to anyone!

HTH,
Chris


On 19/06/2012, at 5:50 AM, Terry Barnum wrote:

> I recently started investigating FS and am curious about what the specific problem is with Macports and FS. I've been using Macports for a couple years and find it to be a great, actively maintained package manager that uses /opt/local, making it a very reliable way to keep production software working through Apple system updates. I've found the Macports team to be very responsive so if there are specific things that I can pass along, it might get fixed. Is there a link or thread that documents what's incompatible with FS?
> 
> Thanks,
> -Terry
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 zoho.com  Tue Jun 19 05:01:07 2012
From: freeswitch at zoho.com (dingdong)
Date: Mon, 18 Jun 2012 18:01:07 -0700
Subject: [Freeswitch-users] PRI + FreeTDM problems
In-Reply-To: <138021aa2d1.-4585004023753801168.3063868342642460335@zoho.com>
References: <138021aa2d1.-4585004023753801168.3063868342642460335@zoho.com>
Message-ID: <1380241dfc4.-7482549776433716373.-2496200701062631123@zoho.com>

heres another logs attached not messed up by the email formatting.





---- On Mon, 18 Jun 2012 17:18:16 -0700 dingdong&lt;freeswitch at zoho.com&gt; wrote ---- 


hello all,   i had this problem months ago and didn't really have the solution,just thinking that maybe someone from here can help.




our primary link for calling outside is a PRI.
so here's what we currently have, when we try to call 18778025246 and  1866-642-5246 - calls are not going through.


but when we call 18775682495 - calls go just fine.




attaching my freetdm.conf.xml too as the provider made me change some fields in there.






pasting the logs provided by our PRI provider telepacific.com,with comparisons for both numbers dialed and labelled as good and bad calls,
----------------------------------------------------------------------------------------------------------------------------------


GOOD ONE S2C4-1091895439 12-06-09 13:49:46 253961 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 3 Call Reference Value Orig 18 M 05 Message Type SETUP I 04 Bearer Capability Length = 3 80 Coding Standard CCITT Info Transfer Cap. Speech 90 Transfer Mode Circuit Transfer Rate 64 kbit/s A2 Layer 1 Protocol G.711 u-law I 18 Channel Identification Length = 3 A9 Interface ID Implicitly Defined Interface Type Primary S2C4-1091895439 12-06-09 13:49:46 253962 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 4 Preferred/Exclusive Exclusive D Channel Indicator Not D Channel Channel Selection As Indicated 83 Coding Standard CCITT Coding Type Number Channel Type B Channel 81 Channel Number 1 I 6C Calling Party Number Length = 13 00 Type of Number Unknown Type Numbering Plan Unknown Numbering Plan 81 Presentation Indicator Allowed S2C4-1091895439 12-06-09 13:49:46 253963 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 5 Screening Indicator User Provided - Passed Number 13231234567 I 70 Called Party Number Length = 12 80 Type of Number Unknown Type Numbering Plan Unknown Numbering Plan Number 18775682495 Raw Hex Data 08 02 00 12 05 04 03 80 90 A2 18 03 A9 83 81 6C 0D 00 81 31 33 32 33 34 35 34 34 30 36 30 70 0C 80 31 38 37 37 35 36 38 32 34 39 35 S2C4-1091895439 12-06-09 13:50:16 253972 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 6 ------------------------------------------------ Event 2 12/06/09 13:47:20 TEI 0 08 Q.931 Call Control Message Call Reference Value Dest 18 M 02 Message Type CALL PROCEEDING I 18 Channel Identification Length = 3 A9 Interface ID Implicitly Defined Interface Type Primary Preferred/Exclusive Exclusive D Channel Indicator Not D Channel Channel Selection As Indicated 83 Coding Standard CCITT Coding Type Number Channel Type B Channel S2C4-1091895439 12-06-09 13:50:17 253973 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 7 81 Channel Number 1 Raw Hex Data 08 02 80 12 02 18 03 A9 83 81 ------------------------------------------------ Event 3 12/06/09 13:47:21 TEI 0 08 Q.931 Call Control Message Call Reference Value Dest 18 M 03 Message Type PROGRESS I 1E Progress Indicator Length = 2 84 Coding Standard CCITT Location Remote Public Network 88 Value In-band Information/Pattern Avail S2C4-1091895439 12-06-09 13:50:17 253974 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 8 Raw Hex Data 08 02 80 12 03 1E 02 84 88 ------------------------------------------------ Event 4 12/06/09 13:47:21 TEI 0 08 Q.931 Call Control Message Call Reference Value Dest 18 M 07 Message Type CONNECT I 1E Progress Indicator Length = 2 84 Coding Standard CCITT Location Remote Public Network 82 Value Destination is not ISDN Raw Hex Data 08 02 80 12 07 1E 02 84 82 S2C4-1091895439 12-06-09 13:50:17 253975 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 9 ------------------------------------------------ Event 5 12/06/09 13:47:33 TEI 0 08 Q.931 Call Control Message Call Reference Value Orig 18 M 45 Message Type DISCONNECT I 08 Cause Length = 2 81 Coding Standard CCITT Location Local Private Network 90 Class Normal Event Value #16, Normal Call Clearing ------------------------------------------------ ------------------------------------------------ --------------------- BAD ONE S2C4-1091895439 12-06-09 13:50:48 253980 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 13 83 Coding Standard CCITT Coding Type Number Channel Type B Channel 81 Channel Number 1 I 6C Calling Party Number Length = 13 00 Type of Number Unknown Type Numbering Plan Unknown Numbering Plan 81 Presentation Indicator Allowed Screening Indicator User Provided - Passed Number 13231234567
 I 70 Called Party Number Length = 12 S2C4-1091895439 12-06-09 13:50:48 253981 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 14 80 Type of Number Unknown Type Numbering Plan Unknown Numbering Plan Number 18778025246 Raw Hex Data 08 02 00 13 05 04 03 80 90 A2 18 03 A9 83 81 6C 0D 00 81 31 33 32 33 34 35 34 34 30 36 30 70 0C 80 31 38 37 37 38 30 32 35 32 34 36 ------------------------------------------------ Event 9 12/06/09 13:48:05 TEI 0 08 Q.931 Call Control Message S2C4-1091895439 12-06-09 13:50:48 253982 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 15 Call Reference Value Dest 19 M 02 Message Type CALL PROCEEDING I 18 Channel Identification Length = 3 A9 Interface ID Implicitly Defined Interface Type Primary Preferred/Exclusive Exclusive D Channel Indicator Not D Channel Channel Selection As Indicated 83 Coding Standard CCITT Coding Type Number Channel Type B Channel 81 Channel Number 1 Raw Hex Data S2C4-1091895439 12-06-09 13:51:18 253993 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 16 08 02 80 13 02 18 03 A9 83 81 ------------------------------------------------ Event 10 12/06/09 13:48:05 TEI 0 08 Q.931 Call Control Message Call Reference Value Dest 19 M 03 Message Type PROGRESS I 1E Progress Indicator Length = 2 8A Coding Standard CCITT Location Beyond Interworking Network 81 Value Call is not End-to-End ISDN Raw Hex Data 08 02 80 13 03 1E 02 8A 81 ------------------------------------------------ S2C4-1091895439 12-06-09 13:51:19 253994 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 17 Event 11 12/06/09 13:48:19 TEI 0 08 Q.931 Call Control Message Call Reference Value Orig 19 M 45 Message Type DISCONNECT I 08 Cause Length = 2 81 Coding Standard CCITT Location Local Private Network 90 Class Normal Event Value #16, Normal Call Clearing Raw Hex Data S2C4-1091895439 12-06-09 13:51:19 253995 MAINT LSANCAGJDS4 M DUMP FILE ALL IN PROGRESS SEGMENT 18 08 02 00 13 45 08 02 81 90 ------------------------------------------------ 






please take note that 18778025246/1866-642-5246 goes thru our sip provider just fine. no problems at all. not with the PRI.
i pressume both numbers are toll free.



i had help from sangoma support no response yet with the 'called number' issue.


wanrouter version


WANPIPE Release: 3.5.27
 i have updated the libsng_isdn to the newest driver as suggested,




pcap sent to sangoma support is saying that when calls go through, i am NOT specifying a 'called number'
but bad calls are having that 'called number' .


now the problem is i am not really setting anything specific (e.g adding that 'called number' field to specific numbers when called)









_________________________________________________________________________ 
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org 
http://www.freeswitchsolutions.com 
 
 
 
 
Official FreeSWITCH Sites 
http://www.freeswitch.org 
http://wiki.freeswitch.org 
http://www.cluecon.com 
 
Join Us At ClueCon - Aug 7-9, 2012 
 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 




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GOOD ONE
 
 
 
 
   S2C4-1091895439 12-06-09 13:49:46 253961 MAINT LSANCAGJDS4
M  DUMP FILE ALL IN PROGRESS  SEGMENT 3
 
                              Call Reference Value              Orig        18
                         M 05 Message Type                               SETUP
                         I 04 Bearer Capability                   Length =   3
                           80   Coding Standard                          CCITT
                                Info Transfer Cap.                      Speech
                           90   Transfer Mode                          Circuit
                                Transfer Rate                        64 kbit/s
                           A2   Layer 1 Protocol                   G.711 u-law
                         I 18 Channel Identification              Length =   3
                           A9   Interface ID                Implicitly Defined
                                Interface Type                         Primary
 
 
   S2C4-1091895439 12-06-09 13:49:46 253962 MAINT LSANCAGJDS4
M  DUMP FILE ALL IN PROGRESS  SEGMENT 4
 
                                Preferred/Exclusive                  Exclusive
                                D Channel Indicator              Not D Channel
                                Channel Selection                 As Indicated
                           83   Coding Standard                          CCITT
                                Coding Type                             Number
                                Channel Type                         B Channel
                           81   Channel Number                               1
                         I 6C Calling Party Number                Length =  13
                           00   Type of Number                    Unknown Type
                                Numbering Plan          Unknown Numbering Plan
                           81   Presentation Indicator                 Allowed
 
 
   S2C4-1091895439 12-06-09 13:49:46 253963 MAINT LSANCAGJDS4
M  DUMP FILE ALL IN PROGRESS  SEGMENT 5
 
                                Screening Indicator     User Provided - Passed
                                Number                             13231234567
                         I 70 Called Party Number                 Length =  12
                           80   Type of Number                    Unknown Type
                                Numbering Plan          Unknown Numbering Plan
                                Number                             18775682495
 
                              Raw Hex Data
                              08 02 00 12 05 04 03 80 90 A2 18 03 A9 83 81 6C
                              0D 00 81 31 33 32 33 34 35 34 34 30 36 30 70 0C
                              80 31 38 37 37 35 36 38 32 34 39 35
 
 
   S2C4-1091895439 12-06-09 13:50:16 253972 MAINT LSANCAGJDS4
M  DUMP FILE ALL IN PROGRESS  SEGMENT 6
 
                              ------------------------------------------------
         Event 2  12/06/09 13:47:20  TEI 0
      08 Q.931 Call Control Message
         Call Reference Value              Dest        18
    M 02 Message Type                     CALL PROCEEDING
    I 18 Channel Identification              Length =   3
      A9   Interface ID                Implicitly Defined
           Interface Type                         Primary
           Preferred/Exclusive                  Exclusive
           D Channel Indicator              Not D Channel
           Channel Selection                 As Indicated
      83   Coding Standard                          CCITT
           Coding Type                             Number
           Channel Type                         B Channel
 
 
   S2C4-1091895439 12-06-09 13:50:17 253973 MAINT LSANCAGJDS4
M  DUMP FILE ALL IN PROGRESS  SEGMENT 7
 
      81   Channel Number                               1
 
         Raw Hex Data
         08 02 80 12 02 18 03 A9 83 81
         ------------------------------------------------
         Event 3  12/06/09 13:47:21  TEI 0
      08 Q.931 Call Control Message
         Call Reference Value              Dest        18
    M 03 Message Type                            PROGRESS
    I 1E Progress Indicator                  Length =   2
      84   Coding Standard                          CCITT
           Location                 Remote Public Network
      88   Value        In-band Information/Pattern Avail
 
 
 
   S2C4-1091895439 12-06-09 13:50:17 253974 MAINT LSANCAGJDS4
M  DUMP FILE ALL IN PROGRESS  SEGMENT 8
 
         Raw Hex Data
         08 02 80 12 03 1E 02 84 88
         ------------------------------------------------
         Event 4  12/06/09 13:47:21  TEI 0
      08 Q.931 Call Control Message
         Call Reference Value              Dest        18
    M 07 Message Type                             CONNECT
    I 1E Progress Indicator                  Length =   2
      84   Coding Standard                          CCITT
           Location                 Remote Public Network
      82   Value                  Destination is not ISDN
 
         Raw Hex Data
         08 02 80 12 07 1E 02 84 82
 
 
   S2C4-1091895439 12-06-09 13:50:17 253975 MAINT LSANCAGJDS4
M  DUMP FILE ALL IN PROGRESS  SEGMENT 9
 
         ------------------------------------------------
                              Event 5  12/06/09 13:47:33  TEI 0
                           08 Q.931 Call Control Message
                              Call Reference Value              Orig        18
                         M 45 Message Type                          DISCONNECT
                         I 08 Cause                               Length =   2
                           81   Coding Standard                          CCITT
                                Location                 Local Private Network
                           90   Class                             Normal Event
                                Value                #16, Normal Call Clearing
 
 
 
  ------------------------------------------------ ------------------------------------------------ ------------------------------------------------
 
BAD ONE
 
 
 
 
 
   S2C4-1091895439 12-06-09 13:50:48 253980 MAINT LSANCAGJDS4
M  DUMP FILE ALL IN PROGRESS  SEGMENT 13
 
                           83   Coding Standard                          CCITT
                                Coding Type                             Number
                                Channel Type                         B Channel
                           81   Channel Number                               1
                         I 6C Calling Party Number                Length =  13
                           00   Type of Number                    Unknown Type
                                Numbering Plan          Unknown Numbering Plan
                           81   Presentation Indicator                 Allowed
                                Screening Indicator     User Provided - Passed
                                Number                             13231234567
                         I 70 Called Party Number                 Length =  12
 
 
   S2C4-1091895439 12-06-09 13:50:48 253981 MAINT LSANCAGJDS4
M  DUMP FILE ALL IN PROGRESS  SEGMENT 14
 
                           80   Type of Number                    Unknown Type
                                Numbering Plan          Unknown Numbering Plan
                                Number                             18778025246
 
                              Raw Hex Data
                              08 02 00 13 05 04 03 80 90 A2 18 03 A9 83 81 6C
                              0D 00 81 31 33 32 33 34 35 34 34 30 36 30 70 0C
                              80 31 38 37 37 38 30 32 35 32 34 36
                              ------------------------------------------------
         Event 9  12/06/09 13:48:05  TEI 0
      08 Q.931 Call Control Message
 
 
   S2C4-1091895439 12-06-09 13:50:48 253982 MAINT LSANCAGJDS4
M  DUMP FILE ALL IN PROGRESS  SEGMENT 15
 
         Call Reference Value              Dest        19
    M 02 Message Type                     CALL PROCEEDING
    I 18 Channel Identification              Length =   3
      A9   Interface ID                Implicitly Defined
           Interface Type                         Primary
           Preferred/Exclusive                  Exclusive
           D Channel Indicator              Not D Channel
           Channel Selection                 As Indicated
      83   Coding Standard                          CCITT
           Coding Type                             Number
           Channel Type                         B Channel
      81   Channel Number                               1
 
         Raw Hex Data
 
 
   S2C4-1091895439 12-06-09 13:51:18 253993 MAINT LSANCAGJDS4
M  DUMP FILE ALL IN PROGRESS  SEGMENT 16
 
         08 02 80 13 02 18 03 A9 83 81
         ------------------------------------------------
         Event 10  12/06/09 13:48:05  TEI 0
      08 Q.931 Call Control Message
         Call Reference Value              Dest        19
    M 03 Message Type                            PROGRESS
    I 1E Progress Indicator                  Length =   2
      8A   Coding Standard                          CCITT
           Location           Beyond Interworking Network
      81   Value              Call is not End-to-End ISDN
 
         Raw Hex Data
         08 02 80 13 03 1E 02 8A 81
         ------------------------------------------------
 
 
   S2C4-1091895439 12-06-09 13:51:19 253994 MAINT LSANCAGJDS4
M  DUMP FILE ALL IN PROGRESS  SEGMENT 17
 
                              Event 11  12/06/09 13:48:19  TEI 0
                           08 Q.931 Call Control Message
                              Call Reference Value              Orig        19
                         M 45 Message Type                          DISCONNECT
                         I 08 Cause                               Length =   2
                           81   Coding Standard                          CCITT
                                Location                 Local Private Network
                           90   Class                             Normal Event
                                Value                #16, Normal Call Clearing
 
                              Raw Hex Data
 
 
   S2C4-1091895439 12-06-09 13:51:19 253995 MAINT LSANCAGJDS4
M  DUMP FILE ALL IN PROGRESS  SEGMENT 18
 
                              08 02 00 13 45 08 02 81 90
                              ------------------------------------------------
 

From terry at digital-outpost.com  Tue Jun 19 05:31:03 2012
From: terry at digital-outpost.com (Terry Barnum)
Date: Mon, 18 Jun 2012 18:31:03 -0700
Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
In-Reply-To: <0AEB7B3A-5591-4694-83B5-E4208D5DA4A0@dons.net.au>
References: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
	<0AEB7B3A-5591-4694-83B5-E4208D5DA4A0@dons.net.au>
Message-ID: <46D48718-2661-41EC-8A59-67BFEBCCF745@digital-outpost.com>


On Jun 18, 2012, at 4:23 PM, Daniel O'Connor wrote:

> On 19/06/2012, at 5:20, Terry Barnum wrote:
>> I recently started investigating FS and am curious about what the specific problem is with Macports and FS. I've been using Macports for a couple years and find it to be a great, actively maintained package manager that uses /opt/local, making it a very reliable way to keep production software working through Apple system updates. I've found the Macports team to be very responsive so if there are specific things that I can pass along, it might get fixed. Is there a link or thread that documents what's incompatible with FS?
> 
> There is no incompatibility per se - I use Macports and have installed FreeSwitch, although the only thing that FS actually uses from MP is libjpeg as it bundles everything else.
> 
> That said, I had trouble getting the bundled PortAudio working so I mangled the mod_portaudio Makefile.am to use the MP PortAudio and it worked fine.

That's really encouraging. Based on the threads I was reading it looked like people and the devs were saying that FS isn't happy with macports installed.

I've tried installing it on a PPC mac mini running 10.5.8 and an Intel i5 iMac running 10.6.8, both with macports installed. I've forgotten now at what build phase the mac mini failed on (I'm not in front of the machine). The iMac fails during make:

libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I.. -I/usr/include/libxml2 -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -msse2 -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DHAVE_VISIBILITY=1 -g -O2 -MT gsm0610_rpe.lo -MD -MP -MF .deps/gsm0610_rpe.Tpo -c gsm0610_rpe.c  -fno-common -DPIC -o gsm0610_rpe.o
/var/folders/B7/B7ea9Uvl2RWJ6k+1YxF5D++++TI/-Tmp-//ccF7gvqy.s:58:suffix or operands invalid for `lea'
/var/folders/B7/B7ea9Uvl2RWJ6k+1YxF5D++++TI/-Tmp-//ccF7gvqy.s:63:suffix or operands invalid for `movq'
make[7]: *** [gsm0610_rpe.lo] Error 1
make[6]: *** [all] Error 2
make[5]: *** [all-recursive] Error 1
make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la] Error 2
make[3]: *** [mod_spandsp-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2

libtool 2.4.2_2 is installed with macports. 
gcc is i686-apple-darwin10-llvm-gcc-4.2 (GCC) 4.2.1 (Based on Apple Inc. build 5658) (LLVM build 2336.1.00)

Any ideas?

Thanks,
-Terry

From terry at digital-outpost.com  Tue Jun 19 05:35:24 2012
From: terry at digital-outpost.com (Terry Barnum)
Date: Mon, 18 Jun 2012 18:35:24 -0700
Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
In-Reply-To: <70B8EC54-1F06-47C3-959E-D4E6C909337B@opencsta.org>
References: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
	<70B8EC54-1F06-47C3-959E-D4E6C909337B@opencsta.org>
Message-ID: <E83CB2E2-86C3-4ABF-BB1A-F9381EF580C7@digital-outpost.com>


On Jun 18, 2012, at 5:59 PM, Chris Mylonas wrote:

> Hi Terry,
> 
> I was having a hissy fit with a short while ago.  It was about libtool being a coit (au slang)
> 
> http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-OSX-Libtool-Macports-td7548764.html
> 
> Excuse the exhaustive gibber, the error messages will mean the most to anyone!
> 
> HTH,
> Chris
> 
> 
> On 19/06/2012, at 5:50 AM, Terry Barnum wrote:
> 
>> I recently started investigating FS and am curious about what the specific problem is with Macports and FS. I've been using Macports for a couple years and find it to be a great, actively maintained package manager that uses /opt/local, making it a very reliable way to keep production software working through Apple system updates. I've found the Macports team to be very responsive so if there are specific things that I can pass along, it might get fixed. Is there a link or thread that documents what's incompatible with FS?
>> 
>> Thanks,
>> -Terry

Hi Chris,

I saw your thread in the list archives and used the subject line for my email hoping that it would continue the discussion.  In that thread it sounds like you had to remove macports to get FS to install. Am I misreading that?

Thanks,
-Terry

From chris at opencsta.org  Tue Jun 19 05:55:24 2012
From: chris at opencsta.org (Chris Mylonas)
Date: Tue, 19 Jun 2012 11:55:24 +1000
Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
In-Reply-To: <E83CB2E2-86C3-4ABF-BB1A-F9381EF580C7@digital-outpost.com>
References: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
	<70B8EC54-1F06-47C3-959E-D4E6C909337B@opencsta.org>
	<E83CB2E2-86C3-4ABF-BB1A-F9381EF580C7@digital-outpost.com>
Message-ID: <A3FF85F4-E701-4D89-A26A-96706B611B92@opencsta.org>

> 
> Hi Chris,

Hi Terry!

> 
> I saw your thread in the list archives and used the subject line for my email hoping that it would continue the discussion.  In that thread it sounds like you had to remove macports to get FS to install. Am I misreading that?

I can't believe you just used it like that!  How dare you :P
OTOH, good thinking.  I nearly didn't read your email but Daniel's response caught my eye.  I remember being able to do that and that's what got me started (I needed to get FS running fast) - and that's macports got the boot unforch.

If you've got good karma with the macports crew then i can certainly muck things up on my laptop again, i know how to do a 180 to get out of it now :)
It will have to wait until July though.

Cheers
Chris

From darius at dons.net.au  Tue Jun 19 09:26:48 2012
From: darius at dons.net.au (Daniel O'Connor)
Date: Tue, 19 Jun 2012 14:56:48 +0930
Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
In-Reply-To: <46D48718-2661-41EC-8A59-67BFEBCCF745@digital-outpost.com>
References: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
	<0AEB7B3A-5591-4694-83B5-E4208D5DA4A0@dons.net.au>
	<46D48718-2661-41EC-8A59-67BFEBCCF745@digital-outpost.com>
Message-ID: <31354FBA-5DBC-47A9-9958-9EC8A515DACD@dons.net.au>


On 19/06/2012, at 11:01, Terry Barnum wrote:
>> That said, I had trouble getting the bundled PortAudio working so I mangled the mod_portaudio Makefile.am to use the MP PortAudio and it worked fine.
> 
> That's really encouraging. Based on the threads I was reading it looked like people and the devs were saying that FS isn't happy with macports installed.
> 
> I've tried installing it on a PPC mac mini running 10.5.8 and an Intel i5 iMac running 10.6.8, both with macports installed. I've forgotten now at what build phase the mac mini failed on (I'm not in front of the machine). The iMac fails during make:
> 
> libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I.. -I/usr/include/libxml2 -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -msse2 -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DHAVE_VISIBILITY=1 -g -O2 -MT gsm0610_rpe.lo -MD -MP -MF .deps/gsm0610_rpe.Tpo -c gsm0610_rpe.c  -fno-common -DPIC -o gsm0610_rpe.o
> /var/folders/B7/B7ea9Uvl2RWJ6k+1YxF5D++++TI/-Tmp-//ccF7gvqy.s:58:suffix or operands invalid for `lea'
> /var/folders/B7/B7ea9Uvl2RWJ6k+1YxF5D++++TI/-Tmp-//ccF7gvqy.s:63:suffix or operands invalid for `movq'
> Any ideas?

I have an Intel mac and configured with..

env CPPFLAGS=-I/opt/local/include LDFLAGS=-L/opt/local/lib ./configure

I am running Lion which uses clang and I think that in this case spandsp doesn't use any asm which is why it probably works.

I don't know why the asm wouldn't work in your case though.

You can probably work around it by applying this patch..
--- a/libs/spandsp/src/gsm0610_rpe.c
+++ b/libs/spandsp/src/gsm0610_rpe.c
@@ -59,7 +59,7 @@
 static void weighting_filter(int16_t x[40],
                              const int16_t *e)      // signal [-5..0.39.44] IN)
 {
-#if defined(__GNUC__)  &&  defined(SPANDSP_USE_MMX)  &&  defined(__x86_64__)
+#if 0 && defined(__GNUC__)  &&  defined(SPANDSP_USE_MMX)  &&  defined(__x86_64__)
     /* Table 4.4   Coefficients of the weighting filter */
     /* This must be padded to a multiple of 4 for MMX to work */
     static const union
@@ -114,7 +114,7 @@ static void weighting_filter(int16_t x[40],
         : "c" (e), "D" (x), [gsm_H] "X" (gsm_H)
         : "rax", "rdx", "rsi", "memory"
     );
-#elif defined(__GNUC__)  &&  defined(SPANDSP_USE_MMX)  &&  defined(__i386__)
+#elif 0 && defined(__GNUC__)  &&  defined(SPANDSP_USE_MMX)  &&  defined(__i386__)
     /* Table 4.4   Coefficients of the weighting filter */
     /* This must be padded to a multiple of 4 for MMX to work */
     static const union

ie disable the asm version and use the C one.

There are probably other places where you will gave to do this though :(

--
Daniel O'Connor software and network engineer
for Genesis Software - http://www.gsoft.com.au
"The nice thing about standards is that there
are so many of them to choose from."
  -- Andrew Tanenbaum
GPG Fingerprint - 5596 B766 97C0 0E94 4347 295E E593 DC20 7B3F CE8C








From curriegrad2004 at gmail.com  Tue Jun 19 10:29:28 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 18 Jun 2012 23:29:28 -0700
Subject: [Freeswitch-users] Gtalk does not ring local phones when making
 outbound calls
In-Reply-To: <1340052514.70031.YahooMailClassic@web161501.mail.bf1.yahoo.com>
References: <1340052514.70031.YahooMailClassic@web161501.mail.bf1.yahoo.com>
Message-ID: <CAMWiMy8znR9E7y=vAHjLuYmyfg6EvJD89=fRTApfOjakNpfP0A@mail.gmail.com>

Do you mean a ringback when you're calling out via google voice?

On Mon, Jun 18, 2012 at 1:48 PM, mtle <getj2k at yahoo.com> wrote:

> Hi,
> I have successfully compiled and installed FS from the latest git as at
> today.  Everything seems OK.  The only issue is that I do not get any rings
> at all with gtalk. I wonder if anyone have had this problem lately and how
> to fix that? Here's my dialplan.
>
> *<extension name="gv_out">*
> *    <condition field="destination_number" expression="^1?(\d{10})$">*
> *        <action application="set" data="hangup_after_bridge=true"/>*
> *        <action application="ring_ready"/>*
> *       <!--<action application="set"
> data="bridge_generate_comfort_noise=true"/>-->*
> *       <!--<action application="set"
> data="execute_on_answer=start_dtmf_generate"/>-->*
> *        <action application="bridge" data="dingaling/gtalk/+1$
> 1 at voice.google.com"/>*
> *    </condition>*
> *</extension>*
>
> Thanks.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hkalyoncu at gmail.com  Tue Jun 19 12:10:53 2012
From: hkalyoncu at gmail.com (huseyin kalyoncu)
Date: Tue, 19 Jun 2012 11:10:53 +0300
Subject: [Freeswitch-users] Actions before CDR writing (b leg only)
In-Reply-To: <BLU162-W235003A2E9EA0381B1212288F80@phx.gbl>
References: <BLU162-W235003A2E9EA0381B1212288F80@phx.gbl>
Message-ID: <CAPf+86VQkOdvXr4Zs+n+53esLG3LSYmO8hybZGwXtoPbPbPOJQ@mail.gmail.com>

arent those two same?

since you set hangup_after_bridge=true, at the end you will have only one
bridge.
so what difference does it make in the CDR?

On Mon, Jun 18, 2012 at 9:15 PM, awais nazir <awais-nazeer at hotmail.com>wrote:

>  Hello Guys
>
> I am using following scenario with series of attempts on gateways on
> failover
>
> <action application="set" data="hangup_after_bridge=true"/>
>     <action application="bridge" data="sofia/local_profile/1111**
> @example1.company.com <1111 at example1.company.com>" />
>     <action application="bridge" data="sofia/local_profile/1111**
> @example2.company.com <1111 at example2.company.com>" />
>
>
> I want to do some actions like setting some variables before CDR (leg B)
> is written like
>
>
> <action application="set" data="hangup_after_bridge=
> **true"/>
>     <action application="bridge" data="sofia/local_profile/1111**
> @example1.company.com <1111 at example1.company.com>" />
> <!-- do some actions-->
> <!-- write CDR-->
>     <action application="bridge" data="sofia/local_profile/1111**
> @example2.company.com <1111 at example2.company.com>" />
> <!-- do some actions-->
> <!-- write CDR-->
> .
> .
> .
> .
>
> Can anybody help me here.
>
> Waisee----
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From alex at thewinelake.com  Tue Jun 19 12:14:19 2012
From: alex at thewinelake.com (Alex)
Date: Tue, 19 Jun 2012 09:14:19 +0100
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <CALNZuwkBOwpBjuKW5kwm7tHfH5VOtGsC5Tj3pHAB8UuwjAjE3A@mail.gmail.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
	<CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
	<4FDF52FA.2070007@thewinelake.com>
	<CALNZuwmhmW7Yp1owGtLK+z8EhG=URxn7c0YpZtkyZvFU24SNMA@mail.gmail.com>
	<4FDF5D9D.4060707@thewinelake.com>
	<4FDF6038.4020606@thewinelake.com>
	<CALNZuwkBOwpBjuKW5kwm7tHfH5VOtGsC5Tj3pHAB8UuwjAjE3A@mail.gmail.com>
Message-ID: <4FE034DB.6070002@thewinelake.com>

Yes, but this works:

function getDestNumber()
         session:execute("log", "INFO mcm2.lua getDestNumber")
         destnum = session:playAndGetDigits(3, 20, 3, 5000, "#", 
"/home/oev/fs-pabx/prompts/dmsb12_mcm3.wav", "", "\\d+\\*?")
         l = string.len(destnum)
         if string.sub(destnum,l)=="*" then
                 destnum = ""
         end
         return destnum
end

Unfortunately, it still doesn't quite work as I'd like. The * doesn't 
act as a "remembered but terminating" key, so the customer still has to 
press # (or, presumably, wait 5s) after pressing * for the message to 
get through!

Is there a DTMF history buffer one can inspect? Then I could make * a 
terminating key and then just see if that was the last DTMF.

Alex



> I didn't need to do any such thing... Did you get an error using 
> myString:len()?
> -Avi
>
>
> On Mon, Jun 18, 2012 at 8:07 PM, Alex <alex at thewinelake.com 
> <mailto:alex at thewinelake.com>> wrote:
>
>     OK, so replying to myself, I think I need to include the string
>     library... I can see in lua.conf.xml that one can have a
>     "module-directory" but I'm not sure where I should get these
>     modules from. Any recommended quick reading on this?
>>     More lua trouble - real beginner stuff, I'm afraid.
>>
>>     Why does this not work:
>>
>>             destnum = session:playAndGetDigits(3, 20, 3, 5000, "#",
>>     "/home/oev/fs-pabx/prompts/dmsb12_mcm3.wav", "", "\\d+\\*")
>>             l = strlen(destnum)
>>
>>     The objection is
>>
>>
>>     2012-06-18 16:52:30.415261 [ERR] mod_lua.cpp:198
>>     /usr/local/freeswitch/scripts/mcm2.lua:13: attempt to call global
>>     'strlen' (a nil value)
>>     stack traceback:
>>             /usr/local/freeswitch/scripts/mcm2.lua:13: in function
>>     'getDestNumber'
>>             /usr/local/freeswitch/scripts/mcm2.lua:49: in function
>>     'unattendedTransfer'
>>             /usr/local/freeswitch/scripts/mcm2.lua:24: in function
>>     'processMcmOption'
>>     /usr/local/freeswitch/scripts/mcm2.lua:153: in main chunk
>>
>>     I was just assuming that lua is lua and that anything in the lua
>>     wiki (eg. http://www.lua.org/manual/2.4/node32.html#exstring)
>>     would work.
>>
>>>     Phrases let you mix and match speak, say, play-file, silence,
>>>     other macros, and even passing in options.
>>>
>>>     http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_phrase
>>>     http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
>>>
>>>     -Avi
>>>
>>>
>>>     _________________________________________________________________________
>>>     Professional FreeSWITCH Consulting Services:
>>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>>     http://www.freeswitchsolutions.com
>>>
>>>     
>>>     
>>>
>>>     Official FreeSWITCH Sites
>>>     http://www.freeswitch.org
>>>     http://wiki.freeswitch.org
>>>     http://www.cluecon.com
>>>
>>>     Join Us At ClueCon - Aug 7-9, 2012
>>>
>>>     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
>>>
>>>
>>>     No virus found in this message.
>>>     Checked by AVG - www.avg.com <http://www.avg.com>
>>>     Version: 2012.0.2177 / Virus Database: 2433/5076 - Release Date:
>>>     06/17/12
>>>
>>
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.com
>>
>>     Join Us At ClueCon - Aug 7-9, 2012
>>
>>     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
>>
>>
>>     No virus found in this message.
>>     Checked by AVG - www.avg.com <http://www.avg.com>
>>     Version: 2012.0.2177 / Virus Database: 2437/5077 - Release Date:
>>     06/18/12 
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2437/5078 - Release Date: 06/18/12
>


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From hkalyoncu at gmail.com  Tue Jun 19 12:18:37 2012
From: hkalyoncu at gmail.com (huseyin kalyoncu)
Date: Tue, 19 Jun 2012 11:18:37 +0300
Subject: [Freeswitch-users] Actions before CDR writing (b leg only)
In-Reply-To: <CAPf+86VQkOdvXr4Zs+n+53esLG3LSYmO8hybZGwXtoPbPbPOJQ@mail.gmail.com>
References: <BLU162-W235003A2E9EA0381B1212288F80@phx.gbl>
	<CAPf+86VQkOdvXr4Zs+n+53esLG3LSYmO8hybZGwXtoPbPbPOJQ@mail.gmail.com>
Message-ID: <CAPf+86VeAyE5OsDYTbybMJX1K7CWMHcDKs+mv-AOFeiMoCpMig@mail.gmail.com>

i forgot to mention that i think you only need to configure log-b-leg=true
in xml_cdr.conf

On Tue, Jun 19, 2012 at 11:10 AM, huseyin kalyoncu <hkalyoncu at gmail.com>wrote:

> arent those two same?
>
> since you set hangup_after_bridge=true, at the end you will have only one
> bridge.
> so what difference does it make in the CDR?
>
> On Mon, Jun 18, 2012 at 9:15 PM, awais nazir <awais-nazeer at hotmail.com>wrote:
>
>>  Hello Guys
>>
>> I am using following scenario with series of attempts on gateways on
>> failover
>>
>> <action application="set" data="hangup_after_bridge=true"/>
>>     <action application="bridge" data="sofia/local_profile/1111**
>> @example1.company.com <1111 at example1.company.com>" />
>>     <action application="bridge" data="sofia/local_profile/1111**
>> @example2.company.com <1111 at example2.company.com>" />
>>
>>
>> I want to do some actions like setting some variables before CDR (leg B)
>> is written like
>>
>>
>> <action application="set" data="hangup_after_bridge=
>> **true"/>
>>     <action application="bridge" data="sofia/local_profile/1111**
>> @example1.company.com <1111 at example1.company.com>" />
>> <!-- do some actions-->
>> <!-- write CDR-->
>>     <action application="bridge" data="sofia/local_profile/1111**
>> @example2.company.com <1111 at example2.company.com>" />
>> <!-- do some actions-->
>> <!-- write CDR-->
>> .
>> .
>> .
>> .
>>
>> Can anybody help me here.
>>
>> Waisee----
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From avi at avimarcus.net  Tue Jun 19 12:49:59 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 19 Jun 2012 11:49:59 +0300
Subject: [Freeswitch-users] lua trouble
In-Reply-To: <4FE034DB.6070002@thewinelake.com>
References: <4FDF3541.4010808@thewinelake.com>
	<CAKzWOxVP4crMTj4B8HLMq9AURzHHXgm1+W82F0p0u6FynNs04w@mail.gmail.com>
	<CALNZuwnQQc2R2i+4+VJXf03oFtHAQ2QX3P-z1idfRnVpT_LY0A@mail.gmail.com>
	<CAKzWOxXpwFjq=1ucaUzLDu5=jefd8zKBjG-7_D2cpU0aS3OP=g@mail.gmail.com>
	<CAM9Gy+9GR+pFEzBu--x9_MaLtJYGKEtNz7H9QGiyXN0YTTXpLA@mail.gmail.com>
	<4FDF52FA.2070007@thewinelake.com>
	<CALNZuwmhmW7Yp1owGtLK+z8EhG=URxn7c0YpZtkyZvFU24SNMA@mail.gmail.com>
	<4FDF5D9D.4060707@thewinelake.com> <4FDF6038.4020606@thewinelake.com>
	<CALNZuwkBOwpBjuKW5kwm7tHfH5VOtGsC5Tj3pHAB8UuwjAjE3A@mail.gmail.com>
	<4FE034DB.6070002@thewinelake.com>
Message-ID: <CALNZuwnmpUUh+PUmgptyJU64818kVxkgs4yawLexASTANDiiwg@mail.gmail.com>

Maybe try: http://wiki.freeswitch.org/wiki/Mod_lua#session:setInputCallback
-Avi


On Tue, Jun 19, 2012 at 11:14 AM, Alex <alex at thewinelake.com> wrote:

>  Yes, but this works:
>
> function getDestNumber()
>         session:execute("log", "INFO mcm2.lua getDestNumber")
>         destnum = session:playAndGetDigits(3, 20, 3, 5000, "#",
> "/home/oev/fs-pabx/prompts/dmsb12_mcm3.wav", "", "\\d+\\*?")
>         l = string.len(destnum)
>         if string.sub(destnum,l)=="*" then
>                 destnum = ""
>         end
>         return destnum
> end
>
> Unfortunately, it still doesn't quite work as I'd like. The * doesn't act
> as a "remembered but terminating" key, so the customer still has to press #
> (or, presumably, wait 5s) after pressing * for the message to get through!
>
> Is there a DTMF history buffer one can inspect? Then I could make * a
> terminating key and then just see if that was the last DTMF.
>
> Alex
>
>
>
>  I didn't need to do any such thing... Did you get an error using
> myString:len()?
> -Avi
>
>
>  On Mon, Jun 18, 2012 at 8:07 PM, Alex <alex at thewinelake.com> wrote:
>
>>  OK, so replying to myself, I think I need to include the string
>> library... I can see in lua.conf.xml that one can have a "module-directory"
>> but I'm not sure where I should get these modules from. Any recommended
>> quick reading on this?
>>
>>  More lua trouble - real beginner stuff, I'm afraid.
>>
>> Why does this not work:
>>
>>         destnum = session:playAndGetDigits(3, 20, 3, 5000, "#",
>> "/home/oev/fs-pabx/prompts/dmsb12_mcm3.wav", "", "\\d+\\*")
>>         l = strlen(destnum)
>>
>> The objection is
>>
>>
>> 2012-06-18 16:52:30.415261 [ERR] mod_lua.cpp:198
>> /usr/local/freeswitch/scripts/mcm2.lua:13: attempt to call global 'strlen'
>> (a nil value)
>> stack traceback:
>>         /usr/local/freeswitch/scripts/mcm2.lua:13: in function
>> 'getDestNumber'
>>         /usr/local/freeswitch/scripts/mcm2.lua:49: in function
>> 'unattendedTransfer'
>>         /usr/local/freeswitch/scripts/mcm2.lua:24: in function
>> 'processMcmOption'
>>         /usr/local/freeswitch/scripts/mcm2.lua:153: in main chunk
>>
>> I was just assuming that lua is lua and that anything in the lua wiki
>> (eg. http://www.lua.org/manual/2.4/node32.html#exstring) would work.
>>
>>  Phrases let you mix and match speak, say, play-file, silence, other
>> macros, and even passing in options.
>>
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_phrase
>> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
>>
>> -Avi
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> No virus found in this message.
>> Checked by AVG - www.avg.com
>> Version: 2012.0.2177 / Virus Database: 2433/5076 - Release Date: 06/17/12
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>>   No virus found in this message.
>> Checked by AVG - www.avg.com
>>  Version: 2012.0.2177 / Virus Database: 2437/5077 - Release Date:
>> 06/18/12
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2177 / Virus Database: 2437/5078 - Release Date: 06/18/12
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From darius at dons.net.au  Tue Jun 19 14:39:05 2012
From: darius at dons.net.au (Daniel O'Connor)
Date: Tue, 19 Jun 2012 20:09:05 +0930
Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
In-Reply-To: <31354FBA-5DBC-47A9-9958-9EC8A515DACD@dons.net.au>
References: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
	<0AEB7B3A-5591-4694-83B5-E4208D5DA4A0@dons.net.au>
	<46D48718-2661-41EC-8A59-67BFEBCCF745@digital-outpost.com>
	<31354FBA-5DBC-47A9-9958-9EC8A515DACD@dons.net.au>
Message-ID: <7A8C8833-8A47-4577-8BFF-123A1DF47EF2@dons.net.au>


On 19/06/2012, at 14:56, Daniel O'Connor wrote:
> ie disable the asm version and use the C one.
> 
> There are probably other places where you will gave to do this though :(


I suppose the correct way is to tell configure to disable MMX etc, however I am not sure which knobs to use for that and how to get them to spandsp's configure script :(

--
Daniel O'Connor software and network engineer
for Genesis Software - http://www.gsoft.com.au
"The nice thing about standards is that there
are so many of them to choose from."
  -- Andrew Tanenbaum
GPG Fingerprint - 5596 B766 97C0 0E94 4347 295E E593 DC20 7B3F CE8C








From dgarcia at anew.com.ve  Tue Jun 19 16:38:50 2012
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Tue, 19 Jun 2012 08:08:50 -0430
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <CAKzWOxWR6PJqPb6NqTLRQa5WMGP6H4zVte1w0i2UZSDN9+fQMg@mail.gmail.com>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
	<4FDFBCA8.1050700@quentustech.com>
	<CAKzWOxWR6PJqPb6NqTLRQa5WMGP6H4zVte1w0i2UZSDN9+fQMg@mail.gmail.com>
Message-ID: <4FE072DA.3040903@anew.com.ve>

Hi Everybody,

Just to be fare, is good to make FS more improved and standardised.It is 
better if in the wiki the devolper or someone involved with rpm update 
the wiki with a "change log", in this way everyone will know what is 
different. This will helpful more than you can expect for FS community.

Shadowym, if you have decided to use FS RPM, you should get more info 
about it and what to expect.

It is just my opinion

On 6/18/2012 8:13 PM, Michael Collins wrote:
>
>
> On Mon, Jun 18, 2012 at 4:41 PM, William King 
> <william.king at quentustech.com <mailto:william.king at quentustech.com>> 
> wrote:
>
>     I believe there was an announcement on this mailing list alerting
>     that the new rpms would be undergoing major changes to include
>     moving the directory locations. Search this list for new threads
>     started by SwK in the past 6 months.
>
>
> There was indeed an announcement by SwK himself. It was as close to 
> FNLoIP (Flashing Neon Lights over IP) as you can get.
> -MC
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2178 / Virus Database: 2437/5079 - Release Date: 06/19/12
>


-- 
Atentamente,
*Dario Garc?a*
Consultor.

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Tel?fono: +58 212 9081842
Cel: +58 412 2221515
dgarcia at anew.com.ve
http://www.anew.com.ve
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From avi at avimarcus.net  Tue Jun 19 16:50:25 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 19 Jun 2012 15:50:25 +0300
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <4FE072DA.3040903@anew.com.ve>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
	<4FDFBCA8.1050700@quentustech.com>
	<CAKzWOxWR6PJqPb6NqTLRQa5WMGP6H4zVte1w0i2UZSDN9+fQMg@mail.gmail.com>
	<4FE072DA.3040903@anew.com.ve>
Message-ID: <CALNZuwnHU7+vbVre0kmrk+P_BU40AorL+4_sDA6-d6f4rLkoew@mail.gmail.com>

A changelog can often be ignored. A "breaking changes" or "not backward
compatible changes" log would be helpful for such things.
Huh, if those changes get tagged in GIT I think they automatically show up
if you're paying attention when doing a pull.
-Avi


On Tue, Jun 19, 2012 at 3:38 PM, Saugort Dario Garcia Tovar <
dgarcia at anew.com.ve> wrote:

>  Hi Everybody,
>
> Just to be fare, is good to make FS more improved and standardised.It is
> better if in the wiki the devolper or someone involved with rpm update the
> wiki with a "change log", in this way everyone will know what is different.
> This will helpful more than you can expect for FS community.
>
> Shadowym, if you have decided to use FS RPM, you should get more info
> about it and what to expect.
>
> It is just my opinion
>
>
> On 6/18/2012 8:13 PM, Michael Collins wrote:
>
>
>
> On Mon, Jun 18, 2012 at 4:41 PM, William King <
> william.king at quentustech.com> wrote:
>
>>  I believe there was an announcement on this mailing list alerting that
>> the new rpms would be undergoing major changes to include moving the
>> directory locations. Search this list for new threads started by SwK in the
>> past 6 months.
>>
>
>  There was indeed an announcement by SwK himself. It was as close to
> FNLoIP (Flashing Neon Lights over IP) as you can get.
> -MC
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2178 / Virus Database: 2437/5079 - Release Date: 06/19/12
>
>
>
> --
> Atentamente,
> *Dario Garc?a*
> Consultor.
>
> CCCT, Nivel C2, Sector Yarey, Mz,
> Ofc. MZ03a.
> Caracas-Venezuela.
> Tel?fono: +58 212 9081842
> Cel: +58 412 2221515
> dgarcia at anew.com.ve
> http://www.anew.com.ve
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jbr at consiglia.dk  Tue Jun 19 16:58:43 2012
From: jbr at consiglia.dk (Jon Bruel)
Date: Tue, 19 Jun 2012 14:58:43 +0200
Subject: [Freeswitch-users] sendevent NOTIFY
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE014608F5348C@SBS2008SERVER.consiglia.local>
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE014608F53494@SBS2008SERVER.consiglia.local>

I have partially solved the issue.

In the dialplan I use a execute_on_ring to run:

    <action application="event" data="Event-Name=NOTIFY,uuid=${uuid},event-string=talk,content-type=application/simple-message-summary"/>

This fires a notify to the phone called, which has nearly the right SIP to- and from-headers.

Unfortunately the tag=xxxx in the to-header is missing. This tag should be the same as the tag in the SIP RINGING response from the phone, but somehow it is not passed on to the NOTIFY.

The result is that the phone cannot determine which call should be answered, and it does not go off-hook.

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jon Bruel
Sent: 18. juni 2012 15:39
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] sendevent NOTIFY

After some time working with BroadSoft, I have returned to the FreeSWITCH, and I'm impressed with the overall improvements in consistency and documentation. Great work!

I have tested the Event Socket sendevent NOTIFY command, and I have tried to check if I could change the SIP From and SIP To headers to suit my requirements. It looks as if there is no way.

So is there a way to specify the From- and To-headers, including the associated tags?

I need this to be able to control the phone sets (set it off-hook).

Thanks /Jon

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From alex at thewinelake.com  Tue Jun 19 17:37:21 2012
From: alex at thewinelake.com (Alex)
Date: Tue, 19 Jun 2012 14:37:21 +0100
Subject: [Freeswitch-users] Interrupting "speak"
Message-ID: <4FE08091.7090005@thewinelake.com>

Is there a way such that if my script is "speak"ing to a user that they 
input DTMF and the speak is terminated immediately?


From avi at avimarcus.net  Tue Jun 19 17:40:16 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 19 Jun 2012 16:40:16 +0300
Subject: [Freeswitch-users] Interrupting "speak"
In-Reply-To: <4FE08091.7090005@thewinelake.com>
References: <4FE08091.7090005@thewinelake.com>
Message-ID: <CALNZuw=JSV13Dz_Au3sKj4Wvf=bLMQ4VGRvdfKLwLEd-C977yA@mail.gmail.com>

Use speak as part of a phrase in play and get digits :)
-Avi


On Tue, Jun 19, 2012 at 4:37 PM, Alex <alex at thewinelake.com> wrote:

> Is there a way such that if my script is "speak"ing to a user that they
> input DTMF and the speak is terminated immediately?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From awais-nazeer at hotmail.com  Tue Jun 19 17:51:58 2012
From: awais-nazeer at hotmail.com (awais nazir)
Date: Tue, 19 Jun 2012 18:51:58 +0500
Subject: [Freeswitch-users] channel variables in show calls api
Message-ID: <BLU162-W21CA24E8CDB1E1F93AF8EF88FF0@phx.gbl>


Hello Guys

Is it possible to do some customization of "show calls" api to show some more channel variables like accountcode?

--waisee
 		 	   		  
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From Hector.Geraldino at ipsoft.com  Tue Jun 19 17:52:19 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Tue, 19 Jun 2012 09:52:19 -0400
Subject: [Freeswitch-users] Interrupting "speak"
In-Reply-To: <CALNZuw=JSV13Dz_Au3sKj4Wvf=bLMQ4VGRvdfKLwLEd-C977yA@mail.gmail.com>
References: <4FE08091.7090005@thewinelake.com>
	<CALNZuw=JSV13Dz_Au3sKj4Wvf=bLMQ4VGRvdfKLwLEd-C977yA@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD022F4EBA6E@NY1-EXMB-01.ip-soft.net>

Or send a "break" command after receiving the DTMF event, when using ESL.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus
Sent: Tuesday, June 19, 2012 9:40 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Interrupting "speak"

Use speak as part of a phrase in play and get digits :)
-Avi

On Tue, Jun 19, 2012 at 4:37 PM, Alex <alex at thewinelake.com<mailto:alex at thewinelake.com>> wrote:
Is there a way such that if my script is "speak"ing to a user that they
input DTMF and the speak is terminated immediately?

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

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From jaasmailing at gmail.com  Tue Jun 19 18:13:07 2012
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Tue, 19 Jun 2012 16:13:07 +0200
Subject: [Freeswitch-users] Spandsp - Cannot send non-existant fax file
	(permissions?)
Message-ID: <4FE088F3.1060207@gmail.com>

Hi all,

I'm in trouble with mod_spandsp and freeswitch executed by user 
"freeswitch".
Even if the file is present and the permissions are ok (readable by 
everyone), the FS console gets:

2012-06-19 16:06:30.098112 [ERR] mod_spandsp_fax.c:1289 Cannot send 
non-existant fax file [/usr/local/freeswitch/storage/fax/test_fax.tif]

FS is able to send if I execute the main process from user "root".


Does spandsp require other special permissions?

Regards,
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From alex at thewinelake.com  Tue Jun 19 19:30:43 2012
From: alex at thewinelake.com (Alex Lake)
Date: Tue, 19 Jun 2012 16:30:43 +0100
Subject: [Freeswitch-users] Interrupting "speak"
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD022F4EBA6E@NY1-EXMB-01.ip-soft.net>
References: <4FE08091.7090005@thewinelake.com>
	<CALNZuw=JSV13Dz_Au3sKj4Wvf=bLMQ4VGRvdfKLwLEd-C977yA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD022F4EBA6E@NY1-EXMB-01.ip-soft.net>
Message-ID: <E349E358-A885-4F70-83E6-D153A6B3184C@digitalmail.com>

Ooooh, that's interesting!
However, I think Avi's right and I'll have to bite the bullet on phrases....



On 19 Jun 2012, at 14:52, Hector Geraldino <Hector.Geraldino at ipsoft.com> wrote:

> Or send a ?break? command after receiving the DTMF event, when using ESL.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus
> Sent: Tuesday, June 19, 2012 9:40 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Interrupting "speak"
>  
> Use speak as part of a phrase in play and get digits :)
> -Avi
>  
> 
> On Tue, Jun 19, 2012 at 4:37 PM, Alex <alex at thewinelake.com> wrote:
> Is there a way such that if my script is "speak"ing to a user that they
> input DTMF and the speak is terminated immediately?
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From ronmccar at gmail.com  Tue Jun 19 19:37:53 2012
From: ronmccar at gmail.com (Ron McCarthy)
Date: Tue, 19 Jun 2012 08:37:53 -0700
Subject: [Freeswitch-users] Multicast RTP collector
Message-ID: <CAMQkVDKhLCPz4o2neLcDeXErFLLZtgt0BNe0XbnjqYpmDcX2fQ@mail.gmail.com>

Can freeswitch subscribe to a multicast group and when it gets a stream
connect that to another channel? Basically looking for a RTP to SIP bridge,
I know a little out of the ordinary :)

So far ive been able to find nothing, but figured it's worth checking.

Thanks
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From jaasmailing at gmail.com  Tue Jun 19 19:38:32 2012
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Tue, 19 Jun 2012 17:38:32 +0200
Subject: [Freeswitch-users] Problem with mod_say_it
Message-ID: <4FE09CF8.5050106@gmail.com>

Hi,

I'm experiencing the following problem with italian language in freeswitch:

2012-06-19 17:34:13.998100 [DEBUG] switch_ivr_play_say.c:67 No language 
specified - Using [it]
2012-06-19 17:34:13.998100 [DEBUG] switch_ivr_play_say.c:244 Handle 
say:[1340118718] (it:it)
2012-06-19 17:34:13.998100 [ERR] mod_say_it.c:446 Unknown Say type=[18]

With mod_say_en I have no problem.
Is it possible to solve this?


Best regards,
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From avi at avimarcus.net  Tue Jun 19 19:38:57 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 19 Jun 2012 18:38:57 +0300
Subject: [Freeswitch-users] Interrupting "speak"
In-Reply-To: <E349E358-A885-4F70-83E6-D153A6B3184C@digitalmail.com>
References: <4FE08091.7090005@thewinelake.com>
	<CALNZuw=JSV13Dz_Au3sKj4Wvf=bLMQ4VGRvdfKLwLEd-C977yA@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD022F4EBA6E@NY1-EXMB-01.ip-soft.net>
	<E349E358-A885-4F70-83E6-D153A6B3184C@digitalmail.com>
Message-ID: <CALNZuwmT3nCGSmVAjJqV3dyU_NvFf-J76e-b_mbJ5tgkk-sN=g@mail.gmail.com>

Once you go over the hurdle and finally understand phrases, you should find
them very simple and quite powerful and easy to get them to do pretty much
whatever you want. As said, they can handle say, speak, pauses, playback
all in one place.
-Avi


On Tue, Jun 19, 2012 at 6:30 PM, Alex Lake <alex at thewinelake.com> wrote:

> Ooooh, that's interesting!
> However, I think Avi's right and I'll have to bite the bullet on
> phrases....
>
>
>
> On 19 Jun 2012, at 14:52, Hector Geraldino <Hector.Geraldino at ipsoft.com>
> wrote:
>
> Or send a ?break? command after receiving the DTMF event, when using ESL.*
> ***
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
> *Sent:* Tuesday, June 19, 2012 9:40 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Interrupting "speak"****
>
> ** **
>
> Use speak as part of a phrase in play and get digits :)
> ****
>
> -Avi****
>
> ** **
>
> On Tue, Jun 19, 2012 at 4:37 PM, Alex <alex at thewinelake.com> wrote:****
>
> Is there a way such that if my script is "speak"ing to a user that they
> input DTMF and the speak is terminated immediately?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mitch.capper at gmail.com  Tue Jun 19 20:50:40 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 19 Jun 2012 09:50:40 -0700
Subject: [Freeswitch-users] Spandsp - Cannot send non-existant fax file
	(permissions?)
In-Reply-To: <4FE088F3.1060207@gmail.com>
References: <4FE088F3.1060207@gmail.com>
Message-ID: <CALn233qSm7rBkF=94CmxkVN209goN4CvmttHCXT99WOcA_yyaw@mail.gmail.com>

Just because the file has readable permissions doesn't mean the entire
path does.  Try doing an: sudo -u freeswitch file
/usr/local/freeswitch/storage/fax/test_fax.tif
if that works.  I would guess not then you can trace up the directory
hopefully to figure out what permission is wrong.

~mitch

On Tue, Jun 19, 2012 at 7:13 AM, Carlo Dimaggio <jaasmailing at gmail.com> wrote:
> Hi all,
>
> I'm in trouble with mod_spandsp and freeswitch executed by user
> "freeswitch".
> Even if the file is present and the permissions are ok (readable by
> everyone), the FS console gets:
>
> 2012-06-19 16:06:30.098112 [ERR] mod_spandsp_fax.c:1289 Cannot send
> non-existant fax file [/usr/local/freeswitch/storage/fax/test_fax.tif]
>
> FS is able to send if I execute the main process from user "root".
>
>
> Does spandsp require other special permissions?
>
> Regards,
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Jun 19 20:51:52 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 19 Jun 2012 09:51:52 -0700 (PDT)
Subject: [Freeswitch-users] Problem with mod_say_it
In-Reply-To: <4FE09CF8.5050106@gmail.com>
References: <4FE09CF8.5050106@gmail.com>
Message-ID: <1340124712425-7580024.post@n2.nabble.com>

This is caused by not having support for SST_SHORT_DATE_TIME for Italian.
Patches would be welcomed for this support. You may look at the en source
for a working implementation to follow.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-mod-say-it-tp7580022p7580024.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From adam.koleszar at virtual-call-center.eu  Tue Jun 19 20:56:46 2012
From: adam.koleszar at virtual-call-center.eu (=?ISO-8859-1?Q?=C1d=E1m_Kolesz=E1r?=)
Date: Tue, 19 Jun 2012 18:56:46 +0200
Subject: [Freeswitch-users] mod_syslog
In-Reply-To: <1744390.8By8XOcLhx@sos>
References: <4FDF181A.6030701@virtual-call-center.eu>
	<161250C0-1AA2-4708-89AA-8EEB3212C544@gmail.com>
	<1744390.8By8XOcLhx@sos>
Message-ID: <CAPZAM1LL3qhTt1SdHQDov8yXeVOOvzWUr61gVPUr0QUFDu1o_A@mail.gmail.com>

Hi,

unix-dgram solved my problem.
Thanks guys!

Br,
Adam

2012/6/18 Sergey Okhapkin <sos at sokhapkin.dyndns.org>

>  Syslog-ng ose handles multiline logs beginning with version 3.2. Here is
> an
> example:
>
> Jun 18 13:26:02 east1 /usr/local/sbin/kamailio[20824]: INFO: <script>:
> incoming reply from udp:217.174.249.226:5062
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 64.85.163.5;branch=z9hG4bK5aed.4b7d8687.0
> From: sip:dispatcher at localhost;tag=119eb1f393742816e68426b045d5a8e7-1177
> To: <sip:217.174.249.226:5062>;tag=1FacrNHveN0ZH
> Call-ID: 30f7ccd477b092dc-20828 at 64.85.163.5
> CSeq: 10 OPTIONS
> Contact: <sip:217.174.249.226:5062>
> User-Agent: CWU SIP GW
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, refer
> Content-Length: 0
>
>
> On Monday 18 June 2012 18:14:40 Steven Ayre wrote:
> > It appears to be a limitation in syslog-ng ose - a quick google turned
> this
> > up which sounds similar to your problem:
> >
> > https://lists.balabit.hu/pipermail/syslog-ng/2011-September/017212.html
> >
> > Steve on iPhone
> >
> > On 18 Jun 2012, at 12:59, ?d?m Kolesz?r<adam.koleszar at virtual-call-
> center.eu> wrote:
> > > Hi,
> > >
> > > A have a little problem with the mod_syslog module. Freeswitch
> generates
> > > multiline log messages (for example: SDP or register messages) which
> > > syslog-ng OSE can't handle. The firs part of the message is sent
> > > correctly but missing the facility and ident from the other part of the
> > > message. Because of this the messages can not be filtered by ident (or
> > > facility) on server side and goes to wrong destination.
> > >
> > > My question is, have you ever noticed this kind of problem?
> > > Thank you for your help in advance.
> > >
> > > Br,
> > > Adam
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > Join Us At ClueCon - Aug 7-9, 2012
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sdevoy at bizfocused.com  Tue Jun 19 21:33:22 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 19 Jun 2012 13:33:22 -0400
Subject: [Freeswitch-users] FaxxBochs hackers?
Message-ID: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com>

Has anyone investigated the new FaxxBochs offering? It appears to handle the
analog fax machine locally, then send the fax as a data TCP connection to
the server (as a TIFF or JPG I would guess).  It totally removes the VOIP
problem with fax machines and is a clever solution.  $30 a month unlimited
in and out is reasonable, but I would rather have that income myself.

 

It would be great if I could resell it and make a cut, but they require 75
device sales the first year.  

 

If someone knows anything about "reconfiguring" one of these, I would love
to read about it.

 

Sean

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From hexade at hotmail.com  Tue Jun 19 21:39:00 2012
From: hexade at hotmail.com (Adelia C.)
Date: Tue, 19 Jun 2012 13:39:00 -0400
Subject: [Freeswitch-users] Call Recording file format(s)
Message-ID: <BLU166-W40BFF0A731759F8822B4F8DAFF0@phx.gbl>


Please help with my call recording question. If you know of a good link/doc with this info, please pass it my way. 

How does call recording work on Freeswitch? Is the recording saved by default in a .wav file? If so, is it always saved as .wav or can I configure in what format I want to save the recording? 

I noticed that .wav, .gsm, mp3 and .ogg formats are supported (http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls).

Are recordings saved in different formats in real time or converted late? Are they converted at the first request and saved for future use? Is this design decision left to the FreeSwitch user - i.e. if converted later, do I need to implement the conversion myself? I scanned through some older questions and sox and lame are mentioned.

My recordings will be around 800K in wave format (raw data uLaw 8b/8kz stereo) so I am not looking to save space but I am getting the request from the front end guys to provide the recording in .mp3 format.

Thank you,
A.C.
 		 	   		  
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From bdfoster at endigotech.com  Tue Jun 19 22:13:25 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 19 Jun 2012 14:13:25 -0400
Subject: [Freeswitch-users] Gtalk does not ring local phones when making
 outbound calls
In-Reply-To: <CAMWiMy8znR9E7y=vAHjLuYmyfg6EvJD89=fRTApfOjakNpfP0A@mail.gmail.com>
References: <1340052514.70031.YahooMailClassic@web161501.mail.bf1.yahoo.com>
	<CAMWiMy8znR9E7y=vAHjLuYmyfg6EvJD89=fRTApfOjakNpfP0A@mail.gmail.com>
Message-ID: <CAKV-jFx=bO+gH6OcWEi6wBt0sPkuDatvcrgiQv28O0NwTbZ22Q@mail.gmail.com>

Google doesn't generate a ringback, you'll have to generate that yourself.
Please see the wiki on how to generate a ringback before audio is
negotiated.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 19, 2012 2:30 AM, "curriegrad2004" <curriegrad2004 at gmail.com> wrote:

> Do you mean a ringback when you're calling out via google voice?
>
> On Mon, Jun 18, 2012 at 1:48 PM, mtle <getj2k at yahoo.com> wrote:
>
>> Hi,
>> I have successfully compiled and installed FS from the latest git as at
>> today.  Everything seems OK.  The only issue is that I do not get any rings
>> at all with gtalk. I wonder if anyone have had this problem lately and how
>> to fix that? Here's my dialplan.
>>
>> *<extension name="gv_out">*
>> *    <condition field="destination_number" expression="^1?(\d{10})$">*
>> *        <action application="set" data="hangup_after_bridge=true"/>*
>> *        <action application="ring_ready"/>*
>> *       <!--<action application="set"
>> data="bridge_generate_comfort_noise=true"/>-->*
>> *       <!--<action application="set"
>> data="execute_on_answer=start_dtmf_generate"/>-->*
>> *        <action application="bridge" data="dingaling/gtalk/+1$
>> 1 at voice.google.com"/>*
>> *    </condition>*
>> *</extension>*
>>
>> Thanks.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Tue Jun 19 22:15:01 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 19 Jun 2012 14:15:01 -0400
Subject: [Freeswitch-users] Multicast RTP collector
In-Reply-To: <CAMQkVDKhLCPz4o2neLcDeXErFLLZtgt0BNe0XbnjqYpmDcX2fQ@mail.gmail.com>
References: <CAMQkVDKhLCPz4o2neLcDeXErFLLZtgt0BNe0XbnjqYpmDcX2fQ@mail.gmail.com>
Message-ID: <CAKV-jFzWwoySR6XSbjMt6+C6a4Awsg78oOQ1EambOM5qoP+DZQ@mail.gmail.com>

Would you by chance be using the polycom PTT stuff in the UC firmware?

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 19, 2012 11:42 AM, "Ron McCarthy" <ronmccar at gmail.com> wrote:

> Can freeswitch subscribe to a multicast group and when it gets a stream
> connect that to another channel? Basically looking for a RTP to SIP bridge,
> I know a little out of the ordinary :)
>
> So far ive been able to find nothing, but figured it's worth checking.
>
> Thanks
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Tue Jun 19 22:18:49 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 19 Jun 2012 14:18:49 -0400
Subject: [Freeswitch-users] FaxxBochs hackers?
In-Reply-To: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com>
References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com>
Message-ID: <CAKV-jFxDmeSh2jRb66RLSw=C_QDe2k91-vXACkWpR+U9rSiCoQ@mail.gmail.com>

Make your own using FS. It wouldn't seem to be that difficult.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 19, 2012 1:36 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> Has anyone investigated the new FaxxBochs offering? It appears to handle
> the analog fax machine locally, then send the fax as a data TCP connection
> to the server (as a TIFF or JPG I would guess).  It totally removes the
> VOIP problem with fax machines and is a clever solution.  $30 a month
> unlimited in and out is reasonable, but I would rather have that income
> myself.****
>
> ** **
>
> It would be great if I could resell it and make a cut, but they require 75
> device sales the first year.  ****
>
> ** **
>
> If someone knows anything about ?reconfiguring? one of these, I would love
> to read about it.****
>
> ** **
>
> Sean****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Tue Jun 19 22:20:20 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 19 Jun 2012 14:20:20 -0400
Subject: [Freeswitch-users] FaxxBochs hackers?
In-Reply-To: <CAKV-jFxDmeSh2jRb66RLSw=C_QDe2k91-vXACkWpR+U9rSiCoQ@mail.gmail.com>
References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com>
	<CAKV-jFxDmeSh2jRb66RLSw=C_QDe2k91-vXACkWpR+U9rSiCoQ@mail.gmail.com>
Message-ID: <CAKV-jFwVEGG+BCsHYi=iiu7J2wZi6Y=41r0ZRV1SKx5qwM_6dA@mail.gmail.com>

P.s. if ur in the US email me offlist. That is something I've had on my
list for quite some time.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 19, 2012 2:18 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> Make your own using FS. It wouldn't seem to be that difficult.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 19, 2012 1:36 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:
>
>> Has anyone investigated the new FaxxBochs offering? It appears to handle
>> the analog fax machine locally, then send the fax as a data TCP connection
>> to the server (as a TIFF or JPG I would guess).  It totally removes the
>> VOIP problem with fax machines and is a clever solution.  $30 a month
>> unlimited in and out is reasonable, but I would rather have that income
>> myself.****
>>
>> ** **
>>
>> It would be great if I could resell it and make a cut, but they require
>> 75 device sales the first year.  ****
>>
>> ** **
>>
>> If someone knows anything about ?reconfiguring? one of these, I would
>> love to read about it.****
>>
>> ** **
>>
>> Sean****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
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From terry at digital-outpost.com  Tue Jun 19 22:21:31 2012
From: terry at digital-outpost.com (Terry Barnum)
Date: Tue, 19 Jun 2012 11:21:31 -0700
Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
In-Reply-To: <31354FBA-5DBC-47A9-9958-9EC8A515DACD@dons.net.au>
References: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
	<0AEB7B3A-5591-4694-83B5-E4208D5DA4A0@dons.net.au>
	<46D48718-2661-41EC-8A59-67BFEBCCF745@digital-outpost.com>
	<31354FBA-5DBC-47A9-9958-9EC8A515DACD@dons.net.au>
Message-ID: <3D137477-E72E-4A59-A625-899DFB15D387@digital-outpost.com>


On Jun 18, 2012, at 10:26 PM, Daniel O'Connor wrote:

> On 19/06/2012, at 11:01, Terry Barnum wrote:
>>> That said, I had trouble getting the bundled PortAudio working so I mangled the mod_portaudio Makefile.am to use the MP PortAudio and it worked fine.
>> 
>> That's really encouraging. Based on the threads I was reading it looked like people and the devs were saying that FS isn't happy with macports installed.
>> 
>> I've tried installing it on a PPC mac mini running 10.5.8 and an Intel i5 iMac running 10.6.8, both with macports installed. I've forgotten now at what build phase the mac mini failed on (I'm not in front of the machine). The iMac fails during make:
>> 
>> libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I.. -I/usr/include/libxml2 -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -msse2 -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DHAVE_VISIBILITY=1 -g -O2 -MT gsm0610_rpe.lo -MD -MP -MF .deps/gsm0610_rpe.Tpo -c gsm0610_rpe.c  -fno-common -DPIC -o gsm0610_rpe.o
>> /var/folders/B7/B7ea9Uvl2RWJ6k+1YxF5D++++TI/-Tmp-//ccF7gvqy.s:58:suffix or operands invalid for `lea'
>> /var/folders/B7/B7ea9Uvl2RWJ6k+1YxF5D++++TI/-Tmp-//ccF7gvqy.s:63:suffix or operands invalid for `movq'
>> Any ideas?
> 
> I have an Intel mac and configured with..
> 
> env CPPFLAGS=-I/opt/local/include LDFLAGS=-L/opt/local/lib ./configure

Hi Daniel,

So I did a make clean and ran your env line and then make. I think it got further but fails on mod_spandsp_la-mod_spandsp_fax.lo

Creating mod_spandsp_la-mod_spandsp_fax.lo
quiet_libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/opt/local/include -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -pipe -no-cpp-precomp -MT mod_spandsp_la-mod_spandsp_fax.lo -MD -MP -MF .deps/mod_spandsp_la-mod_spandsp_fax.Tpo -c mod_spandsp_fax.c  -fno-common -DPIC -o .libs/mod_spandsp_la-mod_spandsp_fax.o
cc1: warnings being treated as errors
mod_spandsp_fax.c: In function 'spanfax_init':
mod_spandsp_fax.c:716: warning: passing argument 2 of 'span_log_set_message_handler' from incompatible pointer type
mod_spandsp_fax.c:716: error: too many arguments to function 'span_log_set_message_handler'
mod_spandsp_fax.c:717: warning: passing argument 2 of 'span_log_set_message_handler' from incompatible pointer type
mod_spandsp_fax.c:717: error: too many arguments to function 'span_log_set_message_handler'
mod_spandsp_fax.c:772: warning: passing argument 2 of 'span_log_set_message_handler' from incompatible pointer type
mod_spandsp_fax.c:772: error: too many arguments to function 'span_log_set_message_handler'
mod_spandsp_fax.c:773: warning: passing argument 2 of 'span_log_set_message_handler' from incompatible pointer type
mod_spandsp_fax.c:773: error: too many arguments to function 'span_log_set_message_handler'
mod_spandsp_fax.c:827: warning: passing argument 2 of 'span_log_set_message_handler' from incompatible pointer type
mod_spandsp_fax.c:827: error: too many arguments to function 'span_log_set_message_handler'
mod_spandsp_fax.c:828: warning: passing argument 2 of 'span_log_set_message_handler' from incompatible pointer type
mod_spandsp_fax.c:828: error: too many arguments to function 'span_log_set_message_handler'
make[4]: *** [mod_spandsp_la-mod_spandsp_fax.lo] Error 1
make[3]: *** [mod_spandsp-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2

Not sure if this is relevant but libjpeg is installed (via macports).

intellimac:~ terry$ ll /opt/local/lib | grep jpeg
-rwxr-xr-x    1 root      admin    243672 Mar  2 12:05 libjpeg.8.dylib*
-rw-r--r--    1 root      admin    288768 Mar  2 12:05 libjpeg.a
lrwxr-xr-x    1 root      admin        15 Mar  2 12:05 libjpeg.dylib@ -> libjpeg.8.dylib
-rwxr-xr-x    1 root      admin       911 Mar  2 12:05 libjpeg.la*
-rwxr-xr-x    1 root      admin    138528 May 14 09:21 libopenjpeg.1.dylib*
-rw-r--r--    1 root      admin    174392 May 14 09:21 libopenjpeg.a
lrwxr-xr-x    1 root      admin        19 May 14 09:21 libopenjpeg.dylib@ -> libopenjpeg.1.dylib
-rwxr-xr-x    1 root      admin       981 May 14 09:21 libopenjpeg.la*

Thanks for any pointers. (Well, not more incompatible pointer types... Badoom ching)

-Terry



From ronmccar at gmail.com  Tue Jun 19 22:37:38 2012
From: ronmccar at gmail.com (Ron McCarthy)
Date: Tue, 19 Jun 2012 11:37:38 -0700
Subject: [Freeswitch-users] Multicast RTP collector
In-Reply-To: <CAKV-jFzWwoySR6XSbjMt6+C6a4Awsg78oOQ1EambOM5qoP+DZQ@mail.gmail.com>
References: <CAMQkVDKhLCPz4o2neLcDeXErFLLZtgt0BNe0XbnjqYpmDcX2fQ@mail.gmail.com>
	<CAKV-jFzWwoySR6XSbjMt6+C6a4Awsg78oOQ1EambOM5qoP+DZQ@mail.gmail.com>
Message-ID: <CAMQkVDLPrv91ua5tn64ZwC2HQSvcDkEkN8KY8_NUfQzYeaNzJA@mail.gmail.com>

No, these are just a IP multicast endpoint.

Thanks

On Tue, Jun 19, 2012 at 11:15 AM, Brian Foster <bdfoster at endigotech.com>wrote:

> Would you by chance be using the polycom PTT stuff in the UC firmware?
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 19, 2012 11:42 AM, "Ron McCarthy" <ronmccar at gmail.com> wrote:
>
>> Can freeswitch subscribe to a multicast group and when it gets a stream
>> connect that to another channel? Basically looking for a RTP to SIP bridge,
>> I know a little out of the ordinary :)
>>
>> So far ive been able to find nothing, but figured it's worth checking.
>>
>> Thanks
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From unau0 at gmx.de  Tue Jun 19 12:30:43 2012
From: unau0 at gmx.de (Mitja)
Date: Tue, 19 Jun 2012 10:30:43 +0200
Subject: [Freeswitch-users] Where are my freetdm sip_headers?
In-Reply-To: <CAA4nhyA44SRLMtsB+HUu-oFWWszzzY7Jh_H0HdW=smDFkforkg@mail.gmail.com>
References: <4FDACBBA.5060302@gmx.de>
	<CAA4nhyA44SRLMtsB+HUu-oFWWszzzY7Jh_H0HdW=smDFkforkg@mail.gmail.com>
Message-ID: <4FE038B3.3060000@gmx.de>

Hello Moises,

works like a charm. Sometimes its the little things that matter most ;)

Thanks a lot!

> On Fri, Jun 15, 2012 at 1:44 AM, Mitja <unau0 at gmx.de 
> <mailto:unau0 at gmx.de>> wrote:
>
>     Hello all,
>
>     I'm have some problems with my freetdm module lately. To be more
>     precise
>     freetdm with ftmod_sangoma_isdn for ISDN handling.
>
>     One of which is, that I cant get early media to be played in
>     pre_answer
>     state (aka progress_media). But I think I saw a Jira on that one,
>     where
>     I will try to get the information/contribute if possible.
>
>     The other problem I have is that on an incoming call I cant
>     reliably get
>     the TON for displaying purpouses. Actually Display would be
>     alright, but
>     calling back the other side is just uncomfortable if your number is
>     missing a leading 0.
>     So I thought I just have to activate the sip_headers in
>     autoloads/freetdm.conf.xml as described here:
>     http://wiki.freeswitch.org/wiki/Mod_freetdm#Channel_Variables
>     (actually
>     right above)
>
>     But when I call my info Application, I cant see any of the headers.
>
>     freetdm.conf.xml: (I set the param sip_headers three times after
>     several
>     tries)
>     <configuration name="freetdm.conf" description="Freetdm
>     Configuration">
>     <settings>
>     <param name="debug" value="0"/>
>     <param name="sip_headers" value="true" />
>
>
>
> This should be <param name="sip-headers" value="true" />, note that 
> there is a dash instead of an underscore.
>
> *Moises Silva
> **/Manager, Software Engineering/***
>
> msilva at sangoma.com <mailto:msilva at sangoma.com>
>
> Sangoma Technologies
>
> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
>
> 	
>
>
> t.   +1 800 388 2475 (N. America)
>
> t.   +1 905 474 1990 x128
>
> f.   +1 905 474 9223
>
> 	
>
> ** 
> <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>
>
> Products 
> <http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | 
> Solutions 
> <http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | 
> Events 
> <http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | 
> Contact 
> <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | 
> Wiki 
> <http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | 
> Facebook 
> <http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | 
> Twitter 
> <http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`| 
> | YouTube 
> <http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>
>


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From michal at noblesolutions.co.uk  Tue Jun 19 12:57:24 2012
From: michal at noblesolutions.co.uk (michal)
Date: Tue, 19 Jun 2012 09:57:24 +0100
Subject: [Freeswitch-users] transfer/bridge in a conference
Message-ID: <1953A95D11D1ED42B24E60E7A59C5AB40B4F47@venus.Noble.local>

What's the best way of implementing this scenario: In conferenceA there
are user1 and user2, in conferenceB there is user3. How to create 3-way
conference with all 3 users and then "transfer?" user2 to confereceB.
User1 has to stay in conferenceA after the "transfer" so next call can
be taken in conference A while user2 and user3 still talk in conference
B. 

 

 I have thought of creating a new conference C where all parties could
talk and "lock" Conferences A and B till user 1 comes back to conference
A and user 3 comes back to conference B (user 3 will hangup the call and
conference C will be closed/finished)

 

Hope this makes sense..

 

I am using JavaScrip/Node.js to develop this. 

 

All hints and help will be appreciated 

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From Hector.Geraldino at ipsoft.com  Tue Jun 19 22:54:30 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Tue, 19 Jun 2012 14:54:30 -0400
Subject: [Freeswitch-users] Call Recording file format(s)
In-Reply-To: <BLU166-W40BFF0A731759F8822B4F8DAFF0@phx.gbl>
References: <BLU166-W40BFF0A731759F8822B4F8DAFF0@phx.gbl>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD022F4EBACC@NY1-EXMB-01.ip-soft.net>

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session#Possible_File_Formats

Encoding is done on-the-fly, and is determined by the file extension. There's the option of post-processing using tools like sox, but native encoding in various formats (including mp3) are supported.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adelia C.
Sent: Tuesday, June 19, 2012 1:39 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Call Recording file format(s)

Please help with my call recording question. If you know of a good link/doc with this info, please pass it my way.

How does call recording work on Freeswitch? Is the recording saved by default in a .wav file? If so, is it always saved as .wav or can I configure in what format I want to save the recording?

I noticed that .wav, .gsm, mp3 and .ogg formats are supported (http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls).

Are recordings saved in different formats in real time or converted late? Are they converted at the first request and saved for future use? Is this design decision left to the FreeSwitch user - i.e. if converted later, do I need to implement the conversion myself? I scanned through some older questions and sox and lame are mentioned.

My recordings will be around 800K in wave format (raw data uLaw 8b/8kz stereo) so I am not looking to save space but I am getting the request from the front end guys to provide the recording in .mp3 format.

Thank you,
A.C.
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From msc at freeswitch.org  Tue Jun 19 23:17:00 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Jun 2012 12:17:00 -0700
Subject: [Freeswitch-users] Call Recording file format(s)
In-Reply-To: <BLU166-W40BFF0A731759F8822B4F8DAFF0@phx.gbl>
References: <BLU166-W40BFF0A731759F8822B4F8DAFF0@phx.gbl>
Message-ID: <CAKzWOxV+7r3SfXhLk+YgFh-WzVL0Cj1QdFVCmO2RPFah1ad7nQ@mail.gmail.com>

All things being equal, we just save as a wav and then do post-processing
to convert it to the flavor du jour desired by the end users. One advantage
of doing this is that you have the option of off-loading the file
post-processing to another system if you so choose. Between sox and lame
you've got all the free tools you need to convert to whatever file format
your circumstances dictate.

-MC

On Tue, Jun 19, 2012 at 10:39 AM, Adelia C. <hexade at hotmail.com> wrote:

>  Please help with my call recording question. If you know of a good
> link/doc with this info, please pass it my way.
>
> How does call recording work on Freeswitch? Is the recording saved by
> default in a .wav file? If so, is it always saved as .wav or can I
> configure in what format I want to save the recording?
>
> I noticed that .wav, .gsm, mp3 and .ogg formats are supported (
> http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls).
>
> Are recordings saved in different formats in real time or converted late?
> Are they converted at the first request and saved for future use? Is this
> design decision left to the FreeSwitch user - i.e. if converted later, do I
> need to implement the conversion myself? I scanned through some older
> questions and sox and lame are mentioned.
>
> My recordings will be around 800K in wave format (raw data uLaw 8b/8kz
> stereo) so I am not looking to save space but I am getting the request from
> the front end guys to provide the recording in .mp3 format.
>
> Thank you,
> A.C.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hexade at hotmail.com  Tue Jun 19 23:55:37 2012
From: hexade at hotmail.com (Adelia C.)
Date: Tue, 19 Jun 2012 15:55:37 -0400
Subject: [Freeswitch-users] Call Recording file format(s)
In-Reply-To: <CAKzWOxV+7r3SfXhLk+YgFh-WzVL0Cj1QdFVCmO2RPFah1ad7nQ@mail.gmail.com>
References: <BLU166-W40BFF0A731759F8822B4F8DAFF0@phx.gbl>,
	<CAKzWOxV+7r3SfXhLk+YgFh-WzVL0Cj1QdFVCmO2RPFah1ad7nQ@mail.gmail.com>
Message-ID: <BLU166-W128ECDAB6E395F0E60AE95DAFF0@phx.gbl>


Thank you all - this is the info I needed.

Date: Tue, 19 Jun 2012 12:17:00 -0700
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Call Recording file format(s)

All things being equal, we just save as a wav and then do post-processing to convert it to the flavor du jour desired by the end users. One advantage of doing this is that you have the option of off-loading the file post-processing to another system if you so choose. Between sox and lame you've got all the free tools you need to convert to whatever file format your circumstances dictate.


-MC

On Tue, Jun 19, 2012 at 10:39 AM, Adelia C. <hexade at hotmail.com> wrote:





Please help with my call recording question. If you know of a good link/doc with this info, please pass it my way. 

How does call recording work on Freeswitch? Is the recording saved by default in a .wav file? If so, is it always saved as .wav or can I configure in what format I want to save the recording? 


I noticed that .wav, .gsm, mp3 and .ogg formats are supported (http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls).


Are recordings saved in different formats in real time or converted late? Are they converted at the first request and saved for future use? Is this design decision left to the FreeSwitch user - i.e. if converted later, do I need to implement the conversion myself? I scanned through some older questions and sox and lame are mentioned.


My recordings will be around 800K in wave format (raw data uLaw 8b/8kz stereo) so I am not looking to save space but I am getting the request from the front end guys to provide the recording in .mp3 format.

Thank you,

A.C.
 		 	   		  

_________________________________________________________________________

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_________________________________________________________________________
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http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
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From megahohol at gmail.com  Wed Jun 20 02:37:51 2012
From: megahohol at gmail.com (Grygoriy Dobrovolskyy)
Date: Wed, 20 Jun 2012 00:37:51 +0200
Subject: [Freeswitch-users] HA setup completed but nothing in the
	database.
In-Reply-To: <CAAfngiHGi=oYeTLRP4OFWRV7R8T8=Le1RphmD=iarG1uN6UGOA@mail.gmail.com>
References: <CAAfngiHGi=oYeTLRP4OFWRV7R8T8=Le1RphmD=iarG1uN6UGOA@mail.gmail.com>
Message-ID: <CAAfngiFimvtSGSbgZqfRgCCYUfvo32B6KN5d0LL+1MY_SRn9Ow@mail.gmail.com>

2012/6/18 Grygoriy Dobrovolskyy <megahohol at gmail.com>

> Hello list, i've completed HA setup with two FS boxes, connected them to
> the ODBC(mysql) and replicated sip_recovery table between servers, the
> problem i face is simple, i have nothing in there when i have calls.
>
> External profile has this option set:
>   <param name="track-calls" value="true"/>
>   <param name="odbc-dsn" value="fsw-cnf:fs-usr:2>sdfjeslkjfd4<gH(^s"/>
>
>
> switch.conf also the required options:
>     <param name="core-db-dsn" value="fsw-cnf:fs-usr:sdfjeslkjfd4<gH(^s"/>
>
>
>
> FS connects fine and have no problems with that, calls, channels are in
> the database, so i am thinking that FS do not "track calls" because of my
> dialplan which is the following:
>
>
>     <condition field="${sip_to_user}" expression="(1234567789).*$"
> break="never">
>       <param name="track-calls" value="true"/>
>       <action inline="true" application="eval" data="URI=${sip_to_user}"/>
>       <action application="set" data="domain_name=$${domain}"/>
>       <action application="bridge"
> data="{absolute_codec_string=PCMA}sofia/internal/1234569787 at 83.164.189.13
> "/>
>     </condition>
>
> So what i do is simple, match the call, and bridge to external URI. I even
> added <param name="track-calls" value="true"/> to test just before bridge.
>
> Dear all please help me to figure this out, how to make FS track this
> calls ?
>


Problem solved, a tiny typo :)
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From msc at freeswitch.org  Wed Jun 20 03:23:15 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Jun 2012 16:23:15 -0700
Subject: [Freeswitch-users] channel variables in show calls api
In-Reply-To: <BLU162-W21CA24E8CDB1E1F93AF8EF88FF0@phx.gbl>
References: <BLU162-W21CA24E8CDB1E1F93AF8EF88FF0@phx.gbl>
Message-ID: <CAKzWOxW=Mzni5KE+JWCfdkGUS2iJur0D7gSzW_daJxPuURTAPQ@mail.gmail.com>

You'd need to do some sql updates and C code hacking for this. I'd
recommend that you utilize the "presence_data" chan var, assuming you're
not already using it for something else. Anything you put into
presence_data should show up just fine in "show channels" and you can even
do "show channels like xxx" and it will look for 'xxx' in the presence_data
for all the active channels.

-MC

On Tue, Jun 19, 2012 at 6:51 AM, awais nazir <awais-nazeer at hotmail.com>wrote:

>  Hello Guys
>
> Is it possible to do some customization of "show calls" api to show some
> more channel variables like accountcode?
>
> --waisee
>
>
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From msc at freeswitch.org  Wed Jun 20 03:32:30 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Jun 2012 16:32:30 -0700
Subject: [Freeswitch-users] transfer/bridge in a conference
In-Reply-To: <1953A95D11D1ED42B24E60E7A59C5AB40B4F47@venus.Noble.local>
References: <1953A95D11D1ED42B24E60E7A59C5AB40B4F47@venus.Noble.local>
Message-ID: <CAKzWOxVw-zq6bMMaHPRFq-exGWF7eMPbCtA2gEcM6h96+bQ38g@mail.gmail.com>

Some context would be helpful here. What is the big picture solution that
you are building? Why are you using conferences at all? Just curious. I'm
sure the gang here will offer suggestions.

-MC

On Tue, Jun 19, 2012 at 1:57 AM, michal <michal at noblesolutions.co.uk> wrote:

>  What's the best way of implementing this scenario: In conferenceA there
> are user1 and user2, in conferenceB there is user3. How to create 3-way
> conference with all 3 users and then "transfer?" user2 to confereceB. User1
> has to stay in conferenceA after the "transfer" so next call can be taken
> in conference A while user2 and user3 still talk in conference B. ****
>
> ** **
>
>  I have thought of creating a new conference C where all parties could
> talk and ?lock? Conferences A and B till user 1 comes back to conference A
> and user 3 comes back to conference B (user 3 will hangup the call and
> conference C will be closed/finished)****
>
> ** **
>
> Hope this makes sense..****
>
> ** **
>
> I am using JavaScrip/Node.js to develop this. ****
>
> ** **
>
> All hints and help will be appreciated ****
>
>
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From gavin.henry at gmail.com  Wed Jun 20 04:36:12 2012
From: gavin.henry at gmail.com (Gavin Henry)
Date: Wed, 20 Jun 2012 01:36:12 +0100
Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch.
In-Reply-To: <CAMbW9oKncUhzW9zsuCMLqUWKKbDjtM+2jhCbW3aGt7Zz8qH1Tw@mail.gmail.com>
References: <CALH5ZrZjrhe5Sq98jLXqhy6EQBhbLoUdD+CPsS27zRZtKf+bcQ@mail.gmail.com>
	<CAMbW9oKncUhzW9zsuCMLqUWKKbDjtM+2jhCbW3aGt7Zz8qH1Tw@mail.gmail.com>
Message-ID: <CAA8_NKC8+6tnSJtj7E1dJimBCfYXhGiE1jLLW4Xy+XTNhMuhnw@mail.gmail.com>

On 29 March 2012 09:49, Andrew Cassidy <andrew at cassidywebservices.co.uk> wrote:
> I use xml_curl, reads config from a central server so is replicated across
> you whole cluster, and directory/dialplan are requested every call (but
> supply filter parameters) and in effect?dynamic?if you generate these from a
> web service.

Hi Andrew,

Out of interest are you using LDAP at all and LDAP filters?

Thanks.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk


From ocset at the800group.com  Wed Jun 20 07:43:18 2012
From: ocset at the800group.com (ocset)
Date: Wed, 20 Jun 2012 11:43:18 +0800
Subject: [Freeswitch-users] Bridge calls hang with GXW410x
Message-ID: <4FE146D6.4010609@the800group.com>

Hi

I have asked this question before with no real solution and then I 
stumbled across this posting on the 3CX website, as they seem to have 
good support manuals for FXO gateways like the GXW410x.

This is exactly the problem I am having. For those of you with a lot 
more in-depth knowledge, how would this translate into FreeSWITCH speak?
*
-----------------------------------

ADDITIONAL NOTES:*

When receiving an inbound call on the GXW4104/4108 which is then 
diverted to an external number via the 3CX configuration, if the 
GXW4104/8 is used for the diverted call the GXW will stop sending SIP 
signalling to 3CX. This is a known issue with the GXW. As a workaround, 
in the 3CX User Interface, edit the Grandstream device, go to the 
"Advanced" tab, and enable the "PBX delivers audio" option.

Please note that applying this workaround will no longer allow inbound 
or outbound faxes to be delivered over the Grandstream device -- so you 
will need a separate dedicated device/line for delivering faxes.

*-----------------------------------
*

Thanks in advance
O
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From matt at neimeyer.org  Wed Jun 20 06:30:06 2012
From: matt at neimeyer.org (Matt Neimeyer)
Date: Tue, 19 Jun 2012 22:30:06 -0400
Subject: [Freeswitch-users] My First FreeSwitch OR Is this easy to do?
Message-ID: <CAMv5evnVG10fKoZmyZVVjh1nqkTCscmGWFYG9d0F5U7W1X40HQ@mail.gmail.com>

I apologize for the length I'm not sure yet what is "relevant"

My partner and I run a small business out of our home and we're
getting big enough that I want to take advantage of some "modern phone
technology" primarily we're interesting in an auto-attendant at this
point but more will come later I'm sure.

We have two analog lines coming into the house (provided via Verizon
FiOS) and then in every room we have two line capable analog phones.
In our FiOS settings I have both lines ring both lines simultaneously
(to emulate a hunt group) so if someone is on line 1 the incoming call
still rings audibly on line 2 instead of just buzzing call waiting.

I will be getting a (hand me down) TDM1600P V2 card which, in its
current configuration, has four FXO and four FXS ports. So my plan was
to install FreeSwitch and "splice in-line" where the phone lines come
out of the FiOS box. So two lines would go into the FreeSwitch Box and
two "extensions" would come out the other side. Whenever I rewire the
house I'll decide then whether to run a data connection to each
"endpoint" and get SIP phones, add more FXS modules to the TDM1600P,
some mix, or whatever but I'm starting small (I think).

As far the auto-attendant goes 95% of what I need now is to filter
"non-humans". So I would set up a simple menu "Press 1 for Tech
Support, Press 2 for Billing, Etc)". This should filter autodialers
that want us to update our Google listing and so on... but I assume we
can still have all of the options (until we hire on more staff) ring
both extensions. Assuming this lets unanswered calls go to voice
mail... On our analog phones though I'd really love to modify the
Caller ID to show which option they selected. So instead of "Bob
Customer / 234-567-8901" it might show "TS - Bob Customer /
234-567-8901" if they selected Tech Support. If possible I'd also like
to "fill gaps" in Caller ID from a database because a few of our
customers only show the number and not a name.

Short term, I'm currently planning on downloading the Blue.box ISO and
using that as my start point. Medium term I'd like to set up email
notification of voice mails. Long term I'd love to integrate Google
Voice and/or Skype. Really long term I might need to "convert" to a
multi-tenant setup for Personal, Company (now with Employees) and Side
Project Company.

I think that's it... Any advice? Comments?

Matt


From terry at digital-outpost.com  Wed Jun 20 11:17:07 2012
From: terry at digital-outpost.com (Terry Barnum)
Date: Wed, 20 Jun 2012 00:17:07 -0700
Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
In-Reply-To: <31354FBA-5DBC-47A9-9958-9EC8A515DACD@dons.net.au>
References: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
	<0AEB7B3A-5591-4694-83B5-E4208D5DA4A0@dons.net.au>
	<46D48718-2661-41EC-8A59-67BFEBCCF745@digital-outpost.com>
	<31354FBA-5DBC-47A9-9958-9EC8A515DACD@dons.net.au>
Message-ID: <51D0A540-AA9B-4223-BFBC-3A2A43CDC759@digital-outpost.com>


On Jun 18, 2012, at 10:26 PM, Daniel O'Connor wrote:

> On 19/06/2012, at 11:01, Terry Barnum wrote:
>>> That said, I had trouble getting the bundled PortAudio working so I mangled the mod_portaudio Makefile.am to use the MP PortAudio and it worked fine.
>> 
>> That's really encouraging. Based on the threads I was reading it looked like people and the devs were saying that FS isn't happy with macports installed.
>> 
>> I've tried installing it on a PPC mac mini running 10.5.8 and an Intel i5 iMac running 10.6.8, both with macports installed. I've forgotten now at what build phase the mac mini failed on (I'm not in front of the machine). The iMac fails during make:
>> 
>> libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I.. -I/usr/include/libxml2 -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -msse2 -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DHAVE_VISIBILITY=1 -g -O2 -MT gsm0610_rpe.lo -MD -MP -MF .deps/gsm0610_rpe.Tpo -c gsm0610_rpe.c  -fno-common -DPIC -o gsm0610_rpe.o
>> /var/folders/B7/B7ea9Uvl2RWJ6k+1YxF5D++++TI/-Tmp-//ccF7gvqy.s:58:suffix or operands invalid for `lea'
>> /var/folders/B7/B7ea9Uvl2RWJ6k+1YxF5D++++TI/-Tmp-//ccF7gvqy.s:63:suffix or operands invalid for `movq'
>> Any ideas?
> 
> I have an Intel mac and configured with..
> 
> env CPPFLAGS=-I/opt/local/include LDFLAGS=-L/opt/local/lib ./configure
> 
> I am running Lion which uses clang and I think that in this case spandsp doesn't use any asm which is why it probably works.
> 
> I don't know why the asm wouldn't work in your case though.
> 
> You can probably work around it by applying this patch..
> --- a/libs/spandsp/src/gsm0610_rpe.c
> +++ b/libs/spandsp/src/gsm0610_rpe.c
> @@ -59,7 +59,7 @@
> static void weighting_filter(int16_t x[40],
>                              const int16_t *e)      // signal [-5..0.39.44] IN)
> {
> -#if defined(__GNUC__)  &&  defined(SPANDSP_USE_MMX)  &&  defined(__x86_64__)
> +#if 0 && defined(__GNUC__)  &&  defined(SPANDSP_USE_MMX)  &&  defined(__x86_64__)
>     /* Table 4.4   Coefficients of the weighting filter */
>     /* This must be padded to a multiple of 4 for MMX to work */
>     static const union
> @@ -114,7 +114,7 @@ static void weighting_filter(int16_t x[40],
>         : "c" (e), "D" (x), [gsm_H] "X" (gsm_H)
>         : "rax", "rdx", "rsi", "memory"
>     );
> -#elif defined(__GNUC__)  &&  defined(SPANDSP_USE_MMX)  &&  defined(__i386__)
> +#elif 0 && defined(__GNUC__)  &&  defined(SPANDSP_USE_MMX)  &&  defined(__i386__)
>     /* Table 4.4   Coefficients of the weighting filter */
>     /* This must be padded to a multiple of 4 for MMX to work */
>     static const union
> 
> ie disable the asm version and use the C one.

So I tried installing on another Mac, an intel xeon MacPro running 10.6.8 and ran into the problem listed on jira <http://jira.freeswitch.org/browse/FS-3992> where ppc code is getting called instead of x86. I did a fresh git clone, applied your above patch and the jira configure.in.diff patch, used your env configure and it compiled and installed! I'll test it out tomorrow. Thanks for you help.

-Terry



From darius at dons.net.au  Wed Jun 20 11:59:56 2012
From: darius at dons.net.au (Daniel O'Connor)
Date: Wed, 20 Jun 2012 17:29:56 +0930
Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
In-Reply-To: <51D0A540-AA9B-4223-BFBC-3A2A43CDC759@digital-outpost.com>
References: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
	<0AEB7B3A-5591-4694-83B5-E4208D5DA4A0@dons.net.au>
	<46D48718-2661-41EC-8A59-67BFEBCCF745@digital-outpost.com>
	<31354FBA-5DBC-47A9-9958-9EC8A515DACD@dons.net.au>
	<51D0A540-AA9B-4223-BFBC-3A2A43CDC759@digital-outpost.com>
Message-ID: <18C04B8C-6DCC-4581-B51C-9E22AD9C1127@dons.net.au>


On 20/06/2012, at 16:47, Terry Barnum wrote:
>> -#elif defined(__GNUC__)  &&  defined(SPANDSP_USE_MMX)  &&  defined(__i386__)
>> +#elif 0 && defined(__GNUC__)  &&  defined(SPANDSP_USE_MMX)  &&  defined(__i386__)
>>    /* Table 4.4   Coefficients of the weighting filter */
>>    /* This must be padded to a multiple of 4 for MMX to work */
>>    static const union
>> 
>> ie disable the asm version and use the C one.
> 
> So I tried installing on another Mac, an intel xeon MacPro running 10.6.8 and ran into the problem listed on jira <http://jira.freeswitch.org/browse/FS-3992> where ppc code is getting called instead of x86. I did a fresh git clone, applied your above patch and the jira configure.in.diff patch, used your env configure and it compiled and installed! I'll test it out tomorrow. Thanks for you help.


Strangely I didn't hit the spidermonkey problem when I compiled (and it did build and run I just checked).

I'm glad it helped, although I am surprised you only needed to modify that one file..

--
Daniel O'Connor software and network engineer
for Genesis Software - http://www.gsoft.com.au
"The nice thing about standards is that there
are so many of them to choose from."
  -- Andrew Tanenbaum
GPG Fingerprint - 5596 B766 97C0 0E94 4347 295E E593 DC20 7B3F CE8C








From darius at dons.net.au  Wed Jun 20 12:05:48 2012
From: darius at dons.net.au (Daniel O'Connor)
Date: Wed, 20 Jun 2012 17:35:48 +0930
Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
In-Reply-To: <3D137477-E72E-4A59-A625-899DFB15D387@digital-outpost.com>
References: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
	<0AEB7B3A-5591-4694-83B5-E4208D5DA4A0@dons.net.au>
	<46D48718-2661-41EC-8A59-67BFEBCCF745@digital-outpost.com>
	<31354FBA-5DBC-47A9-9958-9EC8A515DACD@dons.net.au>
	<3D137477-E72E-4A59-A625-899DFB15D387@digital-outpost.com>
Message-ID: <4A48F47D-7954-496D-A31E-CE5E1AD21E9D@dons.net.au>


On 20/06/2012, at 3:51, Terry Barnum wrote:
> So I did a make clean and ran your env line and then make. I think it got further but fails on mod_spandsp_la-mod_spandsp_fax.lo
> 
> Creating mod_spandsp_la-mod_spandsp_fax.lo
> quiet_libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/opt/local/include -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -pipe -no-cpp-precomp -MT mod_spandsp_la-mod_spandsp_fax.lo -MD -MP -MF .deps/mod_spandsp_la-mod_spandsp_fax.Tpo -c mod_spandsp_fax.c  -fno-common -DPIC -o .libs/mod_spandsp_la-mod_spandsp_fax.o
> cc1: warnings being treated as errors
> mod_spandsp_fax.c: In function 'spanfax_init':
> mod_spandsp_fax.c:716: warning: passing argument 2 of 'span_log_set_message_handler' from incompatible pointer type
> mod_spandsp_fax.c:716: error: too many arguments to function 'span_log_set_message_handler'
> mod_spandsp_fax.c:717: warning: passing argument 2 of 'span_log_set_message_handler' from incompatible pointer type
> mod_spandsp_fax.c:717: error: too many arguments to function 'span_log_set_message_handler'
> mod_spandsp_fax.c:772: warning: passing argument 2 of 'span_log_set_message_handler' from incompatible pointer type
> mod_spandsp_fax.c:772: error: too many arguments to function 'span_log_set_message_handler'
> mod_spandsp_fax.c:773: warning: passing argument 2 of 'span_log_set_message_handler' from incompatible pointer type

I think this is because you have spandsp installed in MacPorts and those includes are being used in preference to the ones in libs/spandsp/..

I think to work around this I just removed spandsp while I built freeswitch (bleh).

IMO the "right" solution is to rename spandsp in freeswitch - it has been modified (looking at the git log) so it is really a fork of the original. That is unless those patches get merged back into spandsp and then macports can get an updated one.

The later solution is still pretty bad though unless there is a way to force freeswitch to not look at the bundled version.

There is a similar problem with portaudio although since it has a stable API it doesn't show up. It doesn't however build on OSX (Intel 10.7 anyway) and in the end I hacked src/mod/endpoint/mod_portaudio to use pkg-config to find the right CFLAGS and LIBS to link to it.

--
Daniel O'Connor software and network engineer
for Genesis Software - http://www.gsoft.com.au
"The nice thing about standards is that there
are so many of them to choose from."
  -- Andrew Tanenbaum
GPG Fingerprint - 5596 B766 97C0 0E94 4347 295E E593 DC20 7B3F CE8C








From andrew at cassidywebservices.co.uk  Wed Jun 20 12:05:49 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 20 Jun 2012 09:05:49 +0100
Subject: [Freeswitch-users] Most Scalable Way To Control FreeSwitch.
In-Reply-To: <CAA8_NKC8+6tnSJtj7E1dJimBCfYXhGiE1jLLW4Xy+XTNhMuhnw@mail.gmail.com>
References: <CALH5ZrZjrhe5Sq98jLXqhy6EQBhbLoUdD+CPsS27zRZtKf+bcQ@mail.gmail.com>
	<CAMbW9oKncUhzW9zsuCMLqUWKKbDjtM+2jhCbW3aGt7Zz8qH1Tw@mail.gmail.com>
	<CAA8_NKC8+6tnSJtj7E1dJimBCfYXhGiE1jLLW4Xy+XTNhMuhnw@mail.gmail.com>
Message-ID: <CAMbW9o+_y6K7pFGHi_4qpVXywJzUhXz2=qYsPdb+8+3SmbY_yg@mail.gmail.com>

Hi Gavin,

I'm not at the moment but you're not the first person to ask me about LDAP.

What do you want to use LDAP for? Just the directory? I believe there is a
mod_ldap which can already handle that, but querying LDAP in <insert
language here> and formatting the result as XML in the form freeswitch is
expecting shouldn't be difficult.

On 20 June 2012 01:36, Gavin Henry <gavin.henry at gmail.com> wrote:

> On 29 March 2012 09:49, Andrew Cassidy <andrew at cassidywebservices.co.uk>
> wrote:
> > I use xml_curl, reads config from a central server so is replicated
> across
> > you whole cluster, and directory/dialplan are requested every call (but
> > supply filter parameters) and in effect dynamic if you generate these
> from a
> > web service.
>
> Hi Andrew,
>
> Out of interest are you using LDAP at all and LDAP filters?
>
> Thanks.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.surevoip.co.uk
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From alex at thewinelake.com  Wed Jun 20 12:24:16 2012
From: alex at thewinelake.com (Alex)
Date: Wed, 20 Jun 2012 09:24:16 +0100
Subject: [Freeswitch-users] lua library
Message-ID: <4FE188B0.6090302@thewinelake.com>

Got some fairly common lua functions that I'd like to use as a library.
I note that there is a "require" feature in lua, but that requires 
modification of the environment variables.
I was wondering if there isn't a simpler "include" type mechanism where 
I can specify the full path of a lua script?


From alex at thewinelake.com  Wed Jun 20 12:32:16 2012
From: alex at thewinelake.com (Alex)
Date: Wed, 20 Jun 2012 09:32:16 +0100
Subject: [Freeswitch-users] Interrupting "speak"
In-Reply-To: <CALNZuw=JSV13Dz_Au3sKj4Wvf=bLMQ4VGRvdfKLwLEd-C977yA@mail.gmail.com>
References: <4FE08091.7090005@thewinelake.com>
	<CALNZuw=JSV13Dz_Au3sKj4Wvf=bLMQ4VGRvdfKLwLEd-C977yA@mail.gmail.com>
Message-ID: <4FE18A90.2060306@thewinelake.com>

Having got to grips with phrases, I can now see that there is a cheating 
way to do this, which maybe should be put in the wiki?

Essentially, the idea is to have file 
<freeswitch>/conf/lang/en/simple/simplesay.xml:

<include>
   <macro name="say">
     <input pattern="(.*)">
       <match>
         <action function="execute" data="sleep(1000)"/>
         <action function="speak-text" data="$1"/>
       </match>
     </input>
   </macro>


Modify <freeswitch>/conf/lang/en/en.xml to include:

<X-PRE-PROCESS cmd="include" data="simple/simplesay.xml"/>

If you don't use cepstral, edit the first line to be:

<language name="en" say-module="en" 
sound-prefix="$${sounds_dir}/en/us/callie" tts-engine="flite" 
tts-voice="kal">


Then in a lua script, you can do something like this:

whatToSay = "Hello World, I love FreeSwitch"
destnum = session:playAndGetDigits(3, 20, 3, 5000, "#*", 
"phrase:say:"..whatToSay, "", "\\d+")

By the way, is there a fix for Kal talking like a chipmunk?


From avi at avimarcus.net  Wed Jun 20 13:44:53 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 20 Jun 2012 12:44:53 +0300
Subject: [Freeswitch-users] lua library
In-Reply-To: <4FE188B0.6090302@thewinelake.com>
References: <4FE188B0.6090302@thewinelake.com>
Message-ID: <CALNZuwmQtLZuhu8_B5povYGx-DJGLV0RG1iCFvaeg6YHBGuGjw@mail.gmail.com>

>
>  dofile "/usr/local/freeswitch/scripts/my_functions.lua"

works. But it seems to only take an absolute path. If you find another way,
that would be great.
Oh, and if you could wikifiy that...

-Avi


On Wed, Jun 20, 2012 at 11:24 AM, Alex <alex at thewinelake.com> wrote:

> Got some fairly common lua functions that I'd like to use as a library.
> I note that there is a "require" feature in lua, but that requires
> modification of the environment variables.
> I was wondering if there isn't a simpler "include" type mechanism where
> I can specify the full path of a lua script?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From alex at thewinelake.com  Wed Jun 20 13:47:24 2012
From: alex at thewinelake.com (Alex)
Date: Wed, 20 Jun 2012 10:47:24 +0100
Subject: [Freeswitch-users] playAndGetDigits incompatible with callback?
Message-ID: <4FE19C2C.1030803@thewinelake.com>

I'm trying to use PlayAndGetDigits to get a sequence of digits, 
terminated with * (abort) or # (proceed).

I thought I had it working by using an InputCallback, but it appears 
that PAGD turns off InputCallbacks. Is this really the case?

Any other suggestions on how to get the result I need?


From alex at thewinelake.com  Wed Jun 20 14:39:46 2012
From: alex at thewinelake.com (Alex)
Date: Wed, 20 Jun 2012 11:39:46 +0100
Subject: [Freeswitch-users] lua library
In-Reply-To: <CALNZuwmQtLZuhu8_B5povYGx-DJGLV0RG1iCFvaeg6YHBGuGjw@mail.gmail.com>
References: <4FE188B0.6090302@thewinelake.com>
	<CALNZuwmQtLZuhu8_B5povYGx-DJGLV0RG1iCFvaeg6YHBGuGjw@mail.gmail.com>
Message-ID: <4FE1A872.6070508@thewinelake.com>

Actually, a little more digging and I found that the standard lua 
environment is OK - it looks in various places by default.
However, I I may well try your thing - I wonder if that doesn't use the 
modular approach though...
I have updated the wiki...

>      dofile "/usr/local/freeswitch/scripts/my_functions.lua"
>
> works. But it seems to only take an absolute path. If you find another 
> way, that would be great.
> Oh, and if you could wikifiy that...
>
> -Avi
>
>
> On Wed, Jun 20, 2012 at 11:24 AM, Alex <alex at thewinelake.com 
> <mailto:alex at thewinelake.com>> wrote:
>
>     Got some fairly common lua functions that I'd like to use as a
>     library.
>     I note that there is a "require" feature in lua, but that requires
>     modification of the environment variables.
>     I was wondering if there isn't a simpler "include" type mechanism
>     where
>     I can specify the full path of a lua script?
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2437/5080 - Release Date: 06/19/12
>


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From alex at thewinelake.com  Wed Jun 20 15:09:52 2012
From: alex at thewinelake.com (Alex)
Date: Wed, 20 Jun 2012 12:09:52 +0100
Subject: [Freeswitch-users] TERMINATOR_USED_VARIABLE
Message-ID: <4FE1AF80.7060800@thewinelake.com>

I note that in switch_ivr_play_say.c there's an intriguing reference to 
SWITCH_READ_TERMINATOR_USED_VARIABLE on line 2015:

         if ((min_digits == 0) && (strlen(digit_buffer) == 0) && 
switch_channel_get_variable(channel, 
SWITCH_READ_TERMINATOR_USED_VARIABLE) != 0)

It seems to me that this suggests that the terminator used in PAGD is 
stored, at least temporarily in a channel variable.

Is there any way I can get my mitts on the value of this following a 
call to PAYG?


From anita.hall at simmortel.com  Wed Jun 20 15:13:01 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Wed, 20 Jun 2012 16:43:01 +0530
Subject: [Freeswitch-users] passing hylafax events via mod_spandsp
Message-ID: <CAGKPtBtNON8YLGBw+YZBrNbyBXSDdsRAqFERj_FjgRp1eEtnPw@mail.gmail.com>

Hi

I am using Hylafax and soft modems emulated via mod_spandsp to receive and
send Fax, details given here http://wiki.freeswitch.org/wiki/HylaFax

Is there a way to pass on the HylaFax events back to FreeSWITCH via
mod_spandsp ? In particular, I would like to communicate the status of
incoming Fax like success / failure / error etc back to FreeSWITCH.

Thanks.

regards,
Anita
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From alex at thewinelake.com  Wed Jun 20 15:18:58 2012
From: alex at thewinelake.com (Alex)
Date: Wed, 20 Jun 2012 12:18:58 +0100
Subject: [Freeswitch-users] TERMINATOR_USED_VARIABLE
In-Reply-To: <4FE1AF80.7060800@thewinelake.com>
References: <4FE1AF80.7060800@thewinelake.com>
Message-ID: <4FE1B1A2.2050601@thewinelake.com>

Oh yeay... Victory is mine!

  lastDigit = session:getVariable("read_terminator_used")

Must put that in the PAGD wiki....
> I note that in switch_ivr_play_say.c there's an intriguing reference to
> SWITCH_READ_TERMINATOR_USED_VARIABLE on line 2015:
>
>           if ((min_digits == 0) && (strlen(digit_buffer) == 0) &&
> switch_channel_get_variable(channel,
> SWITCH_READ_TERMINATOR_USED_VARIABLE) != 0)
>
> It seems to me that this suggests that the terminator used in PAGD is
> stored, at least temporarily in a channel variable.
>
> Is there any way I can get my mitts on the value of this following a
> call to PAYG?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2177 / Virus Database: 2437/5080 - Release Date: 06/19/12
>
>




From saami_mh at ymail.com  Wed Jun 20 15:47:25 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Wed, 20 Jun 2012 04:47:25 -0700 (PDT)
Subject: [Freeswitch-users] how to implement real time in freeswitch ?
Message-ID: <1340192845.93635.YahooMailNeo@web120106.mail.ne1.yahoo.com>

hi guys,
i am going to implement REAL TIME on freeswitch like (ARA) Asterisk Realtime Arcitecture
how can i do that in freeswitch?
plese guide using link address
thanks so much
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From gmaruzz at gmail.com  Wed Jun 20 15:50:57 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 20 Jun 2012 13:50:57 +0200
Subject: [Freeswitch-users] how to implement real time in freeswitch ?
In-Reply-To: <1340192845.93635.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1340192845.93635.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <CALXCt0oSyDfxuJ-Vnxw9WCq8vR+OMaHiF5uO=gSwLuoqeh7zZg@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Xml_curl


On Wed, Jun 20, 2012 at 1:47 PM, Samira Mh <saami_mh at ymail.com> wrote:

> hi guys,
> i am going to implement REAL TIME on freeswitch like (ARA) Asterisk
> Realtime Arcitecture
> how can i do that in freeswitch?
> plese guide using link address
> thanks so much
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From avi at avimarcus.net  Wed Jun 20 15:58:36 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 20 Jun 2012 14:58:36 +0300
Subject: [Freeswitch-users] how to implement real time in freeswitch ?
In-Reply-To: <CALXCt0oSyDfxuJ-Vnxw9WCq8vR+OMaHiF5uO=gSwLuoqeh7zZg@mail.gmail.com>
References: <1340192845.93635.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<CALXCt0oSyDfxuJ-Vnxw9WCq8vR+OMaHiF5uO=gSwLuoqeh7zZg@mail.gmail.com>
Message-ID: <CALNZuwkHizpqkabdbA=w=wV-G9v3Qo+ZKR8aKO9qxNtq30s1-A@mail.gmail.com>

For other asterisk drop-ins, check the rosetta stone wiki page:
http://wiki.freeswitch.org/wiki/Rosetta_stone

-Avi
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From miha at softnet.si  Wed Jun 20 17:47:40 2012
From: miha at softnet.si (Miha)
Date: Wed, 20 Jun 2012 15:47:40 +0200
Subject: [Freeswitch-users] 302 redirect problem
Message-ID: <4FE1D47C.3020700@softnet.si>

Hi,

In sip_profiles I have set to manual redirect.
When call goes in public dialplan and than phone redirect call, FS 
automatically send redirect call to contex redirect.
Is this normal? How to set that call will go to internal dialplan?

Thanks!

2012-06-20 14:55:16.126100 [DEBUG] switch_core_state_machine.c:410 
(sofia/nat/sip:014310000.fs_kabelvoip2 at xxx.xxx.xxx.xxx:1264) State 
Change CS_REPORTING -> CS_DESTROY
2012-06-20 14:55:16.126100 [INFO] mod_dialplan_xml.c:485 Processing 
0038618100100 <0038618100100>->*051357952 in context redirected*
2012-06-20 14:55:16.126100 [DEBUG] switch_core_session.c:1228 Send 
signal sofia/nat/sip:014310000.fs_kabelvoip2 at xxx.xxx.xxx.xxx:1264 [BREAK]
2012-06-20 14:55:16.126100 [DEBUG] switch_core_session.c:1428 Session 2 
(sofia/nat/sip:014310000.fs_kabelvoip2 at xxx.xxx.xxx.xxx:1264) Locked, 
Waiting on external entities
2012-06-20 14:55:16.126100 [NOTICE] switch_core_session.c:1446 Session 2 
(sofia/nat/sip:014310000.fs_kabelvoip2 at xxx.xxx.xxx.xxx:1264) Ended
2012-06-20 14:55:16.126100 [NOTICE] switch_core_session.c:1448 Close 
Channel sofia/nat/sip:014310000.fs_kabelvoip2 at xxx.xxx.xxx.xxx:1264 
[CS_DESTROY]
2012-06-20 14:55:16.126100 [DEBUG] switch_core_state_machine.c:514 
(sofia/nat/sip:014310000.fs_kabelvoip2 at xxx.xxx.xxx.xxx:1264) Callstate 
Change HANGUP -> DOWN
2012-06-20 14:55:16.126100 [DEBUG] switch_core_state_machine.c:517 
(sofia/nat/sip:014310000.fs_kabelvoip2 at xxx.xxx.xxx.xxx:1264) Running 
State Change CS_DESTROY
2012-06-20 14:55:16.126100 [DEBUG] switch_core_state_machine.c:527 
(sofia/nat/sip:014310000.fs_kabelvoip2 at xxx.xxx.xxx.xxx:1264) State DESTROY
2012-06-20 14:55:16.126100 [DEBUG] mod_sofia.c:374 
sofia/nat/sip:014310000.fs_kabelvoip2 at xxx.xxx.xxx.xxx:1264 SOFIA DESTROY
2012-06-20 14:55:16.126100 [DEBUG] switch_core_state_machine.c:86 
sofia/nat/sip:014310000.fs_kabelvoip2 at xxx.xxx.xxx.xxx:1264 Standard DESTROY
2012-06-20 14:55:16.126100 [DEBUG] switch_core_state_machine.c:527 
(sofia/nat/sip:014310000.fs_kabelvoip2 at xxx.xxx.xxx.xxx:1264) State 
DESTROY going to sleep
2012-06-20 14:55:16.126100 [WARNING] mod_dialplan_xml.c:515 *Context 
redirected not found*


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From ken.armstrong.73 at gmail.com  Wed Jun 20 17:58:51 2012
From: ken.armstrong.73 at gmail.com (Ken Armstrong)
Date: Wed, 20 Jun 2012 09:58:51 -0400
Subject: [Freeswitch-users] uuid_simplify from Lua Script
Message-ID: <CA+YB6SQCL5q-PTdf-NfeWAiBP=DLM88xe6ZmMHKQY8sqWzsxCQ@mail.gmail.com>

Hi All,

Can someone give me an example of how to use uuid_simplify from a lua
scripts.

Ken
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From getj2k at yahoo.com  Wed Jun 20 18:41:39 2012
From: getj2k at yahoo.com (mtle)
Date: Wed, 20 Jun 2012 07:41:39 -0700 (PDT)
Subject: [Freeswitch-users] Gtalk does not ring local phones when making
	outbound calls
Message-ID: <1340203299.60346.YahooMailClassic@web161503.mail.bf1.yahoo.com>

Hi Brian,
I am a novice to both FS and VoIP. I just follow the wiki to setup Google voice. ?I thought "ring_ready" will do. ?Could you please give me a pointer to the wiki?Thanks.
Google doesn't generate a ringback, you'll have to generate that yourself. Please see the wiki on how to generate a ringback before audio is negotiated.Brian Foster
Endigo Computer LLCSent from a mobile device.

On Jun 19, 2012 2:30 AM, "curriegrad2004" <curriegrad2004 at gmail.com> wrote:
Do you mean a ringback when you're calling out via google voice?
On Mon, Jun 18, 2012 at 1:48 PM, mtle?<getj2k at yahoo.com>?wrote:
Hi,I have successfully compiled and installed FS from the latest git as at today. ?Everything seems OK.
 ?The only issue is that I do not get any rings at all with gtalk. I wonder if anyone have had this problem lately and how to fix that? Here's my dialplan.
<extension name="gv_out">? ? 
<condition field="destination_number" expression="^1?(\d{10})$">
<action application="set" data="hangup_after_bridge=true"/>? ? ? ? 
<action application="ring_ready"/>
<!--<action application="set" data="bridge_generate_comfort_noise=true"/>-->
<!--<action application="set" data="execute_on_answer=start_dtmf_generate"/>-->
<action application="bridge" data="dingaling/gtalk/+1$1 at voice.google.com"/>
</condition>
</extension>
Thanks.
_________________________________________________________________________
Professional FreeSWITCH Consulting
 Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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_________________________________________________________________________
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consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH
 Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
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http://www.freeswitch.org
--- On Tue, 6/19/12, freeswitch-users-request at lists.freeswitch.org <freeswitch-users-request at lists.freeswitch.org> wrote:

From: freeswitch-users-request at lists.freeswitch.org <freeswitch-users-request at lists.freeswitch.org>
Subject: FreeSWITCH-users Digest, Vol 72, Issue 174
To: freeswitch-users at lists.freeswitch.org
Date: Tuesday, June 19, 2012, 11:19 AM

Send FreeSWITCH-users mailing list submissions to
??? freeswitch-users at lists.freeswitch.org

To subscribe or unsubscribe via the World Wide Web,
 visit
??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
??? freeswitch-users-request at lists.freeswitch.org

You can reach the person managing the list at
??? freeswitch-users-owner at lists.freeswitch.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-users digest..."
Today's Topics:

???1. FaxxBochs hackers? (Sean Devoy)
???2. Call Recording file format(s) (Adelia C.)
???3. Re: Gtalk does not ring local phones when making outbound
? ? ? calls (Brian Foster)
???4. Re: Multicast RTP collector (Brian Foster)
???5. Re: FaxxBochs hackers? (Brian Foster)
_______________________________________________
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 vvenkatar at gmail.com  Wed Jun 20 19:20:20 2012
From: vvenkatar at gmail.com (Venkatesh)
Date: Wed, 20 Jun 2012 08:20:20 -0700
Subject: [Freeswitch-users] How to configure FreeSwitch to use specific
 transport when offering calls to a SIP Gateway
Message-ID: <CAP_69msRp6d8P3HG_tO80bzaRd+RXSt88NBgB9NNsA=JiOqnDg@mail.gmail.com>

Hi !

I am new to freeswitch. Apologies if this question was already answered in
the mailing list. I have a simple IVR application that answers a call and
transfers the caller to a specific number. The transfer target requires
that I offer the call to a gateway. I want to configure freeswitch to use
tcp as the transport for the transfer target. The incoming call arrives
over UDP. I configured the following in the gateway configuration:

        <param name="contact-params" value="transport=tcp"/>

However, I don't see this configuration making FS use TCP. The call
continues to use UDP. In addition, FS adds two contact parameters
(transport=tcp; transport=udp) in the outgoing invite. Would appreciate if
you can let me know if I am missing something.

Venkatesh
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From leon at scarlet-internet.nl  Wed Jun 20 19:36:25 2012
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Wed, 20 Jun 2012 17:36:25 +0200
Subject: [Freeswitch-users] lua library
In-Reply-To: <4FE188B0.6090302@thewinelake.com>
References: <4FE188B0.6090302@thewinelake.com>
Message-ID: <8F61B4E2-D25F-4F91-923C-B287F494FF29@scarlet-internet.nl>

I put my libs in the same dir as the rest of the FS Lua scripts and have this in my lua.conf.xml

	<param name="script-directory" value="$${base_dir}/scripts/?.lua"/>

then I can just

	require 'foo'

in my scripts, without specifying a path

regards,

Leon


On Jun 20, 2012, at 10:24, Alex wrote:

> Got some fairly common lua functions that I'd like to use as a library.
> I note that there is a "require" feature in lua, but that requires 
> modification of the environment variables.
> I was wondering if there isn't a simpler "include" type mechanism where 
> I can specify the full path of a lua script?
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mi.ke at null.net  Wed Jun 20 19:45:40 2012
From: mi.ke at null.net (Mi Ke)
Date: Wed, 20 Jun 2012 11:45:40 -0400
Subject: [Freeswitch-users] gateway registration using TLS
Message-ID: <20120620154541.182990@gmx.com>

Hi All,

 Is it possible to explicitly tell sofia to register as a client using TLS? Here's my TLS-related part of sofia config:

 <profiles>
 <profile name="external">
 <domains>
 <domain name="all" alias="true" parse="false"/>
 </domains>
 <gateways>
 <gateway name="xxx">
 <param name="register" value="true"/>
 <param name="username" value="yyy"/>
 <param name="realm" value="remote_ip_here:5061"/>
 <param name="password" value="zzz"/>
 <param name="register-transport" value="tcp"/>
 <param name="expire-seconds" value="3600"/>
 <param name="retry-seconds" value="60" />
 </gateway>
 </gateways>
 <settings>
 ...
 <param name="context" value="external"/>
 <param name="sip-port" value="5060"/>
 <param name="tls" value="true"/>
 <param name="tls-only" value="false"/>
 <param name="tls-bind-params" value="transport=tls"/>
 <param name="contact-params" value="tport=tcp"/>
 <param name="tls-version" value="sslv23"/>
 <param name="tls-cert-dir" value="/usr/local/freeswitch/conf/ssl"/>
 <param name="tls-sip-port" value="5061"/>
 <param name="dialplan" value="XML"/>
 ...

 </settings>
 </profile>
 </profiles>

 But that does not work...

 tport_connected( 
0x7f9b44014110): events CONNECTED
 tport_send_event(0x7f9b44014110) - ready to send to (tcp/remote_ip_here:5061 http://212.58.166.54:5061 )
 tport_vsend(0x7f9b44014110): 600 bytes of 600 to tcp/remote_ip_here:5061 http://212.58.166.54:5061 
 tport_vsend returned 600
 tport_set_events(0x7f9b44014110): events IN
 tport(0x7f9b44014110): reset timer
 tport_wakeup(0x7f9b44014110): events IN HUP ERR
 tport_release(0x7f9b44014110): 0x7f9b4401a460 by 0x7f9b4402ca70 with (nil)
nta: REGISTER (29733430): Connection reset by peer (104) with tcp/[remote_ip_here]:5061
nua(0x7f9b440196a0): event r_register 503 Service Unavailable
Any ideas what I'm doing wrong?

 Thanks / MiKe
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From shadowym at hotmail.com  Wed Jun 20 18:50:40 2012
From: shadowym at hotmail.com (shadowym)
Date: Wed, 20 Jun 2012 07:50:40 -0700
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <CAKzWOxVAPEESKaW4-Ax2TLRuy03DeTRzkGV0RP1Jm=m76viQJw@mail.gmail.com>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
	<CAKzWOxVAPEESKaW4-Ax2TLRuy03DeTRzkGV0RP1Jm=m76viQJw@mail.gmail.com>
Message-ID: <COL124-DS1764EF4D98EE2B031084E7DDFE0@phx.gbl>

Nothing trollish about a user unhappy about a decision they STRONGLY
disagree with.

 

But it gets even worse again.  Not only have directories been moved and
hierarchy changed.  Some of the default config files haven't been updated to
reflect the moves.  So now you have configs moved to a different directory,
hierarchy changed which makes it non standard from ALL the freeswitch
documentation,  splitting of sound files which were left in original
location, and incorrect directoy setttings in configuration files.

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, June 18, 2012 3:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] freeswitch directories moved

 

Normally I would have slapped this email down for being unnecessarily
trollish, however if we s/emotions//g and get to the core of the issue then
I think we can have a productive discussion. Since I personally was not
involved in the decisions on the RPMs I will simply moderate the discussion.

Please everyone, let's keep the discussion civil and on point. 

-MC

On Mon, Jun 18, 2012 at 3:01 PM, shadowym <shadowym at hotmail.com> wrote:

Who's idea was it to move the freeswitch directories in the RPM and why?

 

Is it too much to ask that directories are standardized and don't change?!

 

Do you know how distruptive it is when you move all the config files from
/opt to /etc.!  Not only that but the hierarchy has been changed.  The /conf
directory has been removed even though it's in all the documentation on the
Freeswitch wiki.  Wtf?  I am used to config files being in /etc for
Asterisk.  Despite all the disadvantages with Asterisk at least they are
sticking with that decision!  Of course if I compile from scratch I suppose
I can choose  the directory myself but for various reasons I used the RPM so
I don't have much choice in the matter unless I want to make my own RPM
which isn't very good use of my time imho.  But wait, it gets worse!
Whoever made this decision decided just moving everything and getting rid of
/conf directory was not disruptive enough.  So they decided to split it up
and leave the sound files where they are in /opt/freeswitch.  Again, wtf?!

 

 

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From shadowym at hotmail.com  Wed Jun 20 18:57:35 2012
From: shadowym at hotmail.com (shadowym)
Date: Wed, 20 Jun 2012 07:57:35 -0700
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <CAKzWOxWR6PJqPb6NqTLRQa5WMGP6H4zVte1w0i2UZSDN9+fQMg@mail.gmail.com>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>	<4FDFBCA8.1050700@quentustech.com>
	<CAKzWOxWR6PJqPb6NqTLRQa5WMGP6H4zVte1w0i2UZSDN9+fQMg@mail.gmail.com>
Message-ID: <COL124-DS1734BA1C8437EE24E71567DDFE0@phx.gbl>

Did the announcement also say they were getting rid of the conf directory
and not bothering to updated directory settings in the default conf files
(that are now in the root directory).  Do they plan to update all the
freeswitch documentation now as well which talks about the conf directory
all over the place?

 

How long ago was this announcement made?  The issue is not about being
caught off guard btw.  It's about why and how it's being done.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, June 18, 2012 5:43 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] freeswitch directories moved

 

 

On Mon, Jun 18, 2012 at 4:41 PM, William King <william.king at quentustech.com>
wrote:

I believe there was an announcement on this mailing list alerting that the
new rpms would be undergoing major changes to include moving the directory
locations. Search this list for new threads started by SwK in the past 6
months.

 

There was indeed an announcement by SwK himself. It was as close to FNLoIP
(Flashing Neon Lights over IP) as you can get.
-MC

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From shadowym at hotmail.com  Wed Jun 20 19:02:02 2012
From: shadowym at hotmail.com (shadowym)
Date: Wed, 20 Jun 2012 08:02:02 -0700
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <CALNZuwnHU7+vbVre0kmrk+P_BU40AorL+4_sDA6-d6f4rLkoew@mail.gmail.com>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>	<4FDFBCA8.1050700@quentustech.com>	<CAKzWOxWR6PJqPb6NqTLRQa5WMGP6H4zVte1w0i2UZSDN9+fQMg@mail.gmail.com>	<4FE072DA.3040903@anew.com.ve>
	<CALNZuwnHU7+vbVre0kmrk+P_BU40AorL+4_sDA6-d6f4rLkoew@mail.gmail.com>
Message-ID: <COL124-DS10B977B37DDE7083722AB3DDFE0@phx.gbl>

Which brings up another good point.  If such a drastic (imho) change is
being made there should be some consideration for backwards compatibility
and conformance to the documentation.  How hard would it have been to add
symlinks?  

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Tuesday, June 19, 2012 5:50 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] freeswitch directories moved

 

A changelog can often be ignored. A "breaking changes" or "not backward
compatible changes" log would be helpful for such things.

Huh, if those changes get tagged in GIT I think they automatically show up
if you're paying attention when doing a pull.

-Avi

 

On Tue, Jun 19, 2012 at 3:38 PM, Saugort Dario Garcia Tovar
<dgarcia at anew.com.ve> wrote:

Hi Everybody,

Just to be fare, is good to make FS more improved and standardised.It is
better if in the wiki the devolper or someone involved with rpm update the
wiki with a "change log", in this way everyone will know what is different.
This will helpful more than you can expect for FS community.

Shadowym, if you have decided to use FS RPM, you should get more info about
it and what to expect.

It is just my opinion



On 6/18/2012 8:13 PM, Michael Collins wrote:

 

On Mon, Jun 18, 2012 at 4:41 PM, William King <william.king at quentustech.com>
wrote:

I believe there was an announcement on this mailing list alerting that the
new rpms would be undergoing major changes to include moving the directory
locations. Search this list for new threads started by SwK in the past 6
months.

 

There was indeed an announcement by SwK himself. It was as close to FNLoIP
(Flashing Neon Lights over IP) as you can get.
-MC



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com
 


 
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
 
Join Us At ClueCon - Aug 7-9, 2012
 
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

No virus found in this message.
Checked by AVG - www.avg.com
Version: 2012.0.2178 / Virus Database: 2437/5079 - Release Date: 06/19/12

 

-- 
Atentamente, 
Dario Garc?a 
Consultor. 

CCCT, Nivel C2, Sector Yarey, Mz, 
Ofc. MZ03a. 
Caracas-Venezuela. 
Tel?fono: +58 212 9081842 <tel:%2B58%20212%209081842>  
Cel: +58 412 2221515 <tel:%2B58%20412%202221515>  
dgarcia at anew.com.ve 
http://www.anew.com.ve


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

 

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From toddb at toddbailey.net  Wed Jun 20 19:08:07 2012
From: toddb at toddbailey.net (toddb at toddbailey.net)
Date: Wed, 20 Jun 2012 08:08:07 -0700
Subject: [Freeswitch-users] Clarification needed on setting up group calling
	parameters
Message-ID: <20120620080807.33e327b490679d2282e332758c73b55b.5f9d927ef5.wbe@email14.secureserver.net>

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From sdevoy at bizfocused.com  Wed Jun 20 20:03:31 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 20 Jun 2012 12:03:31 -0400
Subject: [Freeswitch-users] gateway registration using TLS
In-Reply-To: <20120620154541.182990@gmx.com>
References: <20120620154541.182990@gmx.com>
Message-ID: <08a201cd4efe$3d275520$b775ff60$@bizfocused.com>

Mi Ke,

 

You should put your actual remote provider?s domain name or tip address in where it says ?remote_ip_here? and I assume you have entered you actual username and password where it says ?yyy? and ?zzz?.

It should look SIMILAR to this, but with your provider: <param name="realm" value="sip.voipinnovations.com:5061"/>

Hope that helps,

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mi Ke
Sent: Wednesday, June 20, 2012 11:46 AM
To: Freeswitch Users
Subject: [Freeswitch-users] gateway registration using TLS

 

Hi All,

Is it possible to explicitly tell sofia to register as a client using TLS? Here's my TLS-related part of sofia config:

        <profiles>
            <profile name="external">
                <domains>
                    <domain name="all" alias="true" parse="false"/>
                </domains>
                <gateways>
                        <gateway name="xxx">
                                  <param name="register" value="true"/>
                                  <param name="username" value="yyy"/>
                                  <param name="realm" value="remote_ip_here:5061"/>
                                  <param name="password" value="zzz"/>
                                  <param name="register-transport" value="tcp"/>
                                  <param name="expire-seconds" value="3600"/>
                                  <param name="retry-seconds" value="60" />
                         </gateway>
                </gateways>
                <settings>
                       ...
                       <param name="context" value="external"/>
                       <param name="sip-port" value="5060"/>
                       <param name="tls" value="true"/>
                       <param name="tls-only" value="false"/>
                       <param name="tls-bind-params" value="transport=tls"/>
                       <param name="contact-params" value="tport=tcp"/>
                       <param name="tls-version" value="sslv23"/>
                       <param name="tls-cert-dir" value="/usr/local/freeswitch/conf/ssl"/>
                       <param name="tls-sip-port" value="5061"/>
                       <param name="dialplan" value="XML"/>
                       ...

                </settings>
            </profile>
        </profiles>

But that does not work...

tport_connected( 

0x7f9b44014110): events CONNECTED
tport_send_event(0x7f9b44014110) - ready to send to (tcp/remote_ip_here:5061 <http://212.58.166.54:5061> )
tport_vsend(0x7f9b44014110): 600 bytes of 600 to tcp/remote_ip_here:5061 <http://212.58.166.54:5061> 
tport_vsend returned 600
tport_set_events(0x7f9b44014110): events IN
tport(0x7f9b44014110): reset timer
tport_wakeup(0x7f9b44014110): events IN HUP ERR
tport_release(0x7f9b44014110): 0x7f9b4401a460 by 0x7f9b4402ca70 with (nil)
nta: REGISTER (29733430): Connection reset by peer (104) with tcp/[remote_ip_here]:5061
nua(0x7f9b440196a0): event r_register 503 Service Unavailable 


Any ideas what I'm doing wrong?

Thanks / MiKe

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From msc at freeswitch.org  Wed Jun 20 20:06:42 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Jun 2012 09:06:42 -0700
Subject: [Freeswitch-users] FreeSWITCH Conf Call Today: SIP 101
Message-ID: <CAKzWOxV0HztKFTHYyTNNCPYeL9f14QbrmiDzU_2omiDU_hmJPQ@mail.gmail.com>

Hey all,

Don't forget that at 1pm EDT, 10am PDT we have the FreeSWITCH community
conference call. Darren Schreiber will be doing the SIP 101 discussion. We
look forward to having you all with us.

-Michael
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From lloyd.aloysius at gmail.com  Wed Jun 20 20:15:05 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Wed, 20 Jun 2012 12:15:05 -0400
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <COL124-DS1764EF4D98EE2B031084E7DDFE0@phx.gbl>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
	<CAKzWOxVAPEESKaW4-Ax2TLRuy03DeTRzkGV0RP1Jm=m76viQJw@mail.gmail.com>
	<COL124-DS1764EF4D98EE2B031084E7DDFE0@phx.gbl>
Message-ID: <CAGvuWYZG-g455DK0449i=87Sv+OCHAnwyYjJey6JAktbb13w9w@mail.gmail.com>

Does this directory changes apply to the Git & install. I mean retrieve the
source and build our self.

Thanks
Lloyd





On Wed, Jun 20, 2012 at 10:50 AM, shadowym <shadowym at hotmail.com> wrote:

> Nothing trollish about a user unhappy about a decision they STRONGLY
> disagree with.****
>
> ** **
>
> But it gets even worse again.  Not only have directories been moved and
> hierarchy changed.  Some of the default config files haven?t been updated
> to reflect the moves.  So now you have configs moved to a different
> directory, hierarchy changed which makes it non standard from ALL the
> freeswitch documentation,  splitting of sound files which were left in
> original location, and incorrect directoy setttings in configuration files.
> ****
>
> ** **
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, June 18, 2012 3:13 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] freeswitch directories moved****
>
> ** **
>
> Normally I would have slapped this email down for being unnecessarily
> trollish, however if we s/emotions//g and get to the core of the issue then
> I think we can have a productive discussion. Since I personally was not
> involved in the decisions on the RPMs I will simply moderate the discussion.
>
>
> Please everyone, let's keep the discussion civil and on point.
>
> -MC****
>
> On Mon, Jun 18, 2012 at 3:01 PM, shadowym <shadowym at hotmail.com> wrote:***
> *
>
> Who?s idea was it to move the freeswitch directories in the RPM and why?**
> **
>
>  ****
>
> Is it too much to ask that directories are standardized and don?t change?!
> ****
>
>  ****
>
> Do you know how distruptive it is when you move all the config files from
> /opt to /etc.!  Not only that but the hierarchy has been changed.  The
> /conf directory has been removed even though it?s in all the documentation
> on the Freeswitch wiki.  Wtf?  I am used to config files being in /etc
> for Asterisk.  Despite all the disadvantages with Asterisk at least they
> are sticking with that decision!  Of course if I compile from scratch I
> suppose I can choose  the directory myself but for various reasons I used
> the RPM so I don?t have much choice in the matter unless I want to make
> my own RPM which isn?t very good use of my time imho.  But wait, it gets
> worse!  Whoever made this decision decided just moving everything and
> getting rid of /conf directory was not disruptive enough.  So they
> decided to split it up and leave the sound files where they are in
> /opt/freeswitch.  Again, wtf?!****
>
>  ****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mi.ke at null.net  Wed Jun 20 20:20:19 2012
From: mi.ke at null.net (Mi Ke)
Date: Wed, 20 Jun 2012 12:20:19 -0400
Subject: [Freeswitch-users] gateway registration using TLS
Message-ID: <20120620162020.182950@gmx.com>

Dear Sean,

 It's not that bad - remote_ip, xxx and yyy are only placeholders in my example - actual config has real params ;)

 Did you mean that FS will automatically use TLS for registration when I append :5061 to realm value ?

 Thanks / MiKe

----- Original Message -----
From: Sean Devoy
Sent: 06/20/12 07:03 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] gateway registration using TLS

Mi Ke,
You should put your actual remote provider?s domain name or tip address in where it says ?remote_ip_here? and I assume you have entered you actual username and password where it says ?yyy? and ?zzz?.
It should look SIMILAR to this, but with your provider: <param name="realm" value="sip.voipinnovations.com:5061"/>
Hope that helps,
Sean
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]  *On Behalf Of *Mi Ke
 *Sent:* Wednesday, June 20, 2012 11:46 AM
 *To:* Freeswitch Users
 *Subject:* [Freeswitch-users] gateway registration using TLS

Hi All,

 Is it possible to explicitly tell sofia to register as a client using TLS? Here's my TLS-related part of sofia config:

 <profiles>
 <profile name="external">
 <domains>
 <domain name="all" alias="true" parse="false"/>
 </domains>
 <gateways>
 <gateway name="xxx">
 <param name="register" value="true"/>
 <param name="username" value="yyy"/>
 <param name="realm" value="remote_ip_here:5061"/>
 <param name="password" value="zzz"/>
 <param name="register-transport" value="tcp"/>
 <param name="expire-seconds" value="3600"/>
 <param name="retry-seconds" value="60" />
 </gateway>
 </gateways>
 <settings>
 ...
 <param name="context" value="external"/>
 <param name="sip-port" value="5060"/>
 <param name="tls" value="true"/>
 <param name="tls-only" value="false"/>
 <param name="tls-bind-params" value="transport=tls"/>
 <param name="contact-params" value="tport=tcp"/>
 <param name="tls-version" value="sslv23"/>
 <param name="tls-cert-dir" value="/usr/local/freeswitch/conf/ssl"/>
 <param name="tls-sip-port" value="5061"/>
 <param name="dialplan" value="XML"/>
 ...

 </settings>
 </profile>
 </profiles>

 But that does not work...

 tport_connected( 

0x7f9b44014110): events CONNECTED
 tport_send_event(0x7f9b44014110) - ready to send to (tcp/remote_ip_here:5061 http://212.58.166.54:5061 )
 tport_vsend(0x7f9b44014110): 600 bytes of 600 to tcp/remote_ip_here:5061 http://212.58.166.54:5061 
 tport_vsend returned 600
 tport_set_events(0x7f9b44014110): events IN
 tport(0x7f9b44014110): reset timer
 tport_wakeup(0x7f9b44014110): events IN HUP ERR
 tport_release(0x7f9b44014110): 0x7f9b4401a460 by 0x7f9b4402ca70 with (nil)
nta: REGISTER (29733430): Connection reset by peer (104) with tcp/[remote_ip_here]:5061
nua(0x7f9b440196a0): event r_register 503 Service Unavailable

Any ideas what I'm doing wrong?

 Thanks / MiKe
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From msc at freeswitch.org  Wed Jun 20 20:22:52 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Jun 2012 09:22:52 -0700
Subject: [Freeswitch-users] My First FreeSwitch OR Is this easy to do?
In-Reply-To: <CAMv5evnVG10fKoZmyZVVjh1nqkTCscmGWFYG9d0F5U7W1X40HQ@mail.gmail.com>
References: <CAMv5evnVG10fKoZmyZVVjh1nqkTCscmGWFYG9d0F5U7W1X40HQ@mail.gmail.com>
Message-ID: <CAKzWOxUSHUF3SfvoO9cj-qHQMzVHa=YxYHYzByxudTi-QA1VfA@mail.gmail.com>

Hi Matt,

Welcome to the wacky world of VoIP! It's a wild ride, so hang on tight.

If you're technically inclined but just new to the VoIP scene then your
learning curve won't be so bad. I personally like the idea of a newbie
downloading something like blue.box or FusionPBX in order to get started.
Of the two, blue.box does more abstracting away of the underlying
FreeSWITCH configs. It's really just a personal choice which way to go.
However, if you're looking for a professionally designed system and don't
mind spending a few bucks then you might want to check out
www.cudatel.com- it's also powered by FreeSWITCH.

Happy VoIPing!

-MC

On Tue, Jun 19, 2012 at 7:30 PM, Matt Neimeyer <matt at neimeyer.org> wrote:

> I apologize for the length I'm not sure yet what is "relevant"
>
> My partner and I run a small business out of our home and we're
> getting big enough that I want to take advantage of some "modern phone
> technology" primarily we're interesting in an auto-attendant at this
> point but more will come later I'm sure.
>
> We have two analog lines coming into the house (provided via Verizon
> FiOS) and then in every room we have two line capable analog phones.
> In our FiOS settings I have both lines ring both lines simultaneously
> (to emulate a hunt group) so if someone is on line 1 the incoming call
> still rings audibly on line 2 instead of just buzzing call waiting.
>
> I will be getting a (hand me down) TDM1600P V2 card which, in its
> current configuration, has four FXO and four FXS ports. So my plan was
> to install FreeSwitch and "splice in-line" where the phone lines come
> out of the FiOS box. So two lines would go into the FreeSwitch Box and
> two "extensions" would come out the other side. Whenever I rewire the
> house I'll decide then whether to run a data connection to each
> "endpoint" and get SIP phones, add more FXS modules to the TDM1600P,
> some mix, or whatever but I'm starting small (I think).
>
> As far the auto-attendant goes 95% of what I need now is to filter
> "non-humans". So I would set up a simple menu "Press 1 for Tech
> Support, Press 2 for Billing, Etc)". This should filter autodialers
> that want us to update our Google listing and so on... but I assume we
> can still have all of the options (until we hire on more staff) ring
> both extensions. Assuming this lets unanswered calls go to voice
> mail... On our analog phones though I'd really love to modify the
> Caller ID to show which option they selected. So instead of "Bob
> Customer / 234-567-8901" it might show "TS - Bob Customer /
> 234-567-8901" if they selected Tech Support. If possible I'd also like
> to "fill gaps" in Caller ID from a database because a few of our
> customers only show the number and not a name.
>
> Short term, I'm currently planning on downloading the Blue.box ISO and
> using that as my start point. Medium term I'd like to set up email
> notification of voice mails. Long term I'd love to integrate Google
> Voice and/or Skype. Really long term I might need to "convert" to a
> multi-tenant setup for Personal, Company (now with Employees) and Side
> Project Company.
>
> I think that's it... Any advice? Comments?
>
> Matt
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Jun 20 21:41:10 2012
From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Wed, 20 Jun 2012 14:41:10 -0300
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <COL124-DS1734BA1C8437EE24E71567DDFE0@phx.gbl>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
	<4FDFBCA8.1050700@quentustech.com>
	<CAKzWOxWR6PJqPb6NqTLRQa5WMGP6H4zVte1w0i2UZSDN9+fQMg@mail.gmail.com>
	<COL124-DS1734BA1C8437EE24E71567DDFE0@phx.gbl>
Message-ID: <74D29BFA17294F97A0A8C40A49538E56@freeswitch.org>

Dude, I have been trying not to follow this thread but I can't avoid it because it keeps showing up on my mail client over and over again.

Stop bitching and DO SOMETHING. This is open source, not something you've bought or that someone is trying to sell you. Your demanding tone is irritating to say the very least. You don't like that someone changed a few dirs and did not make the work like you expected, then just freaking FIX IT and contribute the solution. None of the developers have time to write documentation and they struggle to keep their day jobs while making it work for the rest of the world as well and you sit on your comfy place complaining that they are not doing enough? WTF?

SwK is 1 person that has been doing the stable branch as well as trying to package freeswitch to several distros, he could definitely use some help instead of aggressiveness and complaints. Contribute and hop in or live with what you have or just leave quiet but DON'T piss on everyones else's work. You want to be disappointed, that's OK, but you gotta earn it first. What have you done for the project so far?

Sorry to the list for my email but everything has its limits and I tend to be nice most of the times.


--  
Jo?o Mesquita
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Wednesday, June 20, 2012 at 11:57 AM, shadowym wrote:

>  
> Did the announcement also say they were getting rid of the conf directory and not bothering to updated directory settings in the default conf files (that are now in the root directory).  Do they plan to update all the freeswitch documentation now as well which talks about the conf directory all over the place?
>  
>  
>   
>  
>  
> How long ago was this announcement made?  The issue is not about being caught off guard btw.  It?s about why and how it?s being done.
>  
>  
>   
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
> Sent: Monday, June 18, 2012 5:43 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] freeswitch directories moved
>  
>  
>   
>  
>  
>   
>  
>  
> On Mon, Jun 18, 2012 at 4:41 PM, William King <william.king at quentustech.com (mailto:william.king at quentustech.com)> wrote:
>  
>  
> I believe there was an announcement on this mailing list alerting that the new rpms would be undergoing major changes to include moving the directory locations. Search this list for new threads started by SwK in the past 6 months.
>  
>  
>  
>   
>  
>  
>  
> There was indeed an announcement by SwK himself. It was as close to FNLoIP (Flashing Neon Lights over IP) as you can get.
> -MC
>  
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> Join Us At ClueCon - Aug 7-9, 2012
>  
> 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
>  
>  


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From matt at neimeyer.org  Wed Jun 20 21:30:05 2012
From: matt at neimeyer.org (Matt Neimeyer)
Date: Wed, 20 Jun 2012 13:30:05 -0400
Subject: [Freeswitch-users] My First FreeSwitch OR Is this easy to do?
In-Reply-To: <CAKzWOxUSHUF3SfvoO9cj-qHQMzVHa=YxYHYzByxudTi-QA1VfA@mail.gmail.com>
References: <CAMv5evnVG10fKoZmyZVVjh1nqkTCscmGWFYG9d0F5U7W1X40HQ@mail.gmail.com>
	<CAKzWOxUSHUF3SfvoO9cj-qHQMzVHa=YxYHYzByxudTi-QA1VfA@mail.gmail.com>
Message-ID: <CAMv5evmwHJfepMS04EDUT77E9Gf-onYZhnkt2YwzSqUTT7aeaQ@mail.gmail.com>

I am technically inclined... to the point where I debated setting up a
Gentoo box for this because that's what I'm used to administering but
finally decided that the risk of grumbles from the household if it
doesn't "just work" for them necessitated using something a little
more "off the shelf".

As to the blue.box vs FusionPBX... Since this is my first time around
is the extra abstraction a good thing? I assume (dangerous I know)
that if at some point later I change my setup I can copy my
configuration and use it with a new GUI?

Thanks,

Matt

On Wed, Jun 20, 2012 at 12:22 PM, Michael Collins <msc at freeswitch.org> wrote:
> Hi Matt,
>
> Welcome to the wacky world of VoIP! It's a wild ride, so hang on tight.
>
> If you're technically inclined but just new to the VoIP scene then your
> learning curve won't be so bad. I personally like the idea of a newbie
> downloading something like blue.box or FusionPBX in order to get started. Of
> the two, blue.box does more abstracting away of the underlying FreeSWITCH
> configs. It's really just a personal choice which way to go. However, if
> you're looking for a professionally designed system and don't mind spending
> a few bucks then you might want to check out www.cudatel.com - it's also
> powered by FreeSWITCH.
>
> Happy VoIPing!
>
> -MC
>
> On Tue, Jun 19, 2012 at 7:30 PM, Matt Neimeyer <matt at neimeyer.org> wrote:
>>
>> I apologize for the length I'm not sure yet what is "relevant"
>>
>> My partner and I run a small business out of our home and we're
>> getting big enough that I want to take advantage of some "modern phone
>> technology" primarily we're interesting in an auto-attendant at this
>> point but more will come later I'm sure.
>>
>> We have two analog lines coming into the house (provided via Verizon
>> FiOS) and then in every room we have two line capable analog phones.
>> In our FiOS settings I have both lines ring both lines simultaneously
>> (to emulate a hunt group) so if someone is on line 1 the incoming call
>> still rings audibly on line 2 instead of just buzzing call waiting.
>>
>> I will be getting a (hand me down) TDM1600P V2 card which, in its
>> current configuration, has four FXO and four FXS ports. So my plan was
>> to install FreeSwitch and "splice in-line" where the phone lines come
>> out of the FiOS box. So two lines would go into the FreeSwitch Box and
>> two "extensions" would come out the other side. Whenever I rewire the
>> house I'll decide then whether to run a data connection to each
>> "endpoint" and get SIP phones, add more FXS modules to the TDM1600P,
>> some mix, or whatever but I'm starting small (I think).
>>
>> As far the auto-attendant goes 95% of what I need now is to filter
>> "non-humans". So I would set up a simple menu "Press 1 for Tech
>> Support, Press 2 for Billing, Etc)". This should filter autodialers
>> that want us to update our Google listing and so on... but I assume we
>> can still have all of the options (until we hire on more staff) ring
>> both extensions. Assuming this lets unanswered calls go to voice
>> mail... On our analog phones though I'd really love to modify the
>> Caller ID to show which option they selected. So instead of "Bob
>> Customer / 234-567-8901" it might show "TS - Bob Customer /
>> 234-567-8901" if they selected Tech Support. If possible I'd also like
>> to "fill gaps" in Caller ID from a database because a few of our
>> customers only show the number and not a name.
>>
>> Short term, I'm currently planning on downloading the Blue.box ISO and
>> using that as my start point. Medium term I'd like to set up email
>> notification of voice mails. Long term I'd love to integrate Google
>> Voice and/or Skype. Really long term I might need to "convert" to a
>> multi-tenant setup for Personal, Company (now with Employees) and Side
>> Project Company.
>>
>> I think that's it... Any advice? Comments?
>>
>> Matt
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From lazyvirus at gmx.com  Wed Jun 20 22:24:47 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Wed, 20 Jun 2012 20:24:47 +0200
Subject: [Freeswitch-users] My First FreeSwitch OR Is this easy to do?
In-Reply-To: <CAKzWOxUSHUF3SfvoO9cj-qHQMzVHa=YxYHYzByxudTi-QA1VfA@mail.gmail.com>
References: <CAMv5evnVG10fKoZmyZVVjh1nqkTCscmGWFYG9d0F5U7W1X40HQ@mail.gmail.com>
	<CAKzWOxUSHUF3SfvoO9cj-qHQMzVHa=YxYHYzByxudTi-QA1VfA@mail.gmail.com>
Message-ID: <20120620202447.5b230b4d@anubis.defcon1>

On Wed, 20 Jun 2012 09:22:52 -0700
Michael Collins <msc at freeswitch.org> wrote:

>  Of the two, blue.box does more abstracting away of
> the underlying FreeSWITCH configs.

Hmm, you also must be able to install it, which wasn't the case for
me (Debian sid): I never even had the first screen (tested w/ nginx,
lighttpd & apache2:(
On the other hand, FusionPBX installation was a piece of cake.

My ?0.015743

JY
-- 
Hate the sin and love the sinner.
		-- Mahatma Gandhi


From avi at avimarcus.net  Wed Jun 20 22:35:29 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 20 Jun 2012 21:35:29 +0300
Subject: [Freeswitch-users] My First FreeSwitch OR Is this easy to do?
In-Reply-To: <CAMv5evmwHJfepMS04EDUT77E9Gf-onYZhnkt2YwzSqUTT7aeaQ@mail.gmail.com>
References: <CAMv5evnVG10fKoZmyZVVjh1nqkTCscmGWFYG9d0F5U7W1X40HQ@mail.gmail.com>
	<CAKzWOxUSHUF3SfvoO9cj-qHQMzVHa=YxYHYzByxudTi-QA1VfA@mail.gmail.com>
	<CAMv5evmwHJfepMS04EDUT77E9Gf-onYZhnkt2YwzSqUTT7aeaQ@mail.gmail.com>
Message-ID: <CALNZuwn2V9DGBEe09ukQ7bFjapwkjwf1+4itVSNxsZEdj9mj=g@mail.gmail.com>

Neither interface, afaik, can yet suck-in old configs. So you could use the
old configs, but not manage them fully from within the interface.

-Avi


On Wed, Jun 20, 2012 at 8:30 PM, Matt Neimeyer <matt at neimeyer.org> wrote:

> I am technically inclined... to the point where I debated setting up a
> Gentoo box for this because that's what I'm used to administering but
> finally decided that the risk of grumbles from the household if it
> doesn't "just work" for them necessitated using something a little
> more "off the shelf".
>
> As to the blue.box vs FusionPBX... Since this is my first time around
> is the extra abstraction a good thing? I assume (dangerous I know)
> that if at some point later I change my setup I can copy my
> configuration and use it with a new GUI?
>
> Thanks,
>
> Matt
>
> On Wed, Jun 20, 2012 at 12:22 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Hi Matt,
> >
> > Welcome to the wacky world of VoIP! It's a wild ride, so hang on tight.
> >
> > If you're technically inclined but just new to the VoIP scene then your
> > learning curve won't be so bad. I personally like the idea of a newbie
> > downloading something like blue.box or FusionPBX in order to get
> started. Of
> > the two, blue.box does more abstracting away of the underlying FreeSWITCH
> > configs. It's really just a personal choice which way to go. However, if
> > you're looking for a professionally designed system and don't mind
> spending
> > a few bucks then you might want to check out www.cudatel.com - it's also
> > powered by FreeSWITCH.
> >
> > Happy VoIPing!
> >
> > -MC
> >
> > On Tue, Jun 19, 2012 at 7:30 PM, Matt Neimeyer <matt at neimeyer.org>
> wrote:
> >>
> >> I apologize for the length I'm not sure yet what is "relevant"
> >>
> >> My partner and I run a small business out of our home and we're
> >> getting big enough that I want to take advantage of some "modern phone
> >> technology" primarily we're interesting in an auto-attendant at this
> >> point but more will come later I'm sure.
> >>
> >> We have two analog lines coming into the house (provided via Verizon
> >> FiOS) and then in every room we have two line capable analog phones.
> >> In our FiOS settings I have both lines ring both lines simultaneously
> >> (to emulate a hunt group) so if someone is on line 1 the incoming call
> >> still rings audibly on line 2 instead of just buzzing call waiting.
> >>
> >> I will be getting a (hand me down) TDM1600P V2 card which, in its
> >> current configuration, has four FXO and four FXS ports. So my plan was
> >> to install FreeSwitch and "splice in-line" where the phone lines come
> >> out of the FiOS box. So two lines would go into the FreeSwitch Box and
> >> two "extensions" would come out the other side. Whenever I rewire the
> >> house I'll decide then whether to run a data connection to each
> >> "endpoint" and get SIP phones, add more FXS modules to the TDM1600P,
> >> some mix, or whatever but I'm starting small (I think).
> >>
> >> As far the auto-attendant goes 95% of what I need now is to filter
> >> "non-humans". So I would set up a simple menu "Press 1 for Tech
> >> Support, Press 2 for Billing, Etc)". This should filter autodialers
> >> that want us to update our Google listing and so on... but I assume we
> >> can still have all of the options (until we hire on more staff) ring
> >> both extensions. Assuming this lets unanswered calls go to voice
> >> mail... On our analog phones though I'd really love to modify the
> >> Caller ID to show which option they selected. So instead of "Bob
> >> Customer / 234-567-8901" it might show "TS - Bob Customer /
> >> 234-567-8901" if they selected Tech Support. If possible I'd also like
> >> to "fill gaps" in Caller ID from a database because a few of our
> >> customers only show the number and not a name.
> >>
> >> Short term, I'm currently planning on downloading the Blue.box ISO and
> >> using that as my start point. Medium term I'd like to set up email
> >> notification of voice mails. Long term I'd love to integrate Google
> >> Voice and/or Skype. Really long term I might need to "convert" to a
> >> multi-tenant setup for Personal, Company (now with Employees) and Side
> >> Project Company.
> >>
> >> I think that's it... Any advice? Comments?
> >>
> >> Matt
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Jun 20 22:43:10 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Jun 2012 11:43:10 -0700
Subject: [Freeswitch-users] My First FreeSwitch OR Is this easy to do?
In-Reply-To: <20120620202447.5b230b4d@anubis.defcon1>
References: <CAMv5evnVG10fKoZmyZVVjh1nqkTCscmGWFYG9d0F5U7W1X40HQ@mail.gmail.com>
	<CAKzWOxUSHUF3SfvoO9cj-qHQMzVHa=YxYHYzByxudTi-QA1VfA@mail.gmail.com>
	<20120620202447.5b230b4d@anubis.defcon1>
Message-ID: <CAKzWOxVDp8KX+fzw5Ys6xv8hdt3N9qJA-ozHk+YoryH-n42GeA@mail.gmail.com>

My personal opinion is this: if you just want a basic pbx w/ pretty
interface then use blue.box. If you want to learn more about the guts of
FreeSWITCH then use FusionPBX. Fusion is based loosely on the default
configuration of FS - the so-called "vanilla" config. It lets you see a lot
more of what's going on. In some respects it's more "complete" in that it
exposes just about everything you can do in FS. I would actually try them
both out. I don't know how easily they can handle your TDM card, etc. as
I've never done TDM with either of them.

Hope this helps...

-MC

On Wed, Jun 20, 2012 at 11:24 AM, Bzzz <lazyvirus at gmx.com> wrote:

> On Wed, 20 Jun 2012 09:22:52 -0700
> Michael Collins <msc at freeswitch.org> wrote:
>
> >  Of the two, blue.box does more abstracting away of
> > the underlying FreeSWITCH configs.
>
> Hmm, you also must be able to install it, which wasn't the case for
> me (Debian sid): I never even had the first screen (tested w/ nginx,
> lighttpd & apache2:(
> On the other hand, FusionPBX installation was a piece of cake.
>
> My ?0.015743
>
> JY
> --
> Hate the sin and love the sinner.
>                -- Mahatma Gandhi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lazyvirus at gmx.com  Wed Jun 20 22:50:56 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Wed, 20 Jun 2012 20:50:56 +0200
Subject: [Freeswitch-users] My First FreeSwitch OR Is this easy to do?
In-Reply-To: <CAKzWOxVDp8KX+fzw5Ys6xv8hdt3N9qJA-ozHk+YoryH-n42GeA@mail.gmail.com>
References: <CAMv5evnVG10fKoZmyZVVjh1nqkTCscmGWFYG9d0F5U7W1X40HQ@mail.gmail.com>
	<CAKzWOxUSHUF3SfvoO9cj-qHQMzVHa=YxYHYzByxudTi-QA1VfA@mail.gmail.com>
	<20120620202447.5b230b4d@anubis.defcon1>
	<CAKzWOxVDp8KX+fzw5Ys6xv8hdt3N9qJA-ozHk+YoryH-n42GeA@mail.gmail.com>
Message-ID: <20120620205056.29dccd2e@anubis.defcon1>

On Wed, 20 Jun 2012 11:43:10 -0700
Michael Collins <msc at freeswitch.org> wrote:

> My personal opinion is this: if you just want a basic pbx w/ pretty
> interface then use blue.box. If you want to learn more about the
> guts of FreeSWITCH then use FusionPBX.

This was my conclusion too after testing the bb iso.

-- 
Lao Tseu say: Man that scratch his butt at night
awakes with stinking fingers.


From msc at freeswitch.org  Wed Jun 20 22:52:28 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Jun 2012 11:52:28 -0700
Subject: [Freeswitch-users] TERMINATOR_USED_VARIABLE
In-Reply-To: <4FE1B1A2.2050601@thewinelake.com>
References: <4FE1AF80.7060800@thewinelake.com>
	<4FE1B1A2.2050601@thewinelake.com>
Message-ID: <CAKzWOxV=a0vn-m7WqpXOOHS8yt22ra3ykRUpZbFvvehwaXan0Q@mail.gmail.com>

Thanks for doing the leg work and wikifying it!
-MC

On Wed, Jun 20, 2012 at 4:18 AM, Alex <alex at thewinelake.com> wrote:

> Oh yeay... Victory is mine!
>
>  lastDigit = session:getVariable("read_terminator_used")
>
> Must put that in the PAGD wiki....
> > I note that in switch_ivr_play_say.c there's an intriguing reference to
> > SWITCH_READ_TERMINATOR_USED_VARIABLE on line 2015:
> >
> >           if ((min_digits == 0) && (strlen(digit_buffer) == 0) &&
> > switch_channel_get_variable(channel,
> > SWITCH_READ_TERMINATOR_USED_VARIABLE) != 0)
> >
> > It seems to me that this suggests that the terminator used in PAGD is
> > stored, at least temporarily in a channel variable.
> >
> > Is there any way I can get my mitts on the value of this following a
> > call to PAYG?
> >
> >
>
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From terry at digital-outpost.com  Wed Jun 20 22:53:16 2012
From: terry at digital-outpost.com (Terry Barnum)
Date: Wed, 20 Jun 2012 11:53:16 -0700
Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
In-Reply-To: <4A48F47D-7954-496D-A31E-CE5E1AD21E9D@dons.net.au>
References: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
	<0AEB7B3A-5591-4694-83B5-E4208D5DA4A0@dons.net.au>
	<46D48718-2661-41EC-8A59-67BFEBCCF745@digital-outpost.com>
	<31354FBA-5DBC-47A9-9958-9EC8A515DACD@dons.net.au>
	<3D137477-E72E-4A59-A625-899DFB15D387@digital-outpost.com>
	<4A48F47D-7954-496D-A31E-CE5E1AD21E9D@dons.net.au>
Message-ID: <F998E99A-AD61-4F7E-822E-F6A237752940@digital-outpost.com>


On Jun 20, 2012, at 1:05 AM, Daniel O'Connor wrote:

> On 20/06/2012, at 3:51, Terry Barnum wrote:
>> So I did a make clean and ran your env line and then make. I think it got further but fails on mod_spandsp_la-mod_spandsp_fax.lo
>> 
>> Creating mod_spandsp_la-mod_spandsp_fax.lo
>> quiet_libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/opt/local/include -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -pipe -no-cpp-precomp -MT mod_spandsp_la-mod_spandsp_fax.lo -MD -MP -MF .deps/mod_s
> pandsp_la-mod_spandsp_fax.Tpo -c mod_spandsp_fax.c  -fno-common -DPIC -o .libs/mod_spandsp_la-mod_spandsp_fax.o
>> cc1: warnings being treated as errors
>> mod_spandsp_fax.c: In function 'spanfax_init':
>> mod_spandsp_fax.c:716: warning: passing argument 2 of 'span_log_set_message_handler' from incompatible pointer type
>> mod_spandsp_fax.c:716: error: too many arguments to function 'span_log_set_message_handler'
>> mod_spandsp_fax.c:717: warning: passing argument 2 of 'span_log_set_message_handler' from incompatible pointer type
>> mod_spandsp_fax.c:717: error: too many arguments to function 'span_log_set_message_handler'
>> mod_spandsp_fax.c:772: warning: passing argument 2 of 'span_log_set_message_handler' from incompatible pointer type
>> mod_spandsp_fax.c:772: error: too many arguments to function 'span_log_set_message_handler'
>> mod_spandsp_fax.c:773: warning: passing argument 2 of 'span_log_set_message_handler' from incompatible pointer type
> 
> I think this is because you have spandsp installed in MacPorts and those includes are being used in preference to the ones in libs/spandsp/..
> 
> I think to work around this I just removed spandsp while I built freeswitch (bleh).

THanks Daniel. Awhile back I was fooling around with the asterisk macport on this work iMac and spandsp-devel is a dependency. I uninstalled the asterisk port long ago but spandsp-devel was still around.

So it's all in one place, to get freeswitch to compile on this intel iMac running 10.6.8 with macports installed, I uninstalled the spandsp-devel macport, did a fresh git clone, ran your patch <http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-OSX-Libtool-Macports-td7579970.html#a7580004> to force the use of C in gsm0610_rpe.c, ran the jira patch to correctly select x86 instead of ppc <http://jira.freeswitch.org/browse/FS-3992>, ran bootstrap.sh, used your env CPPFLAGS=-I/opt/local/include LDFLAGS=-L/opt/local/lib ./configure and make.

Now on to actually learning to use freeswitch!

> IMO the "right" solution is to rename spandsp in freeswitch - it has been modified (looking at the git log) so it is really a fork of the original. That is unless those patches get merged back into spandsp and then macports can get an updated one.
> 
> The later solution is still pretty bad though unless there is a way to force freeswitch to not look at the bundled version.
> 
> There is a similar problem with portaudio although since it has a stable API it doesn't show up. It doesn't however build on OSX (Intel 10.7 anyway) and in the end I hacked src/mod/endpoint/mod_portaudio to use pkg-config to find the right CFLAGS and LIBS to link to it.

Forgive my ignorance, does the freeswitch includes of things like spandsp and portaudio have to be set globally or can there be separate configs?

-Terry



From sdevoy at bizfocused.com  Wed Jun 20 23:27:57 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 20 Jun 2012 15:27:57 -0400
Subject: [Freeswitch-users] gateway registration using TLS
In-Reply-To: <20120620162020.182950@gmx.com>
References: <20120620162020.182950@gmx.com>
Message-ID: <0a7c01cd4f1a$cc7a3f00$656ebd00$@bizfocused.com>

NO, sorry.  I just saw ?remote_ip_here? in the error message and thought may you actually had used that.

 

Sorry.

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mi Ke
Sent: Wednesday, June 20, 2012 12:20 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] gateway registration using TLS

 

Dear Sean,

It's not that bad - remote_ip, xxx and yyy are only placeholders in my example - actual config has real params ;)

Did you mean that FS will automatically use TLS for registration when I append :5061 to realm value ?

Thanks / MiKe

 

----- Original Message -----

From: Sean Devoy

Sent: 06/20/12 07:03 PM

To: 'FreeSWITCH Users Help'

Subject: Re: [Freeswitch-users] gateway registration using TLS

 

Mi Ke,

 

 

 

 

 

You should put your actual remote provider?s domain name or tip address in where it says ?remote_ip_here? and I assume you have entered you actual username and password where it says ?yyy? and ?zzz?.

 

 

It should look SIMILAR to this, but with your provider: <param name="realm" value="sip.voipinnovations.com:5061"/>

 

 

Hope that helps,

 

 

Sean

 

 

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mi Ke
Sent: Wednesday, June 20, 2012 11:46 AM
To: Freeswitch Users
Subject: [Freeswitch-users] gateway registration using TLS

 

 

 

 

 

Hi All,

Is it possible to explicitly tell sofia to register as a client using TLS? Here's my TLS-related part of sofia config:

        <profiles>
            <profile name="external">
                <domains>
                    <domain name="all" alias="true" parse="false"/>
                </domains>
                <gateways>
                        <gateway name="xxx">
                                  <param name="register" value="true"/>
                                  <param name="username" value="yyy"/>
                                  <param name="realm" value="remote_ip_here:5061"/>
                                  <param name="password" value="zzz"/>
                                  <param name="register-transport" value="tcp"/>
                                  <param name="expire-seconds" value="3600"/>
                                  <param name="retry-seconds" value="60" />
                         </gateway>
                </gateways>
                <settings>
                       ...
                       <param name="context" value="external"/>
                       <param name="sip-port" value="5060"/>
                       <param name="tls" value="true"/>
                       <param name="tls-only" value="false"/>
                       <param name="tls-bind-params" value="transport=tls"/>
                       <param name="contact-params" value="tport=tcp"/>
                       <param name="tls-version" value="sslv23"/>
                       <param name="tls-cert-dir" value="/usr/local/freeswitch/conf/ssl"/>
                       <param name="tls-sip-port" value="5061"/>
                       <param name="dialplan" value="XML"/>
                       ...

                </settings>
            </profile>
        </profiles>

But that does not work...

tport_connected( 

 

 

0x7f9b44014110): events CONNECTED
tport_send_event(0x7f9b44014110) - ready to send to (tcp/remote_ip_here:5061 <http://212.58.166.54:5061> )
tport_vsend(0x7f9b44014110): 600 bytes of 600 to tcp/remote_ip_here:5061 <http://212.58.166.54:5061> 
tport_vsend returned 600
tport_set_events(0x7f9b44014110): events IN
tport(0x7f9b44014110): reset timer
tport_wakeup(0x7f9b44014110): events IN HUP ERR
tport_release(0x7f9b44014110): 0x7f9b4401a460 by 0x7f9b4402ca70 with (nil)
nta: REGISTER (29733430): Connection reset by peer (104) with tcp/[remote_ip_here]:5061
nua(0x7f9b440196a0): event r_register 503 Service Unavailable 

 

 


Any ideas what I'm doing wrong?

Thanks / MiKe

 

 

 

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From sdevoy at bizfocused.com  Wed Jun 20 23:47:22 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 20 Jun 2012 15:47:22 -0400
Subject: [Freeswitch-users] Users need a red flashy light thingy
Message-ID: <0aa601cd4f1d$83117e20$89347a60$@bizfocused.com>

Freeswitch is so much better than my customers deserve.

 

I have the DialPlan set up nicely so they can dial *72 to forward all DID
calls to their answering service and *73 to turn it off.  Today she Dialed
*72 to turn it off 4 different times, and since that is the code to turn it
on --- it did NOT turn off.  She was livid and sent me hate mail about how
"terrible IP phones are and they never work ."

 

I even consoled her saying how *72 and *73 have no inherent indicators to
show which means on and off.   I have offered the obvious solutions of SPEED
DIAL buttons that are labeled "Enable" and "Disable".

 

Now my question.  Anyone have any ideas how to use a line button, or
possibly MWI to indicate that lines are forwarded (a db value).  MWI is
clearly in use so line button is my preference, but if MWI had a flash
function, etc????  I have unused line buttons - How can I make them show
busy (like shared call appearance) from the switch?  Maybe a dummy
user/extension I do something clever with?

 

If it matters the phones are Cisco 504G (4 line).

 

Thanks in advance,

Sean

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From avi at avimarcus.net  Thu Jun 21 00:03:27 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 20 Jun 2012 23:03:27 +0300
Subject: [Freeswitch-users] Users need a red flashy light thingy
In-Reply-To: <0aa601cd4f1d$83117e20$89347a60$@bizfocused.com>
References: <0aa601cd4f1d$83117e20$89347a60$@bizfocused.com>
Message-ID: <CALNZuwmpnsX2CypOSVGu1zus0cFK0CN0UYxqNrLXKbJxTZiXFA@mail.gmail.com>

If that's an actual call.. we have callie recordings for that.
      <prompt phrase="Call forwarding has been cancelled."
filename="ivr/ivr-call_forwarding_has_been_cancelled.wav"/>
      <prompt phrase="Call forwarding has been set."
filename="ivr/ivr-call_forwarding_has_been_set.wav"/>

-Avi


On Wed, Jun 20, 2012 at 10:47 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Freeswitch is so much better than my customers deserve.****
>
> ** **
>
> I have the DialPlan set up nicely so they can dial *72 to forward all DID
> calls to their answering service and *73 to turn it off.  Today she Dialed
> *72 to turn it off 4 different times, and since that is the code to turn it
> on --- it did NOT turn off.  She was livid and sent me hate mail about how
> ?terrible IP phones are and they never work ??****
>
> ** **
>
> I even consoled her saying how *72 and *73 have no inherent indicators to
> show which means on and off.   I have offered the obvious solutions of
> SPEED DIAL buttons that are labeled ?Enable? and ?Disable?.****
>
> ** **
>
> Now my question.  Anyone have any ideas how to use a line button, or
> possibly MWI to indicate that lines are forwarded (a db value).  MWI is
> clearly in use so line button is my preference, but if MWI had a flash
> function, etc????  I have unused line buttons ? How can I make them show
> busy (like shared call appearance) from the switch?  Maybe a dummy
> user/extension I do something clever with?****
>
> ** **
>
> If it matters the phones are Cisco 504G (4 line).****
>
> ** **
>
> Thanks in advance,****
>
> Sean****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sdevoy at bizfocused.com  Thu Jun 21 00:16:44 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 20 Jun 2012 16:16:44 -0400
Subject: [Freeswitch-users] Users need a red flashy light thingy
In-Reply-To: <CALNZuwmpnsX2CypOSVGu1zus0cFK0CN0UYxqNrLXKbJxTZiXFA@mail.gmail.com>
References: <0aa601cd4f1d$83117e20$89347a60$@bizfocused.com>
	<CALNZuwmpnsX2CypOSVGu1zus0cFK0CN0UYxqNrLXKbJxTZiXFA@mail.gmail.com>
Message-ID: <0ad501cd4f21$9ccabda0$d66038e0$@bizfocused.com>

Yes, thanks Avi.  I do  play those recordings when they dial *72 and *73.
One cannot expect his customer to actually listen the prompt you play back,
can they!

 

I just googled some more and found something in one Anthony's books (which I
have now ordered both of)

http://books.google.com/books?id=-6nzkugPG70C
<http://books.google.com/books?id=-6nzkugPG70C&pg=PT540&lpg=PT540&dq=freeswi
tch+event+turn+lights&source=bl&ots=4-5IfOJ3rF&sig=BsSDSKva7FKfzZ0w2HFeWJQ5_
ek&hl=en&sa=X&ei=2S7iT639OYTS6gG2v-0H&sqi=2&ved=0CFEQ6AEwAA#v=onepage&q=free
switch%20event%20turn%20lights&f=false>
&pg=PT540&lpg=PT540&dq=freeswitch+event+turn+lights&source=bl&ots=4-5IfOJ3rF
&sig=BsSDSKva7FKfzZ0w2HFeWJQ5_ek&hl=en&sa=X&ei=2S7iT639OYTS6gG2v-0H&sqi=2&ve
d=0CFEQ6AEwAA#v=onepage&q=freeswitch%20event%20turn%20lights&f=false

 

It appears that I can set a presence event to a dummy user and have the
phone's lines with that user logged in change colors!

 

Now . how to send a presence event in a dial plan????  Google don't fail me
now.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Wednesday, June 20, 2012 4:03 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Users need a red flashy light thingy

 

If that's an actual call.. we have callie recordings for that.

      <prompt phrase="Call forwarding has been cancelled."
filename="ivr/ivr-call_forwarding_has_been_cancelled.wav"/>

      <prompt phrase="Call forwarding has been set."
filename="ivr/ivr-call_forwarding_has_been_set.wav"/>




-Avi

 

On Wed, Jun 20, 2012 at 10:47 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Freeswitch is so much better than my customers deserve.

 

I have the DialPlan set up nicely so they can dial *72 to forward all DID
calls to their answering service and *73 to turn it off.  Today she Dialed
*72 to turn it off 4 different times, and since that is the code to turn it
on --- it did NOT turn off.  She was livid and sent me hate mail about how
"terrible IP phones are and they never work ."

 

I even consoled her saying how *72 and *73 have no inherent indicators to
show which means on and off.   I have offered the obvious solutions of SPEED
DIAL buttons that are labeled "Enable" and "Disable".

 

Now my question.  Anyone have any ideas how to use a line button, or
possibly MWI to indicate that lines are forwarded (a db value).  MWI is
clearly in use so line button is my preference, but if MWI had a flash
function, etc????  I have unused line buttons - How can I make them show
busy (like shared call appearance) from the switch?  Maybe a dummy
user/extension I do something clever with?

 

If it matters the phones are Cisco 504G (4 line).

 

Thanks in advance,

Sean


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

 

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From toddb at toddbailey.net  Thu Jun 21 00:21:39 2012
From: toddb at toddbailey.net (toddb at toddbailey.net)
Date: Wed, 20 Jun 2012 13:21:39 -0700
Subject: [Freeswitch-users] Different Voices for FS
Message-ID: <20120620132139.33e327b490679d2282e332758c73b55b.6e807b7d68.wbe@email14.secureserver.net>

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From toddb at toddbailey.net  Thu Jun 21 00:23:29 2012
From: toddb at toddbailey.net (toddb at toddbailey.net)
Date: Wed, 20 Jun 2012 13:23:29 -0700
Subject: [Freeswitch-users] Users need a red flashy light thingy
Message-ID: <20120620132329.33e327b490679d2282e332758c73b55b.1e0469934d.wbe@email14.secureserver.net>

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From bdfoster at endigotech.com  Thu Jun 21 00:45:45 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 20 Jun 2012 16:45:45 -0400
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <74D29BFA17294F97A0A8C40A49538E56@freeswitch.org>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
	<4FDFBCA8.1050700@quentustech.com>
	<CAKzWOxWR6PJqPb6NqTLRQa5WMGP6H4zVte1w0i2UZSDN9+fQMg@mail.gmail.com>
	<COL124-DS1734BA1C8437EE24E71567DDFE0@phx.gbl>
	<74D29BFA17294F97A0A8C40A49538E56@freeswitch.org>
Message-ID: <CAKV-jFwyH7ybQycVu_DR+K2h_dhijEXiJ+wzrky_terdVpe4Lg@mail.gmail.com>

As I don't think this is going to turn into a productive discussion, the
thread should probably be closed.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 20, 2012 1:44 PM, "Jo?o Mesquita" <jmesquita at freeswitch.org> wrote:

> Dude, I have been trying not to follow this thread but I can't avoid it
> because it keeps showing up on my mail client over and over again.
>
> Stop bitching and DO SOMETHING. This is open source, not something you've
> bought or that someone is trying to sell you. Your demanding tone is
> irritating to say the very least. You don't like that someone changed a few
> dirs and did not make the work like you expected, then just freaking FIX IT
> and contribute the solution. None of the developers have time to write
> documentation and they struggle to keep their day jobs while making it work
> for the rest of the world as well and you sit on your comfy place
> complaining that they are not doing enough? WTF?
>
> SwK is 1 person that has been doing the stable branch as well as trying to
> package freeswitch to several distros, he could definitely use some help
> instead of aggressiveness and complaints. Contribute and hop in or live
> with what you have or just leave quiet but DON'T piss on everyones else's
> work. You want to be disappointed, that's OK, but you gotta earn it first.
> What have you done for the project so far?
>
> Sorry to the list for my email but everything has its limits and I tend to
> be nice most of the times.
>
>
> --
> Jo?o Mesquita
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Wednesday, June 20, 2012 at 11:57 AM, shadowym wrote:
>
> Did the announcement also say they were getting rid of the conf directory
> and not bothering to updated directory settings in the default conf files
> (that are now in the root directory).  Do they plan to update all the
> freeswitch documentation now as well which talks about the conf directory
> all over the place?****
>
> ** **
>
> How long ago was this announcement made?  The issue is not about being
> caught off guard btw.  It?s about why and how it?s being done.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
> *On Behalf Of *Michael Collins
> *Sent:* Monday, June 18, 2012 5:43 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] freeswitch directories moved****
>
> ** **
>
> ** **
>
> On Mon, Jun 18, 2012 at 4:41 PM, William King <
> william.king at quentustech.com> wrote:****
>
> I believe there was an announcement on this mailing list alerting that the
> new rpms would be undergoing major changes to include moving the directory
> locations. Search this list for new threads started by SwK in the past 6
> months.****
>
> ** **
>
> There was indeed an announcement by SwK himself. It was as close to FNLoIP
> (Flashing Neon Lights over IP) as you can get.
> -MC****
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Thu Jun 21 00:46:50 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 20 Jun 2012 16:46:50 -0400
Subject: [Freeswitch-users] Users need a red flashy light thingy
In-Reply-To: <0aa601cd4f1d$83117e20$89347a60$@bizfocused.com>
References: <0aa601cd4f1d$83117e20$89347a60$@bizfocused.com>
Message-ID: <CAKV-jFwYjJM6=G+kzEwGZcA8CwEARmRY6XOgQV0TkEDazOrJEQ@mail.gmail.com>

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 20, 2012 3:49 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> Freeswitch is so much better than my customers deserve.****
>
> ** **
>
> I have the DialPlan set up nicely so they can dial *72 to forward all DID
> calls to their answering service and *73 to turn it off.  Today she Dialed
> *72 to turn it off 4 different times, and since that is the code to turn it
> on --- it did NOT turn off.  She was livid and sent me hate mail about how
> ?terrible IP phones are and they never work ??****
>
> ** **
>
> I even consoled her saying how *72 and *73 have no inherent indicators to
> show which means on and off.   I have offered the obvious solutions of
> SPEED DIAL buttons that are labeled ?Enable? and ?Disable?.****
>
> ** **
>
> Now my question.  Anyone have any ideas how to use a line button, or
> possibly MWI to indicate that lines are forwarded (a db value).  MWI is
> clearly in use so line button is my preference, but if MWI had a flash
> function, etc????  I have unused line buttons ? How can I make them show
> busy (like shared call appearance) from the switch?  Maybe a dummy
> user/extension I do something clever with?****
>
> ** **
>
> If it matters the phones are Cisco 504G (4 line).****
>
> ** **
>
> Thanks in advance,****
>
> Sean****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Thu Jun 21 00:48:48 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 20 Jun 2012 16:48:48 -0400
Subject: [Freeswitch-users] Users need a red flashy light thingy
In-Reply-To: <0aa601cd4f1d$83117e20$89347a60$@bizfocused.com>
References: <0aa601cd4f1d$83117e20$89347a60$@bizfocused.com>
Message-ID: <CAKV-jFxa=FvtGFRtaa3pWkuqWBVkJdfdH1ALCNxttmF7tWz_0g@mail.gmail.com>

Sorry for the blank email. Too many buttons :-)

You might want to put some logic in that checks to see if the DID is
forwarding to the answering service. If so, when she dials the major
extension, she can be notified that it has already been forwarded via a
recording. That's the simplest solution.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 20, 2012 3:49 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> Freeswitch is so much better than my customers deserve.****
>
> ** **
>
> I have the DialPlan set up nicely so they can dial *72 to forward all DID
> calls to their answering service and *73 to turn it off.  Today she Dialed
> *72 to turn it off 4 different times, and since that is the code to turn it
> on --- it did NOT turn off.  She was livid and sent me hate mail about how
> ?terrible IP phones are and they never work ??****
>
> ** **
>
> I even consoled her saying how *72 and *73 have no inherent indicators to
> show which means on and off.   I have offered the obvious solutions of
> SPEED DIAL buttons that are labeled ?Enable? and ?Disable?.****
>
> ** **
>
> Now my question.  Anyone have any ideas how to use a line button, or
> possibly MWI to indicate that lines are forwarded (a db value).  MWI is
> clearly in use so line button is my preference, but if MWI had a flash
> function, etc????  I have unused line buttons ? How can I make them show
> busy (like shared call appearance) from the switch?  Maybe a dummy
> user/extension I do something clever with?****
>
> ** **
>
> If it matters the phones are Cisco 504G (4 line).****
>
> ** **
>
> Thanks in advance,****
>
> Sean****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lazyvirus at gmx.com  Thu Jun 21 00:48:46 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Wed, 20 Jun 2012 22:48:46 +0200
Subject: [Freeswitch-users] Different Voices for FS
In-Reply-To: <20120620132139.33e327b490679d2282e332758c73b55b.6e807b7d68.wbe@email14.secureserver.net>
References: <20120620132139.33e327b490679d2282e332758c73b55b.6e807b7d68.wbe@email14.secureserver.net>
Message-ID: <20120620224846.6ff371dc@anubis.defcon1>

On Wed, 20 Jun 2012 13:21:39 -0700
<toddb at toddbailey.net> wrote:

> My FS default installation provided me voice menus in English with
> USA accent. Can this be changed to perhaps English with
> Austrailian, British or So. Africa accents? What if I want
> spanish, french  or  russian etc? 

This site provide different languages for reasonable rates:
http://www.westany.com/freeswitch

You can also try to use asterisk voices (but I don't know anything
about the licence, though).


And BTW, this isn't a bad question after all; can somebody give
me the best methodology to record my own voices?
Especially: is it better to record long texts and postprod cut/edit
them, or is it better to only record sentences and words?

And is there an URL to download pre-defined texts for that?

JY
-- 
Help save the whales: eat a Japanese


From toddb at toddbailey.net  Thu Jun 21 00:50:27 2012
From: toddb at toddbailey.net (toddb at toddbailey.net)
Date: Wed, 20 Jun 2012 13:50:27 -0700
Subject: [Freeswitch-users] Users need a red flashy light thingy
Message-ID: <20120620135027.33e327b490679d2282e332758c73b55b.16c5a7d48a.wbe@email14.secureserver.net>

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From bdfoster at endigotech.com  Thu Jun 21 00:51:34 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 20 Jun 2012 16:51:34 -0400
Subject: [Freeswitch-users] Users need a red flashy light thingy
In-Reply-To: <0aa601cd4f1d$83117e20$89347a60$@bizfocused.com>
References: <0aa601cd4f1d$83117e20$89347a60$@bizfocused.com>
Message-ID: <CAKV-jFyqVYdVQS6cmZqQS9LNSZnSuVBFVk8nHR_Xo1ny7jE+5Q@mail.gmail.com>

Or have your magic extension forward to an IVR, tells her the current
status, press one to enable, press 2 to disable, then confirm it instead of
two different numbers

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 20, 2012 3:49 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> Freeswitch is so much better than my customers deserve.****
>
> ** **
>
> I have the DialPlan set up nicely so they can dial *72 to forward all DID
> calls to their answering service and *73 to turn it off.  Today she Dialed
> *72 to turn it off 4 different times, and since that is the code to turn it
> on --- it did NOT turn off.  She was livid and sent me hate mail about how
> ?terrible IP phones are and they never work ??****
>
> ** **
>
> I even consoled her saying how *72 and *73 have no inherent indicators to
> show which means on and off.   I have offered the obvious solutions of
> SPEED DIAL buttons that are labeled ?Enable? and ?Disable?.****
>
> ** **
>
> Now my question.  Anyone have any ideas how to use a line button, or
> possibly MWI to indicate that lines are forwarded (a db value).  MWI is
> clearly in use so line button is my preference, but if MWI had a flash
> function, etc????  I have unused line buttons ? How can I make them show
> busy (like shared call appearance) from the switch?  Maybe a dummy
> user/extension I do something clever with?****
>
> ** **
>
> If it matters the phones are Cisco 504G (4 line).****
>
> ** **
>
> Thanks in advance,****
>
> Sean****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sdevoy at bizfocused.com  Thu Jun 21 00:57:27 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 20 Jun 2012 16:57:27 -0400
Subject: [Freeswitch-users] Users need a red flashy light thingy
In-Reply-To: <20120620132329.33e327b490679d2282e332758c73b55b.1e0469934d.wbe@email14.secureserver.net>
References: <20120620132329.33e327b490679d2282e332758c73b55b.1e0469934d.wbe@email14.secureserver.net>
Message-ID: <0b2f01cd4f27$4d2cc170$e7864450$@bizfocused.com>

HI Todd,

 

This too makes perfect sense, but NOT for this user.  The answering service does nothing but take voice messages, type them up and them back as FAXES to the office fax machine.  The users come in the morning and get their FAXES instead of voicemail. 

Can you imagine my frustration with these people?

 

Thanks,

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of toddb at toddbailey.net
Sent: Wednesday, June 20, 2012 4:23 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Users need a red flashy light thingy

 

maybe give up on call forwarding and just email a voice message.

 

-------- Original Message --------
Subject: Re: [Freeswitch-users] Users need a red flashy light thingy
From: "Sean Devoy" < <mailto:sdevoy at bizfocused.com> sdevoy at bizfocused.com>
Date: Wed, June 20, 2012 1:16 pm
To: "'FreeSWITCH Users Help'" < <mailto:freeswitch-users at lists.freeswitch.org> freeswitch-users at lists.freeswitch.org>

Yes, thanks Avi.  I do  play those recordings when they dial *72 and *73.  One cannot expect his customer to actually listen the prompt you play back, can they!

 

I just googled some more and found something in one Anthony?s books (which I have now ordered both of)

 <http://books.google.com/books?id=-6nzkugPG70C&pg=PT540&lpg=PT540&dq=freeswitch+event+turn+lights&source=bl&ots=4-5IfOJ3rF&sig=BsSDSKva7FKfzZ0w2HFeWJQ5_ek&hl=en&sa=X&ei=2S7iT639OYTS6gG2v-0H&sqi=2&ved=0CFEQ6AEwAA#v=onepage&q=freeswitch%20event%20turn%20lights&f=false> http://books.google.com/books?id=-6nzkugPG70C&pg=PT540&lpg=PT540&dq=freeswitch+event+turn+lights&source=bl&ots=4-5IfOJ3rF&sig=BsSDSKva7FKfzZ0w2HFeWJQ5_ek&hl=en&sa=X&ei=2S7iT639OYTS6gG2v-0H&sqi=2&ved=0CFEQ6AEwAA#v=onepage&q=freeswitch%20event%20turn%20lights&f=false

 

It appears that I can set a presence event to a dummy user and have the phone?s lines with that user logged in change colors!

 

Now ? how to send a presence event in a dial plan????  Google don?t fail me now.

 

From:  <mailto:freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org [ <mailto:freeswitch-users-bounces at lists.freeswitch.org> mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus
Sent: Wednesday, June 20, 2012 4:03 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Users need a red flashy light thingy

 

If that's an actual call.. we have callie recordings for that.

      <prompt phrase="Call forwarding has been cancelled." filename="ivr/ivr-call_forwarding_has_been_cancelled.wav"/>

      <prompt phrase="Call forwarding has been set." filename="ivr/ivr-call_forwarding_has_been_set.wav"/>




-Avi

 

On Wed, Jun 20, 2012 at 10:47 PM, Sean Devoy < <mailto:sdevoy at bizfocused.com> sdevoy at bizfocused.com> wrote:

Freeswitch is so much better than my customers deserve.

 

I have the DialPlan set up nicely so they can dial *72 to forward all DID calls to their answering service and *73 to turn it off.  Today she Dialed *72 to turn it off 4 different times, and since that is the code to turn it on --- it did NOT turn off.  She was livid and sent me hate mail about how ?terrible IP phones are and they never work ??

 

I even consoled her saying how *72 and *73 have no inherent indicators to show which means on and off.   I have offered the obvious solutions of SPEED DIAL buttons that are labeled ?Enable? and ?Disable?.

 

Now my question.  Anyone have any ideas how to use a line button, or possibly MWI to indicate that lines are forwarded (a db value).  MWI is clearly in use so line button is my preference, but if MWI had a flash function, etc????  I have unused line buttons ? How can I make them show busy (like shared call appearance) from the switch?  Maybe a dummy user/extension I do something clever with?

 

If it matters the phones are Cisco 504G (4 line).

 

Thanks in advance,

Sean


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
 <mailto:consulting at freeswitch.org> consulting at freeswitch.org
 <http://www.freeswitchsolutions.com/> http://www.freeswitchsolutions.com


 </> 

Official FreeSWITCH Sites
 <http://www.freeswitch.org/> http://www.freeswitch.org
 <http://wiki.freeswitch.org/> http://wiki.freeswitch.org
 <http://www.cluecon.com/> http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
 <mailto: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

 


  _____  


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
 <mailto:consulting at freeswitch.org> consulting at freeswitch.org
 <http://www.freeswitchsolutions.com> http://www.freeswitchsolutions.com


 <> 

Official FreeSWITCH Sites
 <http://www.freeswitch.org> http://www.freeswitch.org
 <http://wiki.freeswitch.org> http://wiki.freeswitch.org
 <http://www.cluecon.com> http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
 <mailto: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
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From sdevoy at bizfocused.com  Thu Jun 21 01:00:34 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 20 Jun 2012 17:00:34 -0400
Subject: [Freeswitch-users] Users need a red flashy light thingy
In-Reply-To: <CAKV-jFyqVYdVQS6cmZqQS9LNSZnSuVBFVk8nHR_Xo1ny7jE+5Q@mail.gmail.com>
References: <0aa601cd4f1d$83117e20$89347a60$@bizfocused.com>
	<CAKV-jFyqVYdVQS6cmZqQS9LNSZnSuVBFVk8nHR_Xo1ny7jE+5Q@mail.gmail.com>
Message-ID: <0b3401cd4f27$bc4c5390$34e4fab0$@bizfocused.com>

HI Brian,

 

This too is an excellent option for implementing the switch over and I will
probably do it.

 

However, nothing trumps a big red line button lit (red) when the phones are
forwarded.

 

Does anyone know how to send a PRESENCE event in an XML dialplan?  

Anthony's Book's  example looks so sweet, but it is from a socket.

http://books.google.com/books?id=-6nzkugPG70C
<http://books.google.com/books?id=-6nzkugPG70C&pg=PT540&lpg=PT540&dq=freeswi
tch+event+turn+lights&source=bl&ots=4-5IfOJ3rF&sig=BsSDSKva7FKfzZ0w2HFeWJQ5_
ek&hl=en&sa=X&ei=2S7iT639OYTS6gG2v-0H&sqi=2&ved=0CFEQ6AEwAA#v=onepage&q=free
switch%20event%20turn%20lights&f=false>
&pg=PT540&lpg=PT540&dq=freeswitch+event+turn+lights&source=bl&ots=4-5IfOJ3rF
&sig=BsSDSKva7FKfzZ0w2HFeWJQ5_ek&hl=en&sa=X&ei=2S7iT639OYTS6gG2v-0H&sqi=2&ve
d=0CFEQ6AEwAA#v=onepage&q=freeswitch%20event%20turn%20lights&f=false


Sean

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
Foster
Sent: Wednesday, June 20, 2012 4:52 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Users need a red flashy light thingy

 

Or have your magic extension forward to an IVR, tells her the current
status, press one to enable, press 2 to disable, then confirm it instead of
two different numbers 

Brian Foster
Endigo Computer LLC

Sent from a mobile device.

On Jun 20, 2012 3:49 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

Freeswitch is so much better than my customers deserve.

 

I have the DialPlan set up nicely so they can dial *72 to forward all DID
calls to their answering service and *73 to turn it off.  Today she Dialed
*72 to turn it off 4 different times, and since that is the code to turn it
on --- it did NOT turn off.  She was livid and sent me hate mail about how
"terrible IP phones are and they never work ."

 

I even consoled her saying how *72 and *73 have no inherent indicators to
show which means on and off.   I have offered the obvious solutions of SPEED
DIAL buttons that are labeled "Enable" and "Disable".

 

Now my question.  Anyone have any ideas how to use a line button, or
possibly MWI to indicate that lines are forwarded (a db value).  MWI is
clearly in use so line button is my preference, but if MWI had a flash
function, etc????  I have unused line buttons - How can I make them show
busy (like shared call appearance) from the switch?  Maybe a dummy
user/extension I do something clever with?

 

If it matters the phones are Cisco 504G (4 line).

 

Thanks in advance,

Sean


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

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From gcd at i.ph  Thu Jun 21 02:00:35 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Thu, 21 Jun 2012 06:00:35 +0800
Subject: [Freeswitch-users] Different Voices for FS
In-Reply-To: <20120620224846.6ff371dc@anubis.defcon1>
References: <20120620132139.33e327b490679d2282e332758c73b55b.6e807b7d68.wbe@email14.secureserver.net>
	<20120620224846.6ff371dc@anubis.defcon1>
Message-ID: <CAHKqMb5hY4Ow-dzjeGnBXe9sx0H+mPi9iU8XTHBg0N3SSgCcNg@mail.gmail.com>

you can find the list of voice prompt phrases under freeswitch source code
directory:

docs/phrase/phrase_en.xml  - for English

phrase list for other languages are available, too.

-nandy

On Thu, Jun 21, 2012 at 4:48 AM, Bzzz <lazyvirus at gmx.com> wrote:

> On Wed, 20 Jun 2012 13:21:39 -0700
> <toddb at toddbailey.net> wrote:
>
> > My FS default installation provided me voice menus in English with
> > USA accent. Can this be changed to perhaps English with
> > Austrailian, British or So. Africa accents? What if I want
> > spanish, french  or  russian etc?
>
> This site provide different languages for reasonable rates:
> http://www.westany.com/freeswitch
>
> You can also try to use asterisk voices (but I don't know anything
> about the licence, though).
>
>
> And BTW, this isn't a bad question after all; can somebody give
> me the best methodology to record my own voices?
> Especially: is it better to record long texts and postprod cut/edit
> them, or is it better to only record sentences and words?
>
> And is there an URL to download pre-defined texts for that?
>
> JY
> --
> Help save the whales: eat a Japanese
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lazyvirus at gmx.com  Thu Jun 21 02:15:18 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Thu, 21 Jun 2012 00:15:18 +0200
Subject: [Freeswitch-users] Different Voices for FS
In-Reply-To: <CAHKqMb5hY4Ow-dzjeGnBXe9sx0H+mPi9iU8XTHBg0N3SSgCcNg@mail.gmail.com>
References: <20120620132139.33e327b490679d2282e332758c73b55b.6e807b7d68.wbe@email14.secureserver.net>
	<20120620224846.6ff371dc@anubis.defcon1>
	<CAHKqMb5hY4Ow-dzjeGnBXe9sx0H+mPi9iU8XTHBg0N3SSgCcNg@mail.gmail.com>
Message-ID: <20120621001518.35b551d6@anubis.defcon1>

On Thu, 21 Jun 2012 06:00:35 +0800
Nandy Dagondon <gcd at i.ph> wrote:

> you can find the list of voice prompt phrases under freeswitch
> source code directory:
> 
> docs/phrase/phrase_en.xml  - for English
> 
> phrase list for other languages are available, too.

Cool, thx.

JY
-- 
I read the newspaper avidly.  It is my one form of continuous
fiction. -- Aneurin Bevan


From gcd at i.ph  Thu Jun 21 02:16:45 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Thu, 21 Jun 2012 06:16:45 +0800
Subject: [Freeswitch-users] Users need a red flashy light thingy
In-Reply-To: <0b3401cd4f27$bc4c5390$34e4fab0$@bizfocused.com>
References: <0aa601cd4f1d$83117e20$89347a60$@bizfocused.com>
	<CAKV-jFyqVYdVQS6cmZqQS9LNSZnSuVBFVk8nHR_Xo1ny7jE+5Q@mail.gmail.com>
	<0b3401cd4f27$bc4c5390$34e4fab0$@bizfocused.com>
Message-ID: <CAHKqMb77L2+g=8n9=HpiKUuoD+_pgFnBcASGm+qbq7TDkS3ViA@mail.gmail.com>

how about playing a message first like "DID call forwarding enabled on this
line" to warn her whenever she makes a call?

-nandy

On Thu, Jun 21, 2012 at 5:00 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> HI Brian,****
>
> ** **
>
> This too is an excellent option for implementing the switch over and I
> will probably do it.****
>
> ** **
>
> However, nothing trumps a big red line button lit (red) when the phones
> are forwarded.****
>
> ** **
>
> Does anyone know how to send a PRESENCE event in an XML dialplan?  ****
>
> Anthony?s Book?s  example looks so sweet, but it is from a socket.****
>
>
> http://books.google.com/books?id=-6nzkugPG70C&pg=PT540&lpg=PT540&dq=freeswitch+event+turn+lights&source=bl&ots=4-5IfOJ3rF&sig=BsSDSKva7FKfzZ0w2HFeWJQ5_ek&hl=en&sa=X&ei=2S7iT639OYTS6gG2v-0H&sqi=2&ved=0CFEQ6AEwAA#v=onepage&q=freeswitch%20event%20turn%20lights&f=false
> ****
>
>
> Sean****
>
> ** **
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian Foster
> *Sent:* Wednesday, June 20, 2012 4:52 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Users need a red flashy light thingy****
>
> ** **
>
> Or have your magic extension forward to an IVR, tells her the current
> status, press one to enable, press 2 to disable, then confirm it instead of
> two different numbers ****
>
> Brian Foster
> Endigo Computer LLC****
>
> Sent from a mobile device.****
>
> On Jun 20, 2012 3:49 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:****
>
> Freeswitch is so much better than my customers deserve.****
>
>  ****
>
> I have the DialPlan set up nicely so they can dial *72 to forward all DID
> calls to their answering service and *73 to turn it off.  Today she Dialed
> *72 to turn it off 4 different times, and since that is the code to turn it
> on --- it did NOT turn off.  She was livid and sent me hate mail about how
> ?terrible IP phones are and they never work ??****
>
>  ****
>
> I even consoled her saying how *72 and *73 have no inherent indicators to
> show which means on and off.   I have offered the obvious solutions of
> SPEED DIAL buttons that are labeled ?Enable? and ?Disable?.****
>
>  ****
>
> Now my question.  Anyone have any ideas how to use a line button, or
> possibly MWI to indicate that lines are forwarded (a db value).  MWI is
> clearly in use so line button is my preference, but if MWI had a flash
> function, etc????  I have unused line buttons ? How can I make them show
> busy (like shared call appearance) from the switch?  Maybe a dummy
> user/extension I do something clever with?****
>
>  ****
>
> If it matters the phones are Cisco 504G (4 line).****
>
>  ****
>
> Thanks in advance,****
>
> Sean****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun 21 02:26:25 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Jun 2012 15:26:25 -0700
Subject: [Freeswitch-users] Users need a red flashy light thingy
In-Reply-To: <0b2f01cd4f27$4d2cc170$e7864450$@bizfocused.com>
References: <20120620132329.33e327b490679d2282e332758c73b55b.1e0469934d.wbe@email14.secureserver.net>
	<0b2f01cd4f27$4d2cc170$e7864450$@bizfocused.com>
Message-ID: <CAKzWOxVhHGVRnkTGSOn+_MYGyX5+b8UKqZAco2xChZUFFsDQWQ@mail.gmail.com>

On Wed, Jun 20, 2012 at 1:57 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> HI Todd,****
>
> ** **
>
> This too makes perfect sense, but NOT for this user.  The answering
> service does nothing but take voice messages, type them up and them back as
> FAXES to the office fax machine.  The users come in the morning and get
> their FAXES instead of voicemail. ****
>
> Can you imagine my frustration with these people?
>

Wow, are you serious? Do they live in Bedrock? I'll bet when they go to the
doctor and get an x-ray there's a little woodpecker that taps out the image
on a stone tablet. I'm glad they're your clients and not mine! :P

-MC
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From jpablolorenzetti at hotmail.com  Thu Jun 21 01:28:45 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Wed, 20 Jun 2012 15:28:45 -0600
Subject: [Freeswitch-users] Rtp traffic
Message-ID: <snt0-eas4143A1B54742009FA785A4DD9FE0@phx.gbl>

Hi, i m trying to connect (establish sip trunks) to huawei MSC (msoft), so i m requesting the information to configure a gateway on the FS side for outbound and inbound, but they are requesting that the RTP traffic be sent to one IP (to the UMG) and the SIP traffic to a differentIP.I have been reading documentation but i have not found how to accomplish this. can someone please give me a hint ? is this possible ? anyone with experience integrating to a huawei msc?

From msc at freeswitch.org  Thu Jun 21 02:31:47 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Jun 2012 15:31:47 -0700
Subject: [Freeswitch-users] Different Voices for FS
In-Reply-To: <20120620132139.33e327b490679d2282e332758c73b55b.6e807b7d68.wbe@email14.secureserver.net>
References: <20120620132139.33e327b490679d2282e332758c73b55b.6e807b7d68.wbe@email14.secureserver.net>
Message-ID: <CAKzWOxU-wTsa1JkCa_OZaAF6fLJmJ_Dqa+e8Mq5RAJddy-oj7g@mail.gmail.com>

Do you have a language set? If so most of the work is done in
conf/lang/en/en.xml where you set the base path for the language. You just
need to make sure that your other language set has the same structure as
the callie sounds we have. Also, if you have an alternate set of English
prompts we'd love to make those globally available, assuming they aren't
under a draconian license...

-MC

On Wed, Jun 20, 2012 at 1:21 PM, <toddb at toddbailey.net> wrote:

> Hey All,
>
> My FS default installation provided me voice menus in English with USA
> accent.
> Can this be changed to perhaps English with Austrailian, British or So.
> Africa accents?
> What if I want spanish, french  or  russian etc?
>
> This is my 12th posting does any one ever respond to questions posted here?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun 21 02:34:45 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Jun 2012 15:34:45 -0700
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <CAKV-jFwyH7ybQycVu_DR+K2h_dhijEXiJ+wzrky_terdVpe4Lg@mail.gmail.com>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
	<4FDFBCA8.1050700@quentustech.com>
	<CAKzWOxWR6PJqPb6NqTLRQa5WMGP6H4zVte1w0i2UZSDN9+fQMg@mail.gmail.com>
	<COL124-DS1734BA1C8437EE24E71567DDFE0@phx.gbl>
	<74D29BFA17294F97A0A8C40A49538E56@freeswitch.org>
	<CAKV-jFwyH7ybQycVu_DR+K2h_dhijEXiJ+wzrky_terdVpe4Lg@mail.gmail.com>
Message-ID: <CAKzWOxUE=wZrdfTXa=szTvwQ9MW2kBpPpyaLe4TrED5KY2N7Gw@mail.gmail.com>

Perhaps. I'll give Shadowym the opportunity to respond to JMesquita's post.
Again, distilling out the emotion it comes down to this: if you want a say
in how things work then get involved. If the OP is willing to do that then
great. Let's see how it shakes out.

-MC

On Wed, Jun 20, 2012 at 1:45 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> As I don't think this is going to turn into a productive discussion, the
> thread should probably be closed.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 20, 2012 1:44 PM, "Jo?o Mesquita" <jmesquita at freeswitch.org> wrote:
>
>> Dude, I have been trying not to follow this thread but I can't avoid it
>> because it keeps showing up on my mail client over and over again.
>>
>> Stop bitching and DO SOMETHING. This is open source, not something you've
>> bought or that someone is trying to sell you. Your demanding tone is
>> irritating to say the very least. You don't like that someone changed a few
>> dirs and did not make the work like you expected, then just freaking FIX IT
>> and contribute the solution. None of the developers have time to write
>> documentation and they struggle to keep their day jobs while making it work
>> for the rest of the world as well and you sit on your comfy place
>> complaining that they are not doing enough? WTF?
>>
>> SwK is 1 person that has been doing the stable branch as well as trying
>> to package freeswitch to several distros, he could definitely use some help
>> instead of aggressiveness and complaints. Contribute and hop in or live
>> with what you have or just leave quiet but DON'T piss on everyones else's
>> work. You want to be disappointed, that's OK, but you gotta earn it first.
>> What have you done for the project so far?
>>
>> Sorry to the list for my email but everything has its limits and I tend
>> to be nice most of the times.
>>
>>
>> --
>> Jo?o Mesquita
>> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>>
>> On Wednesday, June 20, 2012 at 11:57 AM, shadowym wrote:
>>
>> Did the announcement also say they were getting rid of the conf directory
>> and not bothering to updated directory settings in the default conf files
>> (that are now in the root directory).  Do they plan to update all the
>> freeswitch documentation now as well which talks about the conf directory
>> all over the place?****
>>
>> ** **
>>
>> How long ago was this announcement made?  The issue is not about being
>> caught off guard btw.  It?s about why and how it?s being done.****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [
>> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
>> *On Behalf Of *Michael Collins
>> *Sent:* Monday, June 18, 2012 5:43 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] freeswitch directories moved****
>>
>> ** **
>>
>> ** **
>>
>> On Mon, Jun 18, 2012 at 4:41 PM, William King <
>> william.king at quentustech.com> wrote:****
>>
>> I believe there was an announcement on this mailing list alerting that
>> the new rpms would be undergoing major changes to include moving the
>> directory locations. Search this list for new threads started by SwK in the
>> past 6 months.****
>>
>> ** **
>>
>> There was indeed an announcement by SwK himself. It was as close to
>> FNLoIP (Flashing Neon Lights over IP) as you can get.
>> -MC****
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From darius at dons.net.au  Thu Jun 21 03:27:39 2012
From: darius at dons.net.au (Daniel O'Connor)
Date: Thu, 21 Jun 2012 08:57:39 +0930
Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
In-Reply-To: <F998E99A-AD61-4F7E-822E-F6A237752940@digital-outpost.com>
References: <29D49331-2F70-411D-BE42-86CAB8CC181E@digital-outpost.com>
	<0AEB7B3A-5591-4694-83B5-E4208D5DA4A0@dons.net.au>
	<46D48718-2661-41EC-8A59-67BFEBCCF745@digital-outpost.com>
	<31354FBA-5DBC-47A9-9958-9EC8A515DACD@dons.net.au>
	<3D137477-E72E-4A59-A625-899DFB15D387@digital-outpost.com>
	<4A48F47D-7954-496D-A31E-CE5E1AD21E9D@dons.net.au>
	<F998E99A-AD61-4F7E-822E-F6A237752940@digital-outpost.com>
Message-ID: <FCC7A462-A441-4A7B-B521-35421A4EB8F5@dons.net.au>


On 21/06/2012, at 4:23, Terry Barnum wrote:
>> I think to work around this I just removed spandsp while I built freeswitch (bleh).
> 
> THanks Daniel. Awhile back I was fooling around with the asterisk macport on this work iMac and spandsp-devel is a dependency. I uninstalled the asterisk port long ago but spandsp-devel was still around.

OK.

> So it's all in one place, to get freeswitch to compile on this intel iMac running 10.6.8 with macports installed, I uninstalled the spandsp-devel macport, did a fresh git clone, ran your patch <http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-OSX-Libtool-Macports-td7579970.html#a7580004> to force the use of C in gsm0610_rpe.c, ran the jira patch to correctly select x86 instead of ppc <http://jira.freeswitch.org/browse/FS-3992>, ran bootstrap.sh, used your env CPPFLAGS=-I/opt/local/include LDFLAGS=-L/opt/local/lib ./configure and make.
> 
> Now on to actually learning to use freeswitch!

Yes, the real work begins now ;)

>> IMO the "right" solution is to rename spandsp in freeswitch - it has been modified (looking at the git log) so it is really a fork of the original. That is unless those patches get merged back into spandsp and then macports can get an updated one.
>> 
>> The later solution is still pretty bad though unless there is a way to force freeswitch to not look at the bundled version.
>> 
>> There is a similar problem with portaudio although since it has a stable API it doesn't show up. It doesn't however build on OSX (Intel 10.7 anyway) and in the end I hacked src/mod/endpoint/mod_portaudio to use pkg-config to find the right CFLAGS and LIBS to link to it.
> 
> Forgive my ignorance, does the freeswitch includes of things like spandsp and portaudio have to be set globally or can there be separate configs?


I believe it's hard coded into Makefiles and configure scripts :(

--
Daniel O'Connor software and network engineer
for Genesis Software - http://www.gsoft.com.au
"The nice thing about standards is that there
are so many of them to choose from."
  -- Andrew Tanenbaum
GPG Fingerprint - 5596 B766 97C0 0E94 4347 295E E593 DC20 7B3F CE8C








From chris at opencsta.org  Thu Jun 21 03:42:33 2012
From: chris at opencsta.org (Chris Mylonas)
Date: Thu, 21 Jun 2012 09:42:33 +1000
Subject: [Freeswitch-users] My First FreeSwitch OR Is this easy to do?
In-Reply-To: <CAMv5evmwHJfepMS04EDUT77E9Gf-onYZhnkt2YwzSqUTT7aeaQ@mail.gmail.com>
References: <CAMv5evnVG10fKoZmyZVVjh1nqkTCscmGWFYG9d0F5U7W1X40HQ@mail.gmail.com>
	<CAKzWOxUSHUF3SfvoO9cj-qHQMzVHa=YxYHYzByxudTi-QA1VfA@mail.gmail.com>
	<CAMv5evmwHJfepMS04EDUT77E9Gf-onYZhnkt2YwzSqUTT7aeaQ@mail.gmail.com>
Message-ID: <70861CE9-98CC-4082-8E67-8F4768B67FE9@opencsta.org>

> 
> I am technically inclined... to the point where I debated setting up a
> Gentoo box for this because that's what I'm used to administering but
> finally decided that the risk of grumbles from the household if it
> doesn't "just work" for them necessitated using something a little
> more "off the shelf".


hehe .... gentoo rocks!! but for voip systems i run centos because it's just got less update headaches.
there are a bunch of hacks on the net for installing 3rd party scripts or web apps and they are more often than not running on centos as well...so following docs is easy.

you'll really love voip  /s

if you are irc inclined, the fs irc channel is a good place too

:)
chris

From avi at avimarcus.net  Thu Jun 21 08:25:23 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 21 Jun 2012 07:25:23 +0300
Subject: [Freeswitch-users] Debian Packages OR Salt Configuration Files
Message-ID: <CALNZuwnMe7RDaU4YUdH4=wiZR+YuM5ggiTarmObbvkoqhJnvSg@mail.gmail.com>

Hi - I know the centos yum packages are being worked on... is there debian
packages somewhere?

Also, is anyone using salt (http://saltstack.org/) to admin their servers
and/or freeswitch? It's a remote execution tool for a cluster +
configuration version control. I've started using it and it seems pretty
awesome!
If someone has config files, freeswitch modules etc that they use with it
that they'd be willing to share that would be great!

-Avi
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From mayamatakeshi at gmail.com  Thu Jun 21 09:06:56 2012
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 21 Jun 2012 14:06:56 +0900
Subject: [Freeswitch-users] 302 redirect problem
In-Reply-To: <4FE1D47C.3020700@softnet.si>
References: <4FE1D47C.3020700@softnet.si>
Message-ID: <CABaNFCbd_FWM0Dd7wtXPV0TnSaW0iLbu6CjMt+6iPiA2TSCdgw@mail.gmail.com>

On Wed, Jun 20, 2012 at 10:47 PM, Miha <miha at softnet.si> wrote:

>  Hi,
>
> In sip_profiles I have set to manual redirect.
> When call goes in public dialplan and than phone redirect call, FS
> automatically send redirect call to contex redirect.
> Is this normal? How to set that call will go to internal dialplan?
>

I think you missed the explanation in the wiki:

http://wiki.freeswitch.org/wiki/Dialplan_Handling_Incoming_Redirect

*Input variables:

sip_redirect_context

Which context to transfer the call to when handling redirects. Defaults to:
redirected *

Regards,
Takeshi
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From gabe at gundy.org  Thu Jun 21 10:55:30 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 21 Jun 2012 00:55:30 -0600
Subject: [Freeswitch-users] Debian Packages OR Salt Configuration Files
In-Reply-To: <CALNZuwnMe7RDaU4YUdH4=wiZR+YuM5ggiTarmObbvkoqhJnvSg@mail.gmail.com>
References: <CALNZuwnMe7RDaU4YUdH4=wiZR+YuM5ggiTarmObbvkoqhJnvSg@mail.gmail.com>
Message-ID: <CAMwZ-tJ08XQ-kJsYZuuAAb4gb-CzVyf=dFNJUHXnsVbHLRw9Gg@mail.gmail.com>

On Wed, Jun 20, 2012 at 10:25 PM, Avi Marcus <avi at avimarcus.net> wrote:
> Also, is anyone using salt (http://saltstack.org/) to admin their servers
> and/or freeswitch? It's a remote execution tool for a cluster +
> configuration version control. I've started using it and it seems pretty
> awesome!

Salt is the bomb. We're in the process of moving a bunch of
home-brewed Python, Fabric, Django (templating) and json configs over
to Salt. It's much better, faster, secure and sane than what we came
up with.

Anyway, the project we're working on has very specific needs, but
we'll be doing more generic FreeSWITCH stuff that we'll open source.

Good luck and let us know if you create any cool tools.

Best,
Gabe


From miha at softnet.si  Thu Jun 21 11:08:55 2012
From: miha at softnet.si (Miha)
Date: Thu, 21 Jun 2012 09:08:55 +0200
Subject: [Freeswitch-users] 302 redirect problem
In-Reply-To: <CABaNFCbd_FWM0Dd7wtXPV0TnSaW0iLbu6CjMt+6iPiA2TSCdgw@mail.gmail.com>
References: <4FE1D47C.3020700@softnet.si>
	<CABaNFCbd_FWM0Dd7wtXPV0TnSaW0iLbu6CjMt+6iPiA2TSCdgw@mail.gmail.com>
Message-ID: <4FE2C887.1070401@softnet.si>

Thanks Takeshi!

What could be the reason that FS always would like to redirect to contex 
redirected?
In my public dialplan I have set variable:
<action application="set" data="sip_redirect_context=default" />

Thanks!

br,
miha

On 6/21/2012 7:06 AM, mayamatakeshi wrote:
>
>
> On Wed, Jun 20, 2012 at 10:47 PM, Miha <miha at softnet.si 
> <mailto:miha at softnet.si>> wrote:
>
>     Hi,
>
>     In sip_profiles I have set to manual redirect.
>     When call goes in public dialplan and than phone redirect call, FS
>     automatically send redirect call to contex redirect.
>     Is this normal? How to set that call will go to internal dialplan?
>
>
> I think you missed the explanation in the wiki:
>
> http://wiki.freeswitch.org/wiki/Dialplan_Handling_Incoming_Redirect
>
> /Input variables:
>
> sip_redirect_context
>
> Which context to transfer the call to when handling redirects. 
> Defaults to: redirected /
>
> Regards,
> Takeshi
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ocset at the800group.com  Thu Jun 21 11:09:35 2012
From: ocset at the800group.com (ocset)
Date: Thu, 21 Jun 2012 15:09:35 +0800
Subject: [Freeswitch-users] Codec 101 - need help with the basics
Message-ID: <4FE2C8AF.5080805@the800group.com>

Hi

I need some assistance in understanding the codec minefield.

When a stream of audio is sent to FreeSWITCH (FS) using a particular 
codec, does FS just pass it on or is there some level of translation 
occurring?

I guess my question relates to a situation where an FXO gateway 
(GXW410x) uses the G723 codec and the receiing phone does not have a 
matching codec (the phone has G.722, G.711 a law, G.711 u law, G.726 and 
G.729 as options). How is this managed by the phone, FreeSWITCH and the 
FXO device? Should I be setting something in FS to handle this issue?

I have 2 cordless Siemens C610 IP phones an I am having sound quality 
issues (incoming sound is soft) and I was wondering how big a part the 
codec plays in this equation. I have desk phones that are from the same 
manufacturer as the FXO gateway (Grandstream) and they have very good 
sound quality.

Calling internally between all the phones has crystal clear sound.

Thanks in advance
O


From alex at thewinelake.com  Thu Jun 21 12:12:49 2012
From: alex at thewinelake.com (Alex)
Date: Thu, 21 Jun 2012 09:12:49 +0100
Subject: [Freeswitch-users] Interrupting "speak"
In-Reply-To: <CALNZuw=JSV13Dz_Au3sKj4Wvf=bLMQ4VGRvdfKLwLEd-C977yA@mail.gmail.com>
References: <4FE08091.7090005@thewinelake.com>
	<CALNZuw=JSV13Dz_Au3sKj4Wvf=bLMQ4VGRvdfKLwLEd-C977yA@mail.gmail.com>
Message-ID: <4FE2D781.5050604@thewinelake.com>

Having got to grips with phrases, I can now see that there is a cheating 
way to do this, which maybe should be put in the wiki?

Essentially, the idea is to have file 
<freeswitch>/conf/lang/en/simple/simplesay.xml:

<include>
   <macro name="say">
     <input pattern="(.*)">
       <match>
         <action function="execute" data="sleep(1000)"/>
         <action function="speak-text" data="$1"/>
       </match>
     </input>
   </macro>


Modify <freeswitch>/conf/lang/en/en.xml to include:

<X-PRE-PROCESS cmd="include" data="simple/simplesay.xml"/>

If you don't use cepstral, edit the first line to be:

<language name="en" say-module="en" 
sound-prefix="$${sounds_dir}/en/us/callie" tts-engine="flite" 
tts-voice="kal">


Then in a lua script, you can do something like this:

destnum = session:playAndGetDigits(3, 20, 3, 5000, "#*", 
"phrase:sa:"..sayWhatFirst, "", "\\d+")




From paul at cupis.co.uk  Thu Jun 21 12:17:59 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Thu, 21 Jun 2012 09:17:59 +0100
Subject: [Freeswitch-users] Rtp traffic
In-Reply-To: <snt0-eas4143A1B54742009FA785A4DD9FE0@phx.gbl>
References: <snt0-eas4143A1B54742009FA785A4DD9FE0@phx.gbl>
Message-ID: <20120621081759.GA20584@eagle.cupis.co.uk>

On Wed, Jun 20, 2012 at 03:28:45PM -0600, Juan Pablo L. wrote:
> Hi, i m trying to connect (establish sip trunks) to huawei MSC (msoft),
> so i m requesting the information to configure a gateway on the FS side
> for outbound and inbound, but they are requesting that the RTP traffic
> be sent to one IP (to the UMG) and the SIP traffic to a differentIP.
> I have been reading documentation but i have not found how to
> accomplish this. can someone please give me a hint ? is this possible ?
> anyone with experience integrating to a huawei msc?

This shouldn't be a problem - you just setup the SIP side of it and the RTP
should be automatic (the SIP signalling will specify the RTP address/port).

Obviously if you are firewalling RTP you will need to allow the traffic
through your firewall, but you don't need any special configuration on
the FreeSWITCH side to do this.

Regards,



From michal at noblesolutions.co.uk  Thu Jun 21 12:27:50 2012
From: michal at noblesolutions.co.uk (michal)
Date: Thu, 21 Jun 2012 09:27:50 +0100
Subject: [Freeswitch-users] implementation of freeswitch in javascript / node
Message-ID: <1953A95D11D1ED42B24E60E7A59C5AB411B451@venus.Noble.local>

Hi there!

 

Can someone recommend some resources on freeswitch and javascript/node?
Ideally I would like a tutorial or guide on how it all works together
(how to talk to FS)

 

 

 

Thanks in advance! 

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From avi at avimarcus.net  Thu Jun 21 12:35:35 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 21 Jun 2012 11:35:35 +0300
Subject: [Freeswitch-users] Interrupting "speak"
In-Reply-To: <4FE2D781.5050604@thewinelake.com>
References: <4FE08091.7090005@thewinelake.com>
	<CALNZuw=JSV13Dz_Au3sKj4Wvf=bLMQ4VGRvdfKLwLEd-C977yA@mail.gmail.com>
	<4FE2D781.5050604@thewinelake.com>
Message-ID: <CALNZuwmt433CS5Qvg17dwGxKsYVrWZCyybTokNLfKzv+0SZL_g@mail.gmail.com>

You should probably be using this instead of sleep:
<action function="play-file" data="silence_stream://1000"/>

This may be FUD, but I recall that sleep could have unintended side-effects
because it doesn't provide an RTP stream... while this is playing silence.

-Avi

On Thu, Jun 21, 2012 at 11:12 AM, Alex <alex at thewinelake.com> wrote:

> Having got to grips with phrases, I can now see that there is a cheating
> way to do this, which maybe should be put in the wiki?
>
> Essentially, the idea is to have file
> <freeswitch>/conf/lang/en/simple/simplesay.xml:
>
> <include>
>   <macro name="say">
>     <input pattern="(.*)">
>       <match>
>         <action function="execute" data="sleep(1000)"/>
>         <action function="speak-text" data="$1"/>
>       </match>
>     </input>
>   </macro>
>
>
> Modify <freeswitch>/conf/lang/en/en.xml to include:
>
> <X-PRE-PROCESS cmd="include" data="simple/simplesay.xml"/>
>
> If you don't use cepstral, edit the first line to be:
>
> <language name="en" say-module="en"
> sound-prefix="$${sounds_dir}/en/us/callie" tts-engine="flite"
> tts-voice="kal">
>
>
> Then in a lua script, you can do something like this:
>
> destnum = session:playAndGetDigits(3, 20, 3, 5000, "#*",
> "phrase:sa:"..sayWhatFirst, "", "\\d+")
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From avi at avimarcus.net  Thu Jun 21 13:43:08 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 21 Jun 2012 12:43:08 +0300
Subject: [Freeswitch-users] implementation of freeswitch in javascript /
	node
In-Reply-To: <1953A95D11D1ED42B24E60E7A59C5AB411B451@venus.Noble.local>
References: <1953A95D11D1ED42B24E60E7A59C5AB411B451@venus.Noble.local>
Message-ID: <CALNZuwmHbJFe4u_5fCB4CowHynHqVHXqB=5OsZzw3SVwTktn7g@mail.gmail.com>

Me too. I've not seen many examples for node using ESL.. samples (and
explanation?!) of a some code doing some useful things would be great!

-Avi


On Thu, Jun 21, 2012 at 11:27 AM, michal <michal at noblesolutions.co.uk>wrote:

>  Hi there!****
>
> ** **
>
> Can someone recommend some resources on freeswitch and javascript/node?
> Ideally I would like a tutorial or guide on how it all works together (how
> to talk to FS)****
>
> ** **
>
> ** **
>
> ** **
>
> Thanks in advance! ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From alex at thewinelake.com  Thu Jun 21 14:10:13 2012
From: alex at thewinelake.com (Alex)
Date: Thu, 21 Jun 2012 11:10:13 +0100
Subject: [Freeswitch-users] Speccing server for enterprise FS
Message-ID: <4FE2F305.4090808@thewinelake.com>

Looking to setup a new Freeswitch server that will be doing quite a lot 
of traffic.

Trying to get a rough idea of where to spend the money (RAM? CPU? 
Network access?).

There will be perhaps 10,000 simultaneous SIP registrations
Maybe 500 simultaneous calls, some of (say 50%) which will require a-law 
to GSM transcoding.

I know it's a tricky one to answer, but based on experiences here, which 
bits tend to glow first?


From michal at noblesolutions.co.uk  Thu Jun 21 14:28:18 2012
From: michal at noblesolutions.co.uk (michal)
Date: Thu, 21 Jun 2012 11:28:18 +0100
Subject: [Freeswitch-users] implementation of freeswitch in javascript
	/node
References: <1953A95D11D1ED42B24E60E7A59C5AB411B451@venus.Noble.local>
	<CALNZuwmHbJFe4u_5fCB4CowHynHqVHXqB=5OsZzw3SVwTktn7g@mail.gmail.com>
Message-ID: <1953A95D11D1ED42B24E60E7A59C5AB411B468@venus.Noble.local>

Do you have some links to what you have seen?

 

________________________________

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: 21 June 2012 10:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] implementation of freeswitch in
javascript /node

 

Me too. I've not seen many examples for node using ESL.. samples (and
explanation?!) of a some code doing some useful things would be great!




-Avi

 

On Thu, Jun 21, 2012 at 11:27 AM, michal <michal at noblesolutions.co.uk>
wrote:

Hi there!

 

Can someone recommend some resources on freeswitch and javascript/node?
Ideally I would like a tutorial or guide on how it all works together
(how to talk to FS)

 

 

 

Thanks in advance! 


________________________________________________________________________
_
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jos at firstcom.dk  Thu Jun 21 14:54:40 2012
From: jos at firstcom.dk (=?iso-8859-1?Q?Jon_Sch=F8pzinsky?=)
Date: Thu, 21 Jun 2012 12:54:40 +0200
Subject: [Freeswitch-users] Using mod_managed or mod_java
Message-ID: <111509FCE9EBF74BAB2CB38AAC521D9B1337FC0FD8@helle.dk.hq.firstcom.dk>

Hello everybody,

We are designing a new platform, and are planning to use FreeSWITCH for our core services and switching.

We have looked at using the ESL interface, mod_managed og mod_java.

It doesn't seem as though a lot of work is being done on mod_java, and there isn't allot of documentation around. So if anybody could share their experience with using mod_java, that would be great.

mod_managed seems very promising, but we are a bit nervous about running Mono on one of our primary production systems, as our experience with Mono hasn't been all good.
Is anybody running mod_managed on large scale production systems, that can share their experience with doing so?

The ESL interface is working pretty great, but we do like the idea of integrating our core logic directly into FreeSWITCH, instead of having it run in a separate process.



Kind Regards

Jon Leren Sch?pzinsky

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From megahohol at gmail.com  Thu Jun 21 16:04:15 2012
From: megahohol at gmail.com (Grygoriy Dobrovolskyy)
Date: Thu, 21 Jun 2012 14:04:15 +0200
Subject: [Freeswitch-users] Sofia recover and wrong Sip fields when sending
	re-invite
Message-ID: <CAAfngiEZbwGr8TLNG3-RYquLh__46M-8k4VK1dfh5LuFpmnT4w@mail.gmail.com>

Hello list
I was able to setup HA cluster with two FS boxes, and if my primary crashed
sofia recover is triggered on the second box. However i have noticed that
second FS box sending the Re-Invite incorrectly.

My FS boxes setup is extremely simple, when i receive a call i forward it
to the Opensips proxy, so i use bridge like this: (public dialplan)

   <condition field="${sip_to_user}" expression="(123456789).*$"
break="never">
      <action inline="true" application="eval" data="URI=${sip_to_user}"/>
      <action application="set" data="domain_name=$${domain}"/>
      <action application="bridge"
data="{absolute_codec_string=PCMA}sofia/internal/987654333 at 17.2.128.75"/>
    </condition>


So what FS does, its sends the INVITE to the opensips like this: (stripped)

INVITE sip:987654333 at 17.2.128.75 SIP/2.0
Via: SIP/2.0/UDP 17.2.128.147;rport;branch=z9hG4bK5N320Bvar00ZN
Max-Forwards: 68
From: "Grigoriy Dobrovolskyy" <sip:33811740 at 17.2.128.147>;tag=3jNgatZH4D92Q
To: <sip:987654333 at 17.2.128.75>
------stripped-----------


The call is answerred, and all is going fine, i crash this box and trigger
box2 to reinvite the call,  FS2 gets the same ip and send Re-Invite like
this:


INVITE sip:17.2.128.75:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 17.2.128.75;branch=z9hG4bK62a7.6041d237.0;rport
Max-Forwards: 68
From: <sip:987654333 at 17.2.128.75>;tag=B2B.511.202
To: Grigoriy Dobrovolskyy <sip:33811740 at 17.2.128.147>;tag=3jNgatZH4D92Q


Notice the FROM and To field, we are sending the invite to a proxy, from
this proxy, this has no sence, it should be:

From: Grigoriy Dobrovolskyy <sip:33811740 at 17.2.128.147>;tag=3jNgatZH4D92Q
To: <sip:987654333 at 17.2.128.75>;tag=B2B.511.202

Is there any way i can fix this with some kind of param ?

Greg
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From bdfoster at endigotech.com  Thu Jun 21 16:14:15 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 21 Jun 2012 08:14:15 -0400
Subject: [Freeswitch-users] Speccing server for enterprise FS
In-Reply-To: <4FE2F305.4090808@thewinelake.com>
References: <4FE2F305.4090808@thewinelake.com>
Message-ID: <CAKV-jFzwRfZ1BPS-2Dv7Oxt0gsW6ufuD2xY9S5HmhQSs83BbJg@mail.gmail.com>

Hey,
Although I can't share specs for a machine doing that kind of traffic, I
would like to suggest that you should be doing HA and load balancing
between atleast two servers when dealing with that kind of traffic. Its
hard for me to justify putting all that traffic on one server. Just food
for thought. Depending on how many calls you want to risk losing at any
given time, your machine specs might be much lower than expected.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 21, 2012 6:10 AM, "Alex" <alex at thewinelake.com> wrote:

> Looking to setup a new Freeswitch server that will be doing quite a lot
> of traffic.
>
> Trying to get a rough idea of where to spend the money (RAM? CPU?
> Network access?).
>
> There will be perhaps 10,000 simultaneous SIP registrations
> Maybe 500 simultaneous calls, some of (say 50%) which will require a-law
> to GSM transcoding.
>
> I know it's a tricky one to answer, but based on experiences here, which
> bits tend to glow first?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dgarcia at anew.com.ve  Thu Jun 21 16:29:22 2012
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Thu, 21 Jun 2012 07:59:22 -0430
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <COL124-DS10B977B37DDE7083722AB3DDFE0@phx.gbl>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>	<4FDFBCA8.1050700@quentustech.com>	<CAKzWOxWR6PJqPb6NqTLRQa5WMGP6H4zVte1w0i2UZSDN9+fQMg@mail.gmail.com>	<4FE072DA.3040903@anew.com.ve>
	<CALNZuwnHU7+vbVre0kmrk+P_BU40AorL+4_sDA6-d6f4rLkoew@mail.gmail.com>
	<COL124-DS10B977B37DDE7083722AB3DDFE0@phx.gbl>
Message-ID: <4FE313A2.5050109@anew.com.ve>

Hi guys

Shadowym, I think you are overreacting about this. I started with FS 
using GIT, when rpm package where announced I decided to wait until the 
were more mature. You can not expect you will get a new feature or etc 
and it will work as you imagine at the first time.

You found some "issues" with rpm packages then your resolved by your 
self, so, why you make a more positive work and wikify it? Your complain 
is not helping this wonderful project.

I agree with AVI, if the rpm developers could include some sort of 
change log ("big/important changes") will be helpful for all. At the 
end, you are the only responsible for what you do, so, if you decided to 
try rpm packages and find it some differences respect GIT then don't  
complain just inform about it in the best way possible what is wrong, so 
SwK could make the respective corrections, if they apply.



On 6/20/2012 10:32 AM, shadowym wrote:
>
> Which brings up another good point.  If such a drastic (imho) change 
> is being made there should be some consideration for backwards 
> compatibility and conformance to the documentation.  How hard would it 
> have been to add symlinks?
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Avi Marcus
> *Sent:* Tuesday, June 19, 2012 5:50 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] freeswitch directories moved
>
> A changelog can often be ignored. A "breaking changes" or "not 
> backward compatible changes" log would be helpful for such things.
>
> Huh, if those changes get tagged in GIT I think they automatically 
> show up if you're paying attention when doing a pull.
>
> -Avi
>
> On Tue, Jun 19, 2012 at 3:38 PM, Saugort Dario Garcia Tovar 
> <dgarcia at anew.com.ve <mailto:dgarcia at anew.com.ve>> wrote:
>
> Hi Everybody,
>
> Just to be fare, is good to make FS more improved and standardised.It 
> is better if in the wiki the devolper or someone involved with rpm 
> update the wiki with a "change log", in this way everyone will know 
> what is different. This will helpful more than you can expect for FS 
> community.
>
> Shadowym, if you have decided to use FS RPM, you should get more info 
> about it and what to expect.
>
> It is just my opinion
>
>
>
> On 6/18/2012 8:13 PM, Michael Collins wrote:
>
>     On Mon, Jun 18, 2012 at 4:41 PM, William King
>     <william.king at quentustech.com
>     <mailto:william.king at quentustech.com>> wrote:
>
>     I believe there was an announcement on this mailing list alerting
>     that the new rpms would be undergoing major changes to include
>     moving the directory locations. Search this list for new threads
>     started by SwK in the past 6 months.
>
>     There was indeed an announcement by SwK himself. It was as close
>     to FNLoIP (Flashing Neon Lights over IP) as you can get.
>     -MC
>
>     _________________________________________________________________________
>
>     Professional FreeSWITCH Consulting Services:
>
>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>
>     http://www.freeswitchsolutions.com
>
>       
>
>     
>
>     
>
>       
>
>     Official FreeSWITCH Sites
>
>     http://www.freeswitch.org
>
>     http://wiki.freeswitch.org
>
>     http://www.cluecon.com
>
>       
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>       
>
>     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
>
>     No virus found in this message.
>     Checked by AVG - www.avg.com <http://www.avg.com>
>     Version: 2012.0.2178 / Virus Database: 2437/5079 - Release Date:
>     06/19/12
>
> -- 
> Atentamente,
> *Dario Garc?a*
> Consultor.
>
> CCCT, Nivel C2, Sector Yarey, Mz,
> Ofc. MZ03a.
> Caracas-Venezuela.
> Tel?fono: +58 212 9081842 <tel:%2B58%20212%209081842>
> Cel: +58 412 2221515 <tel:%2B58%20412%202221515>
> dgarcia at anew.com.ve <mailto:dgarcia at anew.com.ve>
> http://www.anew.com.ve
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2178 / Virus Database: 2437/5081 - Release Date: 06/20/12
>


-- 
Atentamente,
*Dario Garc?a*
Consultor.

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Tel?fono: +58 212 9081842
Cel: +58 412 2221515
dgarcia at anew.com.ve
http://www.anew.com.ve
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From jpablolorenzetti at hotmail.com  Thu Jun 21 16:40:35 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Thu, 21 Jun 2012 06:40:35 -0600
Subject: [Freeswitch-users] Rtp traffic
Message-ID: <DUB0-P1-EAS51AB57C73DEC6ECA79BBC5D9FD0@phx.gbl>

Oh thank you very much, i thought i had to configure it somewhere, i will try. Thanks a lot.

-----Original Message-----

From: Paul Cupis
Sent: 21 Jun 2012 08:18:25 GMT
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Rtp traffic

On Wed, Jun 20, 2012 at 03:28:45PM -0600, Juan Pablo L. wrote:
> Hi, i m trying to connect (establish sip trunks) to huawei MSC (msoft),
> so i m requesting the information to configure a gateway on the FS side
> for outbound and inbound, but they are requesting that the RTP traffic
> be sent to one IP (to the UMG) and the SIP traffic to a differentIP.
> I have been reading documentation but i have not found how to
> accomplish this. can someone please give me a hint ? is this possible ?
> anyone with experience integrating to a huawei msc?

This shouldn't be a problem - you just setup the SIP side of it and the RTP
should be automatic (the SIP signalling will specify the RTP address/port).

Obviously if you are firewalling RTP you will need to allow the traffic
through your firewall, but you don't need any special configuration on
the FreeSWITCH side to do this.

Regards,


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


From jpablolorenzetti at hotmail.com  Thu Jun 21 17:15:42 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Thu, 21 Jun 2012 07:15:42 -0600
Subject: [Freeswitch-users] Rtp traffic
Message-ID: <col0-p5-eas3113E53F4224FFD4C75122D9FD0@phx.gbl>

Just a little clarification, i had a typo and
I meant csoft
Instead of msoft.

-----Original Message-----

From: Juan Pablo L.
Sent: 21 Jun 2012 12:42:10 GMT
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Rtp traffic

Oh thank you very much, i thought i had to configure it somewhere, i will try. Thanks a lot.

-----Original Message-----

From: Paul Cupis
Sent: 21 Jun 2012 08:18:25 GMT
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Rtp traffic

On Wed, Jun 20, 2012 at 03:28:45PM -0600, Juan Pablo L. wrote:
> Hi, i m trying to connect (establish sip trunks) to huawei MSC (msoft),
> so i m requesting the information to configure a gateway on the FS side
> for outbound and inbound, but they are requesting that the RTP traffic
> be sent to one IP (to the UMG) and the SIP traffic to a differentIP.
> I have been reading documentation but i have not found how to
> accomplish this. can someone please give me a hint ? is this possible ?
> anyone with experience integrating to a huawei msc?

This shouldn't be a problem - you just setup the SIP side of it and the RTP
should be automatic (the SIP signalling will specify the RTP address/port).

Obviously if you are firewalling RTP you will need to allow the traffic
through your firewall, but you don't need any special configuration on
the FreeSWITCH side to do this.

Regards,


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


From mi.ke at null.net  Thu Jun 21 17:17:03 2012
From: mi.ke at null.net (Mi Ke)
Date: Thu, 21 Jun 2012 09:17:03 -0400
Subject: [Freeswitch-users] gateway registration using TLS
Message-ID: <20120621131704.182980@gmx.com>

Hi Sean,

 It seems that setting realm port to 5061 does not set registration mode to TLS - this is how remote server sees our reg attempt:

 REGISTER sip:server_ip:5061;transport=tcp SIP/2.0
 Via: SIP/2.0/TCP our_ip;branch=z9hG4bKXK27gKtt61X9a
 Max-Forwards: 70
 From: <sip:xxxx at server_ip:5061>;tag=Q4ytjXN7t319Q
 To: <sip:xxxx at server_ip:5061>
 Call-ID: 29e638e8-ba6d-11e1-aa8e-e35fbde939c6
 CSeq: 29804746 REGISTER
 Contact: <sip:gw+xxxx at our_ip:5060;transport=tcp;gw=xxxx>
 Expires: 3600
 User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
 Supported: timer, precondition, path, replaces
 Content-Length: 0

 Instead of switching to TLS mode FS just sends plain unencrypted reg request to remote port 5061. After that remote server closes connection.

 Have you practically succeded in placing FS into TLS registration mode?

 Cheers / Mike

----- Original Message -----
From: Sean Devoy
Sent: 06/20/12 10:27 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] gateway registration using TLS

NO, sorry. I just saw ?remote_ip_here? in the error message and thought may you actually had used that.
Sorry.
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]  *On Behalf Of *Mi Ke
 *Sent:* Wednesday, June 20, 2012 12:20 PM
 *To:* FreeSWITCH Users Help
 *Subject:* Re: [Freeswitch-users] gateway registration using TLS
Dear Sean,

 It's not that bad - remote_ip, xxx and yyy are only placeholders in my example - actual config has real params ;)

 Did you mean that FS will automatically use TLS for registration when I append :5061 to realm value ?

 Thanks / MiKe
----- Original Message -----
From: Sean Devoy
Sent: 06/20/12 07:03 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] gateway registration using TLS

Mi Ke,
You should put your actual remote provider?s domain name or tip address in where it says ?remote_ip_here? and I assume you have entered you actual username and password where it says ?yyy? and ?zzz?.
It should look SIMILAR to this, but with your provider: <param name="realm" value="sip.voipinnovations.com:5061"/>
Hope that helps,
Sean
From: freeswitch-users-bounces at lists.freeswitch.org  %5Bmailto:freeswitch-users-bounces at lists.freeswitch.org%5D  *On Behalf Of *Mi Ke
 *Sent:* Wednesday, June 20, 2012 11:46 AM
 *To:* Freeswitch Users
 *Subject:* [Freeswitch-users] gateway registration using TLS

Hi All,

 Is it possible to explicitly tell sofia to register as a client using TLS? Here's my TLS-related part of sofia config:

 <profiles>
 <profile name="external">
 <domains>
 <domain name="all" alias="true" parse="false"/>
 </domains>
 <gateways>
 <gateway name="xxx">
 <param name="register" value="true"/>
 <param name="username" value="yyy"/>
 <param name="realm" value="remote_ip_here:5061"/>
 <param name="password" value="zzz"/>
 <param name="register-transport" value="tcp"/>
 <param name="expire-seconds" value="3600"/>
 <param name="retry-seconds" value="60" />
 </gateway>
 </gateways>
 <settings>
 ...
 <param name="context" value="external"/>
 <param name="sip-port" value="5060"/>
 <param name="tls" value="true"/>
 <param name="tls-only" value="false"/>
 <param name="tls-bind-params" value="transport=tls"/>
 <param name="contact-params" value="tport=tcp"/>
 <param name="tls-version" value="sslv23"/>
 <param name="tls-cert-dir" value="/usr/local/freeswitch/conf/ssl"/>
 <param name="tls-sip-port" value="5061"/>
 <param name="dialplan" value="XML"/>
 ...

 </settings>
 </profile>
 </profiles>

 But that does not work...

 tport_connected( 

0x7f9b44014110): events CONNECTED
 tport_send_event(0x7f9b44014110) - ready to send to (tcp/remote_ip_here:5061 http://212.58.166.54:5061 )
 tport_vsend(0x7f9b44014110): 600 bytes of 600 to tcp/remote_ip_here:5061 http://212.58.166.54:5061 
 tport_vsend returned 600
 tport_set_events(0x7f9b44014110): events IN
 tport(0x7f9b44014110): reset timer
 tport_wakeup(0x7f9b44014110): events IN HUP ERR
 tport_release(0x7f9b44014110): 0x7f9b4401a460 by 0x7f9b4402ca70 with (nil)
nta: REGISTER (29733430): Connection reset by peer (104) with tcp/[remote_ip_here]:5061
nua(0x7f9b440196a0): event r_register 503 Service Unavailable

Any ideas what I'm doing wrong?

 Thanks / MiKe
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From getallad at gmail.com  Thu Jun 21 15:33:58 2012
From: getallad at gmail.com (Sergei)
Date: Thu, 21 Jun 2012 14:33:58 +0300
Subject: [Freeswitch-users] Attended transfer fails when in bypass or proxy
	mode
Message-ID: <001801cd4fa1$c07a9380$416fba80$@gmail.com>

Good day all.

I've recently installed freeswitch in our office and in general everybody is
happy.

The only disappointing thing - is attended transfer. 

I've tried to make it working on Cisco7960 and on Grandstream GXP2000 - but
with no luck. Could please, anybody respond, who successfully makes attended
transfer using bypass mode?

 

In sip profile I use following options:

 

<param name="media-option" value="resume-media-on-hold"/>

<param name="media-option" value="bypass-media-after-att-xfer"/>

 

Scenario is the simplest one: A calls B, B transfers to C (with attended
transfer - like putting A on hold, making outcall to C, bridging two calls
together). 

What I observe on Cisco phone - is following:

 

1.      A calls B. B answers, puts A on hold and calls C. C answers and
talks to B. Up to this time - everything is fine.

2.      B presses 'transfer', in order to bridge two calls together. But,
instead, C hangs up, A stays on hold and B looks like he just put A on hold.
If B right now presses hold button - he will appear in dialog with A again.

 

Did anybody observe such behavior? What could be the advice in such
situation?

 

Thank you in advance, 

 

Siarhei.

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From shcherbatyuk at belrosbank.by  Thu Jun 21 15:42:17 2012
From: shcherbatyuk at belrosbank.by (Eugene Shcherbatyuk)
Date: Thu, 21 Jun 2012 14:42:17 +0300
Subject: [Freeswitch-users] mod_python build error
Message-ID: <4FE30899.10605@belrosbank.by>

Today I could not update mod_python after git pull. Well, I am not quite
certain what to do next except for reporting to the list...

Best regards,
Gene

OS: ubuntu 10.04 LTS server

FreeSWITCH: Version
1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T105634Z

Message:

root at fs:/usr/local/src/freeswitch# make mod_python

making all mod_python
make[4]: *** [mod_python_wrap.cpp] Error 1
make[3]: *** [all] Error 1
make[2]: *** [mod_python-all] Error 1
make[1]: *** [mod_python] Error 2
make: *** [mod_python] Error 2
root at fs:/usr/local/src/freeswitch#



=========================================================

?????? ????????? ? ????? ???????? (???????????) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????.

=========================================================

This message and any attachments (the ?message?) are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC ?BELROSBANK? nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified.

=========================================================


From esander83 at hushmail.com  Thu Jun 21 17:00:21 2012
From: esander83 at hushmail.com (Eric Sander)
Date: Thu, 21 Jun 2012 08:00:21 -0500
Subject: [Freeswitch-users]
	=?utf-8?q?Does_mod=5Ffifo_not_take_in=09effect?=
	=?utf-8?q?ive=5Fcaller=5Fid=5Fname=3F?=
Message-ID: <20120621130021.5286DA6E40@smtp.hushmail.com>

Any plans to export effective_caller_id_name into mod_fifo or is this task considered a bounty?

eric_s

On Saturday, June 16, 2012 at 1:37 AM, Eric Sander <esander83 at hushmail.com> wrote:
>
>Any body know if effective_caller_id_name can be passed on to 
>modfifo. Read the wiki on it and there's no info.  This would be a 
>helpful feature to include as other parts of FS use this to 
>display info.
>
>thanks
>eric_s
>
>On Thursday, June 14, 2012 at 6:35 AM, Andrew Cassidy 
><andrew at cassidywebservices.co.uk> wrote:
>>
>>mod_callcenter also ignores those variables unless you set one 
>>called
>>cc_export_vars. Perhaps there&#39;s a similar mechanism for 
>>mod_fifo?
>>
>>On 14 June 2012 05:31,   wrote:
>> Hi All,
>>I have this basic dialplan: 
>>I can see the call get logged via the first extension but 
>it&#39;s 
>>not
>>passing the info to modfifo.  I can remove the 2nd extension and 
>>put a
>>basic: 
>>and the caller id name info is displayed on the phones but not 
>when
>>routing through mod_fifo.  I have tried both set/export and it 
>>still
>>doesn&#39;t seem to work.
>>Is this correct? 
>>Thanks,eric_s
>>
>>__________________________________________________________________
>_
>>______
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> Join Us At ClueCon - Aug 7-9, 2012
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch
>>-users
>> http://www.freeswitch.org
>>-- 
>>Andrew Cassidy BSc (Hons) MBCS SSCAManaging Director
>>
>> T 03300 100 960  F 03300 100 961 E 
>>andrew at cassidywebservices.co.uk W
>>www.cassidywebservices.co.uk
>
>
>___________________________________________________________________
>______
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>users
>http://www.freeswitch.org



From shcherbatyuk at belrosbank.by  Thu Jun 21 17:18:20 2012
From: shcherbatyuk at belrosbank.by (Eugene Shcherbatyuk)
Date: Thu, 21 Jun 2012 16:18:20 +0300
Subject: [Freeswitch-users] follow-up: mod_python build error
Message-ID: <4FE31F1C.5010508@belrosbank.by>


Sorry  for missing these bits (according to
http://wiki.freeswitch.org/wiki/Report_Issue_Checklist)

root at fs:/usr/local/src/freeswitch# cat /proc/cpuinfo
processor       : 0
vendor_id       : GenuineIntel
cpu family      : 6
model           : 30
model name      : Intel(R) Core(TM) i5 CPU         760  @ 2.80GHz
stepping        : 5
cpu MHz         : 2904.327
cache size      : 6144 KB
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 5
wp              : yes
flags           : fpu vme de pse tsc msr mce cx8 apic mtrr pge mca cmov pat pse36
clflush mmx fxsr sse sse2 constant_tsc up pni monitor ssse3
bogomips        : 5808.65
clflush size    : 64
cache_alignment : 64
address sizes   : 36 bits physical, 48 bits virtual
power management:

root at fs:/usr/local/src/freeswitch# uname -a
72-Ubuntu SMP Fri Jul 29 21:08:37 UTC 2011 i686 GNU/Linux

Yes, situation was reproducible:
a) when updating
b) when trying to make fresh build.
c) when trying to build additional module

No, no modification to build scripts, except for uncommenting mod_python
in modules.conf

> Today I could not update mod_python after git pull. Well, I am not quite certain what to do next except for reporting to the list...
>
> Best regards,
> Gene
>
> OS: ubuntu 10.04 LTS server
>
> FreeSWITCH: Version 1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T105634Z
>
> Message:
>
> root at fs:/usr/local/src/freeswitch# make mod_python
>
> making all mod_python
> make[4]: *** [mod_python_wrap.cpp] Error 1
> make[3]: *** [all] Error 1
> make[2]: *** [mod_python-all] Error 1
> make[1]: *** [mod_python] Error 2
> make: *** [mod_python] Error 2
> root at fs:/usr/local/src/freeswitch#



=========================================================

?????? ????????? ? ????? ???????? (???????????) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????.

=========================================================

This message and any attachments (the ?message?) are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC ?BELROSBANK? nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified.

=========================================================


From sdevoy at bizfocused.com  Thu Jun 21 17:37:53 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 21 Jun 2012 09:37:53 -0400
Subject: [Freeswitch-users] gateway registration using TLS
In-Reply-To: <20120621131704.182980@gmx.com>
References: <20120621131704.182980@gmx.com>
Message-ID: <023d01cd4fb3$0f6cc7e0$2e4657a0$@bizfocused.com>

HI MiKe,

 

I have not, but we have been looking into it.

 

Have you seen this thread:

http://lists.freeswitch.org/pipermail/freeswitch-users/2008-May/003233.html

where Brian West says:

You need to make sure you have openssl-devel when you configure/ 

compile Sofia otherwise it's not enabled.  Also their is more to it  

than "bin/gentls_cert setup"

http://wiki.freeswitch.org/wiki/Tls

 

And Helmut Kuper responds with:

ok works now. I had to install libgnutls-dev befor running ./configure
script. TLS works with default-values of "bin/gentls_cert setup" and
"bin/gentls_cert create".

 

Hope that helps,

Sean

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mi Ke
Sent: Thursday, June 21, 2012 9:17 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] gateway registration using TLS

 

Hi Sean,

It seems that setting realm port to 5061 does not set registration mode to TLS - this is how remote server sees our reg attempt:

REGISTER sip:server_ip:5061;transport=tcp SIP/2.0
Via: SIP/2.0/TCP our_ip;branch=z9hG4bKXK27gKtt61X9a
Max-Forwards: 70
From: <sip:xxxx at server_ip:5061>;tag=Q4ytjXN7t319Q
To: <sip:xxxx at server_ip:5061>
Call-ID: 29e638e8-ba6d-11e1-aa8e-e35fbde939c6
CSeq: 29804746 REGISTER
Contact: <sip:gw+xxxx at our_ip:5060;transport=tcp;gw=xxxx>
Expires: 3600
User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Content-Length: 0

Instead of switching to TLS mode FS just sends plain unencrypted reg request to remote port 5061. After that remote server closes connection.

Have you practically succeded in placing FS into TLS registration mode?

Cheers / Mike

 

----- Original Message -----

From: Sean Devoy

Sent: 06/20/12 10:27 PM

To: 'FreeSWITCH Users Help'

Subject: Re: [Freeswitch-users] gateway registration using TLS

 

NO, sorry.  I just saw ?remote_ip_here? in the error message and thought may you actually had used that.

 

 

 

 

 

Sorry.

 

 

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mi Ke
Sent: Wednesday, June 20, 2012 12:20 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] gateway registration using TLS

 

 

 

 

 

Dear Sean,

It's not that bad - remote_ip, xxx and yyy are only placeholders in my example - actual config has real params ;)

Did you mean that FS will automatically use TLS for registration when I append :5061 to realm value ?

Thanks / MiKe

 

 

 

 

 

----- Original Message -----

 

 

From: Sean Devoy

 

 

Sent: 06/20/12 07:03 PM

 

 

To: 'FreeSWITCH Users Help'

 

 

Subject: Re: [Freeswitch-users] gateway registration using TLS

 

 

 

 

 

Mi Ke,

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

You should put your actual remote provider?s domain name or tip address in where it says ?remote_ip_here? and I assume you have entered you actual username and password where it says ?yyy? and ?zzz?.

 

 

 

 

 

 

 

 

It should look SIMILAR to this, but with your provider: <param name="realm" value="sip.voipinnovations.com:5061"/>

 

 

 

 

 

 

 

 

Hope that helps,

 

 

 

 

 

 

 

 

Sean

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] <mailto:%5Bmailto:freeswitch-users-bounces at lists.freeswitch.org%5D>  On Behalf Of Mi Ke
Sent: Wednesday, June 20, 2012 11:46 AM
To: Freeswitch Users
Subject: [Freeswitch-users] gateway registration using TLS

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Hi All,

Is it possible to explicitly tell sofia to register as a client using TLS? Here's my TLS-related part of sofia config:

        <profiles>
            <profile name="external">
                <domains>
                    <domain name="all" alias="true" parse="false"/>
                </domains>
                <gateways>
                        <gateway name="xxx">
                                  <param name="register" value="true"/>
                                  <param name="username" value="yyy"/>
                                  <param name="realm" value="remote_ip_here:5061"/>
                                  <param name="password" value="zzz"/>
                                  <param name="register-transport" value="tcp"/>
                                  <param name="expire-seconds" value="3600"/>
                                  <param name="retry-seconds" value="60" />
                         </gateway>
                </gateways>
                <settings>
                       ...
                       <param name="context" value="external"/>
                       <param name="sip-port" value="5060"/>
                       <param name="tls" value="true"/>
                       <param name="tls-only" value="false"/>
                       <param name="tls-bind-params" value="transport=tls"/>
                       <param name="contact-params" value="tport=tcp"/>
                       <param name="tls-version" value="sslv23"/>
                       <param name="tls-cert-dir" value="/usr/local/freeswitch/conf/ssl"/>
                       <param name="tls-sip-port" value="5061"/>
                       <param name="dialplan" value="XML"/>
                       ...

                </settings>
            </profile>
        </profiles>

But that does not work...

tport_connected( 

 

 

 

 

 

 

 

 

0x7f9b44014110): events CONNECTED
tport_send_event(0x7f9b44014110) - ready to send to (tcp/remote_ip_here:5061 <http://212.58.166.54:5061> )
tport_vsend(0x7f9b44014110): 600 bytes of 600 to tcp/remote_ip_here:5061 <http://212.58.166.54:5061> 
tport_vsend returned 600
tport_set_events(0x7f9b44014110): events IN
tport(0x7f9b44014110): reset timer
tport_wakeup(0x7f9b44014110): events IN HUP ERR
tport_release(0x7f9b44014110): 0x7f9b4401a460 by 0x7f9b4402ca70 with (nil)
nta: REGISTER (29733430): Connection reset by peer (104) with tcp/[remote_ip_here]:5061
nua(0x7f9b440196a0): event r_register 503 Service Unavailable 

 

 

 

 

 

 

 

 


Any ideas what I'm doing wrong?

Thanks / MiKe

 

 

 

 

 

 

 

 

 

 

 

 

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From lazyvirus at gmx.com  Thu Jun 21 17:48:49 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Thu, 21 Jun 2012 15:48:49 +0200
Subject: [Freeswitch-users] Speccing server for enterprise FS
In-Reply-To: <CAKV-jFzwRfZ1BPS-2Dv7Oxt0gsW6ufuD2xY9S5HmhQSs83BbJg@mail.gmail.com>
References: <4FE2F305.4090808@thewinelake.com>
	<CAKV-jFzwRfZ1BPS-2Dv7Oxt0gsW6ufuD2xY9S5HmhQSs83BbJg@mail.gmail.com>
Message-ID: <20120621154849.5016cb69@anubis.defcon1>

On Thu, 21 Jun 2012 08:14:15 -0400
Brian Foster <bdfoster at endigotech.com> wrote:

> I would like to suggest that you should be doing HA and
> load balancing between atleast two servers when dealing with that
> kind of traffic. Its hard for me to justify putting all that
> traffic on one server. Just food for thought. Depending on how
> many calls you want to risk losing at any given time, your machine
> specs might be much lower than expected.

I hijack (a bit) this thread: with much lower needs than Alex (let's
say for an average size SME's PBX), shall FS perform well on a
machine that is already a svr for other things (lets says, e-mail,
IM, files (samba), etc, or does it need to be on a stand alone?
(in any case w/ its own Ethernet I/F)

JY
-- 
When the shit hits the fan, keep your mouth shut!


From rhuddleston at gmail.com  Thu Jun 21 17:54:28 2012
From: rhuddleston at gmail.com (Robert-IPhone)
Date: Thu, 21 Jun 2012 09:54:28 -0400
Subject: [Freeswitch-users] Speccing server for enterprise FS
In-Reply-To: <20120621154849.5016cb69@anubis.defcon1>
References: <4FE2F305.4090808@thewinelake.com>
	<CAKV-jFzwRfZ1BPS-2Dv7Oxt0gsW6ufuD2xY9S5HmhQSs83BbJg@mail.gmail.com>
	<20120621154849.5016cb69@anubis.defcon1>
Message-ID: <A3A75CB7-C503-4B21-912A-7E7E1D63D255@gmail.com>

I would never share roles with voip. Unless you are running a 2 line office ;)


Sent from my iPhone 4S

On Jun 21, 2012, at 9:48 AM, Bzzz <lazyvirus at gmx.com> wrote:

> On Thu, 21 Jun 2012 08:14:15 -0400
> Brian Foster <bdfoster at endigotech.com> wrote:
> 
>> I would like to suggest that you should be doing HA and
>> load balancing between atleast two servers when dealing with that
>> kind of traffic. Its hard for me to justify putting all that
>> traffic on one server. Just food for thought. Depending on how
>> many calls you want to risk losing at any given time, your machine
>> specs might be much lower than expected.
> 
> I hijack (a bit) this thread: with much lower needs than Alex (let's
> say for an average size SME's PBX), shall FS perform well on a
> machine that is already a svr for other things (lets says, e-mail,
> IM, files (samba), etc, or does it need to be on a stand alone?
> (in any case w/ its own Ethernet I/F)
> 
> JY
> -- 
> When the shit hits the fan, keep your mouth shut!
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From avi at avimarcus.net  Thu Jun 21 18:20:41 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 21 Jun 2012 17:20:41 +0300
Subject: [Freeswitch-users] Speccing server for enterprise FS
In-Reply-To: <A3A75CB7-C503-4B21-912A-7E7E1D63D255@gmail.com>
References: <4FE2F305.4090808@thewinelake.com>
	<CAKV-jFzwRfZ1BPS-2Dv7Oxt0gsW6ufuD2xY9S5HmhQSs83BbJg@mail.gmail.com>
	<20120621154849.5016cb69@anubis.defcon1>
	<A3A75CB7-C503-4B21-912A-7E7E1D63D255@gmail.com>
Message-ID: <CALNZuwmW6UYtLt0Aip2LEDj2o9x2kTibRzFTVmG83UR4Qon5sg@mail.gmail.com>

To clarify: FreeSWITCH doesn't need so much resources such that it needs to
run standalone.

However, as with virtualization... voip calls are meant to be real time and
require real time processing.
E.g. if you're running a ptime of 20 (the norm, it seems) that means there
are 50 packets for each channel each second.
Therefore, if the machine is busy for just 20ms and FS can't process that
packet (on every channel!), stuff starts to happen. So running standalone
it's more about rock-solid stability rather than just having, on average,
enough CPU.

-Avi


On Thu, Jun 21, 2012 at 4:54 PM, Robert-IPhone <rhuddleston at gmail.com>wrote:

> I would never share roles with voip. Unless you are running a 2 line
> office ;)
>
>
> Sent from my iPhone 4S
>
> On Jun 21, 2012, at 9:48 AM, Bzzz <lazyvirus at gmx.com> wrote:
>
> > On Thu, 21 Jun 2012 08:14:15 -0400
> > Brian Foster <bdfoster at endigotech.com> wrote:
> >
> >> I would like to suggest that you should be doing HA and
> >> load balancing between atleast two servers when dealing with that
> >> kind of traffic. Its hard for me to justify putting all that
> >> traffic on one server. Just food for thought. Depending on how
> >> many calls you want to risk losing at any given time, your machine
> >> specs might be much lower than expected.
> >
> > I hijack (a bit) this thread: with much lower needs than Alex (let's
> > say for an average size SME's PBX), shall FS perform well on a
> > machine that is already a svr for other things (lets says, e-mail,
> > IM, files (samba), etc, or does it need to be on a stand alone?
> > (in any case w/ its own Ethernet I/F)
> >
> > JY
> > --
> > When the shit hits the fan, keep your mouth shut!
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From avi at avimarcus.net  Thu Jun 21 18:22:30 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 21 Jun 2012 17:22:30 +0300
Subject: [Freeswitch-users] follow-up: mod_python build error
In-Reply-To: <4FE31F1C.5010508@belrosbank.by>
References: <4FE31F1C.5010508@belrosbank.by>
Message-ID: <CALNZuwkZXoNJ3YiMqy3Rm_bR2VPCZ3k-qSahVDc+wXfDO0zMJw@mail.gmail.com>

Please post your compilation issue on http://jira.freeswitch.org to ensure
it gets followed up on.
Thanks,
-Avi
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From bdfoster at endigotech.com  Thu Jun 21 18:25:41 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 21 Jun 2012 10:25:41 -0400
Subject: [Freeswitch-users] Speccing server for enterprise FS
In-Reply-To: <CALNZuwmW6UYtLt0Aip2LEDj2o9x2kTibRzFTVmG83UR4Qon5sg@mail.gmail.com>
References: <4FE2F305.4090808@thewinelake.com>
	<CAKV-jFzwRfZ1BPS-2Dv7Oxt0gsW6ufuD2xY9S5HmhQSs83BbJg@mail.gmail.com>
	<20120621154849.5016cb69@anubis.defcon1>
	<A3A75CB7-C503-4B21-912A-7E7E1D63D255@gmail.com>
	<CALNZuwmW6UYtLt0Aip2LEDj2o9x2kTibRzFTVmG83UR4Qon5sg@mail.gmail.com>
Message-ID: <CAKV-jFyQuyJA6CMT2Xqrm517Xhm4eibQ0Y22BCt7GM3NWg+5XQ@mail.gmail.com>

There are some things you can do to keep FreeSWITCH as the machines highest
priority but that's probably outside of the scope of this discussion.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 21, 2012 10:22 AM, "Avi Marcus" <avi at avimarcus.net> wrote:

> To clarify: FreeSWITCH doesn't need so much resources such that it needs
> to run standalone.
>
> However, as with virtualization... voip calls are meant to be real time
> and require real time processing.
> E.g. if you're running a ptime of 20 (the norm, it seems) that means there
> are 50 packets for each channel each second.
> Therefore, if the machine is busy for just 20ms and FS can't process that
> packet (on every channel!), stuff starts to happen. So running standalone
> it's more about rock-solid stability rather than just having, on average,
> enough CPU.
>
> -Avi
>
>
> On Thu, Jun 21, 2012 at 4:54 PM, Robert-IPhone <rhuddleston at gmail.com>wrote:
>
>> I would never share roles with voip. Unless you are running a 2 line
>> office ;)
>>
>>
>> Sent from my iPhone 4S
>>
>> On Jun 21, 2012, at 9:48 AM, Bzzz <lazyvirus at gmx.com> wrote:
>>
>> > On Thu, 21 Jun 2012 08:14:15 -0400
>> > Brian Foster <bdfoster at endigotech.com> wrote:
>> >
>> >> I would like to suggest that you should be doing HA and
>> >> load balancing between atleast two servers when dealing with that
>> >> kind of traffic. Its hard for me to justify putting all that
>> >> traffic on one server. Just food for thought. Depending on how
>> >> many calls you want to risk losing at any given time, your machine
>> >> specs might be much lower than expected.
>> >
>> > I hijack (a bit) this thread: with much lower needs than Alex (let's
>> > say for an average size SME's PBX), shall FS perform well on a
>> > machine that is already a svr for other things (lets says, e-mail,
>> > IM, files (samba), etc, or does it need to be on a stand alone?
>> > (in any case w/ its own Ethernet I/F)
>> >
>> > JY
>> > --
>> > When the shit hits the fan, keep your mouth shut!
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > Join Us At ClueCon - Aug 7-9, 2012
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Thu Jun 21 18:32:02 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 21 Jun 2012 10:32:02 -0400
Subject: [Freeswitch-users] Attended transfer fails when in bypass or
 proxy mode
In-Reply-To: <001801cd4fa1$c07a9380$416fba80$@gmail.com>
References: <001801cd4fa1$c07a9380$416fba80$@gmail.com>
Message-ID: <CAKV-jFyFu1xXXr4YuiFamYNvuxoWFkcm1s2DoE8AJns=BbgHug@mail.gmail.com>

As far as I know freeswitch has to do some sort of media handling to get
the attended transfer to work so bypassing media is probably not wise. You
may find out that it's a non issue to proxy media.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 21, 2012 9:25 AM, "Sergei" <getallad at gmail.com> wrote:

> Good day all.****
>
> I?ve recently installed freeswitch in our office and in general everybody
> is happy.****
>
> The only disappointing thing ? is attended transfer. ****
>
> I?ve tried to make it working on Cisco7960 and on Grandstream GXP2000 ?
> but with no luck. Could please, anybody respond, who successfully makes
> attended transfer using bypass mode?****
>
> ** **
>
> In sip profile I use following options:****
>
> ** **
>
> <param name="media-option" value="resume-media-on-hold"/>****
>
> <param name="media-option" value="bypass-media-after-att-xfer"/>****
>
> ** **
>
> Scenario is the simplest one: A calls B, B transfers to C (with attended
> transfer ? like putting A on hold, making outcall to C, bridging two calls
> together). ****
>
> What I observe on Cisco phone ? is following:****
>
> ** **
>
> **1.      **A calls B. B answers, puts A on hold and calls C. C answers
> and talks to B. Up to this time ? everything is fine.****
>
> **2.      **B presses ?transfer?, in order to bridge two calls together.
> But, instead, C hangs up, A stays on hold and B looks like he just put A on
> hold. If B right now presses hold button ? he will appear in dialog with A
> again.****
>
> ** **
>
> Did anybody observe such behavior? What could be the advice in such
> situation?****
>
> ** **
>
> Thank you in advance, ****
>
> ** **
>
> Siarhei.****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lazyvirus at gmx.com  Thu Jun 21 18:37:35 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Thu, 21 Jun 2012 16:37:35 +0200
Subject: [Freeswitch-users] Speccing server for enterprise FS
In-Reply-To: <CAKV-jFyQuyJA6CMT2Xqrm517Xhm4eibQ0Y22BCt7GM3NWg+5XQ@mail.gmail.com>
References: <4FE2F305.4090808@thewinelake.com>
	<CAKV-jFzwRfZ1BPS-2Dv7Oxt0gsW6ufuD2xY9S5HmhQSs83BbJg@mail.gmail.com>
	<20120621154849.5016cb69@anubis.defcon1>
	<A3A75CB7-C503-4B21-912A-7E7E1D63D255@gmail.com>
	<CALNZuwmW6UYtLt0Aip2LEDj2o9x2kTibRzFTVmG83UR4Qon5sg@mail.gmail.com>
	<CAKV-jFyQuyJA6CMT2Xqrm517Xhm4eibQ0Y22BCt7GM3NWg+5XQ@mail.gmail.com>
Message-ID: <20120621163735.67b3a005@anubis.defcon1>

On Thu, 21 Jun 2012 10:25:41 -0400
Brian Foster <bdfoster at endigotech.com> wrote:

Thanks to the three of you; that's what I was thinking
but now I've got the full confirmation.

With the services I'm planning there will be too much risks of
loss, even with a strong renicing of processes.

So, I think I'll use a small separated svr on a separate network
(if possible:(, with an Atom card, that should do the trick.

JY
-- 
The Army needs leaders the way a foot needs a big toe.
		-- Bill Murray


From shcherbatyuk at belrosbank.by  Thu Jun 21 18:17:46 2012
From: shcherbatyuk at belrosbank.by (Eugene Shcherbatyuk)
Date: Thu, 21 Jun 2012 17:17:46 +0300
Subject: [Freeswitch-users] Speccing server for enterprise FS
In-Reply-To: <A3A75CB7-C503-4B21-912A-7E7E1D63D255@gmail.com>
References: <4FE2F305.4090808@thewinelake.com>
	<CAKV-jFzwRfZ1BPS-2Dv7Oxt0gsW6ufuD2xY9S5HmhQSs83BbJg@mail.gmail.com>
	<20120621154849.5016cb69@anubis.defcon1>
	<A3A75CB7-C503-4B21-912A-7E7E1D63D255@gmail.com>
Message-ID: <4FE32D0A.4080802@belrosbank.by>

Having multiple roles/applications on the same server is forced
necessity unless you have enouph boxes. Since we had deployed virtual
infrastructure no one ever wanted to set up multi-purpose servers.
Nevertheless, heavy-duty servers reside on physical, not virtual boxes.
They are dedicated servers though.

On 21/06/12 16:54, Robert-IPhone wrote:
> I would never share roles with voip. Unless you are running a 2 line office ;)
>
>
> Sent from my iPhone 4S
>
> On Jun 21, 2012, at 9:48 AM, Bzzz<lazyvirus at gmx.com>  wrote:
>
>> On Thu, 21 Jun 2012 08:14:15 -0400
>> Brian Foster<bdfoster at endigotech.com>  wrote:
>>
>>> I would like to suggest that you should be doing HA and
>>> load balancing between atleast two servers when dealing with that
>>> kind of traffic. Its hard for me to justify putting all that
>>> traffic on one server. Just food for thought. Depending on how
>>> many calls you want to risk losing at any given time, your machine
>>> specs might be much lower than expected.
>>
>> I hijack (a bit) this thread: with much lower needs than Alex (let's
>> say for an average size SME's PBX), shall FS perform well on a
>> machine that is already a svr for other things (lets says, e-mail,
>> IM, files (samba), etc, or does it need to be on a stand alone?
>> (in any case w/ its own Ethernet I/F)
>>
>> JY
>> --
>> When the shit hits the fan, keep your mouth shut!
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


=========================================================

?????? ????????? ? ????? ???????? (???????????) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????.

=========================================================

This message and any attachments (the ?message?) are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC ?BELROSBANK? nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified.

=========================================================


From shadowym at hotmail.com  Thu Jun 21 03:41:29 2012
From: shadowym at hotmail.com (shadowym)
Date: Wed, 20 Jun 2012 16:41:29 -0700
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <74D29BFA17294F97A0A8C40A49538E56@freeswitch.org>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>	<4FDFBCA8.1050700@quentustech.com>	<CAKzWOxWR6PJqPb6NqTLRQa5WMGP6H4zVte1w0i2UZSDN9+fQMg@mail.gmail.com>	<COL124-DS1734BA1C8437EE24E71567DDFE0@phx.gbl>
	<74D29BFA17294F97A0A8C40A49538E56@freeswitch.org>
Message-ID: <COL124-DS2EAD4B9766C4350798A2FDDFE0@phx.gbl>

Freeswitch is open source?  Really?

 

You get the prize for being the first to post a ?you ungrateful ba***rd? response.  Par for the course.  Thought it would happen a lot sooner.

 

So just because it is open source I am not allowed to be unhappy with a bad decision?  Ok check.  

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita
Sent: Wednesday, June 20, 2012 10:41 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] freeswitch directories moved

 

Dude, I have been trying not to follow this thread but I can't avoid it because it keeps showing up on my mail client over and over again.

 

Stop bitching and DO SOMETHING. This is open source, not something you've bought or that someone is trying to sell you. Your demanding tone is irritating to say the very least. You don't like that someone changed a few dirs and did not make the work like you expected, then just freaking FIX IT and contribute the solution. None of the developers have time to write documentation and they struggle to keep their day jobs while making it work for the rest of the world as well and you sit on your comfy place complaining that they are not doing enough? WTF?

 

SwK is 1 person that has been doing the stable branch as well as trying to package freeswitch to several distros, he could definitely use some help instead of aggressiveness and complaints. Contribute and hop in or live with what you have or just leave quiet but DON'T piss on everyones else's work. You want to be disappointed, that's OK, but you gotta earn it first. What have you done for the project so far?

 

Sorry to the list for my email but everything has its limits and I tend to be nice most of the times.

 

 

-- 

Jo?o Mesquita

Sent with Sparrow <http://www.sparrowmailapp.com/?sig> 

 

On Wednesday, June 20, 2012 at 11:57 AM, shadowym wrote:

Did the announcement also say they were getting rid of the conf directory and not bothering to updated directory settings in the default conf files (that are now in the root directory).  Do they plan to update all the freeswitch documentation now as well which talks about the conf directory all over the place?

 

How long ago was this announcement made?  The issue is not about being caught off guard btw.  It?s about why and how it?s being done.

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Monday, June 18, 2012 5:43 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] freeswitch directories moved

 

 

On Mon, Jun 18, 2012 at 4:41 PM, William King <william.king at quentustech.com> wrote:

I believe there was an announcement on this mailing list alerting that the new rpms would be undergoing major changes to include moving the directory locations. Search this list for new threads started by SwK in the past 6 months.

 

There was indeed an announcement by SwK himself. It was as close to FNLoIP (Flashing Neon Lights over IP) as you can get.
-MC

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com

 





 

Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com

 

Join Us At ClueCon - Aug 7-9, 2012

 

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org

 

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From jaasmailing at gmail.com  Thu Jun 21 18:50:50 2012
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Thu, 21 Jun 2012 16:50:50 +0200
Subject: [Freeswitch-users] Spandsp - Cannot send non-existant fax file
 (permissions?)
In-Reply-To: <CALn233qSm7rBkF=94CmxkVN209goN4CvmttHCXT99WOcA_yyaw@mail.gmail.com>
References: <4FE088F3.1060207@gmail.com>
	<CALn233qSm7rBkF=94CmxkVN209goN4CvmttHCXT99WOcA_yyaw@mail.gmail.com>
Message-ID: <4FE334CA.70005@gmail.com>

Right. There was a directory in the path without the read permission.

Regards,


Il 19/06/12 18.50, Mitch Capper ha scritto:
> Just because the file has readable permissions doesn't mean the entire
> path does.  Try doing an: sudo -u freeswitch file
> /usr/local/freeswitch/storage/fax/test_fax.tif
> if that works.  I would guess not then you can trace up the directory
> hopefully to figure out what permission is wrong.
>
> ~mitch
>
> On Tue, Jun 19, 2012 at 7:13 AM, Carlo Dimaggio<jaasmailing at gmail.com>  wrote:
>> Hi all,
>>
>> I'm in trouble with mod_spandsp and freeswitch executed by user
>> "freeswitch".
>> Even if the file is present and the permissions are ok (readable by
>> everyone), the FS console gets:
>>
>> 2012-06-19 16:06:30.098112 [ERR] mod_spandsp_fax.c:1289 Cannot send
>> non-existant fax file [/usr/local/freeswitch/storage/fax/test_fax.tif]
>>
>> FS is able to send if I execute the main process from user "root".
>>
>>
>> Does spandsp require other special permissions?
>>
>> Regards,
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jnankin at gmail.com  Thu Jun 21 18:52:21 2012
From: jnankin at gmail.com (Joshua Nankin)
Date: Thu, 21 Jun 2012 09:52:21 -0500
Subject: [Freeswitch-users] Spandsp events not being fired
In-Reply-To: <CAJWEF2bRs=qS9Ldep2LN+ADnja53iqDtRX6C+YZ6y1qePc0RgA@mail.gmail.com>
References: <CAAYWZ8mUFBF1rcU_ErEURDgjCpmr8bGeZA8CCCfeQKPj2sOfBg@mail.gmail.com>
	<CAAYWZ8mNw=5=vak_j7XjnkHij+DisxQdVmUg5acZdCPuLTXELw@mail.gmail.com>
	<CAJWEF2bRs=qS9Ldep2LN+ADnja53iqDtRX6C+YZ6y1qePc0RgA@mail.gmail.com>
Message-ID: <CAAYWZ8mST-xjdc9uhfR=eULvYga-+54CP_z_XOwRw7X5EpkO4Q@mail.gmail.com>

Perfect.  Thanks Adam!

On Thu, Jun 14, 2012 at 1:19 PM, Adam Raszynski <netcentrica at gmail.com>wrote:

> I use this and it handles answer, hangup and fax result events:
>
> event plain CHANNEL_ANSWER CHANNEL_HANGUP CUSTOM spandsp::txfaxresult
>
>
> you can also add spandsp::rxfaxresult at the end
>
> 2012/6/14 Joshua Nankin <jnankin at gmail.com>
>
>>  Anyone?
>>
>> I'm using the ESL to listen to events, and I'm listening for ALL.  Not
>> seeing any spandsp events.
>>
>> On Wed, Jun 13, 2012 at 12:50 PM, Joshua Nankin <jnankin at gmail.com>wrote:
>>
>>> I'm sending a fax with freeswitch, but the events listed at
>>> http://wiki.freeswitch.org/wiki/Mod_spandsp#Events do not seem to be
>>> fired.  Do I need to enable them somehow?
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun 21 18:56:41 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jun 2012 07:56:41 -0700
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <COL124-DS2EAD4B9766C4350798A2FDDFE0@phx.gbl>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
	<4FDFBCA8.1050700@quentustech.com>
	<CAKzWOxWR6PJqPb6NqTLRQa5WMGP6H4zVte1w0i2UZSDN9+fQMg@mail.gmail.com>
	<COL124-DS1734BA1C8437EE24E71567DDFE0@phx.gbl>
	<74D29BFA17294F97A0A8C40A49538E56@freeswitch.org>
	<COL124-DS2EAD4B9766C4350798A2FDDFE0@phx.gbl>
Message-ID: <CAKzWOxXBTMm-iy5gKWUbtmyUyheQo6xNDzRa=pyrDuX4LNaHGw@mail.gmail.com>

Okay, this thread is now closed. I had hoped that we could have an adult
discussion on the topic but clearly I was wrong. Please do not reply to
this or any other message in the thread.

-MC

On Wed, Jun 20, 2012 at 4:41 PM, shadowym <shadowym at hotmail.com> wrote:

> Freeswitch is open source?  Really?****
>
> ** **
>
> You get the prize for being the first to post a ?you ungrateful ba***rd?
> response.  Par for the course.  Thought it would happen a lot sooner.****
>
> ** **
>
> So just because it is open source I am not allowed to be unhappy with a
> bad decision?  Ok check.  ****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jo?o
> Mesquita
> *Sent:* Wednesday, June 20, 2012 10:41 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] freeswitch directories moved****
>
> ** **
>
> Dude, I have been trying not to follow this thread but I can't avoid it
> because it keeps showing up on my mail client over and over again.****
>
> ** **
>
> Stop bitching and DO SOMETHING. This is open source, not something you've
> bought or that someone is trying to sell you. Your demanding tone is
> irritating to say the very least. You don't like that someone changed a few
> dirs and did not make the work like you expected, then just freaking FIX IT
> and contribute the solution. None of the developers have time to write
> documentation and they struggle to keep their day jobs while making it work
> for the rest of the world as well and you sit on your comfy place
> complaining that they are not doing enough? WTF?****
>
> ** **
>
> SwK is 1 person that has been doing the stable branch as well as trying to
> package freeswitch to several distros, he could definitely use some help
> instead of aggressiveness and complaints. Contribute and hop in or live
> with what you have or just leave quiet but DON'T piss on everyones else's
> work. You want to be disappointed, that's OK, but you gotta earn it first.
> What have you done for the project so far?****
>
> ** **
>
> Sorry to the list for my email but everything has its limits and I tend to
> be nice most of the times.****
>
> ** **
>
> ** **
>
> -- ****
>
> Jo?o Mesquita****
>
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>****
>
> ** **
>
> On Wednesday, June 20, 2012 at 11:57 AM, shadowym wrote:****
>
> Did the announcement also say they were getting rid of the conf directory
> and not bothering to updated directory settings in the default conf files
> (that are now in the root directory).  Do they plan to update all the
> freeswitch documentation now as well which talks about the conf directory
> all over the place?****
>
>  ****
>
> How long ago was this announcement made?  The issue is not about being
> caught off guard btw.  It?s about why and how it?s being done.****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
> *On Behalf Of *Michael Collins
> *Sent:* Monday, June 18, 2012 5:43 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] freeswitch directories moved
> ****
>
>  ****
>
>  ****
>
> On Mon, Jun 18, 2012 at 4:41 PM, William King <
> william.king at quentustech.com> wrote:****
>
> I believe there was an announcement on this mailing list alerting that the
> new rpms would be undergoing major changes to include moving the directory
> locations. Search this list for new threads started by SwK in the past 6
> months.****
>
>  ****
>
> There was indeed an announcement by SwK himself. It was as close to FNLoIP
> (Flashing Neon Lights over IP) as you can get.
> -MC
>
>
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From jayesh.voip at gmail.com  Thu Jun 21 19:01:14 2012
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Thu, 21 Jun 2012 20:31:14 +0530
Subject: [Freeswitch-users] reading headers in 200 response
Message-ID: <CALvF6vAC+uGZsBY7em0gfeW0R0+qfh8hwqKZcwsWv_cpfOgVCQ@mail.gmail.com>

Hi All,
Is it possible in freeswitch to read headers in 200 OK response. What I am
trying to do is route all calls to an opensips server from my freeswitch
and when a call answers; opensips sends a header in 200 response, for eg:
X-Record: Yes or X-Record. So I was checking if I can read this header in
freeswitch and call record_session to record that call inside freeswitch? I
was trying something similar, can this work?:

            <extension name="call_routing">
            <condition field="destination_number" expression="^(\d+)$">
                <action application="export"
data="execute_on_answer=set_record_status ${sip_h_X-Record}"/>
                <action application="set"
data="execute_on_answer=execute_extension check_recording"/>
                <action application="bridge"
data="{sip_invite_domain=${sip_from_host},ignore_display_updates=true}sofia/${context}/$1@
$mydomain"/>
            </condition>
            </extension>

           <extension name="check_recording">
               <condition field="${record_status}" expression="Yes">
                    <action application="record_session"
data="/usr/local/freeswitch/recordings/extensions/123456.mp3"/>

               </condition>
           </extension>

Can someone help me in understanding if this is possible with freeswitch.

Thanks,

--- Jayesh
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From anthony.minessale at gmail.com  Thu Jun 21 19:14:55 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Jun 2012 10:14:55 -0500
Subject: [Freeswitch-users] freeswitch directories moved
In-Reply-To: <COL124-DS2EAD4B9766C4350798A2FDDFE0@phx.gbl>
References: <006401cd4cfc$09e31900$1da94b00$@hotmail.com>
	<COL124-DS21CAA0505F91B8BB6A7CB1DDF80@phx.gbl>
	<4FDFBCA8.1050700@quentustech.com>
	<CAKzWOxWR6PJqPb6NqTLRQa5WMGP6H4zVte1w0i2UZSDN9+fQMg@mail.gmail.com>
	<COL124-DS1734BA1C8437EE24E71567DDFE0@phx.gbl>
	<74D29BFA17294F97A0A8C40A49538E56@freeswitch.org>
	<COL124-DS2EAD4B9766C4350798A2FDDFE0@phx.gbl>
Message-ID: <CAKbxfG8HKn3TXQixYaT_NJQ7q5dSONnk5byHE4P_zi-j1z1aBw@mail.gmail.com>

You're quite entitled to be unhappy with a decision.  The problem is,
the more times you reenforce your arrogant snotty superiority with the
members of the mailing list, the less productive things get.  Your
last comment about how you were waiting for someone to play the "this
is open source you ungrateful bastard" card is astonishing.  What card
would you expect to play against the "I am an entitlist user of your
software trolling on the list because you have inadvertently forced me
to do some work" card? Just because its spec files doesn't make the
process any different.  If you found a crash bug, would you get mad
because you updated to this broken version or file a JIRA and get it
fixed?  If you are unhappy with a change, you need to report it to the
proper channels and be willing to live with the decisions made to
resolve your issue.  Frankly, I started this project for me.  I have
been working on it for 8 years now and we were recently given the
opportunity to bring in a dedicated individual to work over those spec
files and make them as compliant as possible with all the other angry
people complaining about the ones you like.  We are not going to pass
up this chance to get organized and separate the code into a stable
branch with packages that fit in with most environments.  I don't even
use those packages so I have little concern with where the files go.
I learned long ago to try not to let that bother me or every linux
distro on earth would put me in the loony bin.



On Wed, Jun 20, 2012 at 6:41 PM, shadowym <shadowym at hotmail.com> wrote:
> Freeswitch is open source?? Really?
>
>
>
> You get the prize for being the first to post a ?you ungrateful ba***rd?
> response.? Par for the course.? Thought it would happen a lot sooner.
>
>
>
> So just because it is open source I am not allowed to be unhappy with a bad
> decision?? Ok check.
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o
> Mesquita
> Sent: Wednesday, June 20, 2012 10:41 AM
>
>
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] freeswitch directories moved
>
>
>
> Dude, I have been trying not to follow this thread but I can't avoid it
> because it keeps showing up on my mail client over and over again.
>
>
>
> Stop bitching and DO SOMETHING. This is open source, not something you've
> bought or that someone is trying to sell you. Your demanding tone is
> irritating to say the very least. You don't like that someone changed a few
> dirs and did not make the work like you expected, then just freaking FIX IT
> and contribute the solution. None of the developers have time to write
> documentation and they struggle to keep their day jobs while making it work
> for the rest of the world as well and you sit on your comfy place
> complaining that they are not doing enough? WTF?
>
>
>
> SwK is 1 person that has been doing the stable branch as well as trying to
> package freeswitch to several distros, he could definitely use some help
> instead of aggressiveness and complaints. Contribute and hop in or live with
> what you have or just leave quiet but DON'T piss on everyones else's work.
> You want to be disappointed, that's OK, but you gotta earn it first. What
> have you done for the project so far?
>
>
>
> Sorry to the list for my email but everything has its limits and I tend to
> be nice most of the times.
>
>
>
>
>
> --
>
> Jo?o Mesquita
>
> Sent with Sparrow
>
>
>
> On Wednesday, June 20, 2012 at 11:57 AM, shadowym wrote:
>
> Did the announcement also say they were getting rid of the conf directory
> and not bothering to updated directory settings in the default conf files
> (that are now in the root directory).? Do they plan to update all the
> freeswitch documentation now as well which talks about the conf directory
> all over the place?
>
>
>
> How long ago was this announcement made?? The issue is not about being
> caught off guard btw.? It?s about why and how it?s being done.
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
> Collins
> Sent: Monday, June 18, 2012 5:43 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] freeswitch directories moved
>
>
>
>
>
> On Mon, Jun 18, 2012 at 4:41 PM, William King <william.king at quentustech.com>
> wrote:
>
> I believe there was an announcement on this mailing list alerting that the
> new rpms would be undergoing major changes to include moving the directory
> locations. Search this list for new threads started by SwK in the past 6
> months.
>
>
>
> There was indeed an announcement by SwK himself. It was as close to FNLoIP
> (Flashing Neon Lights over IP) as you can get.
> -MC
>
> _________________________________________________________________________
>
> Professional FreeSWITCH Consulting Services:
>
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
>
>
>
> 
>
> 
>
>
>
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
>
> http://wiki.freeswitch.org
>
> http://www.cluecon.com
>
>
>
> Join Us At ClueCon - Aug 7-9, 2012
>
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From getallad at gmail.com  Thu Jun 21 19:27:24 2012
From: getallad at gmail.com (Sergei)
Date: Thu, 21 Jun 2012 18:27:24 +0300
Subject: [Freeswitch-users] Attended transfer fails when in bypass or
	proxy mode
In-Reply-To: <CAKV-jFyFu1xXXr4YuiFamYNvuxoWFkcm1s2DoE8AJns=BbgHug@mail.gmail.com>
References: <001801cd4fa1$c07a9380$416fba80$@gmail.com>
	<CAKV-jFyFu1xXXr4YuiFamYNvuxoWFkcm1s2DoE8AJns=BbgHug@mail.gmail.com>
Message-ID: <00b801cd4fc2$5c53d4e0$14fb7ea0$@gmail.com>

Brian, thank you for your comment.

In our case - like centralized location of Freeswitch, distributed offices,
and wan channels between them - bypassing it the only option in order not to
saturate these channels. 

I agree, that attended transfer requires media handling - Freeswitch must
put on hold the call in order to make transfer, but this is much lighter
load on network in comparison with flooding it with RTP of every call (call
between two phones in filial will go through central point).

By the same reason, proxying media is not an option too - still puts load on
channels.

 

The question which is still unclear - is this scenario fully supported in
Freeswitch, and, if yes, on which phones and, probably, with which versions
of Freeswitch and with which config options.

 

Siarhei. 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
Foster
Sent: Thursday, June 21, 2012 5:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Attended transfer fails when in bypass or
proxy mode

 

As far as I know freeswitch has to do some sort of media handling to get the
attended transfer to work so bypassing media is probably not wise. You may
find out that it's a non issue to proxy media.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.

On Jun 21, 2012 9:25 AM, "Sergei" <getallad at gmail.com> wrote:

Good day all.

I've recently installed freeswitch in our office and in general everybody is
happy.

The only disappointing thing - is attended transfer. 

I've tried to make it working on Cisco7960 and on Grandstream GXP2000 - but
with no luck. Could please, anybody respond, who successfully makes attended
transfer using bypass mode?

 

In sip profile I use following options:

 

<param name="media-option" value="resume-media-on-hold"/>

<param name="media-option" value="bypass-media-after-att-xfer"/>

 

Scenario is the simplest one: A calls B, B transfers to C (with attended
transfer - like putting A on hold, making outcall to C, bridging two calls
together). 

What I observe on Cisco phone - is following:

 

1.      A calls B. B answers, puts A on hold and calls C. C answers and
talks to B. Up to this time - everything is fine.

2.      B presses 'transfer', in order to bridge two calls together. But,
instead, C hangs up, A stays on hold and B looks like he just put A on hold.
If B right now presses hold button - he will appear in dialog with A again.

 

Did anybody observe such behavior? What could be the advice in such
situation?

 

Thank you in advance, 

 

Siarhei.


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

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From b2m at a-cti.com  Thu Jun 21 19:44:07 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Thu, 21 Jun 2012 21:14:07 +0530
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
Message-ID: <CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>

All,

Is there a way to control recording speed for mod_skypopen? I was trying
to figure out the problem of fast recording/playing

Please find the attached file

Thanks,
Bala
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From bdfoster at endigotech.com  Thu Jun 21 19:46:08 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 21 Jun 2012 11:46:08 -0400
Subject: [Freeswitch-users] Attended transfer fails when in bypass or
 proxy mode
In-Reply-To: <00b801cd4fc2$5c53d4e0$14fb7ea0$@gmail.com>
References: <001801cd4fa1$c07a9380$416fba80$@gmail.com>
	<CAKV-jFyFu1xXXr4YuiFamYNvuxoWFkcm1s2DoE8AJns=BbgHug@mail.gmail.com>
	<00b801cd4fc2$5c53d4e0$14fb7ea0$@gmail.com>
Message-ID: <CAKV-jFxwcxW2oWgumNcKwbXuu-6eYe6MpNJo42x7cQB0QXq3jA@mail.gmail.com>

In that case it will have to be a function of the phone, I THINK Polycom
phones can do this, otherwise im not sure.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 21, 2012 11:40 AM, "Sergei" <getallad at gmail.com> wrote:

> Brian, thank you for your comment.****
>
> In our case ? like centralized location of Freeswitch, distributed
> offices,  and wan channels between them ? bypassing it the only option in
> order not to saturate these channels. ****
>
> I agree, that attended transfer requires media handling ? Freeswitch must
> put on hold the call in order to make transfer, but this is much lighter
> load on network in comparison with flooding it with RTP of every call (call
> between two phones in filial will go through central point).****
>
> By the same reason, proxying media is not an option too ? still puts load
> on channels.****
>
> ** **
>
> The question which is still unclear ? is this scenario fully supported in
> Freeswitch, and, if yes, on which phones and, probably, with which versions
> of Freeswitch and with which config options.****
>
> ** **
>
> Siarhei. ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian Foster
> *Sent:* Thursday, June 21, 2012 5:32 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Attended transfer fails when in bypass
> or proxy mode****
>
> ** **
>
> As far as I know freeswitch has to do some sort of media handling to get
> the attended transfer to work so bypassing media is probably not wise. You
> may find out that it's a non issue to proxy media.****
>
> Brian Foster
> Endigo Computer LLC****
>
> Sent from a mobile device.****
>
> On Jun 21, 2012 9:25 AM, "Sergei" <getallad at gmail.com> wrote:****
>
> Good day all.****
>
> I?ve recently installed freeswitch in our office and in general everybody
> is happy.****
>
> The only disappointing thing ? is attended transfer. ****
>
> I?ve tried to make it working on Cisco7960 and on Grandstream GXP2000 ?
> but with no luck. Could please, anybody respond, who successfully makes
> attended transfer using bypass mode?****
>
>  ****
>
> In sip profile I use following options:****
>
>  ****
>
> <param name="media-option" value="resume-media-on-hold"/>****
>
> <param name="media-option" value="bypass-media-after-att-xfer"/>****
>
>  ****
>
> Scenario is the simplest one: A calls B, B transfers to C (with attended
> transfer ? like putting A on hold, making outcall to C, bridging two calls
> together). ****
>
> What I observe on Cisco phone ? is following:****
>
>  ****
>
> 1.      A calls B. B answers, puts A on hold and calls C. C answers and
> talks to B. Up to this time ? everything is fine.****
>
> 2.      B presses ?transfer?, in order to bridge two calls together. But,
> instead, C hangs up, A stays on hold and B looks like he just put A on
> hold. If B right now presses hold button ? he will appear in dialog with A
> again.****
>
>  ****
>
> Did anybody observe such behavior? What could be the advice in such
> situation?****
>
>  ****
>
> Thank you in advance, ****
>
>  ****
>
> Siarhei.****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Thu Jun 21 19:47:14 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 21 Jun 2012 11:47:14 -0400
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
Message-ID: <CAKV-jFyBnCyrphC1R=tKu=E1PZAjs6NQ1Hp-L5APJoUaeM8Szg@mail.gmail.com>

That one worked.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 21, 2012 11:46 AM, "Balamurugan Mahendran" <b2m at a-cti.com> wrote:

> All,
>
> Is there a way to control recording speed for mod_skypopen? I was trying
> to figure out the problem of fast recording/playing
>
> Please find the attached file
>
> Thanks,
> Bala
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun 21 19:51:28 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jun 2012 08:51:28 -0700
Subject: [Freeswitch-users] New wiki page: Language_Files
Message-ID: <CAKzWOxXJ7cpDMoDmOx6_5DUXfwJ4FAGY8tgapS-6vLjrQy8X8g@mail.gmail.com>

Hello all,

I've created a new wiki page:

http://wiki.freeswitch.org/wiki/Language_Files

I've wanted this for some time and Brian's OCD finally took over and I just
*had* to create this page. :)

I've added languages that I know about or have heard about. *This page is
not complete, and it needs your help.* If you know anything about any
language files then please please go to that page and add what you know. If
you have any issues with editing the wiki then let me know and I will be
happy to assist. In fact, if need be I will do a 20-minute tutorial on wiki
editing on an upcoming community conference call.

Thanks!
-Michael
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From gmaruzz at gmail.com  Thu Jun 21 20:29:52 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 21 Jun 2012 18:29:52 +0200
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
Message-ID: <CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>

On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:

> All,
>
> Is there a way to control recording speed for mod_skypopen? I was trying
> to figure out the problem of fast recording/playing
>
> Please find the attached file
>

Balamurugan,

be aware that mod_skypopen operate at 16khz signed linear (let's say, at HD
audio level; double the number of samples of a normal PCMU or GSM or g711
or G729 call).

So, maybe you have to tweak the recording or playback parameters taking
this into account.

-giovanni





>
> Thanks,
> Bala
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From b2m at a-cti.com  Thu Jun 21 20:50:47 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Thu, 21 Jun 2012 22:20:47 +0530
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
Message-ID: <CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>

Is there a wiki link for these parameters? Thanks!!

Actually it was very good till last week(for about 6 months), suddenly its
broken. I could not trace it yet

Thanks,
Bala


On Thu, Jun 21, 2012 at 9:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

>
>
> On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>
>> All,
>>
>> Is there a way to control recording speed for mod_skypopen? I was trying
>> to figure out the problem of fast recording/playing
>>
>> Please find the attached file
>>
>
> Balamurugan,
>
> be aware that mod_skypopen operate at 16khz signed linear (let's say, at
> HD audio level; double the number of samples of a normal PCMU or GSM or
> g711 or G729 call).
>
> So, maybe you have to tweak the recording or playback parameters taking
> this into account.
>
> -giovanni
>
>
>
>
>
>>
>> Thanks,
>> Bala
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 gmail.com  Thu Jun 21 21:02:06 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 21 Jun 2012 19:02:06 +0200
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
	<CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
Message-ID: <CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>

On Thu, Jun 21, 2012 at 6:50 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:

> Is there a wiki link for these parameters? Thanks!!
>
> Actually it was very good till last week(for about 6 months), suddenly its
> broken. I could not trace it yet
>
>
Have you reinstalled something?

Re-compiled things, or updated config files?

If you have changed nothing, maybe the problem is elsewhere.

Just for test, you can:

1) from FS cli, unload mod_skypopen
2) from ssh killall Xvfb (this will kill all the skype instances too)
3) from ssh relaunch the Xvfb-skype clients (with the script)
4) from FS cli load mod_skypopen

check if it is recording correctly now.

That test is good if you changed nothing.

If you changed something, revert that change :).

-giovanni




> Thanks,
> Bala
>
>
> On Thu, Jun 21, 2012 at 9:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>
>>
>>
>> On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>
>>> All,
>>>
>>> Is there a way to control recording speed for mod_skypopen? I was trying
>>> to figure out the problem of fast recording/playing
>>>
>>> Please find the attached file
>>>
>>
>> Balamurugan,
>>
>> be aware that mod_skypopen operate at 16khz signed linear (let's say, at
>> HD audio level; double the number of samples of a normal PCMU or GSM or
>> g711 or G729 call).
>>
>> So, maybe you have to tweak the recording or playback parameters taking
>> this into account.
>>
>> -giovanni
>>
>>
>>
>>
>>
>>>
>>> Thanks,
>>> Bala
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From b2m at a-cti.com  Thu Jun 21 21:15:01 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Thu, 21 Jun 2012 22:45:01 +0530
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
	<CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
	<CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>
Message-ID: <CAP2fbbeCxNssmEXrmwxd9Z=ddz+CNEzASQ2QRpzispMPt9cjzA@mail.gmail.com>

Thanks for your help!!

Up to my knowledge there was no change.

After that I tried re-installed FS(with latest build) Skype and changed EC2
Instance(different size) but for some reason its still same. I guess some
settings somewhere changed not sure where it is.


Thanks,
Bala

On Thu, Jun 21, 2012 at 10:32 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

>
>
> On Thu, Jun 21, 2012 at 6:50 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>
>> Is there a wiki link for these parameters? Thanks!!
>>
>> Actually it was very good till last week(for about 6
>> months), suddenly its broken. I could not trace it yet
>>
>>
> Have you reinstalled something?
>
> Re-compiled things, or updated config files?
>
> If you have changed nothing, maybe the problem is elsewhere.
>
> Just for test, you can:
>
> 1) from FS cli, unload mod_skypopen
> 2) from ssh killall Xvfb (this will kill all the skype instances too)
> 3) from ssh relaunch the Xvfb-skype clients (with the script)
> 4) from FS cli load mod_skypopen
>
> check if it is recording correctly now.
>
> That test is good if you changed nothing.
>
> If you changed something, revert that change :).
>
> -giovanni
>
>
>
>
>> Thanks,
>> Bala
>>
>>
>> On Thu, Jun 21, 2012 at 9:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>>
>>>
>>>
>>> On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>>
>>>> All,
>>>>
>>>> Is there a way to control recording speed for mod_skypopen? I was
>>>> trying to figure out the problem of fast recording/playing
>>>>
>>>> Please find the attached file
>>>>
>>>
>>> Balamurugan,
>>>
>>> be aware that mod_skypopen operate at 16khz signed linear (let's say, at
>>> HD audio level; double the number of samples of a normal PCMU or GSM or
>>> g711 or G729 call).
>>>
>>> So, maybe you have to tweak the recording or playback parameters taking
>>> this into account.
>>>
>>> -giovanni
>>>
>>>
>>>
>>>
>>>
>>>>
>>>> Thanks,
>>>> Bala
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From b2m at a-cti.com  Thu Jun 21 21:17:36 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Thu, 21 Jun 2012 22:47:36 +0530
Subject: [Freeswitch-users] FreeSWITCH Installer for Debian
In-Reply-To: <CAKV-jFwgmV6Up2RRpe8CBNL9qUN=xV6oBHFjpWcCShvYwPxWXA@mail.gmail.com>
References: <CAKV-jFwgmV6Up2RRpe8CBNL9qUN=xV6oBHFjpWcCShvYwPxWXA@mail.gmail.com>
Message-ID: <CAP2fbbe84p96tgugP-bRPjdvjsrnEx42-ecR0GKn_9o6ijsoRQ@mail.gmail.com>

Is this outdated link?   files.endigovoip.com/freeswitch
/fs-debian-installer/fs-debian-installer_lastest.sh  (looks broken now)

Thanks,
Bala

On Fri, Mar 23, 2012 at 12:07 AM, Brian Foster <bdfoster at endigotech.com>wrote:

> Alright, so I've managed to put together a BASH script to install FS on
> top of Debian (Squeeze). I took the information on the wiki, along with
> some best practices and threw them all together for an automated installer.
> I came up with this really because I'm 'lazy'.
>
> I'm on version 0.0.2, and it's not considered extremely safe due to almost
> no error checking. I'd like to add some of that in. Plans are in the works
> to do some command line switches, put some options in so that you can
> customize certain things about the FS install, as well as having a guided
> install. The guided install would really be for the beginners, and would do
> such things as setting the hostname, setting timezone data, basically build
> FreeSWITCH from an absolutely clean install. When Ken's Debian packages
> come out for  FreeSWITCH, I plan on adding the ability to install from
> those packages as well as just installing from git head and (eventually)
> the 1.2 branch.
>
> I'd like to get some testers from the community and get some feedback on
> what they would like to see in this installer. It's still VERY basic, and
> that's a good thing! This way, I can add features from the get-go from
> users that are used to installing FS in a certain way.
>
>
> http://files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_latest.sh<< Current Version as of this writing is 0.0.2
>
> To start, do this:
> cd
> wget
> files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_lastest.sh-O fs-debian-installer.sh
> chmod +x fs-debian-installer.sh
> ./fs-debian-installer.sh
>
> Let me know what you think!
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 gmaruzz at gmail.com  Thu Jun 21 21:29:00 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 21 Jun 2012 19:29:00 +0200
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CAP2fbbeCxNssmEXrmwxd9Z=ddz+CNEzASQ2QRpzispMPt9cjzA@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
	<CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
	<CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>
	<CAP2fbbeCxNssmEXrmwxd9Z=ddz+CNEzASQ2QRpzispMPt9cjzA@mail.gmail.com>
Message-ID: <CALXCt0oLTuNuL4XO92BFwMUWCs+tp_fcJxjjtib2ffktdFMLCw@mail.gmail.com>

On Thu, Jun 21, 2012 at 7:15 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:

> Thanks for your help!!
>
> Up to my knowledge there was no change.
>
> After that I tried re-installed FS(with latest build) Skype and changed
> EC2 Instance(different size) but for some reason its still same. I guess
> some settings somewhere changed not sure where it is.
>
>
I just checked your mp3, it says it's recorded at 8khz, but it's not true.
It's 16khz. If you play it at 16khz it sounds ok. So, you have to look into
the mp3 converter you used and configure it to not lie about the samples
per second ;).

-giovanni





>
> Thanks,
> Bala
>
> On Thu, Jun 21, 2012 at 10:32 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>
>>
>>
>> On Thu, Jun 21, 2012 at 6:50 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>
>>> Is there a wiki link for these parameters? Thanks!!
>>>
>>> Actually it was very good till last week(for about 6
>>> months), suddenly its broken. I could not trace it yet
>>>
>>>
>> Have you reinstalled something?
>>
>> Re-compiled things, or updated config files?
>>
>> If you have changed nothing, maybe the problem is elsewhere.
>>
>> Just for test, you can:
>>
>> 1) from FS cli, unload mod_skypopen
>> 2) from ssh killall Xvfb (this will kill all the skype instances too)
>> 3) from ssh relaunch the Xvfb-skype clients (with the script)
>> 4) from FS cli load mod_skypopen
>>
>> check if it is recording correctly now.
>>
>> That test is good if you changed nothing.
>>
>> If you changed something, revert that change :).
>>
>> -giovanni
>>
>>
>>
>>
>>> Thanks,
>>> Bala
>>>
>>>
>>> On Thu, Jun 21, 2012 at 9:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>>>
>>>>
>>>>
>>>> On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>>>
>>>>> All,
>>>>>
>>>>> Is there a way to control recording speed for mod_skypopen? I was
>>>>> trying to figure out the problem of fast recording/playing
>>>>>
>>>>> Please find the attached file
>>>>>
>>>>
>>>> Balamurugan,
>>>>
>>>> be aware that mod_skypopen operate at 16khz signed linear (let's say,
>>>> at HD audio level; double the number of samples of a normal PCMU or GSM or
>>>> g711 or G729 call).
>>>>
>>>> So, maybe you have to tweak the recording or playback parameters taking
>>>> this into account.
>>>>
>>>> -giovanni
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>>
>>>>> Thanks,
>>>>> Bala
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From b2m at a-cti.com  Thu Jun 21 21:37:01 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Thu, 21 Jun 2012 23:07:01 +0530
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CALXCt0oLTuNuL4XO92BFwMUWCs+tp_fcJxjjtib2ffktdFMLCw@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
	<CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
	<CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>
	<CAP2fbbeCxNssmEXrmwxd9Z=ddz+CNEzASQ2QRpzispMPt9cjzA@mail.gmail.com>
	<CALXCt0oLTuNuL4XO92BFwMUWCs+tp_fcJxjjtib2ffktdFMLCw@mail.gmail.com>
Message-ID: <CAP2fbbfXAk9WKwKFQ0_iOx8-WEYUqbLw9zrhvpZs1Wo96oQjZA@mail.gmail.com>

I am afraid, I did not do anything outside freeswitch.

just have the following for recording in my dialplan


    <action application="set" data="RECORD_COPYRIGHT=(c) freeswitch"/>
    <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
    <action application="set" data="RECORD_ARTIST=auto"/>
    <action application="set" data="RECORD_COMMENT=auto recording"/>
    <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d
%H:%M)}"/>
    <action application="set" data="RECORD_STEREO=true"/>
    <action application="set" data="record_sample_rate=11000"/>
    <action application="set" data="media_bug_answer_req=true"/>
    <action application="record_session"
data="$${base_dir}/recordings/${uuid}.mp3"/>
    <action application="set" data="ignore_early_media=false"/>
    <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,ORIGINATOR_CANCEL,CANCEL,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>


Thanks,
Bala


On Thu, Jun 21, 2012 at 10:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> On Thu, Jun 21, 2012 at 7:15 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>
>> Thanks for your help!!
>>
>> Up to my knowledge there was no change.
>>
>> After that I tried re-installed FS(with latest build) Skype and changed
>> EC2 Instance(different size) but for some reason its still same. I guess
>> some settings somewhere changed not sure where it is.
>>
>>
> I just checked your mp3, it says it's recorded at 8khz, but it's not true.
> It's 16khz. If you play it at 16khz it sounds ok. So, you have to look into
> the mp3 converter you used and configure it to not lie about the samples
> per second ;).
>
> -giovanni
>
>
>
>
>
>>
>> Thanks,
>> Bala
>>
>> On Thu, Jun 21, 2012 at 10:32 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>>
>>>
>>>
>>> On Thu, Jun 21, 2012 at 6:50 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>>
>>>> Is there a wiki link for these parameters? Thanks!!
>>>>
>>>> Actually it was very good till last week(for about 6
>>>> months), suddenly its broken. I could not trace it yet
>>>>
>>>>
>>> Have you reinstalled something?
>>>
>>> Re-compiled things, or updated config files?
>>>
>>> If you have changed nothing, maybe the problem is elsewhere.
>>>
>>> Just for test, you can:
>>>
>>> 1) from FS cli, unload mod_skypopen
>>> 2) from ssh killall Xvfb (this will kill all the skype instances too)
>>> 3) from ssh relaunch the Xvfb-skype clients (with the script)
>>> 4) from FS cli load mod_skypopen
>>>
>>> check if it is recording correctly now.
>>>
>>> That test is good if you changed nothing.
>>>
>>> If you changed something, revert that change :).
>>>
>>> -giovanni
>>>
>>>
>>>
>>>
>>>> Thanks,
>>>> Bala
>>>>
>>>>
>>>> On Thu, Jun 21, 2012 at 9:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com
>>>> > wrote:
>>>>
>>>>>
>>>>>
>>>>> On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>>>>
>>>>>> All,
>>>>>>
>>>>>> Is there a way to control recording speed for mod_skypopen? I was
>>>>>> trying to figure out the problem of fast recording/playing
>>>>>>
>>>>>> Please find the attached file
>>>>>>
>>>>>
>>>>> Balamurugan,
>>>>>
>>>>> be aware that mod_skypopen operate at 16khz signed linear (let's say,
>>>>> at HD audio level; double the number of samples of a normal PCMU or GSM or
>>>>> g711 or G729 call).
>>>>>
>>>>> So, maybe you have to tweak the recording or playback parameters
>>>>> taking this into account.
>>>>>
>>>>> -giovanni
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>>
>>>>>> Thanks,
>>>>>> Bala
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From b2m at a-cti.com  Thu Jun 21 21:45:51 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Thu, 21 Jun 2012 23:15:51 +0530
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CAP2fbbfXAk9WKwKFQ0_iOx8-WEYUqbLw9zrhvpZs1Wo96oQjZA@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
	<CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
	<CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>
	<CAP2fbbeCxNssmEXrmwxd9Z=ddz+CNEzASQ2QRpzispMPt9cjzA@mail.gmail.com>
	<CALXCt0oLTuNuL4XO92BFwMUWCs+tp_fcJxjjtib2ffktdFMLCw@mail.gmail.com>
	<CAP2fbbfXAk9WKwKFQ0_iOx8-WEYUqbLw9zrhvpZs1Wo96oQjZA@mail.gmail.com>
Message-ID: <CAP2fbbfYdUrE=GYgVH5WwcbgW7s4=_f+81=DM=aK8dpYJ+kQnw@mail.gmail.com>

I just changed  <action application="set" data="record_sample_rate=16000"/>,
still looks same.


Thanks,
Bala


On Thu, Jun 21, 2012 at 11:07 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:

> I am afraid, I did not do anything outside freeswitch.
>
> just have the following for recording in my dialplan
>
>
>     <action application="set" data="RECORD_COPYRIGHT=(c) freeswitch"/>
>     <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
>     <action application="set" data="RECORD_ARTIST=auto"/>
>     <action application="set" data="RECORD_COMMENT=auto recording"/>
>     <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d
> %H:%M)}"/>
>     <action application="set" data="RECORD_STEREO=true"/>
>     <action application="set" data="record_sample_rate=11000"/>
>     <action application="set" data="media_bug_answer_req=true"/>
>     <action application="record_session"
> data="$${base_dir}/recordings/${uuid}.mp3"/>
>     <action application="set" data="ignore_early_media=false"/>
>     <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,ORIGINATOR_CANCEL,CANCEL,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>
>
>
> Thanks,
> Bala
>
>
> On Thu, Jun 21, 2012 at 10:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>
>> On Thu, Jun 21, 2012 at 7:15 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>
>>> Thanks for your help!!
>>>
>>> Up to my knowledge there was no change.
>>>
>>> After that I tried re-installed FS(with latest build) Skype and changed
>>> EC2 Instance(different size) but for some reason its still same. I guess
>>> some settings somewhere changed not sure where it is.
>>>
>>>
>> I just checked your mp3, it says it's recorded at 8khz, but it's not
>> true. It's 16khz. If you play it at 16khz it sounds ok. So, you have to
>> look into the mp3 converter you used and configure it to not lie about the
>> samples per second ;).
>>
>> -giovanni
>>
>>
>>
>>
>>
>>>
>>> Thanks,
>>> Bala
>>>
>>> On Thu, Jun 21, 2012 at 10:32 PM, Giovanni Maruzzelli <gmaruzz at gmail.com
>>> > wrote:
>>>
>>>>
>>>>
>>>> On Thu, Jun 21, 2012 at 6:50 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>>>
>>>>> Is there a wiki link for these parameters? Thanks!!
>>>>>
>>>>> Actually it was very good till last week(for about 6
>>>>> months), suddenly its broken. I could not trace it yet
>>>>>
>>>>>
>>>> Have you reinstalled something?
>>>>
>>>> Re-compiled things, or updated config files?
>>>>
>>>> If you have changed nothing, maybe the problem is elsewhere.
>>>>
>>>> Just for test, you can:
>>>>
>>>> 1) from FS cli, unload mod_skypopen
>>>> 2) from ssh killall Xvfb (this will kill all the skype instances too)
>>>> 3) from ssh relaunch the Xvfb-skype clients (with the script)
>>>> 4) from FS cli load mod_skypopen
>>>>
>>>> check if it is recording correctly now.
>>>>
>>>> That test is good if you changed nothing.
>>>>
>>>> If you changed something, revert that change :).
>>>>
>>>> -giovanni
>>>>
>>>>
>>>>
>>>>
>>>>> Thanks,
>>>>> Bala
>>>>>
>>>>>
>>>>> On Thu, Jun 21, 2012 at 9:59 PM, Giovanni Maruzzelli <
>>>>> gmaruzz at gmail.com> wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran <b2m at a-cti.com
>>>>>> > wrote:
>>>>>>
>>>>>>> All,
>>>>>>>
>>>>>>> Is there a way to control recording speed for mod_skypopen? I was
>>>>>>> trying to figure out the problem of fast recording/playing
>>>>>>>
>>>>>>> Please find the attached file
>>>>>>>
>>>>>>
>>>>>> Balamurugan,
>>>>>>
>>>>>> be aware that mod_skypopen operate at 16khz signed linear (let's say,
>>>>>> at HD audio level; double the number of samples of a normal PCMU or GSM or
>>>>>> g711 or G729 call).
>>>>>>
>>>>>> So, maybe you have to tweak the recording or playback parameters
>>>>>> taking this into account.
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>> Thanks,
>>>>>>> Bala
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at 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
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From covici at ccs.covici.com  Thu Jun 21 21:54:28 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 21 Jun 2012 13:54:28 -0400
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CAP2fbbfYdUrE=GYgVH5WwcbgW7s4=_f+81=DM=aK8dpYJ+kQnw@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
	<CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
	<CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>
	<CAP2fbbeCxNssmEXrmwxd9Z=ddz+CNEzASQ2QRpzispMPt9cjzA@mail.gmail.com>
	<CALXCt0oLTuNuL4XO92BFwMUWCs+tp_fcJxjjtib2ffktdFMLCw@mail.gmail.com>
	<CAP2fbbfXAk9WKwKFQ0_iOx8-WEYUqbLw9zrhvpZs1Wo96oQjZA@mail.gmail.com>
	<CAP2fbbfYdUrE=GYgVH5WwcbgW7s4=_f+81=DM=aK8dpYJ+kQnw@mail.gmail.com>
Message-ID: <9123.1340301268@ccs.covici.com>

How did you convert to mp3 -- fs normally records .wav at whatever
sampling rate determined by the codec -- I did not know that record
parameter did anything.

Balamurugan Mahendran <b2m at a-cti.com> wrote:

> I just changed  <action application="set" data="record_sample_rate=16000"/>,
> still looks same.
> 
> 
> Thanks,
> Bala
> 
> 
> On Thu, Jun 21, 2012 at 11:07 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
> 
> > I am afraid, I did not do anything outside freeswitch.
> >
> > just have the following for recording in my dialplan
> >
> >
> >     <action application="set" data="RECORD_COPYRIGHT=(c) freeswitch"/>
> >     <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
> >     <action application="set" data="RECORD_ARTIST=auto"/>
> >     <action application="set" data="RECORD_COMMENT=auto recording"/>
> >     <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d
> > %H:%M)}"/>
> >     <action application="set" data="RECORD_STEREO=true"/>
> >     <action application="set" data="record_sample_rate=11000"/>
> >     <action application="set" data="media_bug_answer_req=true"/>
> >     <action application="record_session"
> > data="$${base_dir}/recordings/${uuid}.mp3"/>
> >     <action application="set" data="ignore_early_media=false"/>
> >     <action application="set"
> > data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,ORIGINATOR_CANCEL,CANCEL,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>
> >
> >
> > Thanks,
> > Bala
> >
> >
> > On Thu, Jun 21, 2012 at 10:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
> >
> >> On Thu, Jun 21, 2012 at 7:15 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
> >>
> >>> Thanks for your help!!
> >>>
> >>> Up to my knowledge there was no change.
> >>>
> >>> After that I tried re-installed FS(with latest build) Skype and changed
> >>> EC2 Instance(different size) but for some reason its still same. I guess
> >>> some settings somewhere changed not sure where it is.
> >>>
> >>>
> >> I just checked your mp3, it says it's recorded at 8khz, but it's not
> >> true. It's 16khz. If you play it at 16khz it sounds ok. So, you have to
> >> look into the mp3 converter you used and configure it to not lie about the
> >> samples per second ;).
> >>
> >> -giovanni
> >>
> >>
> >>
> >>
> >>
> >>>
> >>> Thanks,
> >>> Bala
> >>>
> >>> On Thu, Jun 21, 2012 at 10:32 PM, Giovanni Maruzzelli <gmaruzz at gmail.com
> >>> > wrote:
> >>>
> >>>>
> >>>>
> >>>> On Thu, Jun 21, 2012 at 6:50 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
> >>>>
> >>>>> Is there a wiki link for these parameters? Thanks!!
> >>>>>
> >>>>> Actually it was very good till last week(for about 6
> >>>>> months), suddenly its broken. I could not trace it yet
> >>>>>
> >>>>>
> >>>> Have you reinstalled something?
> >>>>
> >>>> Re-compiled things, or updated config files?
> >>>>
> >>>> If you have changed nothing, maybe the problem is elsewhere.
> >>>>
> >>>> Just for test, you can:
> >>>>
> >>>> 1) from FS cli, unload mod_skypopen
> >>>> 2) from ssh killall Xvfb (this will kill all the skype instances too)
> >>>> 3) from ssh relaunch the Xvfb-skype clients (with the script)
> >>>> 4) from FS cli load mod_skypopen
> >>>>
> >>>> check if it is recording correctly now.
> >>>>
> >>>> That test is good if you changed nothing.
> >>>>
> >>>> If you changed something, revert that change :).
> >>>>
> >>>> -giovanni
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>> Thanks,
> >>>>> Bala
> >>>>>
> >>>>>
> >>>>> On Thu, Jun 21, 2012 at 9:59 PM, Giovanni Maruzzelli <
> >>>>> gmaruzz at gmail.com> wrote:
> >>>>>
> >>>>>>
> >>>>>>
> >>>>>> On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran <b2m at a-cti.com
> >>>>>> > wrote:
> >>>>>>
> >>>>>>> All,
> >>>>>>>
> >>>>>>> Is there a way to control recording speed for mod_skypopen? I was
> >>>>>>> trying to figure out the problem of fast recording/playing
> >>>>>>>
> >>>>>>> Please find the attached file
> >>>>>>>
> >>>>>>
> >>>>>> Balamurugan,
> >>>>>>
> >>>>>> be aware that mod_skypopen operate at 16khz signed linear (let's say,
> >>>>>> at HD audio level; double the number of samples of a normal PCMU or GSM or
> >>>>>> g711 or G729 call).
> >>>>>>
> >>>>>> So, maybe you have to tweak the recording or playback parameters
> >>>>>> taking this into account.
> >>>>>>
> >>>>>> -giovanni
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>
> >>>>>>> Thanks,
> >>>>>>> Bala
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>> _________________________________________________________________________
> >>>>>>> Professional FreeSWITCH Consulting Services:
> >>>>>>> consulting at freeswitch.org
> >>>>>>> http://www.freeswitchsolutions.com
> >>>>>>>
> >>>>>>> 
> >>>>>>> 
> >>>>>>>
> >>>>>>> Official FreeSWITCH Sites
> >>>>>>> http://www.freeswitch.org
> >>>>>>> http://wiki.freeswitch.org
> >>>>>>> http://www.cluecon.com
> >>>>>>>
> >>>>>>> Join Us At ClueCon - Aug 7-9, 2012
> >>>>>>>
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> FreeSWITCH-users at 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
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> _________________________________________________________________________
> >>>>>> Professional FreeSWITCH Consulting Services:
> >>>>>> consulting at freeswitch.org
> >>>>>> http://www.freeswitchsolutions.com
> >>>>>>
> >>>>>> 
> >>>>>> 
> >>>>>>
> >>>>>> Official FreeSWITCH Sites
> >>>>>> http://www.freeswitch.org
> >>>>>> http://wiki.freeswitch.org
> >>>>>> http://www.cluecon.com
> >>>>>>
> >>>>>> Join Us At ClueCon - Aug 7-9, 2012
> >>>>>>
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>> UNSUBSCRIBE:
> >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>> http://www.freeswitch.org
> >>>>>>
> >>>>>>
> >>>>>
> >>>>>
> >>>>> _________________________________________________________________________
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>>
> >>>>> 
> >>>>> 
> >>>>>
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://wiki.freeswitch.org
> >>>>> http://www.cluecon.com
> >>>>>
> >>>>> Join Us At ClueCon - Aug 7-9, 2012
> >>>>>
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at 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
> >>>>
> >>>>
> >>>>
> >>>> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> Join Us At ClueCon - Aug 7-9, 2012
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>
> >>> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> Join Us At ClueCon - Aug 7-9, 2012
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at 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
> >>
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From gmaruzz at gmail.com  Thu Jun 21 21:55:33 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 21 Jun 2012 19:55:33 +0200
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CAP2fbbfYdUrE=GYgVH5WwcbgW7s4=_f+81=DM=aK8dpYJ+kQnw@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
	<CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
	<CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>
	<CAP2fbbeCxNssmEXrmwxd9Z=ddz+CNEzASQ2QRpzispMPt9cjzA@mail.gmail.com>
	<CALXCt0oLTuNuL4XO92BFwMUWCs+tp_fcJxjjtib2ffktdFMLCw@mail.gmail.com>
	<CAP2fbbfXAk9WKwKFQ0_iOx8-WEYUqbLw9zrhvpZs1Wo96oQjZA@mail.gmail.com>
	<CAP2fbbfYdUrE=GYgVH5WwcbgW7s4=_f+81=DM=aK8dpYJ+kQnw@mail.gmail.com>
Message-ID: <CALXCt0pad5LY3N9ecsAzjdmqZRPMqCeQdBwbzhoQLUY73T22eA@mail.gmail.com>

On Thu, Jun 21, 2012 at 7:45 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:

> I just changed  <action application="set" data="record_sample_rate=16000"/>,
> still looks same.
>
>
Just for the sake of it, try with record_sample_rate=8000 and check how it
is... Anyway, problem is there, somewhere.

Maybe you can post again to the mailing list putting: "recording problem"
in the subject (eg: without naming mod_skypopen), so maybe someone can
chime in.

I cannot check it out right now, if You have not solved in 12 hours, write
again, I'll look into that.

-giovanni



>
> Thanks,
> Bala
>
>
> On Thu, Jun 21, 2012 at 11:07 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>
>> I am afraid, I did not do anything outside freeswitch.
>>
>> just have the following for recording in my dialplan
>>
>>
>>     <action application="set" data="RECORD_COPYRIGHT=(c) freeswitch"/>
>>     <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
>>     <action application="set" data="RECORD_ARTIST=auto"/>
>>     <action application="set" data="RECORD_COMMENT=auto recording"/>
>>     <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d
>> %H:%M)}"/>
>>     <action application="set" data="RECORD_STEREO=true"/>
>>     <action application="set" data="record_sample_rate=11000"/>
>>     <action application="set" data="media_bug_answer_req=true"/>
>>     <action application="record_session"
>> data="$${base_dir}/recordings/${uuid}.mp3"/>
>>     <action application="set" data="ignore_early_media=false"/>
>>     <action application="set"
>> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,ORIGINATOR_CANCEL,CANCEL,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>
>>
>>
>> Thanks,
>> Bala
>>
>>
>> On Thu, Jun 21, 2012 at 10:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>>
>>> On Thu, Jun 21, 2012 at 7:15 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>>
>>>> Thanks for your help!!
>>>>
>>>> Up to my knowledge there was no change.
>>>>
>>>> After that I tried re-installed FS(with latest build) Skype and changed
>>>> EC2 Instance(different size) but for some reason its still same. I guess
>>>> some settings somewhere changed not sure where it is.
>>>>
>>>>
>>> I just checked your mp3, it says it's recorded at 8khz, but it's not
>>> true. It's 16khz. If you play it at 16khz it sounds ok. So, you have to
>>> look into the mp3 converter you used and configure it to not lie about the
>>> samples per second ;).
>>>
>>> -giovanni
>>>
>>>
>>>
>>>
>>>
>>>>
>>>> Thanks,
>>>> Bala
>>>>
>>>> On Thu, Jun 21, 2012 at 10:32 PM, Giovanni Maruzzelli <
>>>> gmaruzz at gmail.com> wrote:
>>>>
>>>>>
>>>>>
>>>>> On Thu, Jun 21, 2012 at 6:50 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>>>>
>>>>>> Is there a wiki link for these parameters? Thanks!!
>>>>>>
>>>>>> Actually it was very good till last week(for about 6
>>>>>> months), suddenly its broken. I could not trace it yet
>>>>>>
>>>>>>
>>>>> Have you reinstalled something?
>>>>>
>>>>> Re-compiled things, or updated config files?
>>>>>
>>>>> If you have changed nothing, maybe the problem is elsewhere.
>>>>>
>>>>> Just for test, you can:
>>>>>
>>>>> 1) from FS cli, unload mod_skypopen
>>>>> 2) from ssh killall Xvfb (this will kill all the skype instances too)
>>>>> 3) from ssh relaunch the Xvfb-skype clients (with the script)
>>>>> 4) from FS cli load mod_skypopen
>>>>>
>>>>> check if it is recording correctly now.
>>>>>
>>>>> That test is good if you changed nothing.
>>>>>
>>>>> If you changed something, revert that change :).
>>>>>
>>>>> -giovanni
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>> Thanks,
>>>>>> Bala
>>>>>>
>>>>>>
>>>>>> On Thu, Jun 21, 2012 at 9:59 PM, Giovanni Maruzzelli <
>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran <
>>>>>>> b2m at a-cti.com> wrote:
>>>>>>>
>>>>>>>> All,
>>>>>>>>
>>>>>>>> Is there a way to control recording speed for mod_skypopen? I was
>>>>>>>> trying to figure out the problem of fast recording/playing
>>>>>>>>
>>>>>>>> Please find the attached file
>>>>>>>>
>>>>>>>
>>>>>>> Balamurugan,
>>>>>>>
>>>>>>> be aware that mod_skypopen operate at 16khz signed linear (let's
>>>>>>> say, at HD audio level; double the number of samples of a normal PCMU or
>>>>>>> GSM or g711 or G729 call).
>>>>>>>
>>>>>>> So, maybe you have to tweak the recording or playback parameters
>>>>>>> taking this into account.
>>>>>>>
>>>>>>> -giovanni
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>> Thanks,
>>>>>>>> Bala
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at 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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From b2m at a-cti.com  Thu Jun 21 22:01:14 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Thu, 21 Jun 2012 23:31:14 +0530
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CALXCt0pad5LY3N9ecsAzjdmqZRPMqCeQdBwbzhoQLUY73T22eA@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
	<CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
	<CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>
	<CAP2fbbeCxNssmEXrmwxd9Z=ddz+CNEzASQ2QRpzispMPt9cjzA@mail.gmail.com>
	<CALXCt0oLTuNuL4XO92BFwMUWCs+tp_fcJxjjtib2ffktdFMLCw@mail.gmail.com>
	<CAP2fbbfXAk9WKwKFQ0_iOx8-WEYUqbLw9zrhvpZs1Wo96oQjZA@mail.gmail.com>
	<CAP2fbbfYdUrE=GYgVH5WwcbgW7s4=_f+81=DM=aK8dpYJ+kQnw@mail.gmail.com>
	<CALXCt0pad5LY3N9ecsAzjdmqZRPMqCeQdBwbzhoQLUY73T22eA@mail.gmail.com>
Message-ID: <CAP2fbbdq=LuzVNJGJJH-7yw1dxraLw00M-91-99BVg6NQfU-WA@mail.gmail.com>

Thanks for your help!!

This is happening only for Skype calls, all other endpoint call recordings
are very fine.

Thanks,
Bala


On Thu, Jun 21, 2012 at 11:25 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> On Thu, Jun 21, 2012 at 7:45 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>
>> I just changed  <action application="set" data="record_sample_rate=16000"/>,
>> still looks same.
>>
>>
> Just for the sake of it, try with record_sample_rate=8000 and check how
> it is... Anyway, problem is there, somewhere.
>
> Maybe you can post again to the mailing list putting: "recording problem"
> in the subject (eg: without naming mod_skypopen), so maybe someone can
> chime in.
>
> I cannot check it out right now, if You have not solved in 12 hours, write
> again, I'll look into that.
>
> -giovanni
>
>
>
>>
>> Thanks,
>> Bala
>>
>>
>> On Thu, Jun 21, 2012 at 11:07 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>
>>> I am afraid, I did not do anything outside freeswitch.
>>>
>>> just have the following for recording in my dialplan
>>>
>>>
>>>     <action application="set" data="RECORD_COPYRIGHT=(c) freeswitch"/>
>>>     <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
>>>     <action application="set" data="RECORD_ARTIST=auto"/>
>>>     <action application="set" data="RECORD_COMMENT=auto recording"/>
>>>     <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d
>>> %H:%M)}"/>
>>>     <action application="set" data="RECORD_STEREO=true"/>
>>>     <action application="set" data="record_sample_rate=11000"/>
>>>     <action application="set" data="media_bug_answer_req=true"/>
>>>     <action application="record_session"
>>> data="$${base_dir}/recordings/${uuid}.mp3"/>
>>>     <action application="set" data="ignore_early_media=false"/>
>>>     <action application="set"
>>> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,ORIGINATOR_CANCEL,CANCEL,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>
>>>
>>>
>>> Thanks,
>>> Bala
>>>
>>>
>>> On Thu, Jun 21, 2012 at 10:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com
>>> > wrote:
>>>
>>>> On Thu, Jun 21, 2012 at 7:15 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>>>
>>>>> Thanks for your help!!
>>>>>
>>>>> Up to my knowledge there was no change.
>>>>>
>>>>> After that I tried re-installed FS(with latest build) Skype and
>>>>> changed EC2 Instance(different size) but for some reason its still same. I
>>>>> guess some settings somewhere changed not sure where it is.
>>>>>
>>>>>
>>>> I just checked your mp3, it says it's recorded at 8khz, but it's not
>>>> true. It's 16khz. If you play it at 16khz it sounds ok. So, you have to
>>>> look into the mp3 converter you used and configure it to not lie about the
>>>> samples per second ;).
>>>>
>>>> -giovanni
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>>
>>>>> Thanks,
>>>>> Bala
>>>>>
>>>>> On Thu, Jun 21, 2012 at 10:32 PM, Giovanni Maruzzelli <
>>>>> gmaruzz at gmail.com> wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Jun 21, 2012 at 6:50 PM, Balamurugan Mahendran <b2m at a-cti.com
>>>>>> > wrote:
>>>>>>
>>>>>>> Is there a wiki link for these parameters? Thanks!!
>>>>>>>
>>>>>>> Actually it was very good till last week(for about 6
>>>>>>> months), suddenly its broken. I could not trace it yet
>>>>>>>
>>>>>>>
>>>>>> Have you reinstalled something?
>>>>>>
>>>>>> Re-compiled things, or updated config files?
>>>>>>
>>>>>> If you have changed nothing, maybe the problem is elsewhere.
>>>>>>
>>>>>> Just for test, you can:
>>>>>>
>>>>>> 1) from FS cli, unload mod_skypopen
>>>>>> 2) from ssh killall Xvfb (this will kill all the skype instances too)
>>>>>> 3) from ssh relaunch the Xvfb-skype clients (with the script)
>>>>>> 4) from FS cli load mod_skypopen
>>>>>>
>>>>>> check if it is recording correctly now.
>>>>>>
>>>>>> That test is good if you changed nothing.
>>>>>>
>>>>>> If you changed something, revert that change :).
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>> Thanks,
>>>>>>> Bala
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Jun 21, 2012 at 9:59 PM, Giovanni Maruzzelli <
>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran <
>>>>>>>> b2m at a-cti.com> wrote:
>>>>>>>>
>>>>>>>>> All,
>>>>>>>>>
>>>>>>>>> Is there a way to control recording speed for mod_skypopen? I was
>>>>>>>>> trying to figure out the problem of fast recording/playing
>>>>>>>>>
>>>>>>>>> Please find the attached file
>>>>>>>>>
>>>>>>>>
>>>>>>>> Balamurugan,
>>>>>>>>
>>>>>>>> be aware that mod_skypopen operate at 16khz signed linear (let's
>>>>>>>> say, at HD audio level; double the number of samples of a normal PCMU or
>>>>>>>> GSM or g711 or G729 call).
>>>>>>>>
>>>>>>>> So, maybe you have to tweak the recording or playback parameters
>>>>>>>> taking this into account.
>>>>>>>>
>>>>>>>> -giovanni
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>>
>>>>>>>>> Thanks,
>>>>>>>>> Bala
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at 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
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at 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
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Thu Jun 21 22:42:13 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 21 Jun 2012 14:42:13 -0400
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CAP2fbbdq=LuzVNJGJJH-7yw1dxraLw00M-91-99BVg6NQfU-WA@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
	<CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
	<CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>
	<CAP2fbbeCxNssmEXrmwxd9Z=ddz+CNEzASQ2QRpzispMPt9cjzA@mail.gmail.com>
	<CALXCt0oLTuNuL4XO92BFwMUWCs+tp_fcJxjjtib2ffktdFMLCw@mail.gmail.com>
	<CAP2fbbfXAk9WKwKFQ0_iOx8-WEYUqbLw9zrhvpZs1Wo96oQjZA@mail.gmail.com>
	<CAP2fbbfYdUrE=GYgVH5WwcbgW7s4=_f+81=DM=aK8dpYJ+kQnw@mail.gmail.com>
	<CALXCt0pad5LY3N9ecsAzjdmqZRPMqCeQdBwbzhoQLUY73T22eA@mail.gmail.com>
	<CAP2fbbdq=LuzVNJGJJH-7yw1dxraLw00M-91-99BVg6NQfU-WA@mail.gmail.com>
Message-ID: <CAKV-jFxr_cMcm-sq_2oQkqiNywpruDwNsETH-G1UCe-jniVHDg@mail.gmail.com>

It's an issue outside of mod_skypopen, which although its helpful to
mention in the body of your email, mentioning it in the subject might scare
off potential users/devs who might know what the problem is.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 21, 2012 2:02 PM, "Balamurugan Mahendran" <b2m at a-cti.com> wrote:

> Thanks for your help!!
>
> This is happening only for Skype calls, all other endpoint call recordings
> are very fine.
>
> Thanks,
> Bala
>
>
> On Thu, Jun 21, 2012 at 11:25 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>
>> On Thu, Jun 21, 2012 at 7:45 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>
>>> I just changed  <action application="set" data="record_sample_rate=16000"/>,
>>> still looks same.
>>>
>>>
>> Just for the sake of it, try with record_sample_rate=8000 and check how
>> it is... Anyway, problem is there, somewhere.
>>
>> Maybe you can post again to the mailing list putting: "recording problem"
>> in the subject (eg: without naming mod_skypopen), so maybe someone can
>> chime in.
>>
>> I cannot check it out right now, if You have not solved in 12 hours,
>> write again, I'll look into that.
>>
>> -giovanni
>>
>>
>>
>>>
>>> Thanks,
>>> Bala
>>>
>>>
>>> On Thu, Jun 21, 2012 at 11:07 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>>
>>>> I am afraid, I did not do anything outside freeswitch.
>>>>
>>>> just have the following for recording in my dialplan
>>>>
>>>>
>>>>     <action application="set" data="RECORD_COPYRIGHT=(c) freeswitch"/>
>>>>     <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
>>>>     <action application="set" data="RECORD_ARTIST=auto"/>
>>>>     <action application="set" data="RECORD_COMMENT=auto recording"/>
>>>>     <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d
>>>> %H:%M)}"/>
>>>>     <action application="set" data="RECORD_STEREO=true"/>
>>>>     <action application="set" data="record_sample_rate=11000"/>
>>>>     <action application="set" data="media_bug_answer_req=true"/>
>>>>     <action application="record_session"
>>>> data="$${base_dir}/recordings/${uuid}.mp3"/>
>>>>     <action application="set" data="ignore_early_media=false"/>
>>>>     <action application="set"
>>>> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,ORIGINATOR_CANCEL,CANCEL,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>
>>>>
>>>>
>>>> Thanks,
>>>> Bala
>>>>
>>>>
>>>> On Thu, Jun 21, 2012 at 10:59 PM, Giovanni Maruzzelli <
>>>> gmaruzz at gmail.com> wrote:
>>>>
>>>>> On Thu, Jun 21, 2012 at 7:15 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>>>>
>>>>>> Thanks for your help!!
>>>>>>
>>>>>> Up to my knowledge there was no change.
>>>>>>
>>>>>> After that I tried re-installed FS(with latest build) Skype and
>>>>>> changed EC2 Instance(different size) but for some reason its still same. I
>>>>>> guess some settings somewhere changed not sure where it is.
>>>>>>
>>>>>>
>>>>> I just checked your mp3, it says it's recorded at 8khz, but it's not
>>>>> true. It's 16khz. If you play it at 16khz it sounds ok. So, you have to
>>>>> look into the mp3 converter you used and configure it to not lie about the
>>>>> samples per second ;).
>>>>>
>>>>> -giovanni
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>>
>>>>>> Thanks,
>>>>>> Bala
>>>>>>
>>>>>> On Thu, Jun 21, 2012 at 10:32 PM, Giovanni Maruzzelli <
>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Jun 21, 2012 at 6:50 PM, Balamurugan Mahendran <
>>>>>>> b2m at a-cti.com> wrote:
>>>>>>>
>>>>>>>> Is there a wiki link for these parameters? Thanks!!
>>>>>>>>
>>>>>>>> Actually it was very good till last week(for about 6
>>>>>>>> months), suddenly its broken. I could not trace it yet
>>>>>>>>
>>>>>>>>
>>>>>>> Have you reinstalled something?
>>>>>>>
>>>>>>> Re-compiled things, or updated config files?
>>>>>>>
>>>>>>> If you have changed nothing, maybe the problem is elsewhere.
>>>>>>>
>>>>>>> Just for test, you can:
>>>>>>>
>>>>>>> 1) from FS cli, unload mod_skypopen
>>>>>>> 2) from ssh killall Xvfb (this will kill all the skype instances too)
>>>>>>> 3) from ssh relaunch the Xvfb-skype clients (with the script)
>>>>>>> 4) from FS cli load mod_skypopen
>>>>>>>
>>>>>>> check if it is recording correctly now.
>>>>>>>
>>>>>>> That test is good if you changed nothing.
>>>>>>>
>>>>>>> If you changed something, revert that change :).
>>>>>>>
>>>>>>> -giovanni
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> Thanks,
>>>>>>>> Bala
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, Jun 21, 2012 at 9:59 PM, Giovanni Maruzzelli <
>>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran <
>>>>>>>>> b2m at a-cti.com> wrote:
>>>>>>>>>
>>>>>>>>>> All,
>>>>>>>>>>
>>>>>>>>>> Is there a way to control recording speed for mod_skypopen? I was
>>>>>>>>>> trying to figure out the problem of fast recording/playing
>>>>>>>>>>
>>>>>>>>>> Please find the attached file
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Balamurugan,
>>>>>>>>>
>>>>>>>>> be aware that mod_skypopen operate at 16khz signed linear (let's
>>>>>>>>> say, at HD audio level; double the number of samples of a normal PCMU or
>>>>>>>>> GSM or g711 or G729 call).
>>>>>>>>>
>>>>>>>>> So, maybe you have to tweak the recording or playback parameters
>>>>>>>>> taking this into account.
>>>>>>>>>
>>>>>>>>> -giovanni
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Thanks,
>>>>>>>>>> Bala
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.com
>>>>>>>>>>
>>>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at 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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at 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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Thu Jun 21 22:43:28 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 21 Jun 2012 14:43:28 -0400
Subject: [Freeswitch-users] FreeSWITCH Installer for Debian
In-Reply-To: <CAP2fbbe84p96tgugP-bRPjdvjsrnEx42-ecR0GKn_9o6ijsoRQ@mail.gmail.com>
References: <CAKV-jFwgmV6Up2RRpe8CBNL9qUN=xV6oBHFjpWcCShvYwPxWXA@mail.gmail.com>
	<CAP2fbbe84p96tgugP-bRPjdvjsrnEx42-ecR0GKn_9o6ijsoRQ@mail.gmail.com>
Message-ID: <CAKV-jFxB3OEfZFZS9BY=KvNnD-Uwc49acrhf+j3i5sOMzq6Zqg@mail.gmail.com>

It's broken, didn't seem like there was much interest in this so I only use
it internally.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 21, 2012 1:19 PM, "Balamurugan Mahendran" <b2m at a-cti.com> wrote:

> Is this outdated link?   files.endigovoip.com/freeswitch
> /fs-debian-installer/fs-debian-installer_lastest.sh  (looks broken now)
>
> Thanks,
> Bala
>
> On Fri, Mar 23, 2012 at 12:07 AM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> Alright, so I've managed to put together a BASH script to install FS on
>> top of Debian (Squeeze). I took the information on the wiki, along with
>> some best practices and threw them all together for an automated installer.
>> I came up with this really because I'm 'lazy'.
>>
>> I'm on version 0.0.2, and it's not considered extremely safe due to
>> almost no error checking. I'd like to add some of that in. Plans are in the
>> works to do some command line switches, put some options in so that you can
>> customize certain things about the FS install, as well as having a guided
>> install. The guided install would really be for the beginners, and would do
>> such things as setting the hostname, setting timezone data, basically build
>> FreeSWITCH from an absolutely clean install. When Ken's Debian packages
>> come out for  FreeSWITCH, I plan on adding the ability to install from
>> those packages as well as just installing from git head and (eventually)
>> the 1.2 branch.
>>
>> I'd like to get some testers from the community and get some feedback on
>> what they would like to see in this installer. It's still VERY basic, and
>> that's a good thing! This way, I can add features from the get-go from
>> users that are used to installing FS in a certain way.
>>
>>
>> http://files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_latest.sh<< Current Version as of this writing is 0.0.2
>>
>> To start, do this:
>> cd
>> wget
>> files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_lastest.sh-O fs-debian-installer.sh
>> chmod +x fs-debian-installer.sh
>> ./fs-debian-installer.sh
>>
>> Let me know what you think!
>> --
>> Brian D. Foster
>> Endigo Computer LLC
>> Email: bdfoster at endigotech.com
>> Phone: 317-800-7876
>> Indianapolis, Indiana, USA
>>
>> This message contains confidential information and is intended for those
>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>> you are not the intended recipient you are notified that disclosing,
>> copying, distributing or taking any action in reliance on the contents of
>> this information is strictly prohibited. E-mail transmission cannot be
>> guaranteed to be secure or error-free as information could be intercepted,
>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>> The sender therefore does not accept liability for any errors or omissions
>> in the contents of this message, which arise as a result of e-mail
>> transmission. If verification is required please request a hard-copy
>> version.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.king at quentustech.com  Thu Jun 21 22:46:29 2012
From: william.king at quentustech.com (William King)
Date: Thu, 21 Jun 2012 11:46:29 -0700
Subject: [Freeswitch-users] FreeSWITCH Installer for Debian
In-Reply-To: <CAKV-jFxB3OEfZFZS9BY=KvNnD-Uwc49acrhf+j3i5sOMzq6Zqg@mail.gmail.com>
References: <CAKV-jFwgmV6Up2RRpe8CBNL9qUN=xV6oBHFjpWcCShvYwPxWXA@mail.gmail.com>
	<CAP2fbbe84p96tgugP-bRPjdvjsrnEx42-ecR0GKn_9o6ijsoRQ@mail.gmail.com>
	<CAKV-jFxB3OEfZFZS9BY=KvNnD-Uwc49acrhf+j3i5sOMzq6Zqg@mail.gmail.com>
Message-ID: <4FE36C05.1040701@quentustech.com>

The debian packages have been updated. So it is a very simple process to 
build the package locally, and install from that.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 06/21/2012 11:43 AM, Brian Foster wrote:
>
> It's broken, didn't seem like there was much interest in this so I 
> only use it internally.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
>
> On Jun 21, 2012 1:19 PM, "Balamurugan Mahendran" <b2m at a-cti.com 
> <mailto:b2m at a-cti.com>> wrote:
>
>     Is this outdated link?
>     files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_lastest.sh
>     <http://files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_lastest.sh>  (looks
>     broken now)
>
>     Thanks,
>     Bala
>
>     On Fri, Mar 23, 2012 at 12:07 AM, Brian Foster
>     <bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>> wrote:
>
>         Alright, so I've managed to put together a BASH script to
>         install FS on top of Debian (Squeeze). I took the information
>         on the wiki, along with some best practices and threw them all
>         together for an automated installer. I came up with this
>         really because I'm 'lazy'.
>
>         I'm on version 0.0.2, and it's not considered extremely safe
>         due to almost no error checking. I'd like to add some of that
>         in. Plans are in the works to do some command line switches,
>         put some options in so that you can customize certain things
>         about the FS install, as well as having a guided install. The
>         guided install would really be for the beginners, and would do
>         such things as setting the hostname, setting timezone data,
>         basically build FreeSWITCH from an absolutely clean install.
>         When Ken's Debian packages come out for  FreeSWITCH, I plan on
>         adding the ability to install from those packages as well as
>         just installing from git head and (eventually) the 1.2 branch.
>
>         I'd like to get some testers from the community and get some
>         feedback on what they would like to see in this installer.
>         It's still VERY basic, and that's a good thing! This way, I
>         can add features from the get-go from users that are used to
>         installing FS in a certain way.
>
>         http://files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_latest.sh
>         << Current Version as of this writing is 0.0.2
>
>         To start, do this:
>         cd
>         wget
>         files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_lastest.sh
>         <http://files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_lastest.sh>
>         -O fs-debian-installer.sh
>         chmod +x fs-debian-installer.sh
>         ./fs-debian-installer.sh
>
>         Let me know what you think!
>         -- 
>         Brian D. Foster
>         Endigo Computer LLC
>         Email: bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>
>         Phone: 317-800-7876 <tel:317-800-7876>
>         Indianapolis, Indiana, USA
>
>         This message contains confidential information and is intended
>         for those listed in the "To:", "CC:", and/or "BCC:" fields of
>         the message header. If you are not the intended recipient you
>         are notified that disclosing, copying, distributing or taking
>         any action in reliance on the contents of this information is
>         strictly prohibited. E-mail transmission cannot be guaranteed
>         to be secure or error-free as information could be
>         intercepted, corrupted, lost, destroyed, arrive late or
>         incomplete, or contain viruses. The sender therefore does not
>         accept liability for any errors or omissions in the contents
>         of this message, which arise as a result of e-mail
>         transmission. If verification is required please request a
>         hard-copy version.
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.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
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From bdfoster at endigotech.com  Thu Jun 21 22:54:01 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 21 Jun 2012 14:54:01 -0400
Subject: [Freeswitch-users] FreeSWITCH Installer for Debian
In-Reply-To: <4FE36C05.1040701@quentustech.com>
References: <CAKV-jFwgmV6Up2RRpe8CBNL9qUN=xV6oBHFjpWcCShvYwPxWXA@mail.gmail.com>
	<CAP2fbbe84p96tgugP-bRPjdvjsrnEx42-ecR0GKn_9o6ijsoRQ@mail.gmail.com>
	<CAKV-jFxB3OEfZFZS9BY=KvNnD-Uwc49acrhf+j3i5sOMzq6Zqg@mail.gmail.com>
	<4FE36C05.1040701@quentustech.com>
Message-ID: <CAKV-jFxmfYfMbmaMKncZ-0xg3UKS6OHdNMvC3J+m0wOT3J089g@mail.gmail.com>

That was one point of the script, but there were other reasons to have it
around as well, such as automated installs.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 21, 2012 2:46 PM, "William King" <william.king at quentustech.com>
wrote:

>  The debian packages have been updated. So it is a very simple process to
> build the package locally, and install from that.
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518william.king at quentustech.com
>
>
> On 06/21/2012 11:43 AM, Brian Foster wrote:
>
> It's broken, didn't seem like there was much interest in this so I only
> use it internally.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 21, 2012 1:19 PM, "Balamurugan Mahendran" <b2m at a-cti.com> wrote:
>
>> Is this outdated link?   files.endigovoip.com/freeswitch
>> /fs-debian-installer/fs-debian-installer_lastest.sh  (looks broken now)
>>
>>  Thanks,
>> Bala
>>
>>  On Fri, Mar 23, 2012 at 12:07 AM, Brian Foster <bdfoster at endigotech.com>wrote:
>>
>>> Alright, so I've managed to put together a BASH script to install FS on
>>> top of Debian (Squeeze). I took the information on the wiki, along with
>>> some best practices and threw them all together for an automated installer.
>>> I came up with this really because I'm 'lazy'.
>>>
>>>  I'm on version 0.0.2, and it's not considered extremely safe due to
>>> almost no error checking. I'd like to add some of that in. Plans are in the
>>> works to do some command line switches, put some options in so that you can
>>> customize certain things about the FS install, as well as having a guided
>>> install. The guided install would really be for the beginners, and would do
>>> such things as setting the hostname, setting timezone data, basically build
>>> FreeSWITCH from an absolutely clean install. When Ken's Debian packages
>>> come out for  FreeSWITCH, I plan on adding the ability to install from
>>> those packages as well as just installing from git head and (eventually)
>>> the 1.2 branch.
>>>
>>>  I'd like to get some testers from the community and get some feedback
>>> on what they would like to see in this installer. It's still VERY basic,
>>> and that's a good thing! This way, I can add features from the get-go from
>>> users that are used to installing FS in a certain way.
>>>
>>>
>>> http://files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_latest.sh<< Current Version as of this writing is 0.0.2
>>>
>>>  To start, do this:
>>> cd
>>> wget
>>> files.endigovoip.com/freeswitch/fs-debian-installer/fs-debian-installer_lastest.sh-O fs-debian-installer.sh
>>> chmod +x fs-debian-installer.sh
>>> ./fs-debian-installer.sh
>>>
>>>  Let me know what you think!
>>>  --
>>> Brian D. Foster
>>> Endigo Computer LLC
>>> Email: bdfoster at endigotech.com
>>> Phone: 317-800-7876
>>> Indianapolis, Indiana, USA
>>>
>>> This message contains confidential information and is intended for those
>>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>>> you are not the intended recipient you are notified that disclosing,
>>> copying, distributing or taking any action in reliance on the contents of
>>> this information is strictly prohibited. E-mail transmission cannot be
>>> guaranteed to be secure or error-free as information could be intercepted,
>>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>>> The sender therefore does not accept liability for any errors or omissions
>>> in the contents of this message, which arise as a result of e-mail
>>> transmission. If verification is required please request a hard-copy
>>> version.
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mytemike72 at gmail.com  Fri Jun 22 00:27:31 2012
From: mytemike72 at gmail.com (Michael Lutz)
Date: Thu, 21 Jun 2012 22:27:31 +0200
Subject: [Freeswitch-users] Randomly DTMF events when using inband detection
	using START_DTMF
Message-ID: <CAPu0Fr3hJEZOjAEQF1gUe6HH9n38q5s3_+6YYpqV6kq-DVo9nA@mail.gmail.com>

Hi All,

I have multiple providers delivering calls to my switches. Most of
them use RFC2833 and I have no issues.
However one of them is not supporting RFC2833 and is using inband dtmf.
For this specific provider I have 'START_DTMF' as one of the first
commands in my dialplan.
This way I can detect digits pressed from a call comming in from that
provider too.

The problem is that it seems on those specific calls I have two weird
things happening:

1. Sometimes it just seems to stop recognize digits.
2. Even worse, sometimes it just seems to randomly (recognize)
generate DTMF tones that are not pressed.

Of course it could be something that is generated by the provider
gateway, as I can also 'hear' a DTMF at that specific moment.

Does anyone have experience with issues like this or have pointers?


Thanks in advance,

Regards,
Mike.


From mi.ke at null.net  Fri Jun 22 01:42:55 2012
From: mi.ke at null.net (Mi Ke)
Date: Thu, 21 Jun 2012 17:42:55 -0400
Subject: [Freeswitch-users] gateway registration using TLS
Message-ID: <20120621214255.182960@gmx.com>

Hi Sean,

 Thank you for your links, I've double checked my setup and found that both openssl-dev and opentls-dev were in place before compiling. I presume that my SSL profile wouldn't start w/o these libs at all but it does:

 root at freeswitch /usr/local/freeswitch/bin # netstat -lpn |grep free
 tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 22156/freeswitch
 tcp 0 0 78.46.44.138:5060 0.0.0.0:* LISTEN 22156/freeswitch
tcp 0 0 78.46.44.138:5061 0.0.0.0:* LISTEN 22156/freeswitch
 udp 0 0 78.46.44.138:5060 0.0.0.0:* 22156/freeswitch

 Also, as an extra test - when I remove cafile.pem and agent.pem profiles do not start so my ssl part seem to be ok.

 I would be very appreciated if FreeSwitch developers could shed some light on that matter i.e. can FS do TLS registration as a client and how to configure it ?

 Thanks / Mike


----- Original Message -----
From: Sean Devoy
Sent: 06/21/12 04:37 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] gateway registration using TLS

HI MiKe,
I have not, but we have been looking into it.
Have you seen this thread:
http://lists.freeswitch.org/pipermail/freeswitch-users/2008-May/003233.html 
 where Brian West says:
You need to make sure you have openssl-devel when you configure/ 
compile Sofia otherwise it's not enabled. Also their is more to it 
than "bin/gentls_cert setup"
http://wiki.freeswitch.org/wiki/Tls 
 And Helmut Kuper responds with:
ok works now. I had to install libgnutls-dev befor running ./configure
script. TLS works with default-values of "bin/gentls_cert setup" and
"bin/gentls_cert create".
 Hope that helps,
 Sean
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]  *On Behalf Of *Mi Ke  *Sent:* Thursday, June 21, 2012 9:17 AM  *To:* FreeSWITCH Users Help  *Subject:* Re: [Freeswitch-users] gateway registration using TLS
Hi Sean, It seems that setting realm port to 5061 does not set registration mode to TLS - this is how remote server sees our reg attempt: REGISTER sip:server_ip:5061;transport=tcp  SIP/2.0 Via: SIP/2.0/TCP our_ip;branch=z9hG4bKXK27gKtt61X9a Max-Forwards: 70 From: <sip:xxxx at server_ip:5061 >;tag=Q4ytjXN7t319Q To: <sip:xxxx at server_ip:5061 > Call-ID: 29e638e8-ba6d-11e1-aa8e-e35fbde939c6 CSeq: 29804746 REGISTER Contact: <sip:gw+xxxx at our_ip:5060;transport=tcp;gw=xxxx > Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 Instead of switching to TLS mode FS just sends plain unencrypted reg request to remote port 5061. After that remote server closes connection. Have you practically succeded in placing FS into TLS registration mode? Cheers / Mike
----- Original Message -----
From: Sean Devoy
Sent: 06/20/12 10:27 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] gateway registration using TLS

NO, sorry. I just saw ?remote_ip_here? in the error message and thought may you actually had used that.
Sorry.
From: freeswitch-users-bounces at lists.freeswitch.org  %5Bmailto:freeswitch-users-bounces at lists.freeswitch.org%5D  *On Behalf Of *Mi Ke  *Sent:* Wednesday, June 20, 2012 12:20 PM  *To:* FreeSWITCH Users Help  *Subject:* Re: [Freeswitch-users] gateway registration using TLS
Dear Sean, It's not that bad - remote_ip, xxx and yyy are only placeholders in my example - actual config has real params ;) Did you mean that FS will automatically use TLS for registration when I append :5061 to realm value ? Thanks / MiKe
----- Original Message -----
From: Sean Devoy
Sent: 06/20/12 07:03 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] gateway registration using TLS

Mi Ke,
You should put your actual remote provider?s domain name or tip address in where it says ?remote_ip_here? and I assume you have entered you actual username and password where it says ?yyy? and ?zzz?.
It should look SIMILAR to this, but with your provider: <param name="realm" value="sip.voipinnovations.com:5061"/>
Hope that helps,
Sean
From: freeswitch-users-bounces at lists.freeswitch.org  %5Bmailto:freeswitch-users-bounces at lists.freeswitch.org%5D  *On Behalf Of *Mi Ke  *Sent:* Wednesday, June 20, 2012 11:46 AM  *To:* Freeswitch Users  *Subject:* [Freeswitch-users] gateway registration using TLS

Hi All, Is it possible to explicitly tell sofia to register as a client using TLS? Here's my TLS-related part of sofia config: <profiles> <profile name="external"> <domains> <domain name="all" alias="true" parse="false"/> </domains> <gateways> <gateway name="xxx"> <param name="register" value="true"/> <param name="username" value="yyy"/> <param name="realm" value="remote_ip_here:5061"/> <param name="password" value="zzz"/> <param name="register-transport" value="tcp"/> <param name="expire-seconds" value="3600"/> <param name="retry-seconds" value="60" /> </gateway> </gateways> <settings> ... <param name="context" value="external"/> <param name="sip-port" value="5060"/> <param name="tls" value="true"/> <param name="tls-only" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="contact-params" value="tport=tcp"/> <param name="tls-version" value="sslv23"/> <param name="tls-cert-dir" value="/usr/local/freeswitch/conf/ssl"/> <param name="tls-sip-port" value="5061"/> <param name="dialplan" value="XML"/> ... </settings> </profile> </profiles> But that does not work... tport_connected( 

0x7f9b44014110): events CONNECTED tport_send_event(0x7f9b44014110) - ready to send to (tcp/remote_ip_here:5061 http://212.58.166.54:5061 ) tport_vsend(0x7f9b44014110): 600 bytes of 600 to tcp/remote_ip_here:5061 http://212.58.166.54:5061  tport_vsend returned 600 tport_set_events(0x7f9b44014110): events IN tport(0x7f9b44014110): reset timer tport_wakeup(0x7f9b44014110): events IN HUP ERR tport_release(0x7f9b44014110): 0x7f9b4401a460 by 0x7f9b4402ca70 with (nil) nta: REGISTER (29733430): Connection reset by peer (104) with tcp/[remote_ip_here]:5061nua(0x7f9b440196a0): event r_register 503 Service Unavailable
Any ideas what I'm doing wrong? Thanks / MiKe
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From b2m at a-cti.com  Fri Jun 22 09:54:05 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Fri, 22 Jun 2012 11:24:05 +0530
Subject: [Freeswitch-users] Call recording issue
Message-ID: <CAP2fbbeVmMz8NMLkro88ffAkaKTtVp4gDd2-q2NACUb_qejCjA@mail.gmail.com>

All,


I was digging for an call recording issue with Skype calls for the past one
week, looking here for any clue/help.

In a different thread I was advised that its out of mod_skyopen module.
Also it seems there is a problem related to file conversion and bit rating.

Looks like call recorded with 16k rate but named as 8k.

Is there a way to control this parameter? or where should I focus for this.


note:

I tried different hardware, latest source, and new configuration but could
not trace it. It was good till last Friday, and there was no change in any
settings as per our knowledge.

Thanks for your help!!


Thanks,
Bala
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From sherifomran2000 at yahoo.com  Fri Jun 22 13:39:03 2012
From: sherifomran2000 at yahoo.com (Sherif Omran)
Date: Fri, 22 Jun 2012 02:39:03 -0700 (PDT)
Subject: [Freeswitch-users] Invalid application transfer help
In-Reply-To: <CAP2fbbeVmMz8NMLkro88ffAkaKTtVp4gDd2-q2NACUb_qejCjA@mail.gmail.com>
Message-ID: <1340357943.10949.YahooMailClassic@web110804.mail.gq1.yahoo.com>

Hi Guys,

My server was working fine till today, It does not transfer any call. When I call an extension i get? a not connected tone and in the log I see invalid application

2012-06-22 09:37:26.414815 [ERR] switch_core_session.c:2156 Invalid Application transfer

I wonder what this could be? I was trying yesterday to install some packages. Could this be related to that?

kind regards,
Sherif

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From peter.olsson at visionutveckling.se  Fri Jun 22 14:18:03 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 22 Jun 2012 10:18:03 +0000
Subject: [Freeswitch-users] Invalid application transfer help
In-Reply-To: <1340357943.10949.YahooMailClassic@web110804.mail.gq1.yahoo.com>
References: <CAP2fbbeVmMz8NMLkro88ffAkaKTtVp4gDd2-q2NACUb_qejCjA@mail.gmail.com>,
	<1340357943.10949.YahooMailClassic@web110804.mail.gq1.yahoo.com>
Message-ID: <F8AE005E-E72A-437A-9F44-4A54BF0568B7@visionutveckling.se>

Seems you don't have mod_dptools loaded anymore.

/Peter

22 jun 2012 kl. 11:45 skrev "Sherif Omran" <sherifomran2000 at yahoo.com<mailto:sherifomran2000 at yahoo.com>>:

Hi Guys,

My server was working fine till today, It does not transfer any call. When I call an extension i get  a not connected tone and in the log I see invalid application

2012-06-22 09:37:26.414815 [ERR] switch_core_session.c:2156 Invalid Application transfer

I wonder what this could be? I was trying yesterday to install some packages. Could this be related to that?

kind regards,
Sherif


!DSPAM:4fe43b9a32761615820512!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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
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!DSPAM:4fe43b9a32761615820512!


From admin at blindi.net  Fri Jun 22 17:21:39 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Fri, 22 Jun 2012 15:21:39 +0200 (CEST)
Subject: [Freeswitch-users] Extension limit per session?
In-Reply-To: <F8AE005E-E72A-437A-9F44-4A54BF0568B7@visionutveckling.se>
References: <CAP2fbbeVmMz8NMLkro88ffAkaKTtVp4gDd2-q2NACUb_qejCjA@mail.gmail.com>,
	<1340357943.10949.YahooMailClassic@web110804.mail.gq1.yahoo.com>
	<F8AE005E-E72A-437A-9F44-4A54BF0568B7@visionutveckling.se>
Message-ID: <Pine.LNX.4.64.1206221516050.22290@tomback.antagus.de>

Hi all,

i have setup a complex ivr and extensionstructure for incoming callers.

I have the following error:

[[mfreeswitch at internal> [[mM[[21CM[[K[[31m2012-06-22 15:09:23.942083
[ERR] switch_core_session.c:2363 Error 
sofia/external/08925007676 at arcor.de
too many stacked extensionsM


Can you help please to increase the limit higher?

Thanks.

---------------
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blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From cmrienzo at gmail.com  Fri Jun 22 18:42:49 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 22 Jun 2012 10:42:49 -0400
Subject: [Freeswitch-users] Extension limit per session?
In-Reply-To: <Pine.LNX.4.64.1206221516050.22290@tomback.antagus.de>
References: <CAP2fbbeVmMz8NMLkro88ffAkaKTtVp4gDd2-q2NACUb_qejCjA@mail.gmail.com>
	<1340357943.10949.YahooMailClassic@web110804.mail.gq1.yahoo.com>
	<F8AE005E-E72A-437A-9F44-4A54BF0568B7@visionutveckling.se>
	<Pine.LNX.4.64.1206221516050.22290@tomback.antagus.de>
Message-ID: <CAD-qRkhDRdnydoEHOykSVfTW7Ob5KMW7K2bGqobgn7AGKMzVQQ@mail.gmail.com>

Your approach might not be optimal since I've never heard of anybody
needing to mess with this value before.  Do you really need to execute more
than 16 extensions?

If so, increase the value of SWITCH_MAX_STACKS in
src/include/switch_types.h from 16 to whatever you need.  Recomple and
re-install.


Chris


On Fri, Jun 22, 2012 at 9:21 AM, Thomas Hoellriegel <admin at blindi.net>wrote:

> Hi all,
>
> i have setup a complex ivr and extensionstructure for incoming callers.
>
> I have the following error:
>
> [[mfreeswitch at internal> [[mM[[21CM[[K[[31m2012-06-22 15:09:23.942083
> [ERR] switch_core_session.c:2363 Error sofia/external/08925007676@**
> arcor.de <08925007676 at arcor.de>
> too many stacked extensionsM
>
>
> Can you help please to increase the limit higher?
>
> Thanks.
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/**listinfo/blinde-misc<http://www.blindi.net/mailman/listinfo/blinde-misc>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vallimamod.abdullah at imtelecom.fr  Fri Jun 22 19:44:25 2012
From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH)
Date: Fri, 22 Jun 2012 17:44:25 +0200
Subject: [Freeswitch-users] Alert-Info header not understood by SPA phones
Message-ID: <A0B516FD-73C1-4F9E-B928-4C4AA54BD393@imtelecom.fr>

Hi List,

I am trying to setup a custom ringtone for internal calls for cisco SPA504G through the Alert-Info header. 
I have put <action application="export" data="alert_info=Office"/> before the bridge as stated on the wiki but the sip header does not seems to be recognized by the phone.  The invite the alert-info header is sent correctly in the sip invite but it contains extra brackets (Alert-Info: <Office>) wich confuses the phone as it is expecting "Alert-Info: Office".
I have also tried <action application="set" data="sip_h_Alert-Info=Office"/> but it does not make any difference.

Is there any way to remove these brackets ?

Thanks you!

-- 
Best Regards,
- vma.

From jpablolorenzetti at hotmail.com  Fri Jun 22 19:57:59 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Fri, 22 Jun 2012 15:57:59 +0000
Subject: [Freeswitch-users] variable not being replaced in dialplan
Message-ID: <DUB109-W10261C32EBA6DDECA887211D9FC0@phx.gbl>


Good morning, i have the following entry in my dialplan to dialout
using a gateway:

<extension name="gw_outbound_call">
??????? <condition field="destination_number" expression="[\d]{7,}?$">
??????????????? <action application="set" data="effective_caller_id_number=10000"/>
??????????????? <action application="bridge" data="sofia/gateway/gw/$1"/>
??????? </condition>
??? </extension>

the dialplan runs well and when FS has to bridge the call it opens a channel
to "sofia/gateway/gw/$1", it does not replace the $1 by the actual number
being dialed, i have the same in other parts of the dialplan and it works
good, i can not find what i m doing wrong here, i would appreciate any hint
about this issue. 
thanks!
 		 	   		  

From mario_fs at mgtech.com  Fri Jun 22 20:09:04 2012
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 22 Jun 2012 09:09:04 -0700
Subject: [Freeswitch-users] Alert-Info header not understood by SPA
	phones
In-Reply-To: <A0B516FD-73C1-4F9E-B928-4C4AA54BD393@imtelecom.fr>
References: <A0B516FD-73C1-4F9E-B928-4C4AA54BD393@imtelecom.fr>
Message-ID: <39F586EF-0B6B-4243-A385-900D7CD870D7@mgtech.com>

I don't know how similar they are but on my SPA962 the line below works. I think you need to use the settings of the name (office) not the name.
Mario G

<action application="export" data="alert_info=n=Simple-5;w=2;c=5"/>
   
On Jun 22, 2012, at 8:44 AM, Vallimamod ABDULLAH wrote:

> Hi List,
> 
> I am trying to setup a custom ringtone for internal calls for cisco SPA504G through the Alert-Info header. 
> I have put <action application="export" data="alert_info=Office"/> before the bridge as stated on the wiki but the sip header does not seems to be recognized by the phone.  The invite the alert-info header is sent correctly in the sip invite but it contains extra brackets (Alert-Info: <Office>) wich confuses the phone as it is expecting "Alert-Info: Office".
> I have also tried <action application="set" data="sip_h_Alert-Info=Office"/> but it does not make any difference.
> 
> Is there any way to remove these brackets ?
> 
> Thanks you!
> 
> -- 
> Best Regards,
> - vma.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From alex at thewinelake.com  Fri Jun 22 20:15:32 2012
From: alex at thewinelake.com (Alex)
Date: Fri, 22 Jun 2012 17:15:32 +0100
Subject: [Freeswitch-users] variable not being replaced in dialplan
In-Reply-To: <DUB109-W10261C32EBA6DDECA887211D9FC0@phx.gbl>
References: <DUB109-W10261C32EBA6DDECA887211D9FC0@phx.gbl>
Message-ID: <4FE49A24.7050004@thewinelake.com>

Is there a reason you're not just doing

<action application="bridge" data="sofia/gateway/gw/${destination_number}"/>



From haloha201 at gmail.com  Fri Jun 22 20:16:25 2012
From: haloha201 at gmail.com (haloha)
Date: Fri, 22 Jun 2012 23:16:25 +0700
Subject: [Freeswitch-users] the problem with mod_xml_curl
Message-ID: <CAMBhhBJg437_c7OOgzsP=4QfFvZjrNLZ7YhQ946G2vj7eZvdCQ@mail.gmail.com>

hi all

i do a test with mod_curl_xml and i found strange thing with mod_curl_xml
my pastebin: http://pastebin.freeswitch.org/19347

let me explain:
1. the sip client send register to freeswitch
2. freeswitch send to web server ask for user infomation
3. web server respone to freeswitch the user infomation but there is no
password of user
4. sip client register successfull -- this maybe the wrong thing


could  someone please help

thank you
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From vipkilla at gmail.com  Fri Jun 22 20:22:22 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 22 Jun 2012 12:22:22 -0400
Subject: [Freeswitch-users] the problem with mod_xml_curl
In-Reply-To: <CAMBhhBJg437_c7OOgzsP=4QfFvZjrNLZ7YhQ946G2vj7eZvdCQ@mail.gmail.com>
References: <CAMBhhBJg437_c7OOgzsP=4QfFvZjrNLZ7YhQ946G2vj7eZvdCQ@mail.gmail.com>
Message-ID: <CAC-LwPMhe3yqWZpb_P9nrwyhc-Obu5MzkV3hq48FZ6gmSOdkeQ@mail.gmail.com>

If there is no password parameter, then one is not required for that
user therefore anybody can register as the user without a password.

On Fri, Jun 22, 2012 at 12:16 PM, haloha <haloha201 at gmail.com> wrote:
> hi all
>
> i do a test with mod_curl_xml and i found strange thing with mod_curl_xml
> my pastebin: http://pastebin.freeswitch.org/19347
>
> let me explain:
> 1. the sip client send register to freeswitch
> 2. freeswitch send to web server ask for user infomation
> 3. web server respone to freeswitch the user infomation but there is no
> password of user
> 4. sip client register successfull -- this maybe the wrong thing
>
>
> could? someone please help
>
> thank you
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From drk at drkngs.net  Fri Jun 22 20:43:17 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Fri, 22 Jun 2012 09:43:17 -0700
Subject: [Freeswitch-users] Using mod_managed or mod_java
In-Reply-To: <111509FCE9EBF74BAB2CB38AAC521D9B1337FC0FD8@helle.dk.hq.firstcom.dk>
Message-ID: <20120622164317.117c74ac@mail.tritonwest.net>

I have a lot of experience with mod_managed, however it's mostly with the  M$ .NET CLR implmentation. I have heard from others that it also works pretty well using the MONO flavor.  
   
I use the Windows implmentation in a comericial product, that is ia full carrier grade softswitch. I have customers that have a lot of traffic, 400+ call setups per second, and have no problems with being able to keep up.  
   
I also have a number of custom solutions for clients for both large call center applications, and auto-dial outbound applicatons. Some of these applications have more then 1000+ calls per box, and I 've never had a production problem.  
   
If you need any examples or help please feel free to contact me off the list.  
   
--Dave      _____  

  From: Jon Sch?pzinsky [mailto:jos at firstcom.dk]
To: freeswitch-users at lists.freeswitch.org [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Thu, 21 Jun 2012 03:54:40 -0700
Subject: [Freeswitch-users] Using mod_managed or mod_java

  
  
Hello everybody,  
   
We are designing a new platform, and are planning to use FreeSWITCH for our core services and switching.  
   
We have looked at using the ESL interface, mod_managed og mod_java.  
   
It doesn?t seem as though a lot of work is being done on mod_java, and there isn?t allot of documentation around. So if anybody could share their experience with using mod_java, that would be great.  
   
mod_managed seems very promising, but we are a bit nervous about running Mono on one of our primary production systems, as our experience with Mono hasn?t been all good.  
Is anybody running mod_managed on large scale production systems, that can share their experience with doing so?  
   
The ESL interface is working pretty great, but we do like the idea of integrating our core logic directly into FreeSWITCH, instead of having it run in a separate process.  
   
   
   
Kind Regards  
   
Jon Leren Sch?pzinsky  
       
   
 
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From paul at cupis.co.uk  Fri Jun 22 20:54:50 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Fri, 22 Jun 2012 17:54:50 +0100
Subject: [Freeswitch-users] variable not being replaced in dialplan
In-Reply-To: <DUB109-W10261C32EBA6DDECA887211D9FC0@phx.gbl>
References: <DUB109-W10261C32EBA6DDECA887211D9FC0@phx.gbl>
Message-ID: <20120622165450.GA24108@eagle.cupis.co.uk>

On Fri, Jun 22, 2012 at 03:57:59PM +0000, Juan Pablo L. wrote:
> Good morning, i have the following entry in my dialplan to dialout
> using a gateway:
> 
> <extension name="gw_outbound_call">
>   <condition field="destination_number" expression="[\d]{7,}?$">
>     <action application="set" data="effective_caller_id_number=10000"/>
>     <action application="bridge" data="sofia/gateway/gw/$1"/>
>   </condition>
> </extension>
> 
> the dialplan runs well and when FS has to bridge the call it opens a channel
> to "sofia/gateway/gw/$1", it does not replace the $1 by the actual number
> being dialed, i have the same in other parts of the dialplan and it works
> good, i can not find what i m doing wrong here, i would appreciate any hint
> about this issue. 

You are not setting $1.

   <condition field="destination_number" expression="[\d]{7,}?$">

should be:

   <condition field="destination_number" expression="([\d]{7,}?)$">

take note of the parentheses () which set the $1 variable.

Regards,



From jpablolorenzetti at hotmail.com  Fri Jun 22 21:09:19 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Fri, 22 Jun 2012 17:09:19 +0000
Subject: [Freeswitch-users] variable not being replaced in dialplan
In-Reply-To: <4FE49A24.7050004@thewinelake.com>
References: <DUB109-W10261C32EBA6DDECA887211D9FC0@phx.gbl>,
	<4FE49A24.7050004@thewinelake.com>
Message-ID: <DUB109-W4600FFC4461A37E1616774D9FC0@phx.gbl>


Hi thanks, i also tried as in the default configurations:

<extension name="gw_outbound_call">
??????? <condition field="destination_number" expression="^[\d]{7}?$">
?????????????? <action application="export" data="dialed_extension=$1"/>
??????????????? <action application="set" data="effective_caller_id_number=1000"/>
??????????????? <action application="bridge" data="sofia/gateway/gw/${dialed_extension}"/>
??????? </condition>
??? </extension>

but still it replaces the ${dialed_extension} by $1 ... i have not tried what you suggest, i will try it
but still cant find a reason why the above does not work ... thanks!

----------------------------------------
> Date: Fri, 22 Jun 2012 17:15:32 +0100
> From: alex at thewinelake.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] variable not being replaced in dialplan
>
> Is there a reason you're not just doing
>
> <action application="bridge" data="sofia/gateway/gw/${destination_number}"/>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  

From jpablolorenzetti at hotmail.com  Fri Jun 22 21:12:35 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Fri, 22 Jun 2012 17:12:35 +0000
Subject: [Freeswitch-users] variable not being replaced in dialplan
In-Reply-To: <DUB109-W4600FFC4461A37E1616774D9FC0@phx.gbl>
References: <DUB109-W10261C32EBA6DDECA887211D9FC0@phx.gbl>,
	<4FE49A24.7050004@thewinelake.com>,
	<DUB109-W4600FFC4461A37E1616774D9FC0@phx.gbl>
Message-ID: <DUB109-W2338DED148B2C85ADD7BB7D9FC0@phx.gbl>


Alex, thanks for your suggestion, that worked like a charm, i still have
lots to learn, thanks very much!!!

----------------------------------------
> From: jpablolorenzetti at hotmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: RE: [Freeswitch-users] variable not being replaced in dialplan
> Date: Fri, 22 Jun 2012 17:09:19 +0000
>
>
> Hi thanks, i also tried as in the default configurations:
>
> <extension name="gw_outbound_call">
>         <condition field="destination_number" expression="^[\d]{7}?$">
>                <action application="export" data="dialed_extension=$1"/>
>                 <action application="set" data="effective_caller_id_number=1000"/>
>                 <action application="bridge" data="sofia/gateway/gw/${dialed_extension}"/>
>         </condition>
>     </extension>
>
> but still it replaces the ${dialed_extension} by $1 ... i have not tried what you suggest, i will try it
> but still cant find a reason why the above does not work ... thanks!
>
> ----------------------------------------
> > Date: Fri, 22 Jun 2012 17:15:32 +0100
> > From: alex at thewinelake.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] variable not being replaced in dialplan
> >
> > Is there a reason you're not just doing
> >
> > <action application="bridge" data="sofia/gateway/gw/${destination_number}"/>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
 		 	   		  

From jpablolorenzetti at hotmail.com  Fri Jun 22 21:13:40 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Fri, 22 Jun 2012 17:13:40 +0000
Subject: [Freeswitch-users] variable not being replaced in dialplan
In-Reply-To: <20120622165450.GA24108@eagle.cupis.co.uk>
References: <DUB109-W10261C32EBA6DDECA887211D9FC0@phx.gbl>,
	<20120622165450.GA24108@eagle.cupis.co.uk>
Message-ID: <DUB109-W10093C80B1E749D33DF0D78D9FC0@phx.gbl>


thank you very much, that worked too, i m learning and still a lot ahead but thanks
for your support.

----------------------------------------
> Date: Fri, 22 Jun 2012 17:54:50 +0100
> From: paul at cupis.co.uk
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] variable not being replaced in dialplan
>
> On Fri, Jun 22, 2012 at 03:57:59PM +0000, Juan Pablo L. wrote:
> > Good morning, i have the following entry in my dialplan to dialout
> > using a gateway:
> >
> > <extension name="gw_outbound_call">
> > <condition field="destination_number" expression="[\d]{7,}?$">
> > <action application="set" data="effective_caller_id_number=10000"/>
> > <action application="bridge" data="sofia/gateway/gw/$1"/>
> > </condition>
> > </extension>
> >
> > the dialplan runs well and when FS has to bridge the call it opens a channel
> > to "sofia/gateway/gw/$1", it does not replace the $1 by the actual number
> > being dialed, i have the same in other parts of the dialplan and it works
> > good, i can not find what i m doing wrong here, i would appreciate any hint
> > about this issue.
>
> You are not setting $1.
>
> <condition field="destination_number" expression="[\d]{7,}?$">
>
> should be:
>
> <condition field="destination_number" expression="([\d]{7,}?)$">
>
> take note of the parentheses () which set the $1 variable.
>
> Regards,
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  

From amusingparsnip at me.com  Fri Jun 22 14:47:44 2012
From: amusingparsnip at me.com (Jeremy Sharp)
Date: Fri, 22 Jun 2012 11:47:44 +0100
Subject: [Freeswitch-users] Cutting out after 30 seconds
Message-ID: <AFF35AA0-1E32-4C0F-A2FE-E49D8A55458D@me.com>

I have some Siemens DECT sip handsets, which are cutting off on outbound calls after 30 seconds. Doesn't seem to affect other phones, or inbound calls.

I have tried setting the transport to TCP and adding {ignore_display_updates=true} to the bridge action but this hasn't helped.

What could it be?

2012-06-22 11:43:52.978442 [DEBUG] switch_core_session.c:923 Send signal sofia/internal/1002 at 192.168.1.1 [BREAK]
2012-06-22 11:43:52.978442 [DEBUG] switch_core_session.c:923 Send signal sofia/internal/1002 at 192.168.1.1 [BREAK]
2012-06-22 11:43:52.998446 [DEBUG] sofia.c:5838 Channel sofia/internal/1002 at 192.168.1.1 entering state [terminating][0]
2012-06-22 11:43:52.998446 [DEBUG] switch_channel.c:2899 (sofia/internal/1002 at 192.168.1.1) Callstate Change ACTIVE -> HANGUP
2012-06-22 11:43:52.998446 [NOTICE] sofia.c:6628 Hangup sofia/internal/1002 at 192.168.1.1 [CS_EXECUTE] [NORMAL_UNSPECIFIED]
2012-06-22 11:43:52.998446 [DEBUG] switch_channel.c:2922 Send signal sofia/internal/1002 at 192.168.1.1 [KILL]
2012-06-22 11:43:52.998446 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/1002 at 192.168.1.1 [BREAK]
2012-06-22 11:43:52.998446 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD DONE [sofia/internal/1002 at 192.168.1.1]
2012-06-22 11:43:52.998446 [DEBUG] switch_ivr_bridge.c:613 Send signal sofia/external/01234567890 [BREAK]
2012-06-22 11:43:53.018396 [DEBUG] switch_ivr_bridge.c:501 sofia/internal/1002 at 192.168.1.1 ending bridge by request from write function
2012-06-22 11:43:53.018396 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD DONE [sofia/external/01234567890]
2012-06-22 11:43:53.018396 [DEBUG] switch_ivr_bridge.c:613 Send signal sofia/internal/1002 at 192.168.1.1 [BREAK]
2012-06-22 11:43:53.018396 [DEBUG] switch_channel.c:2899 (sofia/external/01234567890) Callstate Change ACTIVE -> HANGUP
2012-06-22 11:43:53.018396 [NOTICE] switch_ivr_bridge.c:676 Hangup sofia/external/01234567890 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2012-06-22 11:43:53.018396 [DEBUG] switch_channel.c:2922 Send signal sofia/external/01234567890 [KILL]
2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:1228 Send signal sofia/external/01234567890 [BREAK]
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:443 (sofia/external/01234567890) State EXCHANGE_MEDIA going to sleep
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:385 (sofia/external/01234567890) Running State Change CS_HANGUP
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:625 (sofia/external/01234567890) State HANGUP
2012-06-22 11:43:53.018396 [DEBUG] mod_sofia.c:469 Channel sofia/external/01234567890 hanging up, cause: NORMAL_CLEARING
2012-06-22 11:43:53.018396 [DEBUG] mod_sofia.c:517 Sending BYE to sofia/external/01234567890
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:47 sofia/external/01234567890 Standard HANGUP, cause: NORMAL_CLEARING
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:625 (sofia/external/01234567890) State HANGUP going to sleep
2012-06-22 11:43:53.018396 [DEBUG] switch_ivr_bridge.c:1456 sofia/external/01234567890 skip receive message [UNBRIDGE] (channel is hungup already)
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:416 (sofia/external/01234567890) State Change CS_HANGUP -> CS_REPORTING
2012-06-22 11:43:53.018396 [DEBUG] switch_ivr_bridge.c:1459 sofia/internal/1002 at 192.168.1.1 skip receive message [UNBRIDGE] (channel is hungup already)
2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:1228 Send signal sofia/external/01234567890 [BREAK]
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:385 (sofia/external/01234567890) Running State Change CS_REPORTING
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:685 (sofia/external/01234567890) State REPORTING
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:79 sofia/external/01234567890 Standard REPORTING, cause: NORMAL_CLEARING
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:685 (sofia/external/01234567890) State REPORTING going to sleep
2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:2333 sofia/internal/1002 at 192.168.1.1 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1002 at 192.168.1.1) State EXECUTE going to sleep
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/1002 at 192.168.1.1) Running State Change CS_HANGUP
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:410 (sofia/external/01234567890) State Change CS_REPORTING -> CS_DESTROY
2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:1228 Send signal sofia/external/01234567890 [BREAK]
2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:1428 Session 45 (sofia/external/01234567890) Locked, Waiting on external entities
2012-06-22 11:43:53.018396 [NOTICE] switch_core_session.c:1446 Session 45 (sofia/external/01234567890) Ended
2012-06-22 11:43:53.018396 [NOTICE] switch_core_session.c:1448 Close Channel sofia/external/01234567890 [CS_DESTROY]
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:514 (sofia/external/01234567890) Callstate Change HANGUP -> DOWN
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:517 (sofia/external/01234567890) Running State Change CS_DESTROY
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:527 (sofia/external/01234567890) State DESTROY
2012-06-22 11:43:53.018396 [DEBUG] mod_sofia.c:374 sofia/external/01234567890 SOFIA DESTROY
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:86 sofia/external/01234567890 Standard DESTROY
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:527 (sofia/external/01234567890) State DESTROY going to sleep
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1002 at 192.168.1.1) State HANGUP
2012-06-22 11:43:53.018396 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1002 at 192.168.1.1 hanging up, cause: NORMAL_UNSPECIFIED
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1002 at 192.168.1.1 Standard HANGUP, cause: NORMAL_UNSPECIFIED
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1002 at 192.168.1.1) State HANGUP going to sleep
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/1002 at 192.168.1.1) State Change CS_HANGUP -> CS_REPORTING
2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/1002 at 192.168.1.1 [BREAK]
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/1002 at 192.168.1.1) Running State Change CS_REPORTING
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/1002 at 192.168.1.1) State REPORTING
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1002 at 192.168.1.1 Standard REPORTING, cause: NORMAL_UNSPECIFIED
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/1002 at 192.168.1.1) State REPORTING going to sleep
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 192.168.1.1) State Change CS_REPORTING -> CS_DESTROY
2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/1002 at 192.168.1.1 [BREAK]
2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:1428 Session 44 (sofia/internal/1002 at 192.168.1.1) Locked, Waiting on external entities
2012-06-22 11:43:53.018396 [NOTICE] switch_core_session.c:1446 Session 44 (sofia/internal/1002 at 192.168.1.1) Ended
2012-06-22 11:43:53.018396 [NOTICE] switch_core_session.c:1448 Close Channel sofia/internal/1002 at 192.168.1.1 [CS_DESTROY]
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/1002 at 192.168.1.1) Callstate Change HANGUP -> DOWN
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/1002 at 192.168.1.1) Running State Change CS_DESTROY
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/1002 at 192.168.1.1) State DESTROY
2012-06-22 11:43:53.018396 [DEBUG] mod_sofia.c:374 sofia/internal/1002 at 192.168.1.1 SOFIA DESTROY
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1002 at 192.168.1.1 Standard DESTROY
2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/1002 at 192.168.1.1) State DESTROY going to sleep
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From cmrienzo at gmail.com  Fri Jun 22 21:41:47 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 22 Jun 2012 13:41:47 -0400
Subject: [Freeswitch-users] the problem with mod_xml_curl
In-Reply-To: <CAC-LwPMhe3yqWZpb_P9nrwyhc-Obu5MzkV3hq48FZ6gmSOdkeQ@mail.gmail.com>
References: <CAMBhhBJg437_c7OOgzsP=4QfFvZjrNLZ7YhQ946G2vj7eZvdCQ@mail.gmail.com>
	<CAC-LwPMhe3yqWZpb_P9nrwyhc-Obu5MzkV3hq48FZ6gmSOdkeQ@mail.gmail.com>
Message-ID: <CAD-qRkhpM1K+kpOkqtPvbVsKBv5ZfHgwJ=CE2A+=jME70GtL0A@mail.gmail.com>

As instructed on IRC try:

 <param name="allow-empty-password" value="false"/>

in the directory.

Chris


On Fri, Jun 22, 2012 at 12:22 PM, Vik Killa <vipkilla at gmail.com> wrote:

> If there is no password parameter, then one is not required for that
> user therefore anybody can register as the user without a password.
>
> On Fri, Jun 22, 2012 at 12:16 PM, haloha <haloha201 at gmail.com> wrote:
> > hi all
> >
> > i do a test with mod_curl_xml and i found strange thing with mod_curl_xml
> > my pastebin: http://pastebin.freeswitch.org/19347
> >
> > let me explain:
> > 1. the sip client send register to freeswitch
> > 2. freeswitch send to web server ask for user infomation
> > 3. web server respone to freeswitch the user infomation but there is no
> > password of user
> > 4. sip client register successfull -- this maybe the wrong thing
> >
> >
> > could  someone please help
> >
> > thank you
> >
> >
>
>
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From vallimamod.abdullah at imtelecom.fr  Fri Jun 22 22:17:30 2012
From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH)
Date: Fri, 22 Jun 2012 20:17:30 +0200
Subject: [Freeswitch-users] Alert-Info header not understood by SPA
	phones
In-Reply-To: <39F586EF-0B6B-4243-A385-900D7CD870D7@mgtech.com>
References: <A0B516FD-73C1-4F9E-B928-4C4AA54BD393@imtelecom.fr>
	<39F586EF-0B6B-4243-A385-900D7CD870D7@mgtech.com>
Message-ID: <FAA4F224-4805-4A5F-A42F-EB0CA683FC44@imtelecom.fr>

Hi Mario,

Thanks for your answer. After your mail I have tested with a SPA962 and indeed it works with both your syntax and mine!
So looks like cisco have changed the way it works on their new SPA50XG models?

- vma.


On Jun 22, 2012, at 6:09 PM, Mario G wrote:

> I don't know how similar they are but on my SPA962 the line below works. I think you need to use the settings of the name (office) not the name.
> Mario G
> 
> <action application="export" data="alert_info=n=Simple-5;w=2;c=5"/>
>    
> On Jun 22, 2012, at 8:44 AM, Vallimamod ABDULLAH wrote:
> 
>> Hi List,
>> 
>> I am trying to setup a custom ringtone for internal calls for cisco SPA504G through the Alert-Info header. 
>> I have put <action application="export" data="alert_info=Office"/> before the bridge as stated on the wiki but the sip header does not seems to be recognized by the phone.  The invite the alert-info header is sent correctly in the sip invite but it contains extra brackets (Alert-Info: <Office>) wich confuses the phone as it is expecting "Alert-Info: Office".
>> I have also tried <action application="set" data="sip_h_Alert-Info=Office"/> but it does not make any difference.
>> 
>> Is there any way to remove these brackets ?
>> 
>> Thanks you!



From toddb at toddbailey.net  Fri Jun 22 22:24:07 2012
From: toddb at toddbailey.net (toddb at toddbailey.net)
Date: Fri, 22 Jun 2012 11:24:07 -0700
Subject: [Freeswitch-users] Clarification needed on setting up group
 calling parameters
Message-ID: <20120622112407.33e327b490679d2282e332758c73b55b.43c97e5644.wbe@email14.secureserver.net>

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From bdfoster at endigotech.com  Fri Jun 22 22:33:54 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 22 Jun 2012 14:33:54 -0400
Subject: [Freeswitch-users] Clarification needed on setting up group
 calling parameters
In-Reply-To: <20120622112407.33e327b490679d2282e332758c73b55b.43c97e5644.wbe@email14.secureserver.net>
References: <20120622112407.33e327b490679d2282e332758c73b55b.43c97e5644.wbe@email14.secureserver.net>
Message-ID: <CAKV-jFxMdmCNVh8WZNeV+iF+5XeCCswL1WG0SFhFOH6A2XE-2A@mail.gmail.com>

Sorry for the delay. Here's a link for you (not sure what you're looking
at):

http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups

See if that clarifies anything for you.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 22, 2012 2:25 PM, <toddb at toddbailey.net> wrote:

> Anyone ?  anyone at all care to step in on this ?
>
>
> -------- Original Message --------
> Subject: [Freeswitch-users] Clarification needed on setting up group
> calling parameters
> From: <toddb at toddbailey.net>
> Date: Wed, June 20, 2012 8:08 am
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>
>  Hi All,
>
>
> read through the documentation a few times in setting up call groups and
> can't seem to make it happen.
>
> 1 problem is the doc's don't make it easy by stating which file I need to
> be editing.
>
>
> Perhaps a call group isn't what I need,  so here the scenario
>
> I'd like to set up in-bound calls to automatically go to extensions 1002 -
> 1007 where the fist to answer gets the call.
>
> I'm using a spa3102 to route ptsn calls to voip extension 1001,  so ext
> 1001 needs to ring on 1002-1007.
>
> Q: what method does this and what files & locations do I need to edit?
>
>
> thx.
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From monemran at gmail.com  Fri Jun 22 22:35:17 2012
From: monemran at gmail.com (Mohammad Emran)
Date: Sat, 23 Jun 2012 00:35:17 +0600
Subject: [Freeswitch-users] Clarification needed on setting up group
	calling parameters
In-Reply-To: <20120622112407.33e327b490679d2282e332758c73b55b.43c97e5644.wbe@email14.secureserver.net>
References: <20120622112407.33e327b490679d2282e332758c73b55b.43c97e5644.wbe@email14.secureserver.net>
Message-ID: <DA4B97EB-ADCD-48AB-8841-A7FA93D79346@gmail.com>

Try this
http://wiki.freeswitch.org/wiki/Call_Groups


Sent from my iPad

On Jun 23, 2012, at 12:24 AM, <toddb at toddbailey.net> wrote:

> Anyone ?  anyone at all care to step in on this ? 
>  
> -------- Original Message --------
> Subject: [Freeswitch-users] Clarification needed on setting up group
> calling parameters
> From: <toddb at toddbailey.net>
> Date: Wed, June 20, 2012 8:08 am
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> 
>  Hi All,
>  
>  
> read through the documentation a few times in setting up call groups and can't seem to make it happen.
>  
> 1 problem is the doc's don't make it easy by stating which file I need to be editing.
>  
>  
> Perhaps a call group isn't what I need,  so here the scenario
>  
> I'd like to set up in-bound calls to automatically go to extensions 1002 - 1007 where the fist to answer gets the call.
>  
> I'm using a spa3102 to route ptsn calls to voip extension 1001,  so ext 1001 needs to ring on 1002-1007.
>  
> Q: what method does this and what files & locations do I need to edit?
>  
>  
> thx.
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From sherifomran2000 at yahoo.com  Fri Jun 22 23:37:23 2012
From: sherifomran2000 at yahoo.com (Sherif Omran)
Date: Fri, 22 Jun 2012 12:37:23 -0700 (PDT)
Subject: [Freeswitch-users] Invalid application transfer help
In-Reply-To: <F8AE005E-E72A-437A-9F44-4A54BF0568B7@visionutveckling.se>
Message-ID: <1340393843.11740.YahooMailClassic@web110804.mail.gq1.yahoo.com>

how can I load it? should i recompile and reinstall

thanks
Sherif


--- On Fri, 6/22/12, Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> From: Peter Olsson <peter.olsson at visionutveckling.se>
> Subject: Re: [Freeswitch-users] Invalid application transfer help
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Friday, June 22, 2012, 1:18 PM
> Seems you don't have mod_dptools
> loaded anymore.
> 
> /Peter
> 
> 22 jun 2012 kl. 11:45 skrev "Sherif Omran" <sherifomran2000 at yahoo.com<mailto:sherifomran2000 at yahoo.com>>:
> 
> Hi Guys,
> 
> My server was working fine till today, It does not transfer
> any call. When I call an extension i get? a not
> connected tone and in the log I see invalid application
> 
> 2012-06-22 09:37:26.414815 [ERR] switch_core_session.c:2156
> Invalid Application transfer
> 
> I wonder what this could be? I was trying yesterday to
> install some packages. Could this be related to that?
> 
> kind regards,
> Sherif
> 
> 
> !DSPAM:4fe43b9a32761615820512!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> 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
> 
> 
> !DSPAM:4fe43b9a32761615820512!
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From Hector.Geraldino at ipsoft.com  Sat Jun 23 01:08:52 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Fri, 22 Jun 2012 17:08:52 -0400
Subject: [Freeswitch-users] Invalid application transfer help
In-Reply-To: <1340393843.11740.YahooMailClassic@web110804.mail.gq1.yahoo.com>
References: <F8AE005E-E72A-437A-9F44-4A54BF0568B7@visionutveckling.se>
	<1340393843.11740.YahooMailClassic@web110804.mail.gq1.yahoo.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD022F4EBD50@NY1-EXMB-01.ip-soft.net>

Check if module is loaded:

>mod_exists mod_dptools

If not, load it:

>load mod_dptools

Or reload it:

>reload mod_dptools

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sherif Omran
Sent: Friday, June 22, 2012 3:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Invalid application transfer help

how can I load it? should i recompile and reinstall

thanks
Sherif


--- On Fri, 6/22/12, Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> From: Peter Olsson <peter.olsson at visionutveckling.se>
> Subject: Re: [Freeswitch-users] Invalid application transfer help
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Friday, June 22, 2012, 1:18 PM
> Seems you don't have mod_dptools
> loaded anymore.
> 
> /Peter
> 
> 22 jun 2012 kl. 11:45 skrev "Sherif Omran" <sherifomran2000 at yahoo.com<mailto:sherifomran2000 at yahoo.com>>:
> 
> Hi Guys,
> 
> My server was working fine till today, It does not transfer any call. 
> When I call an extension i get? a not connected tone and in the log I 
> see invalid application
> 
> 2012-06-22 09:37:26.414815 [ERR] switch_core_session.c:2156 Invalid 
> Application transfer
> 
> I wonder what this could be? I was trying yesterday to install some 
> packages. Could this be related to that?
> 
> kind regards,
> Sherif
> 
> 
> !DSPAM:4fe43b9a32761615820512!
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
> 
>  
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.fr
> eeswitch.org> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
> 
> 
> !DSPAM:4fe43b9a32761615820512!
> 
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
>  
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users 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
> 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


From elliott at zoogmedia.com  Sat Jun 23 02:28:49 2012
From: elliott at zoogmedia.com (Elliott Vogel)
Date: Fri, 22 Jun 2012 22:28:49 +0000
Subject: [Freeswitch-users] dtmf dialplan
Message-ID: <461821C20115054F9056ED0E52EAC4DA104B8CCB@SN2PRD0710MB384.namprd07.prod.outlook.com>

How can i transfer a call if '#' is pressed while a audio file is playing from a dialplan?

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From alex at thewinelake.com  Sat Jun 23 02:48:40 2012
From: alex at thewinelake.com (Alexander Lake)
Date: Fri, 22 Jun 2012 23:48:40 +0100
Subject: [Freeswitch-users] dtmf dialplan
In-Reply-To: <461821C20115054F9056ED0E52EAC4DA104B8CCB@SN2PRD0710MB384.namprd07.prod.outlook.com>
References: <461821C20115054F9056ED0E52EAC4DA104B8CCB@SN2PRD0710MB384.namprd07.prod.outlook.com>
Message-ID: <DBC65773-98F1-4F68-ABC5-66960B8EA093@thewinelake.com>

You could either setup a callback, or use PlayAndGetDigits

On 22 Jun 2012, at 23:28, Elliott Vogel wrote:

> How can i transfer a call if '#' is pressed while a audio file is playing from a dialplan?
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mario_fs at mgtech.com  Sat Jun 23 03:11:12 2012
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 22 Jun 2012 16:11:12 -0700
Subject: [Freeswitch-users] Alert-Info header not understood by SPA
	phones
In-Reply-To: <FAA4F224-4805-4A5F-A42F-EB0CA683FC44@imtelecom.fr>
References: <A0B516FD-73C1-4F9E-B928-4C4AA54BD393@imtelecom.fr>
	<39F586EF-0B6B-4243-A385-900D7CD870D7@mgtech.com>
	<FAA4F224-4805-4A5F-A42F-EB0CA683FC44@imtelecom.fr>
Message-ID: <98DE356E-FEFF-4732-906C-BADBF119E891@mgtech.com>

Interesting, I could not get the "name" to work, my syntax also allows you to set the W and C variables from FreeSwitch instead of using the ones set on the phone which is more flexible.

Since you have SPA962s I would like to ask a question regarding a problem I can't solve: 30-40 percent of the time when an SPA962 answers an outside call, some of the other SPA962s keep ringing for a while or you answer/hangup. They are not using shared lines. Ever run into that? Been months on this issue and no one here responded to the issue when I posted it. Thanks,
Mario G

On Jun 22, 2012, at 11:17 AM, Vallimamod ABDULLAH wrote:

> Hi Mario,
> 
> Thanks for your answer. After your mail I have tested with a SPA962 and indeed it works with both your syntax and mine!
> So looks like cisco have changed the way it works on their new SPA50XG models?
> 
> - vma.
> 
> 
> On Jun 22, 2012, at 6:09 PM, Mario G wrote:
> 
>> I don't know how similar they are but on my SPA962 the line below works. I think you need to use the settings of the name (office) not the name.
>> Mario G
>> 
>> <action application="export" data="alert_info=n=Simple-5;w=2;c=5"/>
>> 
>> On Jun 22, 2012, at 8:44 AM, Vallimamod ABDULLAH wrote:
>> 
>>> Hi List,
>>> 
>>> I am trying to setup a custom ringtone for internal calls for cisco SPA504G through the Alert-Info header. 
>>> I have put <action application="export" data="alert_info=Office"/> before the bridge as stated on the wiki but the sip header does not seems to be recognized by the phone.  The invite the alert-info header is sent correctly in the sip invite but it contains extra brackets (Alert-Info: <Office>) wich confuses the phone as it is expecting "Alert-Info: Office".
>>> I have also tried <action application="set" data="sip_h_Alert-Info=Office"/> but it does not make any difference.
>>> 
>>> Is there any way to remove these brackets ?
>>> 
>>> Thanks you!
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From tahir at ictinnovations.com  Sat Jun 23 05:11:23 2012
From: tahir at ictinnovations.com (tahir almas)
Date: Sat, 23 Jun 2012 06:11:23 +0500
Subject: [Freeswitch-users] ICTFAX Version 2.0 beta released ,
	an Open Source FOIP software solution
Message-ID: <CAOL0t+LZ4wV9P2+3SLneQgLQ0x2JvWLt9uxu0-YmkkQoiLWeQQ@mail.gmail.com>

ICT Innovations  achieved big milestone by releasing   ICTFAX Ver 2.0 beta,
an open source Foip software supporting Email to Fax , Fax to Email and Web
to Fax

Earlier version of ICTFAX was based on Asterisk and  Drupal Ver  4.7
causing issues to new users , We at ICT Innovations devoted months on this
project and rewritten all code in Drupal 7.0 as well as ported back-end to
Plivo communication framework , Now ICTFAX using open source Freeswitch as
back-end communication engine making it advance and stable open source FOIP
solution

For more information, please visit  http://www.ictfax.org

*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT

****************************************************************************************************************
NOTICE OF CONFIDENTIALITY
This communication including any information transmitted with it is
intended only for the use of the addressees and is confidential and may be
protected by legal privilege . If you are not an intended recipient, be
aware that any disclosure, copying, distribution or use of this e-mail or
any attachment is prohibited. If you have received this e-mail in error,
please notify us immediately by returning it to the sender and delete this
copy from your system. Thank you for your cooperation.
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From saami_mh at ymail.com  Sat Jun 23 09:39:50 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Fri, 22 Jun 2012 22:39:50 -0700 (PDT)
Subject: [Freeswitch-users] is it possible to use lua in mod_xml_curl ?
Message-ID: <1340429990.6414.YahooMailNeo@web120105.mail.ne1.yahoo.com>

the supported languages in mod_xml_cur include:
perl,python,php,c++,c# , but can i use lua ?
lua is meet our requirements for implementing mod_xml_curl
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From saami_mh at ymail.com  Sat Jun 23 09:48:14 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Fri, 22 Jun 2012 22:48:14 -0700 (PDT)
Subject: [Freeswitch-users] some questions on mod_xml_curl?
Message-ID: <1340430494.72582.YahooMailNeo@web120106.mail.ne1.yahoo.com>

hi,
i have some questions on mod_xml_curl ..
1- for implementing mod_xml_cur must to running webserver?
2-is it possible to use mysql directly without webserver ,like ASterisk realtime architecture?
in other hand after sip request use sql request instead of http request?
3- if we want to use python or c++ or perl langueges in mod_xml_curl so the werb server should to be implement via php
because using python we couldn't implement web server ?
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From sherifomran2000 at yahoo.com  Sat Jun 23 12:05:29 2012
From: sherifomran2000 at yahoo.com (Sherif Omran)
Date: Sat, 23 Jun 2012 01:05:29 -0700 (PDT)
Subject: [Freeswitch-users] Invalid application transfer help
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD022F4EBD50@NY1-EXMB-01.ip-soft.net>
Message-ID: <1340438729.30883.YahooMailClassic@web110813.mail.gq1.yahoo.com>

I get the following ... Please take a look

freeswitch at internal> mod_exists
-ERR mod_exists Command not found!

freeswitch at internal> mod_exists mod_dptools
-ERR mod_exists mod_dptools Command not found!

freeswitch at internal> load mod_dptools
+OK Reloading XML
-ERR [module load file routine returned an error]

2012-06-23 08:04:58.914388 [INFO] mod_enum.c:812 ENUM Reloaded
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449 Adding tone_descriptor: 1
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: CED_TONE(0)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: SIT(1)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (950, 0, 256, 400)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (1400, 0, 256, 400)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (1800, 0, 256, 400)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: REORDER_TONE(2)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (480, 620, 224, 272)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (0, 0, 224, 272)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: BUSY_TONE(3)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (480, 620, 464, 516)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (0, 0, 464, 516)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449 Adding tone_descriptor: 44
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: CED_TONE(0)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: SIT(1)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (950, 0, 256, 400)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (1400, 0, 256, 400)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (1800, 0, 256, 400)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: REORDER_TONE(2)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 368, 416)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 336, 368)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 256, 288)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 512, 544)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: BUSY_TONE(3)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449 Adding tone_descriptor: 49
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: CED_TONE(0)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: CED_TONE(0), element (2100, 0, 500, 0)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: SIT(1)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (900, 0, 256, 400)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (1400, 0, 256, 400)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (1800, 0, 256, 400)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: REORDER_TONE(2)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (425, 0, 224, 272)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (0, 0, 224, 272)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: BUSY_TONE(3)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (425, 0, 464, 516)
2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (0, 0, 464, 516)
2012-06-23 08:04:58.914388 [INFO] switch_time.c:1123 Timezone reloaded 530 definitions
2012-06-23 08:04:58.914388 [CRIT] switch_loadable_module.c:1300 Error Loading module /home/vBilling/freeswitch/mod/mod_dptools.so
**/home/vBilling/freeswitch/mod/mod_dptools.so: undefined symbol: switch_core_media_bug_remove_all**



Thanks


--- On Sat, 6/23/12, Hector Geraldino <Hector.Geraldino at ipsoft.com> wrote:

> From: Hector Geraldino <Hector.Geraldino at ipsoft.com>
> Subject: Re: [Freeswitch-users] Invalid application transfer help
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Saturday, June 23, 2012, 12:08 AM
> Check if module is loaded:
> 
> >mod_exists mod_dptools
> 
> If not, load it:
> 
> >load mod_dptools
> 
> Or reload it:
> 
> >reload mod_dptools
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
> On Behalf Of Sherif Omran
> Sent: Friday, June 22, 2012 3:37 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Invalid application transfer
> help
> 
> how can I load it? should i recompile and reinstall
> 
> thanks
> Sherif
> 
> 
> --- On Fri, 6/22/12, Peter Olsson <peter.olsson at visionutveckling.se>
> wrote:
> 
> > From: Peter Olsson <peter.olsson at visionutveckling.se>
> > Subject: Re: [Freeswitch-users] Invalid application
> transfer help
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Date: Friday, June 22, 2012, 1:18 PM
> > Seems you don't have mod_dptools
> > loaded anymore.
> > 
> > /Peter
> > 
> > 22 jun 2012 kl. 11:45 skrev "Sherif Omran" <sherifomran2000 at yahoo.com<mailto:sherifomran2000 at yahoo.com>>:
> > 
> > Hi Guys,
> > 
> > My server was working fine till today, It does not
> transfer any call. 
> > When I call an extension i get? a not connected tone
> and in the log I 
> > see invalid application
> > 
> > 2012-06-22 09:37:26.414815 [ERR]
> switch_core_session.c:2156 Invalid 
> > Application transfer
> > 
> > I wonder what this could be? I was trying yesterday to
> install some 
> > packages. Could this be related to that?
> > 
> > kind regards,
> > Sherif
> > 
> > 
> > !DSPAM:4fe43b9a32761615820512!
> >
> ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> > http://www.freeswitchsolutions.com
> > 
> > FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > Join Us At ClueCon - Aug 7-9, 2012
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.fr
> > eeswitch.org> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
> > 
> > 
> > !DSPAM:4fe43b9a32761615820512!
> > 
> >
> ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > Join Us At ClueCon - Aug 7-9, 2012
> > 
> > FreeSWITCH-users 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
> > 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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> 
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
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> 
> Join Us At ClueCon - Aug 7-9, 2012
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> 


From peter.olsson at visionutveckling.se  Sat Jun 23 15:53:02 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 23 Jun 2012 11:53:02 +0000
Subject: [Freeswitch-users] Invalid application transfer help
In-Reply-To: <1340438729.30883.YahooMailClassic@web110813.mail.gq1.yahoo.com>
References: <6A6B4C284AD15042B429EB9D904544AD022F4EBD50@NY1-EXMB-01.ip-soft.net>,
	<1340438729.30883.YahooMailClassic@web110813.mail.gq1.yahoo.com>
Message-ID: <B5E2C029-C331-42A5-B4FE-80709E668683@visionutveckling.se>

You must make sure you use the same version of the FS core library and it's modules. You have an error for a missing symbol when loading the module.

/Peter

23 jun 2012 kl. 10:12 skrev "Sherif Omran" <sherifomran2000 at yahoo.com>:

> I get the following ... Please take a look
> 
> freeswitch at internal> mod_exists
> -ERR mod_exists Command not found!
> 
> freeswitch at internal> mod_exists mod_dptools
> -ERR mod_exists mod_dptools Command not found!
> 
> freeswitch at internal> load mod_dptools
> +OK Reloading XML
> -ERR [module load file routine returned an error]
> 
> 2012-06-23 08:04:58.914388 [INFO] mod_enum.c:812 ENUM Reloaded
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449 Adding tone_descriptor: 1
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: CED_TONE(0)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: CED_TONE(0), element (2100, 0, 500, 0)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: SIT(1)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (950, 0, 256, 400)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (1400, 0, 256, 400)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: SIT(1), element (1800, 0, 256, 400)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: REORDER_TONE(2)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (480, 620, 224, 272)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: REORDER_TONE(2), element (0, 0, 224, 272)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 1, tone: BUSY_TONE(3)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (480, 620, 464, 516)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (0, 0, 464, 516)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449 Adding tone_descriptor: 44
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: CED_TONE(0)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: CED_TONE(0), element (2100, 0, 500, 0)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: SIT(1)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (950, 0, 256, 400)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (1400, 0, 256, 400)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: SIT(1), element (1800, 0, 256, 400)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: REORDER_TONE(2)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 368, 416)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 336, 368)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (400, 0, 256, 288)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: REORDER_TONE(2), element (0, 0, 512, 544)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 44, tone: BUSY_TONE(3)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (400, 0, 352, 384)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0, 0, 352, 384)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449 Adding tone_descriptor: 49
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: CED_TONE(0)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: CED_TONE(0), element (2100, 0, 500, 0)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: SIT(1)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (900, 0, 256, 400)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (1400, 0, 256, 400)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: SIT(1), element (1800, 0, 256, 400)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: REORDER_TONE(2)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (425, 0, 224, 272)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: REORDER_TONE(2), element (0, 0, 224, 272)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: 49, tone: BUSY_TONE(3)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (425, 0, 464, 516)
> 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (0, 0, 464, 516)
> 2012-06-23 08:04:58.914388 [INFO] switch_time.c:1123 Timezone reloaded 530 definitions
> 2012-06-23 08:04:58.914388 [CRIT] switch_loadable_module.c:1300 Error Loading module /home/vBilling/freeswitch/mod/mod_dptools.so
> **/home/vBilling/freeswitch/mod/mod_dptools.so: undefined symbol: switch_core_media_bug_remove_all**
> 
> 
> 
> Thanks
> 
> 
> --- On Sat, 6/23/12, Hector Geraldino <Hector.Geraldino at ipsoft.com> wrote:
> 
>> From: Hector Geraldino <Hector.Geraldino at ipsoft.com>
>> Subject: Re: [Freeswitch-users] Invalid application transfer help
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Date: Saturday, June 23, 2012, 12:08 AM
>> Check if module is loaded:
>> 
>>> mod_exists mod_dptools
>> 
>> If not, load it:
>> 
>>> load mod_dptools
>> 
>> Or reload it:
>> 
>>> reload mod_dptools
>> 
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
>> On Behalf Of Sherif Omran
>> Sent: Friday, June 22, 2012 3:37 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Invalid application transfer
>> help
>> 
>> how can I load it? should i recompile and reinstall
>> 
>> thanks
>> Sherif
>> 
>> 
>> --- On Fri, 6/22/12, Peter Olsson <peter.olsson at visionutveckling.se>
>> wrote:
>> 
>>> From: Peter Olsson <peter.olsson at visionutveckling.se>
>>> Subject: Re: [Freeswitch-users] Invalid application
>> transfer help
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Date: Friday, June 22, 2012, 1:18 PM
>>> Seems you don't have mod_dptools
>>> loaded anymore.
>>> 
>>> /Peter
>>> 
>>> 22 jun 2012 kl. 11:45 skrev "Sherif Omran" <sherifomran2000 at yahoo.com<mailto:sherifomran2000 at yahoo.com>>:
>>> 
>>> Hi Guys,
>>> 
>>> My server was working fine till today, It does not
>> transfer any call. 
>>> When I call an extension i get  a not connected tone
>> and in the log I 
>>> see invalid application
>>> 
>>> 2012-06-22 09:37:26.414815 [ERR]
>> switch_core_session.c:2156 Invalid 
>>> Application transfer
>>> 
>>> I wonder what this could be? I was trying yesterday to
>> install some 
>>> packages. Could this be related to that?
>>> 
>>> kind regards,
>>> Sherif
>>> 
>>> 
>>> 
>>> 
>> ______________________________________________________________________
>>> ___ Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>> 
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> Join Us At ClueCon - Aug 7-9, 2012
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.fr
>>> eeswitch.org> 
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>> rs
>>> http://www.freeswitch.org
>>> 
>>> 
>>> !DSPAM:4fe43b9a32761615820512!
>>> 
>>> 
>> ______________________________________________________________________
>>> ___ Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> Join Us At ClueCon - Aug 7-9, 2012
>>> 
>>> FreeSWITCH-users 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
>>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
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>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
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>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> 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
>> http://www.freeswitch.org
>> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> !DSPAM:4fe5774732761352082344!
> 


From steveayre at gmail.com  Sat Jun 23 16:11:23 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 23 Jun 2012 13:11:23 +0100
Subject: [Freeswitch-users] some questions on mod_xml_curl?
In-Reply-To: <1340430494.72582.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1340430494.72582.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <B2D8FFB9-807E-48F9-A13C-0FBEE70BE324@gmail.com>

> 1- for implementing mod_xml_cur must to running webserver?


The purpose of mod_xml_curl is to download configuration files via HTTP, so yes it does require running a webserver. That doesn't have to run on the same machine though, it just needs to be reachable.

> 3- if we want to use python or c++ or perl langueges in mod_xml_curl so the werb server should to be implement via php
> because using python we couldn't implement web server ?

You can implement the scripts running on the webserver in any language you like (any of the above including python can be used), the script just needs to capable of outputting the correct XML.

> 2-is it possible to use mysql directly without webserver ,like ASterisk realtime architecture?
> in other hand after sip request use sql request instead of http request 

This isn't enough information for us to answer your question. Some parts of FS can indeed talk to a DB directly, others generally shouldn't (CDR). What specifically are you wanting to store in the database and access directly from FS?


Steve on iPhone



On 23 Jun 2012, at 06:48, Samira Mh <saami_mh at ymail.com> wrote:

> hi,
> i have some questions on mod_xml_curl ..
> 1- for implementing mod_xml_cur must to running webserver?
> 2-is it possible to use mysql directly without webserver ,like ASterisk realtime architecture?
> in other hand after sip request use sql request instead of http request 
> 3- if we want to use python or c++ or perl langueges in mod_xml_curl so the werb server should to be implement via php
> because using python we couldn't implement web server ?
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveayre at gmail.com  Sat Jun 23 16:12:45 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 23 Jun 2012 13:12:45 +0100
Subject: [Freeswitch-users] is it possible to use lua in mod_xml_curl ?
In-Reply-To: <1340429990.6414.YahooMailNeo@web120105.mail.ne1.yahoo.com>
References: <1340429990.6414.YahooMailNeo@web120105.mail.ne1.yahoo.com>
Message-ID: <073BDA71-2741-42A4-A85E-13A4A4846C68@gmail.com>

This isn't a FS question.

You can use any language your webserver allows you to use. FS only cares about the XML output.

If your webserver supports lua, then yes you can use lua.

Steve on iPhone



On 23 Jun 2012, at 06:39, Samira Mh <saami_mh at ymail.com> wrote:

> the supported languages in mod_xml_cur include:
> perl,python,php,c++,c# , but can i use lua ?
> lua is meet our requirements for implementing mod_xml_curl
> 
>  
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From b2m at a-cti.com  Sat Jun 23 16:34:03 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Sat, 23 Jun 2012 18:04:03 +0530
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CALXCt0pad5LY3N9ecsAzjdmqZRPMqCeQdBwbzhoQLUY73T22eA@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
	<CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
	<CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>
	<CAP2fbbeCxNssmEXrmwxd9Z=ddz+CNEzASQ2QRpzispMPt9cjzA@mail.gmail.com>
	<CALXCt0oLTuNuL4XO92BFwMUWCs+tp_fcJxjjtib2ffktdFMLCw@mail.gmail.com>
	<CAP2fbbfXAk9WKwKFQ0_iOx8-WEYUqbLw9zrhvpZs1Wo96oQjZA@mail.gmail.com>
	<CAP2fbbfYdUrE=GYgVH5WwcbgW7s4=_f+81=DM=aK8dpYJ+kQnw@mail.gmail.com>
	<CALXCt0pad5LY3N9ecsAzjdmqZRPMqCeQdBwbzhoQLUY73T22eA@mail.gmail.com>
Message-ID: <CAP2fbbcAuJ6t_xw2CKDKeqf6FLE=MZ1QbC9wA4rB5xQPEXSVvA@mail.gmail.com>

Sorry, I could not trace it yet, it would be very helpful if you find time
for me.

Thanks,
Bala

On Thu, Jun 21, 2012 at 11:25 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> On Thu, Jun 21, 2012 at 7:45 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>
>> I just changed  <action application="set" data="record_sample_rate=16000"/>,
>> still looks same.
>>
>>
> Just for the sake of it, try with record_sample_rate=8000 and check how
> it is... Anyway, problem is there, somewhere.
>
> Maybe you can post again to the mailing list putting: "recording problem"
> in the subject (eg: without naming mod_skypopen), so maybe someone can
> chime in.
>
> I cannot check it out right now, if You have not solved in 12 hours, write
> again, I'll look into that.
>
> -giovanni
>
>
>
>>
>> Thanks,
>> Bala
>>
>>
>> On Thu, Jun 21, 2012 at 11:07 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>
>>> I am afraid, I did not do anything outside freeswitch.
>>>
>>> just have the following for recording in my dialplan
>>>
>>>
>>>     <action application="set" data="RECORD_COPYRIGHT=(c) freeswitch"/>
>>>     <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
>>>     <action application="set" data="RECORD_ARTIST=auto"/>
>>>     <action application="set" data="RECORD_COMMENT=auto recording"/>
>>>     <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d
>>> %H:%M)}"/>
>>>     <action application="set" data="RECORD_STEREO=true"/>
>>>     <action application="set" data="record_sample_rate=11000"/>
>>>     <action application="set" data="media_bug_answer_req=true"/>
>>>     <action application="record_session"
>>> data="$${base_dir}/recordings/${uuid}.mp3"/>
>>>     <action application="set" data="ignore_early_media=false"/>
>>>     <action application="set"
>>> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,ORIGINATOR_CANCEL,CANCEL,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>
>>>
>>>
>>> Thanks,
>>> Bala
>>>
>>>
>>> On Thu, Jun 21, 2012 at 10:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com
>>> > wrote:
>>>
>>>> On Thu, Jun 21, 2012 at 7:15 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>>>
>>>>> Thanks for your help!!
>>>>>
>>>>> Up to my knowledge there was no change.
>>>>>
>>>>> After that I tried re-installed FS(with latest build) Skype and
>>>>> changed EC2 Instance(different size) but for some reason its still same. I
>>>>> guess some settings somewhere changed not sure where it is.
>>>>>
>>>>>
>>>> I just checked your mp3, it says it's recorded at 8khz, but it's not
>>>> true. It's 16khz. If you play it at 16khz it sounds ok. So, you have to
>>>> look into the mp3 converter you used and configure it to not lie about the
>>>> samples per second ;).
>>>>
>>>> -giovanni
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>>
>>>>> Thanks,
>>>>> Bala
>>>>>
>>>>> On Thu, Jun 21, 2012 at 10:32 PM, Giovanni Maruzzelli <
>>>>> gmaruzz at gmail.com> wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Jun 21, 2012 at 6:50 PM, Balamurugan Mahendran <b2m at a-cti.com
>>>>>> > wrote:
>>>>>>
>>>>>>> Is there a wiki link for these parameters? Thanks!!
>>>>>>>
>>>>>>> Actually it was very good till last week(for about 6
>>>>>>> months), suddenly its broken. I could not trace it yet
>>>>>>>
>>>>>>>
>>>>>> Have you reinstalled something?
>>>>>>
>>>>>> Re-compiled things, or updated config files?
>>>>>>
>>>>>> If you have changed nothing, maybe the problem is elsewhere.
>>>>>>
>>>>>> Just for test, you can:
>>>>>>
>>>>>> 1) from FS cli, unload mod_skypopen
>>>>>> 2) from ssh killall Xvfb (this will kill all the skype instances too)
>>>>>> 3) from ssh relaunch the Xvfb-skype clients (with the script)
>>>>>> 4) from FS cli load mod_skypopen
>>>>>>
>>>>>> check if it is recording correctly now.
>>>>>>
>>>>>> That test is good if you changed nothing.
>>>>>>
>>>>>> If you changed something, revert that change :).
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>> Thanks,
>>>>>>> Bala
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Jun 21, 2012 at 9:59 PM, Giovanni Maruzzelli <
>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran <
>>>>>>>> b2m at a-cti.com> wrote:
>>>>>>>>
>>>>>>>>> All,
>>>>>>>>>
>>>>>>>>> Is there a way to control recording speed for mod_skypopen? I was
>>>>>>>>> trying to figure out the problem of fast recording/playing
>>>>>>>>>
>>>>>>>>> Please find the attached file
>>>>>>>>>
>>>>>>>>
>>>>>>>> Balamurugan,
>>>>>>>>
>>>>>>>> be aware that mod_skypopen operate at 16khz signed linear (let's
>>>>>>>> say, at HD audio level; double the number of samples of a normal PCMU or
>>>>>>>> GSM or g711 or G729 call).
>>>>>>>>
>>>>>>>> So, maybe you have to tweak the recording or playback parameters
>>>>>>>> taking this into account.
>>>>>>>>
>>>>>>>> -giovanni
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>>
>>>>>>>>> Thanks,
>>>>>>>>> Bala
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at 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
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at 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
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 gmail.com  Sat Jun 23 16:52:50 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 23 Jun 2012 14:52:50 +0200
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CAP2fbbcAuJ6t_xw2CKDKeqf6FLE=MZ1QbC9wA4rB5xQPEXSVvA@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
	<CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
	<CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>
	<CAP2fbbeCxNssmEXrmwxd9Z=ddz+CNEzASQ2QRpzispMPt9cjzA@mail.gmail.com>
	<CALXCt0oLTuNuL4XO92BFwMUWCs+tp_fcJxjjtib2ffktdFMLCw@mail.gmail.com>
	<CAP2fbbfXAk9WKwKFQ0_iOx8-WEYUqbLw9zrhvpZs1Wo96oQjZA@mail.gmail.com>
	<CAP2fbbfYdUrE=GYgVH5WwcbgW7s4=_f+81=DM=aK8dpYJ+kQnw@mail.gmail.com>
	<CALXCt0pad5LY3N9ecsAzjdmqZRPMqCeQdBwbzhoQLUY73T22eA@mail.gmail.com>
	<CAP2fbbcAuJ6t_xw2CKDKeqf6FLE=MZ1QbC9wA4rB5xQPEXSVvA@mail.gmail.com>
Message-ID: <CALXCt0oC8OSO=QiFFc7MpaM_sp9KXGdxDSnyrr5Ptk1PoUAYEw@mail.gmail.com>

Please send me in private all I need to exactly duplicate your problem: eg:
config file, dialplan, etc. All. thx :)


On Sat, Jun 23, 2012 at 2:34 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:

> Sorry, I could not trace it yet, it would be very helpful if you find time
> for me.
>
> Thanks,
> Bala
>
>
> On Thu, Jun 21, 2012 at 11:25 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>
>> On Thu, Jun 21, 2012 at 7:45 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>
>>> I just changed  <action application="set" data="record_sample_rate=16000"/>,
>>> still looks same.
>>>
>>>
>> Just for the sake of it, try with record_sample_rate=8000 and check how
>> it is... Anyway, problem is there, somewhere.
>>
>> Maybe you can post again to the mailing list putting: "recording problem"
>> in the subject (eg: without naming mod_skypopen), so maybe someone can
>> chime in.
>>
>> I cannot check it out right now, if You have not solved in 12 hours,
>> write again, I'll look into that.
>>
>> -giovanni
>>
>>
>>
>>>
>>> Thanks,
>>> Bala
>>>
>>>
>>> On Thu, Jun 21, 2012 at 11:07 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>>
>>>> I am afraid, I did not do anything outside freeswitch.
>>>>
>>>> just have the following for recording in my dialplan
>>>>
>>>>
>>>>     <action application="set" data="RECORD_COPYRIGHT=(c) freeswitch"/>
>>>>     <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
>>>>     <action application="set" data="RECORD_ARTIST=auto"/>
>>>>     <action application="set" data="RECORD_COMMENT=auto recording"/>
>>>>     <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d
>>>> %H:%M)}"/>
>>>>     <action application="set" data="RECORD_STEREO=true"/>
>>>>     <action application="set" data="record_sample_rate=11000"/>
>>>>     <action application="set" data="media_bug_answer_req=true"/>
>>>>     <action application="record_session"
>>>> data="$${base_dir}/recordings/${uuid}.mp3"/>
>>>>     <action application="set" data="ignore_early_media=false"/>
>>>>     <action application="set"
>>>> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,ORIGINATOR_CANCEL,CANCEL,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>
>>>>
>>>>
>>>> Thanks,
>>>> Bala
>>>>
>>>>
>>>> On Thu, Jun 21, 2012 at 10:59 PM, Giovanni Maruzzelli <
>>>> gmaruzz at gmail.com> wrote:
>>>>
>>>>> On Thu, Jun 21, 2012 at 7:15 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>>>>
>>>>>> Thanks for your help!!
>>>>>>
>>>>>> Up to my knowledge there was no change.
>>>>>>
>>>>>> After that I tried re-installed FS(with latest build) Skype and
>>>>>> changed EC2 Instance(different size) but for some reason its still same. I
>>>>>> guess some settings somewhere changed not sure where it is.
>>>>>>
>>>>>>
>>>>> I just checked your mp3, it says it's recorded at 8khz, but it's not
>>>>> true. It's 16khz. If you play it at 16khz it sounds ok. So, you have to
>>>>> look into the mp3 converter you used and configure it to not lie about the
>>>>> samples per second ;).
>>>>>
>>>>> -giovanni
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>>
>>>>>> Thanks,
>>>>>> Bala
>>>>>>
>>>>>> On Thu, Jun 21, 2012 at 10:32 PM, Giovanni Maruzzelli <
>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Jun 21, 2012 at 6:50 PM, Balamurugan Mahendran <
>>>>>>> b2m at a-cti.com> wrote:
>>>>>>>
>>>>>>>> Is there a wiki link for these parameters? Thanks!!
>>>>>>>>
>>>>>>>> Actually it was very good till last week(for about 6
>>>>>>>> months), suddenly its broken. I could not trace it yet
>>>>>>>>
>>>>>>>>
>>>>>>> Have you reinstalled something?
>>>>>>>
>>>>>>> Re-compiled things, or updated config files?
>>>>>>>
>>>>>>> If you have changed nothing, maybe the problem is elsewhere.
>>>>>>>
>>>>>>> Just for test, you can:
>>>>>>>
>>>>>>> 1) from FS cli, unload mod_skypopen
>>>>>>> 2) from ssh killall Xvfb (this will kill all the skype instances too)
>>>>>>> 3) from ssh relaunch the Xvfb-skype clients (with the script)
>>>>>>> 4) from FS cli load mod_skypopen
>>>>>>>
>>>>>>> check if it is recording correctly now.
>>>>>>>
>>>>>>> That test is good if you changed nothing.
>>>>>>>
>>>>>>> If you changed something, revert that change :).
>>>>>>>
>>>>>>> -giovanni
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> Thanks,
>>>>>>>> Bala
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, Jun 21, 2012 at 9:59 PM, Giovanni Maruzzelli <
>>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran <
>>>>>>>>> b2m at a-cti.com> wrote:
>>>>>>>>>
>>>>>>>>>> All,
>>>>>>>>>>
>>>>>>>>>> Is there a way to control recording speed for mod_skypopen? I was
>>>>>>>>>> trying to figure out the problem of fast recording/playing
>>>>>>>>>>
>>>>>>>>>> Please find the attached file
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Balamurugan,
>>>>>>>>>
>>>>>>>>> be aware that mod_skypopen operate at 16khz signed linear (let's
>>>>>>>>> say, at HD audio level; double the number of samples of a normal PCMU or
>>>>>>>>> GSM or g711 or G729 call).
>>>>>>>>>
>>>>>>>>> So, maybe you have to tweak the recording or playback parameters
>>>>>>>>> taking this into account.
>>>>>>>>>
>>>>>>>>> -giovanni
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Thanks,
>>>>>>>>>> Bala
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.com
>>>>>>>>>>
>>>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at 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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at 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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
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From frankjr at mcpeekdodge.com  Sat Jun 23 18:29:49 2012
From: frankjr at mcpeekdodge.com (Frank Busalacchi Jr)
Date: Sat, 23 Jun 2012 14:29:49 +0000
Subject: [Freeswitch-users] Sharing:  Call Intercept
Message-ID: <C066384B0C1F1E4B9B870F050D7F1A9D402150F5@ADAMA.mcpeekdodge.local>

Over the last 6 months, I have received all kinds of support and help from the freeswitch community (both IRC and this list) in learning to control this incredible piece of software.  Im truly grateful, thanks everyone.



I run freeswitch at my car dealership.  We have approximately 40 sip extensions, an ATT voip connection, and 6 analog outbound fax lines provided by freetdm.  Prior to freeswitch, I was running an old Lucent/Avaya Merlin "key" system.  One of the features of the Merlin was that it was pretty easy to do "directed callpick".  In essence, you would dial *7 + extension you wanted to pickup.  Provided the call was either on hold or ringing at that extension, you would pick it up.  When I first started using Freeswitch, I was very disappointed in the directed callpick.  Before I realized that the call pickup that was built into the sample dialplan was really just an example to show a simple possibility for using intercept.  I was convinced that it was a totally useful feature.  I couldn't believe that was as good as it got...  My problem with the directed pickup that is in the sample dialplan is that it can steal a live call right from another extension, as well as it really only works with the most recent call on that extension.



Then I woke up, and rolled my own 8-) .  In an effort to provide just another example, I wanted to share the piece of code I've developed.  I will warn that this was the first perl script I have ever written, and probably is pretty ugly, but it seems to work perfectly in our environment.  Bottom line, do what you want with it...I'm just trying to help other beginner's like myself:



To all beginners:  You will need to build and enable mod_perl for this to work...



First, we need an entry in your XML dialplan to see the user's request to interecept a "held" or "ringing" extension.  The only 4 digit extensions at my company are internal ones so I match on ** + 4-digit-extension.



In the xML dialplan:



    <extension name="intercept-ext">
      <condition field="destination_number" expression="^\*\*(\d{4})$">
        <action application="answer"/>
        <action application="perl" data="/usr/local/freeswitch/scripts/myintercept.pl $1"/>
        <action application="sleep" data="2000"/>
      </condition>
    </extension>

Then, we need the actual perl script:

/usr/local/freeswitch/scripts/myintercept.pl:

# This is code to allow us to use intercept to grab only calls that are on hold or ringing.
#
# by Frank Busalacchi 2012-06-13

use strict;
use Switch;
use Data::Dumper;
use XML::Simple;
use POSIX qw(strftime);
our $session;

{
    my $dest_target = $ARGV[0];

    my $api = new freeswitch::API();
    my $xml = new XML::Simple;
    my $chan_xml = $xml->XMLin($api->executeString("show detailed_calls as xml"),KeyAttr => 'row', ForceArray => ['row']);

    my %candidates;
    my $result;

    foreach my $row (@{$chan_xml->{row}}) {
        if ($row->{callstate} eq "RINGING" && $row->{dest} eq $dest_target) {
            $candidates{ $row->{created_epoch} - 1000 }  = $row->{call_uuid};     #In the case of a tie, we want to pickup ringing calls before held calls...
            next;
        }

        elsif ($row->{callstate} eq "HELD" && $row->{b_sent_callee_num} eq $dest_target ) {
            $candidates{ $row->{created_epoch} } = $row->{b_uuid};
            next;
        }

        elsif ($row->{b_callstate} eq "HELD" && $row->{b_callee_num} eq $dest_target ) {
            $candidates{ $row->{b_created_epoch} } = $row->{uuid};
            next;
        }
    }

    if( keys( %candidates ) == 0) {
        $result = "[Intercept] No suitable channel to intercept on extension $dest_target.";
        freeswitch::consoleLog("INFO","$result\n");
    }

    else {
        my $user = $session->getVariable('caller_id_name');
        my $cid_num = $session->getVariable('caller_id_number');

        for my $winner ( sort keys %candidates ) {

            my $uuid= $candidates{$winner};

            $result = "[Intercept]User:$user" . "[" . "$cid_num" . "] intercepting Extension:$dest_target" . ".  UUID:" . "$uuid" . "\n";
            freeswitch::consoleLog("INFO","$result\n");
            $session->execute("intercept",$uuid);
            last;
        }
    }
    1;
}



Enjoy!



-Frank




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From vallimamod.abdullah at imtelecom.fr  Sat Jun 23 18:36:44 2012
From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH)
Date: Sat, 23 Jun 2012 16:36:44 +0200
Subject: [Freeswitch-users] Alert-Info header not understood by SPA
	phones
In-Reply-To: <98DE356E-FEFF-4732-906C-BADBF119E891@mgtech.com>
References: <A0B516FD-73C1-4F9E-B928-4C4AA54BD393@imtelecom.fr>
	<39F586EF-0B6B-4243-A385-900D7CD870D7@mgtech.com>
	<FAA4F224-4805-4A5F-A42F-EB0CA683FC44@imtelecom.fr>
	<98DE356E-FEFF-4732-906C-BADBF119E891@mgtech.com>
Message-ID: <A9BD2F94-DC39-4EA4-9E8E-2081102B50AF@imtelecom.fr>

On Jun 23, 2012, at 1:11 AM, Mario G wrote:

> Interesting, I could not get the "name" to work, my syntax also allows you to set the W and C variables from FreeSwitch instead of using the ones set on the phone which is more flexible.

Yes, it is definitely better. For those who may come across the same issue, I have posted a question on the cisco forum: https://supportforums.cisco.com/message/3666436.
My temporary workaround was to declare all the local extensions on each phone's personal directory and associate the desired ringtone with it.

> Since you have SPA962s I would like to ask a question regarding a problem I can't solve: 30-40 percent of the time when an SPA962 answers an outside call, some of the other SPA962s keep ringing for a while or you answer/hangup. They are not using shared lines. Ever run into that? Been months on this issue and no one here responded to the issue when I posted it.

I have never encountered this kind issue but for what I know, the spa962 works very bad with sip alg, worse than the spa942. So if there is nat in your call path, you should look there. Also make a sip capture with ngrep (command line: ngrep -qt -W byline -d <network interface> host <phone ip> and port 5060) for the phone that is continuing to ring and pastebin it, I'll have a look on it to check if it receives the correct CANCEL and how it responds to it.

-- 
Best regards,
- vma
.



From saami_mh at ymail.com  Sat Jun 23 19:37:16 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Sat, 23 Jun 2012 08:37:16 -0700 (PDT)
Subject: [Freeswitch-users] some questions on mod_xml_curl?
In-Reply-To: <B2D8FFB9-807E-48F9-A13C-0FBEE70BE324@gmail.com>
References: <1340430494.72582.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<B2D8FFB9-807E-48F9-A13C-0FBEE70BE324@gmail.com>
Message-ID: <1340465836.59184.YahooMailNeo@web120102.mail.ne1.yahoo.com>



hi Steven,
it is kind of you answer my question in the way that meet my need,
but i need some helps, ?by reading wiki on mod_xml_curl i am not able to understand what exactly must to do for implement realtime on freeswitch
what is the next step for implement realtime after set binding directory ?in?/usr/local/freeswitch/conf/autoload_configs/xml_curl.conf.xml ?as follow and load mod_xml_curl ?
<configuration name="xml_curl.conf" description="cURL XML Gateway">
? <bindings>
? ? <binding name="directory_fetcher">
? ?? ? ?<param name="gateway-url" value="http://192.168.10.89:8080" bindings="directory"/>
? ? ? ? <param name="disable-100-continue" value="true"/>
? </binding>
? </bindings>
</configuration>

i think the next step is to put my script -on any programming languges-in the below path ?
/usr/local/freeswitch/htdocs/

is it possible for you to describe me ?only the steps for implement raltime to findout what exactly should to do?

thanks so much



________________________________
 From: Steven Ayre <steveayre at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Cc: Free SWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Saturday, June 23, 2012 4:41 PM
Subject: Re: [Freeswitch-users] some questions on mod_xml_curl?
 

1- for implementing mod_xml_cur must to running webserver?
The purpose of mod_xml_curl is to download configuration files via HTTP, so yes it does require running a webserver. That doesn't have to run on the same machine though, it just needs to be reachable.
3- if we want to use python or c++ or perl langueges in mod_xml_curl so the werb server should to be implement via php
>because using python we couldn't implement web server ?
You can implement the scripts running on the webserver in any language you like (any of the above including python can be used), the script just needs to capable of outputting the correct XML.


2-is it possible to use mysql directly without webserver ,like ASterisk realtime architecture?
>in other hand after sip request use sql request instead of http request?
This isn't enough information for us to answer your question. Some parts of FS can indeed talk to a DB directly, others generally shouldn't (CDR). What specifically are you wanting to store in the database and access directly from FS?




Steve on iPhone



On 23 Jun 2012, at 06:48, Samira Mh <saami_mh at ymail.com> wrote:


hi,
>i have some questions on mod_xml_curl ..
>1- for implementing mod_xml_cur must to running webserver?
>2-is it possible to use mysql directly without webserver ,like ASterisk realtime architecture?
>in other hand after sip request use sql request instead of http request?
>3- if we want to use python or c++ or perl langueges in mod_xml_curl so the werb server should to be implement via php
>because using python we couldn't implement web server ?
>
>
_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 gmail.com  Sat Jun 23 20:34:15 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 23 Jun 2012 18:34:15 +0200
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CAP2fbbcAuJ6t_xw2CKDKeqf6FLE=MZ1QbC9wA4rB5xQPEXSVvA@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
	<CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
	<CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>
	<CAP2fbbeCxNssmEXrmwxd9Z=ddz+CNEzASQ2QRpzispMPt9cjzA@mail.gmail.com>
	<CALXCt0oLTuNuL4XO92BFwMUWCs+tp_fcJxjjtib2ffktdFMLCw@mail.gmail.com>
	<CAP2fbbfXAk9WKwKFQ0_iOx8-WEYUqbLw9zrhvpZs1Wo96oQjZA@mail.gmail.com>
	<CAP2fbbfYdUrE=GYgVH5WwcbgW7s4=_f+81=DM=aK8dpYJ+kQnw@mail.gmail.com>
	<CALXCt0pad5LY3N9ecsAzjdmqZRPMqCeQdBwbzhoQLUY73T22eA@mail.gmail.com>
	<CAP2fbbcAuJ6t_xw2CKDKeqf6FLE=MZ1QbC9wA4rB5xQPEXSVvA@mail.gmail.com>
Message-ID: <CALXCt0qP7mt0H86_77vA3tJQLY85VRk6hxmd85+JGo7_i52Hiw@mail.gmail.com>

On Sat, Jun 23, 2012 at 2:34 PM, Balamurugan Mahendran <b2m at a-cti.com> wrote:
>
> Sorry, I could not trace it yet, it would be very helpful if you find time
> for me.


I just done what you do, but for me it works without any problem:

1) I compiled and added mod_shout to a freeswitch just compiled and
installed as per the mod_skypopen wiki page :
http://wiki.freeswitch.org/wiki/Skypopen#Building.2C_Configuration_and_Operation

2) I added the following extension to
/usr/local/freeswitch/conf/dialplan/default.xml

  <extension name="bala">
  <condition field="destination_number" expression="^12345678$">

  <action application="set" data="RECORD_TITLE=Recording
${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d
%H:%M:%S)}"/>
  <action application="set" data="ringback=${us-ring}"/>
  <action application="set" data="RECORD_COPYRIGHT=(c) freeswitch.org"/>
  <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
  <action application="set" data="RECORD_ARTIST=auto"/>
  <action application="set" data="RECORD_COMMENT=auto recording"/>
  <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/>
  <action application="set" data="RECORD_STEREO=true"/>
  <action application="set" data="record_sample_rate=16000"/>
  <action application="set" data="media_bug_answer_req=true"/>
  <action application="record_session"
data="$${base_dir}/recordings/${uuid}.mp3"/>
  <action application="set" data="ignore_early_media=false"/>
  <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,ORIGINATOR_CANCEL,CANCEL,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>
  <action application="set" data="call_timeout=23"/>
  <action application="set" data="hangup_after_bridge=true"/>
  <action application="bridge"
data="{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}skypopen/skype102/myskypename"/>
  </condition>
  </extension>

Called from a SIP phone, received the call on skype, talked, hangup,
downloaded the mp3 file, it sounds perfect.

You got some other problem.


--
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From bdfoster at endigotech.com  Sat Jun 23 22:51:58 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 23 Jun 2012 14:51:58 -0400
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CAP2fbbfXAk9WKwKFQ0_iOx8-WEYUqbLw9zrhvpZs1Wo96oQjZA@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
	<CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
	<CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>
	<CAP2fbbeCxNssmEXrmwxd9Z=ddz+CNEzASQ2QRpzispMPt9cjzA@mail.gmail.com>
	<CALXCt0oLTuNuL4XO92BFwMUWCs+tp_fcJxjjtib2ffktdFMLCw@mail.gmail.com>
	<CAP2fbbfXAk9WKwKFQ0_iOx8-WEYUqbLw9zrhvpZs1Wo96oQjZA@mail.gmail.com>
Message-ID: <CAKV-jFyeQm33PxY3Z=7MR8SoXab7frssopwpLXeMhS7oEJk3yw@mail.gmail.com>

take a look at your record sample rate parameter, its set to 11000.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 21, 2012 1:39 PM, "Balamurugan Mahendran" <b2m at a-cti.com> wrote:

> I am afraid, I did not do anything outside freeswitch.
>
> just have the following for recording in my dialplan
>
>
>     <action application="set" data="RECORD_COPYRIGHT=(c) freeswitch"/>
>     <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
>     <action application="set" data="RECORD_ARTIST=auto"/>
>     <action application="set" data="RECORD_COMMENT=auto recording"/>
>     <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d
> %H:%M)}"/>
>     <action application="set" data="RECORD_STEREO=true"/>
>     <action application="set" data="record_sample_rate=11000"/>
>     <action application="set" data="media_bug_answer_req=true"/>
>     <action application="record_session"
> data="$${base_dir}/recordings/${uuid}.mp3"/>
>     <action application="set" data="ignore_early_media=false"/>
>     <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,ORIGINATOR_CANCEL,CANCEL,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>
>
>
> Thanks,
> Bala
>
>
> On Thu, Jun 21, 2012 at 10:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>
>> On Thu, Jun 21, 2012 at 7:15 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>
>>> Thanks for your help!!
>>>
>>> Up to my knowledge there was no change.
>>>
>>> After that I tried re-installed FS(with latest build) Skype and changed
>>> EC2 Instance(different size) but for some reason its still same. I guess
>>> some settings somewhere changed not sure where it is.
>>>
>>>
>> I just checked your mp3, it says it's recorded at 8khz, but it's not
>> true. It's 16khz. If you play it at 16khz it sounds ok. So, you have to
>> look into the mp3 converter you used and configure it to not lie about the
>> samples per second ;).
>>
>> -giovanni
>>
>>
>>
>>
>>
>>>
>>> Thanks,
>>> Bala
>>>
>>> On Thu, Jun 21, 2012 at 10:32 PM, Giovanni Maruzzelli <gmaruzz at gmail.com
>>> > wrote:
>>>
>>>>
>>>>
>>>> On Thu, Jun 21, 2012 at 6:50 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:
>>>>
>>>>> Is there a wiki link for these parameters? Thanks!!
>>>>>
>>>>> Actually it was very good till last week(for about 6
>>>>> months), suddenly its broken. I could not trace it yet
>>>>>
>>>>>
>>>> Have you reinstalled something?
>>>>
>>>> Re-compiled things, or updated config files?
>>>>
>>>> If you have changed nothing, maybe the problem is elsewhere.
>>>>
>>>> Just for test, you can:
>>>>
>>>> 1) from FS cli, unload mod_skypopen
>>>> 2) from ssh killall Xvfb (this will kill all the skype instances too)
>>>> 3) from ssh relaunch the Xvfb-skype clients (with the script)
>>>> 4) from FS cli load mod_skypopen
>>>>
>>>> check if it is recording correctly now.
>>>>
>>>> That test is good if you changed nothing.
>>>>
>>>> If you changed something, revert that change :).
>>>>
>>>> -giovanni
>>>>
>>>>
>>>>
>>>>
>>>>> Thanks,
>>>>> Bala
>>>>>
>>>>>
>>>>> On Thu, Jun 21, 2012 at 9:59 PM, Giovanni Maruzzelli <
>>>>> gmaruzz at gmail.com> wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran <b2m at a-cti.com
>>>>>> > wrote:
>>>>>>
>>>>>>> All,
>>>>>>>
>>>>>>> Is there a way to control recording speed for mod_skypopen? I was
>>>>>>> trying to figure out the problem of fast recording/playing
>>>>>>>
>>>>>>> Please find the attached file
>>>>>>>
>>>>>>
>>>>>> Balamurugan,
>>>>>>
>>>>>> be aware that mod_skypopen operate at 16khz signed linear (let's say,
>>>>>> at HD audio level; double the number of samples of a normal PCMU or GSM or
>>>>>> g711 or G729 call).
>>>>>>
>>>>>> So, maybe you have to tweak the recording or playback parameters
>>>>>> taking this into account.
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>> Thanks,
>>>>>>> Bala
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at 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
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 gmail.com  Sat Jun 23 23:50:51 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 23 Jun 2012 21:50:51 +0200
Subject: [Freeswitch-users] Fwd: Mod_skypopen : recording
In-Reply-To: <CAKV-jFyeQm33PxY3Z=7MR8SoXab7frssopwpLXeMhS7oEJk3yw@mail.gmail.com>
References: <CAP2fbbeGtThiJqAGHaquEpO8xRtAY3O0aveca8FXK91YrO7ozA@mail.gmail.com>
	<CAP2fbbfrghjBG22g92rPhcYuv9k5RmoZYcu=RSLM7Lw_vcqvdA@mail.gmail.com>
	<CALXCt0qFPZbeWExW_is-TBHMda25RUASv+e3kace+ZB5n_GjkQ@mail.gmail.com>
	<CAP2fbbc_PbTZoSyD-MKVL4FSWMkCOjVNFb5SwTmv3p9k-Q3hBA@mail.gmail.com>
	<CALXCt0oRugp86LxwZMCXsfQDfJ5B8Wnr3vFe9aU5_yepTs+_-g@mail.gmail.com>
	<CAP2fbbeCxNssmEXrmwxd9Z=ddz+CNEzASQ2QRpzispMPt9cjzA@mail.gmail.com>
	<CALXCt0oLTuNuL4XO92BFwMUWCs+tp_fcJxjjtib2ffktdFMLCw@mail.gmail.com>
	<CAP2fbbfXAk9WKwKFQ0_iOx8-WEYUqbLw9zrhvpZs1Wo96oQjZA@mail.gmail.com>
	<CAKV-jFyeQm33PxY3Z=7MR8SoXab7frssopwpLXeMhS7oEJk3yw@mail.gmail.com>
Message-ID: <CALXCt0oUFgK_WvUCs9ovSa_-mQbSDj7jmmNpckJtf33yCPp5DQ@mail.gmail.com>

no Brian, he tested with 16k too. It's something else.


On 6/23/12, Brian Foster <bdfoster at endigotech.com> wrote:
> take a look at your record sample rate parameter, its set to 11000.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Jun 21, 2012 1:39 PM, "Balamurugan Mahendran" <b2m at a-cti.com> wrote:
>
>> I am afraid, I did not do anything outside freeswitch.
>>
>> just have the following for recording in my dialplan
>>
>>
>>     <action application="set" data="RECORD_COPYRIGHT=(c) freeswitch"/>
>>     <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
>>     <action application="set" data="RECORD_ARTIST=auto"/>
>>     <action application="set" data="RECORD_COMMENT=auto recording"/>
>>     <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d
>> %H:%M)}"/>
>>     <action application="set" data="RECORD_STEREO=true"/>
>>     <action application="set" data="record_sample_rate=11000"/>
>>     <action application="set" data="media_bug_answer_req=true"/>
>>     <action application="record_session"
>> data="$${base_dir}/recordings/${uuid}.mp3"/>
>>     <action application="set" data="ignore_early_media=false"/>
>>     <action application="set"
>> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,ORIGINATOR_CANCEL,CANCEL,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>
>>
>>
>> Thanks,
>> Bala
>>
>>
>> On Thu, Jun 21, 2012 at 10:59 PM, Giovanni Maruzzelli
>> <gmaruzz at gmail.com>wrote:
>>
>>> On Thu, Jun 21, 2012 at 7:15 PM, Balamurugan Mahendran
>>> <b2m at a-cti.com>wrote:
>>>
>>>> Thanks for your help!!
>>>>
>>>> Up to my knowledge there was no change.
>>>>
>>>> After that I tried re-installed FS(with latest build) Skype and changed
>>>> EC2 Instance(different size) but for some reason its still same. I
>>>> guess
>>>> some settings somewhere changed not sure where it is.
>>>>
>>>>
>>> I just checked your mp3, it says it's recorded at 8khz, but it's not
>>> true. It's 16khz. If you play it at 16khz it sounds ok. So, you have to
>>> look into the mp3 converter you used and configure it to not lie about
>>> the
>>> samples per second ;).
>>>
>>> -giovanni
>>>
>>>
>>>
>>>
>>>
>>>>
>>>> Thanks,
>>>> Bala
>>>>
>>>> On Thu, Jun 21, 2012 at 10:32 PM, Giovanni Maruzzelli
>>>> <gmaruzz at gmail.com
>>>> > wrote:
>>>>
>>>>>
>>>>>
>>>>> On Thu, Jun 21, 2012 at 6:50 PM, Balamurugan Mahendran
>>>>> <b2m at a-cti.com>wrote:
>>>>>
>>>>>> Is there a wiki link for these parameters? Thanks!!
>>>>>>
>>>>>> Actually it was very good till last week(for about 6
>>>>>> months), suddenly its broken. I could not trace it yet
>>>>>>
>>>>>>
>>>>> Have you reinstalled something?
>>>>>
>>>>> Re-compiled things, or updated config files?
>>>>>
>>>>> If you have changed nothing, maybe the problem is elsewhere.
>>>>>
>>>>> Just for test, you can:
>>>>>
>>>>> 1) from FS cli, unload mod_skypopen
>>>>> 2) from ssh killall Xvfb (this will kill all the skype instances too)
>>>>> 3) from ssh relaunch the Xvfb-skype clients (with the script)
>>>>> 4) from FS cli load mod_skypopen
>>>>>
>>>>> check if it is recording correctly now.
>>>>>
>>>>> That test is good if you changed nothing.
>>>>>
>>>>> If you changed something, revert that change :).
>>>>>
>>>>> -giovanni
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>> Thanks,
>>>>>> Bala
>>>>>>
>>>>>>
>>>>>> On Thu, Jun 21, 2012 at 9:59 PM, Giovanni Maruzzelli <
>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Jun 21, 2012 at 5:44 PM, Balamurugan Mahendran
>>>>>>> <b2m at a-cti.com
>>>>>>> > wrote:
>>>>>>>
>>>>>>>> All,
>>>>>>>>
>>>>>>>> Is there a way to control recording speed for mod_skypopen? I was
>>>>>>>> trying to figure out the problem of fast recording/playing
>>>>>>>>
>>>>>>>> Please find the attached file
>>>>>>>>
>>>>>>>
>>>>>>> Balamurugan,
>>>>>>>
>>>>>>> be aware that mod_skypopen operate at 16khz signed linear (let's
>>>>>>> say,
>>>>>>> at HD audio level; double the number of samples of a normal PCMU or
>>>>>>> GSM or
>>>>>>> g711 or G729 call).
>>>>>>>
>>>>>>> So, maybe you have to tweak the recording or playback parameters
>>>>>>> taking this into account.
>>>>>>>
>>>>>>> -giovanni
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>> Thanks,
>>>>>>>> Bala
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at 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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 avi at avimarcus.net  Sun Jun 24 00:01:01 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 23 Jun 2012 23:01:01 +0300
Subject: [Freeswitch-users] is it possible to use lua in mod_xml_curl ?
In-Reply-To: <073BDA71-2741-42A4-A85E-13A4A4846C68@gmail.com>
References: <1340429990.6414.YahooMailNeo@web120105.mail.ne1.yahoo.com>
	<073BDA71-2741-42A4-A85E-13A4A4846C68@gmail.com>
Message-ID: <CALNZuw=d+aqku6kbwGe0fhTad8V8xnP50bFOZtXnSY1UD2FsCQ@mail.gmail.com>

Lua can serve xml_curl directly without a webserver.
See: http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration

-Avi



On Sat, Jun 23, 2012 at 3:12 PM, Steven Ayre <steveayre at gmail.com> wrote:

> This isn't a FS question.
>
> You can use any language your webserver allows you to use. FS only cares
> about the XML output.
>
> If your webserver supports lua, then yes you can use lua.
>
> Steve on iPhone
>
>
>
> On 23 Jun 2012, at 06:39, Samira Mh <saami_mh at ymail.com> wrote:
>
> the supported languages in mod_xml_cur include:
> perl,python,php,c++,c# , but can i use lua ?
> lua is meet our requirements for implementing mod_xml_curl
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun 24 02:04:14 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 23 Jun 2012 17:04:14 -0500
Subject: [Freeswitch-users] ICTFAX Version 2.0 beta released ,
 an Open Source FOIP software solution
In-Reply-To: <CAOL0t+LZ4wV9P2+3SLneQgLQ0x2JvWLt9uxu0-YmkkQoiLWeQQ@mail.gmail.com>
References: <CAOL0t+LZ4wV9P2+3SLneQgLQ0x2JvWLt9uxu0-YmkkQoiLWeQQ@mail.gmail.com>
Message-ID: <CAKbxfG90w3+S59AcddEpzNxyV7Jfw4Bev3kjN3h04vCpbvELYw@mail.gmail.com>

Cool,

You should sponsor ClueCon 2012!

On Fri, Jun 22, 2012 at 8:11 PM, tahir almas <tahir at ictinnovations.com> wrote:
> ICT Innovations? achieved big milestone by releasing?? ICTFAX Ver 2.0 beta,
> an open source Foip software supporting Email to Fax , Fax to Email and Web
> to Fax
>
> Earlier version of ICTFAX was based on Asterisk and? Drupal Ver? 4.7 causing
> issues to new users , We at ICT Innovations devoted months on this project
> and rewritten all code in Drupal 7.0 as well as ported back-end to Plivo
> communication framework , Now ICTFAX using open source Freeswitch as
> back-end communication engine making it advance and stable open source FOIP
> solution
>
> For more information, please visit? http://www.ictfax.org
>
> Tahir Almas
>
> Managing Partner
> ICT Innovations
> http://www.ictinnovations.com
> Leveraging open source in ICT
>
> ****************************************************************************************************************
> NOTICE OF CONFIDENTIALITY
> This communication including any information transmitted with it is intended
> only for the use of the addressees and is confidential and may be protected
> by legal privilege . If you are not an intended recipient, be aware that any
> disclosure, copying, distribution or use of this e-mail or any attachment is
> prohibited. If you have received this e-mail in error, please notify us
> immediately by returning it to the sender and delete this copy from your
> system. Thank you for your cooperation.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From royj at yandex.ru  Sun Jun 24 02:09:59 2012
From: royj at yandex.ru (royj)
Date: Sun, 24 Jun 2012 02:09:59 +0400
Subject: [Freeswitch-users] mod_gsmopen on FreeBSD
In-Reply-To: <CALXCt0qNKnn9UB2ONf+X3effOW3OyQhU18MU2XXzrrgA6K=z2Q@mail.gmail.com>
References: <20120611233123.ebed9bee.royj@yandex.ru>
	<CALXCt0qNKnn9UB2ONf+X3effOW3OyQhU18MU2XXzrrgA6K=z2Q@mail.gmail.com>
Message-ID: <20120624020959.eb725706.royj@yandex.ru>


I was able to adapt the code (a few changes in gsmopen_protocol.cpp and a little more commentary in serport.cpp) and use mod_gsmopen on FreeBSD. It is ought to be port libctb, but at the moment there is a patches (http://jira.freeswitch.org/browse/FS-4338)

After some changes
<param name="controldevice_name" value="/dev/cuaU0.2"/>
<param name="controldevice_audio_name" value="/dev/cuaU0.1"/>
works for me

P.S. How to get access to http://pastebin.freeswitch.org?

Regards
royj

On Tue, 12 Jun 2012 10:03:46 +0200
Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> On Mon, Jun 11, 2012 at 9:31 PM, royj <royj at yandex.ru> wrote:
> 
> >
> > Hi all
> > Does anyone use mod_gsmopen on FreeBSD? I have tried to make according to
> > the instructions - http://wiki.freeswitch.org/wiki/GSMopen#Building, and
> > got an error when compiling libctb-0.16 which is only for Linux and Win32 (
> > https://iftools.com/download/ctb/0.14/refman.pdf). Is there any solution
> > other than changing OS )
> >
> 
> I believe would be easy to adapt libctb to compile on FreeBSD. Why don't
> you do that, and contribute it back with a patch on jira.freeswitch.org ? I
> would happily integrate your code into mainline.
> If you're not able to, maybe you can ask some of the FreeBSD guys in the
> mailing list or in IRC, will probably take 15mins or so...
> 
> -giovanni
> 
> 
> 
> 
> > --
> > royj
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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


-- 
royj


From shishko69 at gmail.com  Sat Jun 23 19:03:07 2012
From: shishko69 at gmail.com (Shishko)
Date: Sat, 23 Jun 2012 17:03:07 +0200
Subject: [Freeswitch-users] Changing FROM header
Message-ID: <CAHT_yt-n7YsZw=pC0KSPFz0iMKjp5CxAsPeQJUYY+F70WTrcYw@mail.gmail.com>

Hi,

is there a way I can change From header in bridge application line?
In a-leg I get something as "xxxxxx <sip:xxxxxx at domain.tld>", and I'd
like to somehow change it to "yyyyyy <sip:yyyyyy at domain.tld>" in
b-leg.
I tried with sip_h_From and sip_from_uri, but no success...

Thanks


From aam-003 at cox.net  Sat Jun 23 21:23:07 2012
From: aam-003 at cox.net (aam-003 at cox.net)
Date: Sat, 23 Jun 2012 10:23:07 -0700
Subject: [Freeswitch-users] Hangup not detected after fifo
Message-ID: <178F92A1-846D-480D-898A-66EBD7CCA0EA@cox.net>

I am having issues with calls not being terminated correctly - FS doesn't seem to notice when the call is hang up.

Here is my scenario:

Extension 201 places a call which gets routed as follows:

<extension name="Callback_Start" >
  <condition field="caller_id_number" expression="^201$" >
  </condition>
  <condition field="destination_number" expression="^\+?1([2-9]\d{9})$" break="never" >
      <action application="set" data="target_number=+1$1" inline="true"/>
  </condition>
  <condition field="destination_number" expression="^(?:\+|011)(\d+)$" break="never" >
      <action application="set" data="target_number=+$1" inline="true"/>
  </condition>
  <condition field="${target_number}" expression="^\+\d+$" >
      <action application="answer" />
      <action application="system" data="/usr/local/freeswitch/bin/my-callback.php ${target_number}" />
      <action application="set" data="hangup_after_bridge=true" />
      <action application="fifo" data="MY_CALLBACK in" />
  </condition>
</extension>

This causes a callback being initiated via external SIP account which then gets processed with the following dial plan:

<extension name="Callback_Answer" >
  <condition field="context" expression="public" />
  <condition field="${fifo(count MY_CALLBACK)}" expression="^MY_CALLBACK\:[0-9]\:[1-9]" >
      <action application="answer" />
      <action application="fifo" data="MY_CALLBACK out nowait" />
  </condition>
</extension>

Everything works as expected - extension 201 gets connected to the callback and all media is working correctly. However, when extension 201 hangs up, my FS server doesn't seem to realize that that the call is over and the call seems to be hanging around in limbo for another 100 seconds or so.

How can I convince the server to terminate the call as soon as my internal extension hangs up? Is there a better way to achieve this callback scenario instead?

Thanks for any pointers!

From sherifomran2000 at yahoo.com  Sun Jun 24 03:06:31 2012
From: sherifomran2000 at yahoo.com (Sherif Omran)
Date: Sat, 23 Jun 2012 16:06:31 -0700 (PDT)
Subject: [Freeswitch-users] Invalid application transfer help
In-Reply-To: <B5E2C029-C331-42A5-B4FE-80709E668683@visionutveckling.se>
Message-ID: <1340492791.29895.YahooMailClassic@web110802.mail.gq1.yahoo.com>

how do i test the versions?

thanks
Sherif


--- On Sat, 6/23/12, Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> From: Peter Olsson <peter.olsson at visionutveckling.se>
> Subject: Re: [Freeswitch-users] Invalid application transfer help
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Saturday, June 23, 2012, 2:53 PM
> You must make sure you use the same
> version of the FS core library and it's modules. You have an
> error for a missing symbol when loading the module.
> 
> /Peter
> 
> 23 jun 2012 kl. 10:12 skrev "Sherif Omran" <sherifomran2000 at yahoo.com>:
> 
> > I get the following ... Please take a look
> > 
> > freeswitch at internal> mod_exists
> > -ERR mod_exists Command not found!
> > 
> > freeswitch at internal> mod_exists mod_dptools
> > -ERR mod_exists mod_dptools Command not found!
> > 
> > freeswitch at internal> load mod_dptools
> > +OK Reloading XML
> > -ERR [module load file routine returned an error]
> > 
> > 2012-06-23 08:04:58.914388 [INFO] mod_enum.c:812 ENUM
> Reloaded
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449
> Adding tone_descriptor: 1
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> Adding tone_descriptor: 1, tone: CED_TONE(0)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 1, tone: CED_TONE(0), element (2100,
> 0, 500, 0)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> Adding tone_descriptor: 1, tone: SIT(1)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 1, tone: SIT(1), element (950, 0,
> 256, 400)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 1, tone: SIT(1), element (1400, 0,
> 256, 400)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 1, tone: SIT(1), element (1800, 0,
> 256, 400)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> Adding tone_descriptor: 1, tone: REORDER_TONE(2)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 1, tone: REORDER_TONE(2), element
> (480, 620, 224, 272)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 1, tone: REORDER_TONE(2), element
> (0, 0, 224, 272)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> Adding tone_descriptor: 1, tone: BUSY_TONE(3)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (480,
> 620, 464, 516)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (0,
> 0, 464, 516)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449
> Adding tone_descriptor: 44
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> Adding tone_descriptor: 44, tone: CED_TONE(0)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 44, tone: CED_TONE(0), element
> (2100, 0, 500, 0)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> Adding tone_descriptor: 44, tone: SIT(1)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 44, tone: SIT(1), element (950, 0,
> 256, 400)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 44, tone: SIT(1), element (1400, 0,
> 256, 400)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 44, tone: SIT(1), element (1800, 0,
> 256, 400)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> Adding tone_descriptor: 44, tone: REORDER_TONE(2)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 44, tone: REORDER_TONE(2), element
> (400, 0, 368, 416)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 44, tone: REORDER_TONE(2), element
> (0, 0, 336, 368)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 44, tone: REORDER_TONE(2), element
> (400, 0, 256, 288)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 44, tone: REORDER_TONE(2), element
> (0, 0, 512, 544)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> Adding tone_descriptor: 44, tone: BUSY_TONE(3)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 44, tone: BUSY_TONE(3), element
> (400, 0, 352, 384)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0,
> 0, 352, 384)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 44, tone: BUSY_TONE(3), element
> (400, 0, 352, 384)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0,
> 0, 352, 384)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449
> Adding tone_descriptor: 49
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> Adding tone_descriptor: 49, tone: CED_TONE(0)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 49, tone: CED_TONE(0), element
> (2100, 0, 500, 0)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> Adding tone_descriptor: 49, tone: SIT(1)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 49, tone: SIT(1), element (900, 0,
> 256, 400)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 49, tone: SIT(1), element (1400, 0,
> 256, 400)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 49, tone: SIT(1), element (1800, 0,
> 256, 400)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> Adding tone_descriptor: 49, tone: REORDER_TONE(2)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 49, tone: REORDER_TONE(2), element
> (425, 0, 224, 272)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 49, tone: REORDER_TONE(2), element
> (0, 0, 224, 272)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> Adding tone_descriptor: 49, tone: BUSY_TONE(3)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 49, tone: BUSY_TONE(3), element
> (425, 0, 464, 516)
> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (0,
> 0, 464, 516)
> > 2012-06-23 08:04:58.914388 [INFO] switch_time.c:1123
> Timezone reloaded 530 definitions
> > 2012-06-23 08:04:58.914388 [CRIT]
> switch_loadable_module.c:1300 Error Loading module
> /home/vBilling/freeswitch/mod/mod_dptools.so
> > **/home/vBilling/freeswitch/mod/mod_dptools.so:
> undefined symbol: switch_core_media_bug_remove_all**
> > 
> > 
> > 
> > Thanks
> > 
> > 
> > --- On Sat, 6/23/12, Hector Geraldino <Hector.Geraldino at ipsoft.com>
> wrote:
> > 
> >> From: Hector Geraldino <Hector.Geraldino at ipsoft.com>
> >> Subject: Re: [Freeswitch-users] Invalid application
> transfer help
> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >> Date: Saturday, June 23, 2012, 12:08 AM
> >> Check if module is loaded:
> >> 
> >>> mod_exists mod_dptools
> >> 
> >> If not, load it:
> >> 
> >>> load mod_dptools
> >> 
> >> Or reload it:
> >> 
> >>> reload mod_dptools
> >> 
> >> -----Original Message-----
> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
> >> On Behalf Of Sherif Omran
> >> Sent: Friday, June 22, 2012 3:37 PM
> >> To: FreeSWITCH Users Help
> >> Subject: Re: [Freeswitch-users] Invalid application
> transfer
> >> help
> >> 
> >> how can I load it? should i recompile and
> reinstall
> >> 
> >> thanks
> >> Sherif
> >> 
> >> 
> >> --- On Fri, 6/22/12, Peter Olsson <peter.olsson at visionutveckling.se>
> >> wrote:
> >> 
> >>> From: Peter Olsson <peter.olsson at visionutveckling.se>
> >>> Subject: Re: [Freeswitch-users] Invalid
> application
> >> transfer help
> >>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >>> Date: Friday, June 22, 2012, 1:18 PM
> >>> Seems you don't have mod_dptools
> >>> loaded anymore.
> >>> 
> >>> /Peter
> >>> 
> >>> 22 jun 2012 kl. 11:45 skrev "Sherif Omran"
> <sherifomran2000 at yahoo.com<mailto:sherifomran2000 at yahoo.com>>:
> >>> 
> >>> Hi Guys,
> >>> 
> >>> My server was working fine till today, It does
> not
> >> transfer any call. 
> >>> When I call an extension i get? a not
> connected tone
> >> and in the log I 
> >>> see invalid application
> >>> 
> >>> 2012-06-22 09:37:26.414815 [ERR]
> >> switch_core_session.c:2156 Invalid 
> >>> Application transfer
> >>> 
> >>> I wonder what this could be? I was trying
> yesterday to
> >> install some 
> >>> packages. Could this be related to that?
> >>> 
> >>> kind regards,
> >>> Sherif
> >>> 
> >>> 
> >>> 
> >>> 
> >>
> ______________________________________________________________________
> >>> ___ Professional FreeSWITCH Consulting
> Services:
> >>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> >>> http://www.freeswitchsolutions.com
> >>> 
> >>> FreeSWITCH-powered IP PBX: The CudaTel
> Communication
> >> Server 
> >>> 
> >>> 
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>> 
> >>> Join Us At ClueCon - Aug 7-9, 2012
> >>> 
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.fr
> >>> eeswitch.org> 
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >>> rs
> >>> http://www.freeswitch.org
> >>> 
> >>> 
> >>> !DSPAM:4fe43b9a32761615820512!
> >>> 
> >>> 
> >>
> ______________________________________________________________________
> >>> ___ Professional FreeSWITCH Consulting
> Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>> 
> >>> FreeSWITCH-powered IP PBX: The CudaTel
> Communication
> >> Server 
> >>> 
> >>> 
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>> 
> >>> Join Us At ClueCon - Aug 7-9, 2012
> >>> 
> >>> FreeSWITCH-users 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
> >>> 
> >> 
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >> 
> >> FreeSWITCH-powered IP PBX: The CudaTel
> Communication Server
> >> 
> >> 
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >> 
> >> Join Us At ClueCon - Aug 7-9, 2012
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >> 
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >> 
> >> FreeSWITCH-powered IP PBX: The CudaTel
> Communication Server
> >> 
> >> 
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >> 
> >> Join Us At ClueCon - Aug 7-9, 2012
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >> 
> > 
> >
> _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > Join Us At ClueCon - Aug 7-9, 2012
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > !DSPAM:4fe5774732761352082344!
> > 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From jpablolorenzetti at hotmail.com  Sun Jun 24 05:52:36 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Sun, 24 Jun 2012 01:52:36 +0000
Subject: [Freeswitch-users] internal extension is unregistered when
 registering from remote location
Message-ID: <DUB109-W8194139EEDA5F62A96753DD9E20@phx.gbl>



Hi, i have an internal extension 2000 registering from a remote location, extension 2000 can make calls
to internal extension and through gateways but nobody can call the 2000 extension, when someone tries to reach extension
2000 there is this error: 
"switch_ivr_originate.c:2492 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]"

and voicemail invites to leave a message.

when extension 2000 registers i get this in the FS logs:

sofia_reg.c:1445 SIP auth challenge (REGISTER) on sofia profile 'internal-hq' for [2000 at 10.19.0.8] from ip 190.242.32.118

so it is registering, when i check in the database this extension is not there. i would appreciate any hints. 

thanks!
 		 	   		  

From moises.silva at gmail.com  Sun Jun 24 07:47:20 2012
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 23 Jun 2012 23:47:20 -0400
Subject: [Freeswitch-users] Codec 101 - need help with the basics
In-Reply-To: <4FE2C8AF.5080805@the800group.com>
References: <4FE2C8AF.5080805@the800group.com>
Message-ID: <CAA4nhyALnwXmd2pVeFXborYtTtbY=9o5yggyi1p3y1KOQjrqFg@mail.gmail.com>

On Thu, Jun 21, 2012 at 3:09 AM, ocset <ocset at the800group.com> wrote:

> Hi
>
> I need some assistance in understanding the codec minefield.
>
> When a stream of audio is sent to FreeSWITCH (FS) using a particular
> codec, does FS just pass it on or is there some level of translation
> occurring?
>

In some situations audio may be just passed thru, but most real life
scenarios will involve some level of translation (transcoding).



> I guess my question relates to a situation where an FXO gateway
> (GXW410x) uses the G723 codec and the receiing phone does not have a
> matching codec (the phone has G.722, G.711 a law, G.711 u law, G.726 and
> G.729 as options). How is this managed by the phone, FreeSWITCH and the
> FXO device? Should I be setting something in FS to handle this issue?
>

Whenever FreeSWITCH detects format mismatch between the parties it will
perform transcoding (translation of one audio format to another). You have
several configuration options to force a given list of codecs to be
offered.

Now, in some shameless self-promotion, I suggest you to read or watch my
presentation on FreeSWITCH media handling from here:

http://www.moythreads.com/congresos/cluecon2011/media-handling-freeswitch-final.pdf
http://www.youtube.com/watch?v=508jpSQk8qE

That should make things a bit clearer for you.


> I have 2 cordless Siemens C610 IP phones an I am having sound quality
> issues (incoming sound is soft) and I was wondering how big a part the
> codec plays in this equation. I have desk phones that are from the same
> manufacturer as the FXO gateway (Grandstream) and they have very good
> sound quality.
>

I don't think the codec has a role in your issue, if the audio is soft or
quiet it has more to do with the gain control. If you place calls to those
Siemens phones without using the gateway (ie, calling in to play a file) is
the audio still low?

*Moises Silva
**Manager, Software Engineering***

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



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From msc at freeswitch.org  Sun Jun 24 08:05:54 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 23 Jun 2012 21:05:54 -0700
Subject: [Freeswitch-users] Invalid application transfer help
In-Reply-To: <1340492791.29895.YahooMailClassic@web110802.mail.gq1.yahoo.com>
References: <B5E2C029-C331-42A5-B4FE-80709E668683@visionutveckling.se>
	<1340492791.29895.YahooMailClassic@web110802.mail.gq1.yahoo.com>
Message-ID: <CAKzWOxV9joHQ5K6p+LJRMK9WoNE-mrO8YUYzdLnnoiFF=VTHNw@mail.gmail.com>

this sounds like a problem with your build. i'd do "make current" from your
freeswitch source directory and then try again.
-MC

On Sat, Jun 23, 2012 at 4:06 PM, Sherif Omran <sherifomran2000 at yahoo.com>wrote:

> how do i test the versions?
>
> thanks
> Sherif
>
>
> --- On Sat, 6/23/12, Peter Olsson <peter.olsson at visionutveckling.se>
> wrote:
>
> > From: Peter Olsson <peter.olsson at visionutveckling.se>
> > Subject: Re: [Freeswitch-users] Invalid application transfer help
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Date: Saturday, June 23, 2012, 2:53 PM
> > You must make sure you use the same
> > version of the FS core library and it's modules. You have an
> > error for a missing symbol when loading the module.
> >
> > /Peter
> >
> > 23 jun 2012 kl. 10:12 skrev "Sherif Omran" <sherifomran2000 at yahoo.com>:
> >
> > > I get the following ... Please take a look
> > >
> > > freeswitch at internal> mod_exists
> > > -ERR mod_exists Command not found!
> > >
> > > freeswitch at internal> mod_exists mod_dptools
> > > -ERR mod_exists mod_dptools Command not found!
> > >
> > > freeswitch at internal> load mod_dptools
> > > +OK Reloading XML
> > > -ERR [module load file routine returned an error]
> > >
> > > 2012-06-23 08:04:58.914388 [INFO] mod_enum.c:812 ENUM
> > Reloaded
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449
> > Adding tone_descriptor: 1
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> > Adding tone_descriptor: 1, tone: CED_TONE(0)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 1, tone: CED_TONE(0), element (2100,
> > 0, 500, 0)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> > Adding tone_descriptor: 1, tone: SIT(1)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 1, tone: SIT(1), element (950, 0,
> > 256, 400)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 1, tone: SIT(1), element (1400, 0,
> > 256, 400)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 1, tone: SIT(1), element (1800, 0,
> > 256, 400)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> > Adding tone_descriptor: 1, tone: REORDER_TONE(2)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 1, tone: REORDER_TONE(2), element
> > (480, 620, 224, 272)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 1, tone: REORDER_TONE(2), element
> > (0, 0, 224, 272)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> > Adding tone_descriptor: 1, tone: BUSY_TONE(3)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (480,
> > 620, 464, 516)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (0,
> > 0, 464, 516)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449
> > Adding tone_descriptor: 44
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> > Adding tone_descriptor: 44, tone: CED_TONE(0)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 44, tone: CED_TONE(0), element
> > (2100, 0, 500, 0)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> > Adding tone_descriptor: 44, tone: SIT(1)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 44, tone: SIT(1), element (950, 0,
> > 256, 400)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 44, tone: SIT(1), element (1400, 0,
> > 256, 400)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 44, tone: SIT(1), element (1800, 0,
> > 256, 400)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> > Adding tone_descriptor: 44, tone: REORDER_TONE(2)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 44, tone: REORDER_TONE(2), element
> > (400, 0, 368, 416)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 44, tone: REORDER_TONE(2), element
> > (0, 0, 336, 368)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 44, tone: REORDER_TONE(2), element
> > (400, 0, 256, 288)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 44, tone: REORDER_TONE(2), element
> > (0, 0, 512, 544)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> > Adding tone_descriptor: 44, tone: BUSY_TONE(3)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 44, tone: BUSY_TONE(3), element
> > (400, 0, 352, 384)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0,
> > 0, 352, 384)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 44, tone: BUSY_TONE(3), element
> > (400, 0, 352, 384)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0,
> > 0, 352, 384)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449
> > Adding tone_descriptor: 49
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> > Adding tone_descriptor: 49, tone: CED_TONE(0)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 49, tone: CED_TONE(0), element
> > (2100, 0, 500, 0)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> > Adding tone_descriptor: 49, tone: SIT(1)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 49, tone: SIT(1), element (900, 0,
> > 256, 400)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 49, tone: SIT(1), element (1400, 0,
> > 256, 400)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 49, tone: SIT(1), element (1800, 0,
> > 256, 400)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> > Adding tone_descriptor: 49, tone: REORDER_TONE(2)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 49, tone: REORDER_TONE(2), element
> > (425, 0, 224, 272)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 49, tone: REORDER_TONE(2), element
> > (0, 0, 224, 272)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470
> > Adding tone_descriptor: 49, tone: BUSY_TONE(3)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 49, tone: BUSY_TONE(3), element
> > (425, 0, 464, 516)
> > > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505
> > Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (0,
> > 0, 464, 516)
> > > 2012-06-23 08:04:58.914388 [INFO] switch_time.c:1123
> > Timezone reloaded 530 definitions
> > > 2012-06-23 08:04:58.914388 [CRIT]
> > switch_loadable_module.c:1300 Error Loading module
> > /home/vBilling/freeswitch/mod/mod_dptools.so
> > > **/home/vBilling/freeswitch/mod/mod_dptools.so:
> > undefined symbol: switch_core_media_bug_remove_all**
> > >
> > >
> > >
> > > Thanks
> > >
> > >
> > > --- On Sat, 6/23/12, Hector Geraldino <Hector.Geraldino at ipsoft.com>
> > wrote:
> > >
> > >> From: Hector Geraldino <Hector.Geraldino at ipsoft.com>
> > >> Subject: Re: [Freeswitch-users] Invalid application
> > transfer help
> > >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > >> Date: Saturday, June 23, 2012, 12:08 AM
> > >> Check if module is loaded:
> > >>
> > >>> mod_exists mod_dptools
> > >>
> > >> If not, load it:
> > >>
> > >>> load mod_dptools
> > >>
> > >> Or reload it:
> > >>
> > >>> reload mod_dptools
> > >>
> > >> -----Original Message-----
> > >> From: freeswitch-users-bounces at lists.freeswitch.org
> > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
> > >> On Behalf Of Sherif Omran
> > >> Sent: Friday, June 22, 2012 3:37 PM
> > >> To: FreeSWITCH Users Help
> > >> Subject: Re: [Freeswitch-users] Invalid application
> > transfer
> > >> help
> > >>
> > >> how can I load it? should i recompile and
> > reinstall
> > >>
> > >> thanks
> > >> Sherif
> > >>
> > >>
> > >> --- On Fri, 6/22/12, Peter Olsson <peter.olsson at visionutveckling.se>
> > >> wrote:
> > >>
> > >>> From: Peter Olsson <peter.olsson at visionutveckling.se>
> > >>> Subject: Re: [Freeswitch-users] Invalid
> > application
> > >> transfer help
> > >>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > >>> Date: Friday, June 22, 2012, 1:18 PM
> > >>> Seems you don't have mod_dptools
> > >>> loaded anymore.
> > >>>
> > >>> /Peter
> > >>>
> > >>> 22 jun 2012 kl. 11:45 skrev "Sherif Omran"
> > <sherifomran2000 at yahoo.com<mailto:sherifomran2000 at yahoo.com>>:
> > >>>
> > >>> Hi Guys,
> > >>>
> > >>> My server was working fine till today, It does
> > not
> > >> transfer any call.
> > >>> When I call an extension i get  a not
> > connected tone
> > >> and in the log I
> > >>> see invalid application
> > >>>
> > >>> 2012-06-22 09:37:26.414815 [ERR]
> > >> switch_core_session.c:2156 Invalid
> > >>> Application transfer
> > >>>
> > >>> I wonder what this could be? I was trying
> > yesterday to
> > >> install some
> > >>> packages. Could this be related to that?
> > >>>
> > >>> kind regards,
> > >>> Sherif
> > >>>
> > >>>
> > >>>
> > >>>
> > >>
> > ______________________________________________________________________
> > >>> ___ Professional FreeSWITCH Consulting
> > Services:
> > >>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> > >>> http://www.freeswitchsolutions.com
> > >>>
> > >>> FreeSWITCH-powered IP PBX: The CudaTel
> > Communication
> > >> Server
> > >>> 
> > >>>
> > >>> Official FreeSWITCH Sites
> > >>> http://www.freeswitch.org
> > >>> http://wiki.freeswitch.org
> > >>> http://www.cluecon.com
> > >>>
> > >>> Join Us At ClueCon - Aug 7-9, 2012
> > >>>
> > >>> FreeSWITCH-users mailing list
> > >>> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.fr
> > >>> eeswitch.org>
> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-use
> > >>> rs
> > >>> http://www.freeswitch.org
> > >>>
> > >>>
> > >>> !DSPAM:4fe43b9a32761615820512!
> > >>>
> > >>>
> > >>
> > ______________________________________________________________________
> > >>> ___ Professional FreeSWITCH Consulting
> > Services:
> > >>> consulting at freeswitch.org
> > >>> http://www.freeswitchsolutions.com
> > >>>
> > >>> FreeSWITCH-powered IP PBX: The CudaTel
> > Communication
> > >> Server
> > >>> 
> > >>>
> > >>> Official FreeSWITCH Sites
> > >>> http://www.freeswitch.org
> > >>> http://wiki.freeswitch.org
> > >>> http://www.cluecon.com
> > >>>
> > >>> Join Us At ClueCon - Aug 7-9, 2012
> > >>>
> > >>> FreeSWITCH-users 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
> > >>>
> > >>
> > >>
> > _________________________________________________________________________
> > >> Professional FreeSWITCH Consulting Services:
> > >> consulting at freeswitch.org
> > >> http://www.freeswitchsolutions.com
> > >>
> > >> FreeSWITCH-powered IP PBX: The CudaTel
> > Communication Server
> > >> 
> > >>
> > >> Official FreeSWITCH Sites
> > >> http://www.freeswitch.org
> > >> http://wiki.freeswitch.org
> > >> http://www.cluecon.com
> > >>
> > >> Join Us At ClueCon - Aug 7-9, 2012
> > >>
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >>
> > >>
> > _________________________________________________________________________
> > >> Professional FreeSWITCH Consulting Services:
> > >> consulting at freeswitch.org
> > >> http://www.freeswitchsolutions.com
> > >>
> > >> FreeSWITCH-powered IP PBX: The CudaTel
> > Communication Server
> > >> 
> > >>
> > >> Official FreeSWITCH Sites
> > >> http://www.freeswitch.org
> > >> http://wiki.freeswitch.org
> > >> http://www.cluecon.com
> > >>
> > >> Join Us At ClueCon - Aug 7-9, 2012
> > >>
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >>
> > >
> > >
> > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > FreeSWITCH-powered IP PBX: The CudaTel Communication
> > Server
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > Join Us At ClueCon - Aug 7-9, 2012
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > > !DSPAM:4fe5774732761352082344!
> > >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sherifomran2000 at yahoo.com  Sun Jun 24 11:00:08 2012
From: sherifomran2000 at yahoo.com (Sherif Omran)
Date: Sun, 24 Jun 2012 00:00:08 -0700 (PDT)
Subject: [Freeswitch-users] Invalid application transfer help
In-Reply-To: <CAKzWOxV9joHQ5K6p+LJRMK9WoNE-mrO8YUYzdLnnoiFF=VTHNw@mail.gmail.com>
Message-ID: <1340521208.15683.YahooMailClassic@web110814.mail.gq1.yahoo.com>

thank you 
It worked

--- On Sun, 6/24/12, Michael Collins <msc at freeswitch.org> wrote:

From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Invalid application transfer help
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Sunday, June 24, 2012, 7:05 AM

this sounds like a problem with your build. i'd do "make current" from your freeswitch source directory and then try again.
-MC

On Sat, Jun 23, 2012 at 4:06 PM, Sherif Omran <sherifomran2000 at yahoo.com> wrote:

how do i test the versions?



thanks

Sherif





--- On Sat, 6/23/12, Peter Olsson <peter.olsson at visionutveckling.se> wrote:



> From: Peter Olsson <peter.olsson at visionutveckling.se>

> Subject: Re: [Freeswitch-users] Invalid application transfer help

> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

> Date: Saturday, June 23, 2012, 2:53 PM

> You must make sure you use the same

> version of the FS core library and it's modules. You have an

> error for a missing symbol when loading the module.

>

> /Peter

>

> 23 jun 2012 kl. 10:12 skrev "Sherif Omran" <sherifomran2000 at yahoo.com>:

>

> > I get the following ... Please take a look

> >

> > freeswitch at internal> mod_exists

> > -ERR mod_exists Command not found!

> >

> > freeswitch at internal> mod_exists mod_dptools

> > -ERR mod_exists mod_dptools Command not found!

> >

> > freeswitch at internal> load mod_dptools

> > +OK Reloading XML

> > -ERR [module load file routine returned an error]

> >

> > 2012-06-23 08:04:58.914388 [INFO] mod_enum.c:812 ENUM

> Reloaded

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449

> Adding tone_descriptor: 1

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470

> Adding tone_descriptor: 1, tone: CED_TONE(0)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 1, tone: CED_TONE(0), element (2100,

> 0, 500, 0)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470

> Adding tone_descriptor: 1, tone: SIT(1)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 1, tone: SIT(1), element (950, 0,

> 256, 400)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 1, tone: SIT(1), element (1400, 0,

> 256, 400)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 1, tone: SIT(1), element (1800, 0,

> 256, 400)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470

> Adding tone_descriptor: 1, tone: REORDER_TONE(2)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 1, tone: REORDER_TONE(2), element

> (480, 620, 224, 272)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 1, tone: REORDER_TONE(2), element

> (0, 0, 224, 272)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470

> Adding tone_descriptor: 1, tone: BUSY_TONE(3)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (480,

> 620, 464, 516)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 1, tone: BUSY_TONE(3), element (0,

> 0, 464, 516)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449

> Adding tone_descriptor: 44

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470

> Adding tone_descriptor: 44, tone: CED_TONE(0)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 44, tone: CED_TONE(0), element

> (2100, 0, 500, 0)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470

> Adding tone_descriptor: 44, tone: SIT(1)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 44, tone: SIT(1), element (950, 0,

> 256, 400)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 44, tone: SIT(1), element (1400, 0,

> 256, 400)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 44, tone: SIT(1), element (1800, 0,

> 256, 400)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470

> Adding tone_descriptor: 44, tone: REORDER_TONE(2)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 44, tone: REORDER_TONE(2), element

> (400, 0, 368, 416)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 44, tone: REORDER_TONE(2), element

> (0, 0, 336, 368)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 44, tone: REORDER_TONE(2), element

> (400, 0, 256, 288)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 44, tone: REORDER_TONE(2), element

> (0, 0, 512, 544)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470

> Adding tone_descriptor: 44, tone: BUSY_TONE(3)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 44, tone: BUSY_TONE(3), element

> (400, 0, 352, 384)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0,

> 0, 352, 384)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 44, tone: BUSY_TONE(3), element

> (400, 0, 352, 384)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 44, tone: BUSY_TONE(3), element (0,

> 0, 352, 384)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:449

> Adding tone_descriptor: 49

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470

> Adding tone_descriptor: 49, tone: CED_TONE(0)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 49, tone: CED_TONE(0), element

> (2100, 0, 500, 0)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470

> Adding tone_descriptor: 49, tone: SIT(1)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 49, tone: SIT(1), element (900, 0,

> 256, 400)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 49, tone: SIT(1), element (1400, 0,

> 256, 400)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 49, tone: SIT(1), element (1800, 0,

> 256, 400)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470

> Adding tone_descriptor: 49, tone: REORDER_TONE(2)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 49, tone: REORDER_TONE(2), element

> (425, 0, 224, 272)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 49, tone: REORDER_TONE(2), element

> (0, 0, 224, 272)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:470

> Adding tone_descriptor: 49, tone: BUSY_TONE(3)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 49, tone: BUSY_TONE(3), element

> (425, 0, 464, 516)

> > 2012-06-23 08:04:58.914388 [DEBUG] mod_spandsp.c:505

> Adding tone_descriptor: 49, tone: BUSY_TONE(3), element (0,

> 0, 464, 516)

> > 2012-06-23 08:04:58.914388 [INFO] switch_time.c:1123

> Timezone reloaded 530 definitions

> > 2012-06-23 08:04:58.914388 [CRIT]

> switch_loadable_module.c:1300 Error Loading module

> /home/vBilling/freeswitch/mod/mod_dptools.so

> > **/home/vBilling/freeswitch/mod/mod_dptools.so:

> undefined symbol: switch_core_media_bug_remove_all**

> >

> >

> >

> > Thanks

> >

> >

> > --- On Sat, 6/23/12, Hector Geraldino <Hector.Geraldino at ipsoft.com>

> wrote:

> >

> >> From: Hector Geraldino <Hector.Geraldino at ipsoft.com>

> >> Subject: Re: [Freeswitch-users] Invalid application

> transfer help

> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

> >> Date: Saturday, June 23, 2012, 12:08 AM

> >> Check if module is loaded:

> >>

> >>> mod_exists mod_dptools

> >>

> >> If not, load it:

> >>

> >>> load mod_dptools

> >>

> >> Or reload it:

> >>

> >>> reload mod_dptools

> >>

> >> -----Original Message-----

> >> From: freeswitch-users-bounces at lists.freeswitch.org

> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org]

> >> On Behalf Of Sherif Omran

> >> Sent: Friday, June 22, 2012 3:37 PM

> >> To: FreeSWITCH Users Help

> >> Subject: Re: [Freeswitch-users] Invalid application

> transfer

> >> help

> >>

> >> how can I load it? should i recompile and

> reinstall

> >>

> >> thanks

> >> Sherif

> >>

> >>

> >> --- On Fri, 6/22/12, Peter Olsson <peter.olsson at visionutveckling.se>

> >> wrote:

> >>

> >>> From: Peter Olsson <peter.olsson at visionutveckling.se>

> >>> Subject: Re: [Freeswitch-users] Invalid

> application

> >> transfer help

> >>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

> >>> Date: Friday, June 22, 2012, 1:18 PM

> >>> Seems you don't have mod_dptools

> >>> loaded anymore.

> >>>

> >>> /Peter

> >>>

> >>> 22 jun 2012 kl. 11:45 skrev "Sherif Omran"

> <sherifomran2000 at yahoo.com<mailto:sherifomran2000 at yahoo.com>>:

> >>>

> >>> Hi Guys,

> >>>

> >>> My server was working fine till today, It does

> not

> >> transfer any call.

> >>> When I call an extension i get? a not

> connected tone

> >> and in the log I

> >>> see invalid application

> >>>

> >>> 2012-06-22 09:37:26.414815 [ERR]

> >> switch_core_session.c:2156 Invalid

> >>> Application transfer

> >>>

> >>> I wonder what this could be? I was trying

> yesterday to

> >> install some

> >>> packages. Could this be related to that?

> >>>

> >>> kind regards,

> >>> Sherif

> >>>

> >>>

> >>>

> >>>

> >>

> ______________________________________________________________________

> >>> ___ Professional FreeSWITCH Consulting

> Services:

> >>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>

> >>> http://www.freeswitchsolutions.com

> >>>

> >>> FreeSWITCH-powered IP PBX: The CudaTel

> Communication

> >> Server

> >>> 

> >>>

> >>> Official FreeSWITCH Sites

> >>> http://www.freeswitch.org

> >>> http://wiki.freeswitch.org

> >>> http://www.cluecon.com

> >>>

> >>> Join Us At ClueCon - Aug 7-9, 2012

> >>>

> >>> FreeSWITCH-users mailing list

> >>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.fr

> >>> eeswitch.org>

> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use

> >>> rs

> >>> http://www.freeswitch.org

> >>>

> >>>

> >>> !DSPAM:4fe43b9a32761615820512!

> >>>

> >>>

> >>

> ______________________________________________________________________

> >>> ___ Professional FreeSWITCH Consulting

> Services:

> >>> consulting at freeswitch.org

> >>> http://www.freeswitchsolutions.com

> >>>

> >>> FreeSWITCH-powered IP PBX: The CudaTel

> Communication

> >> Server

> >>> 

> >>>

> >>> Official FreeSWITCH Sites

> >>> http://www.freeswitch.org

> >>> http://wiki.freeswitch.org

> >>> http://www.cluecon.com

> >>>

> >>> Join Us At ClueCon - Aug 7-9, 2012

> >>>

> >>> FreeSWITCH-users 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

> >>>

> >>

> >>

> _________________________________________________________________________

> >> Professional FreeSWITCH Consulting Services:

> >> consulting at freeswitch.org

> >> http://www.freeswitchsolutions.com

> >>

> >> FreeSWITCH-powered IP PBX: The CudaTel

> Communication Server

> >> 

> >>

> >> Official FreeSWITCH Sites

> >> http://www.freeswitch.org

> >> http://wiki.freeswitch.org

> >> http://www.cluecon.com

> >>

> >> Join Us At ClueCon - Aug 7-9, 2012

> >>

> >> FreeSWITCH-users mailing list

> >> FreeSWITCH-users at lists.freeswitch.org

> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

> >> http://www.freeswitch.org

> >>

> >>

> _________________________________________________________________________

> >> Professional FreeSWITCH Consulting Services:

> >> consulting at freeswitch.org

> >> http://www.freeswitchsolutions.com

> >>

> >> FreeSWITCH-powered IP PBX: The CudaTel

> Communication Server

> >> 

> >>

> >> Official FreeSWITCH Sites

> >> http://www.freeswitch.org

> >> http://wiki.freeswitch.org

> >> http://www.cluecon.com

> >>

> >> Join Us At ClueCon - Aug 7-9, 2012

> >>

> >> FreeSWITCH-users mailing list

> >> FreeSWITCH-users at lists.freeswitch.org

> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

> >> http://www.freeswitch.org

> >>

> >

> >

> _________________________________________________________________________

> > Professional FreeSWITCH Consulting Services:

> > consulting at freeswitch.org

> > http://www.freeswitchsolutions.com

> >

> > FreeSWITCH-powered IP PBX: The CudaTel Communication

> Server

> > 

> >

> > Official FreeSWITCH Sites

> > http://www.freeswitch.org

> > http://wiki.freeswitch.org

> > http://www.cluecon.com

> >

> > Join Us At ClueCon - Aug 7-9, 2012

> >

> > FreeSWITCH-users mailing list

> > FreeSWITCH-users at lists.freeswitch.org

> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

> > http://www.freeswitch.org

> >

> > !DSPAM:4fe5774732761352082344!

> >

>

> _________________________________________________________________________

> Professional FreeSWITCH Consulting Services:

> consulting at freeswitch.org

> http://www.freeswitchsolutions.com

>

> 

> 

>

> Official FreeSWITCH Sites

> http://www.freeswitch.org

> http://wiki.freeswitch.org

> http://www.cluecon.com

>

> Join Us At ClueCon - Aug 7-9, 2012

>

> FreeSWITCH-users mailing list

> FreeSWITCH-users at lists.freeswitch.org

> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

> http://www.freeswitch.org

>



_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com



Join Us At ClueCon - Aug 7-9, 2012



FreeSWITCH-users mailing list

FreeSWITCH-users at 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-----

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at 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 chris at opencsta.org  Sun Jun 24 11:22:12 2012
From: chris at opencsta.org (Chris Mylonas)
Date: Sun, 24 Jun 2012 17:22:12 +1000
Subject: [Freeswitch-users] Cutting out after 30 seconds
In-Reply-To: <AFF35AA0-1E32-4C0F-A2FE-E49D8A55458D@me.com>
References: <AFF35AA0-1E32-4C0F-A2FE-E49D8A55458D@me.com>
Message-ID: <6332A1E4-90AE-4804-9F22-FEAD6CC30FF8@opencsta.org>

Hi Jeremy,

You may get better results with a SIP trace from the network - either tcpdump/wireshark/ngrep.
Could be SIP session timers as a Sunday afternoon guess.

With ngrep, the command for port 5060 is:
ngrep -W byline port 5060

Compare the difference between your working phones and your siemens ones.  There is probably a setting on the siemens for responding to critical packets or something of that ilk.

Cheers
Chris



On 22/06/2012, at 8:47 PM, Jeremy Sharp wrote:

> I have some Siemens DECT sip handsets, which are cutting off on outbound calls after 30 seconds. Doesn't seem to affect other phones, or inbound calls.
> 
> I have tried setting the transport to TCP and adding {ignore_display_updates=true} to the bridge action but this hasn't helped.
> 
> What could it be?
> 
> 2012-06-22 11:43:52.978442 [DEBUG] switch_core_session.c:923 Send signal sofia/internal/1002 at 192.168.1.1 [BREAK]
> 2012-06-22 11:43:52.978442 [DEBUG] switch_core_session.c:923 Send signal sofia/internal/1002 at 192.168.1.1 [BREAK]
> 2012-06-22 11:43:52.998446 [DEBUG] sofia.c:5838 Channel sofia/internal/1002 at 192.168.1.1 entering state [terminating][0]
> 2012-06-22 11:43:52.998446 [DEBUG] switch_channel.c:2899 (sofia/internal/1002 at 192.168.1.1) Callstate Change ACTIVE -> HANGUP
> 2012-06-22 11:43:52.998446 [NOTICE] sofia.c:6628 Hangup sofia/internal/1002 at 192.168.1.1 [CS_EXECUTE] [NORMAL_UNSPECIFIED]
> 2012-06-22 11:43:52.998446 [DEBUG] switch_channel.c:2922 Send signal sofia/internal/1002 at 192.168.1.1 [KILL]
> 2012-06-22 11:43:52.998446 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/1002 at 192.168.1.1 [BREAK]
> 2012-06-22 11:43:52.998446 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD DONE [sofia/internal/1002 at 192.168.1.1]
> 2012-06-22 11:43:52.998446 [DEBUG] switch_ivr_bridge.c:613 Send signal sofia/external/01234567890 [BREAK]
> 2012-06-22 11:43:53.018396 [DEBUG] switch_ivr_bridge.c:501 sofia/internal/1002 at 192.168.1.1 ending bridge by request from write function
> 2012-06-22 11:43:53.018396 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD DONE [sofia/external/01234567890]
> 2012-06-22 11:43:53.018396 [DEBUG] switch_ivr_bridge.c:613 Send signal sofia/internal/1002 at 192.168.1.1 [BREAK]
> 2012-06-22 11:43:53.018396 [DEBUG] switch_channel.c:2899 (sofia/external/01234567890) Callstate Change ACTIVE -> HANGUP
> 2012-06-22 11:43:53.018396 [NOTICE] switch_ivr_bridge.c:676 Hangup sofia/external/01234567890 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2012-06-22 11:43:53.018396 [DEBUG] switch_channel.c:2922 Send signal sofia/external/01234567890 [KILL]
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:1228 Send signal sofia/external/01234567890 [BREAK]
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:443 (sofia/external/01234567890) State EXCHANGE_MEDIA going to sleep
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:385 (sofia/external/01234567890) Running State Change CS_HANGUP
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:625 (sofia/external/01234567890) State HANGUP
> 2012-06-22 11:43:53.018396 [DEBUG] mod_sofia.c:469 Channel sofia/external/01234567890 hanging up, cause: NORMAL_CLEARING
> 2012-06-22 11:43:53.018396 [DEBUG] mod_sofia.c:517 Sending BYE to sofia/external/01234567890
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:47 sofia/external/01234567890 Standard HANGUP, cause: NORMAL_CLEARING
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:625 (sofia/external/01234567890) State HANGUP going to sleep
> 2012-06-22 11:43:53.018396 [DEBUG] switch_ivr_bridge.c:1456 sofia/external/01234567890 skip receive message [UNBRIDGE] (channel is hungup already)
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:416 (sofia/external/01234567890) State Change CS_HANGUP -> CS_REPORTING
> 2012-06-22 11:43:53.018396 [DEBUG] switch_ivr_bridge.c:1459 sofia/internal/1002 at 192.168.1.1 skip receive message [UNBRIDGE] (channel is hungup already)
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:1228 Send signal sofia/external/01234567890 [BREAK]
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:385 (sofia/external/01234567890) Running State Change CS_REPORTING
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:685 (sofia/external/01234567890) State REPORTING
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:79 sofia/external/01234567890 Standard REPORTING, cause: NORMAL_CLEARING
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:685 (sofia/external/01234567890) State REPORTING going to sleep
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:2333 sofia/internal/1002 at 192.168.1.1 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1002 at 192.168.1.1) State EXECUTE going to sleep
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/1002 at 192.168.1.1) Running State Change CS_HANGUP
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:410 (sofia/external/01234567890) State Change CS_REPORTING -> CS_DESTROY
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:1228 Send signal sofia/external/01234567890 [BREAK]
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:1428 Session 45 (sofia/external/01234567890) Locked, Waiting on external entities
> 2012-06-22 11:43:53.018396 [NOTICE] switch_core_session.c:1446 Session 45 (sofia/external/01234567890) Ended
> 2012-06-22 11:43:53.018396 [NOTICE] switch_core_session.c:1448 Close Channel sofia/external/01234567890 [CS_DESTROY]
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:514 (sofia/external/01234567890) Callstate Change HANGUP -> DOWN
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:517 (sofia/external/01234567890) Running State Change CS_DESTROY
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:527 (sofia/external/01234567890) State DESTROY
> 2012-06-22 11:43:53.018396 [DEBUG] mod_sofia.c:374 sofia/external/01234567890 SOFIA DESTROY
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:86 sofia/external/01234567890 Standard DESTROY
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:527 (sofia/external/01234567890) State DESTROY going to sleep
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1002 at 192.168.1.1) State HANGUP
> 2012-06-22 11:43:53.018396 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1002 at 192.168.1.1 hanging up, cause: NORMAL_UNSPECIFIED
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1002 at 192.168.1.1 Standard HANGUP, cause: NORMAL_UNSPECIFIED
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1002 at 192.168.1.1) State HANGUP going to sleep
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/1002 at 192.168.1.1) State Change CS_HANGUP -> CS_REPORTING
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/1002 at 192.168.1.1 [BREAK]
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/1002 at 192.168.1.1) Running State Change CS_REPORTING
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/1002 at 192.168.1.1) State REPORTING
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1002 at 192.168.1.1 Standard REPORTING, cause: NORMAL_UNSPECIFIED
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/1002 at 192.168.1.1) State REPORTING going to sleep
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 192.168.1.1) State Change CS_REPORTING -> CS_DESTROY
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/1002 at 192.168.1.1 [BREAK]
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_session.c:1428 Session 44 (sofia/internal/1002 at 192.168.1.1) Locked, Waiting on external entities
> 2012-06-22 11:43:53.018396 [NOTICE] switch_core_session.c:1446 Session 44 (sofia/internal/1002 at 192.168.1.1) Ended
> 2012-06-22 11:43:53.018396 [NOTICE] switch_core_session.c:1448 Close Channel sofia/internal/1002 at 192.168.1.1 [CS_DESTROY]
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/1002 at 192.168.1.1) Callstate Change HANGUP -> DOWN
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/1002 at 192.168.1.1) Running State Change CS_DESTROY
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/1002 at 192.168.1.1) State DESTROY
> 2012-06-22 11:43:53.018396 [DEBUG] mod_sofia.c:374 sofia/internal/1002 at 192.168.1.1 SOFIA DESTROY
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1002 at 192.168.1.1 Standard DESTROY
> 2012-06-22 11:43:53.018396 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/1002 at 192.168.1.1) State DESTROY going to sleep
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sherifomran2000 at yahoo.com  Sun Jun 24 11:35:33 2012
From: sherifomran2000 at yahoo.com (Sherif Omran)
Date: Sun, 24 Jun 2012 00:35:33 -0700 (PDT)
Subject: [Freeswitch-users] Codec 101 - need help with the basics
In-Reply-To: <CAA4nhyALnwXmd2pVeFXborYtTtbY=9o5yggyi1p3y1KOQjrqFg@mail.gmail.com>
Message-ID: <1340523333.29621.YahooMailClassic@web110815.mail.gq1.yahoo.com>

try using greedy instead of generous? :) it will work


--- On Sun, 6/24/12, Moises Silva <moises.silva at gmail.com> wrote:

From: Moises Silva <moises.silva at gmail.com>
Subject: Re: [Freeswitch-users] Codec 101 - need help with the basics
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Sunday, June 24, 2012, 6:47 AM

On Thu, Jun 21, 2012 at 3:09 AM, ocset <ocset at the800group.com> wrote:

Hi



I need some assistance in understanding the codec minefield.



When a stream of audio is sent to FreeSWITCH (FS) using a particular

codec, does FS just pass it on or is there some level of translation

occurring?

In some situations audio may be just passed thru, but most real life scenarios will involve some level of translation (transcoding).
?

I guess my question relates to a situation where an FXO gateway

(GXW410x) uses the G723 codec and the receiing phone does not have a

matching codec (the phone has G.722, G.711 a law, G.711 u law, G.726 and

G.729 as options). How is this managed by the phone, FreeSWITCH and the

FXO device? Should I be setting something in FS to handle this issue?

Whenever FreeSWITCH detects format mismatch between the parties it will perform transcoding (translation of one audio format to another). You have several configuration options to force a given list of codecs to be offered.?

Now, in some shameless self-promotion, I suggest you to read or watch my presentation on FreeSWITCH media handling from here:
http://www.moythreads.com/congresos/cluecon2011/media-handling-freeswitch-final.pdf
http://www.youtube.com/watch?v=508jpSQk8qE
That should make things a bit clearer for you.?

I have 2 cordless Siemens C610 IP phones an I am having sound quality

issues (incoming sound is soft) and I was wondering how big a part the

codec plays in this equation. I have desk phones that are from the same

manufacturer as the FXO gateway (Grandstream) and they have very good

sound quality.

I don't think the codec has a role in your issue, if the audio is soft or quiet it has more to do with the gain control. If you place calls to those Siemens phones without using the gateway (ie, calling in to play a file) is the audio still low?


Moises Silva
Manager, Software Engineering
msilva at sangoma.com
Sangoma Technologies100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.?? +1 800 388 2475 (N. America)t.?? +1 905 474 1990 x128
f.?? +1 905 474 9223?


Products?|?Solutions?|?Events?|?Contact?|?Wiki?|?Facebook?|?Twitter`| |?YouTube
?

-----Inline Attachment Follows-----

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing 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 alex at thewinelake.com  Sun Jun 24 12:12:56 2012
From: alex at thewinelake.com (Alexander Lake)
Date: Sun, 24 Jun 2012 09:12:56 +0100
Subject: [Freeswitch-users] internal extension is unregistered when
	registering from remote location
In-Reply-To: <DUB109-W8194139EEDA5F62A96753DD9E20@phx.gbl>
References: <DUB109-W8194139EEDA5F62A96753DD9E20@phx.gbl>
Message-ID: <90E5CD1D-4124-4DE1-9D60-BC30B3CBAA9A@thewinelake.com>

...can you confirm that the registration sticks with this from fs_cl:

sofia_contact 2000

We sometimes get similar results when there are client LAN problems.

On 24 Jun 2012, at 02:52, Juan Pablo L. wrote:

> 
> 
> Hi, i have an internal extension 2000 registering from a remote location, extension 2000 can make calls
> to internal extension and through gateways but nobody can call the 2000 extension, when someone tries to reach extension
> 2000 there is this error: 
> "switch_ivr_originate.c:2492 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]"
> 
> and voicemail invites to leave a message.
> 
> when extension 2000 registers i get this in the FS logs:
> 
> sofia_reg.c:1445 SIP auth challenge (REGISTER) on sofia profile 'internal-hq' for [2000 at 10.19.0.8] from ip 190.242.32.118
> 
> so it is registering, when i check in the database this extension is not there. i would appreciate any hints. 
> 
> thanks!
> 		 	   		  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From xyangni at gmail.com  Sun Jun 24 12:44:29 2012
From: xyangni at gmail.com (Yihui Li)
Date: Sun, 24 Jun 2012 09:44:29 +0100
Subject: [Freeswitch-users] FIFO from public context
Message-ID: <CAO1DH1_OQWZTKDCc8MHbWkwrQwRDdpN3UynuOAjNc2sS9=J9BQ@mail.gmail.com>

Hi all,

I am trying to use FIFO and followed the simple call center example here:
http://wiki.freeswitch.org/wiki/Simple_call_center_using_mod_fifo

When the dialplan is in default context and the call is made from a
registered client, it works fine. However, if I move the dialplan to public
context and make the call through sip call on another FS server or let my
services provider forward the call my the sip address, FIFO will not work
and no agent is ringed. Checked the log, it says "INCOMPATIBLE_DESTINATION"
when trying to call agent.

Can anyone help? Thanks.
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From ocset at the800group.com  Sun Jun 24 17:17:48 2012
From: ocset at the800group.com (ocset)
Date: Sun, 24 Jun 2012 21:17:48 +0800
Subject: [Freeswitch-users] Strange behaviour when sending to voicemail
Message-ID: <4FE7137C.4000905@the800group.com>

Hi

I have a FreeSWITCH install that uses a Faktortel SIP gateway for 
incoming/outgoing calls. The gateway codec is g.722 and everything works 
fine except for redirection to voicemail. (I have hard coded the codec 
for the gateway since it would not work otherwise.)

Here is the conf/dialplan/public/01_line1.xml file with the dialplan for 
incoming calls

<extension name="line1">
<condition field="destination_number" expression="^(5555555555)$">
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="call_timeout=20"/>
<action application="bridge" data="group/sales@${domain_name}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="default ${domain} 1000"/>
</condition>
</extension>

When the call is redirected to voicemail, half of the playback message 
is lost and the first thing the user hears is "to listen to the message 
press 1..."

If I change the dialplan to transfer to the "1000" extension instead, 
allowing FreeSWITCH to redirect to voicemail automatically when ext 1001 
does not answer, then voicemail works correctly.

Any suggestions?

Thanks
O.







From ocset at the800group.com  Sun Jun 24 17:34:23 2012
From: ocset at the800group.com (ocset)
Date: Sun, 24 Jun 2012 21:34:23 +0800
Subject: [Freeswitch-users] Bind extension to an IP address
Message-ID: <4FE7175F.90209@the800group.com>

Hi

Is it possible to bind an user/extension to a specific IP address? (e.g. 
extension 1001 can only log in from IP address 192.168.0.101)

Thanks
O.


From saami_mh at ymail.com  Sun Jun 24 17:50:09 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Sun, 24 Jun 2012 06:50:09 -0700 (PDT)
Subject: [Freeswitch-users] question on mod_lua for
	serving_configurations(implement realtime in freeswitch)?
Message-ID: <1340545809.7895.YahooMailNeo@web120105.mail.ne1.yahoo.com>

hi,
in mod_lua for Serving_Configuration (?http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration?)?i have defined the settings as follows:
1-

?<configuration name="lua.conf" description="LUA Configuration">

? <settings>

? ?? ? <param name="xml-handler-script" value="dp.lua"/>
? ? <param name="xml-handler-bindings" value="directory"/>
? ? ? ?
? </settings>
</configuration>

2-?/usr/local/freeswitch/scripts/dp.lua
freeswitch.consoleLog("INFO", "SECTION ? ?" .. XML_REQUEST["section"] .. "\n")
freeswitch.consoleLog("INFO", "TAG_NAME ? " .. XML_REQUEST["tag_name"] .. "\n")
freeswitch.consoleLog("INFO", "KEY_NAME ? " .. XML_REQUEST["key_name"] .. "\n")
freeswitch.consoleLog("INFO", "KEY_VALUE ?" .. XML_REQUEST["key_value"] .. "\n")

freeswitch.consoleLog("INFO",params:serialize())
io.write("TEST \n" .. params:serialize("xml") .. "\n");
--io.write("SIP_AUTH_METHOD\n" .. params:getHeader("sip_auth_method") .. "\n");
--io.write("sip_profile\n" .. params:getHeader("sip_profile") .. "\n");
local sipprofile = params:getHeader("sip_profile");
print ("sip profile is:" .. sipprofile);

3- the output of the above lua file is :



TEST
<event>
? <headers>
? ? <Event-Name>REQUEST_PARAMS</Event-Name>
? ? <Core-UUID>1d640ebe-be01-11e1-b2cd-4109894802cc</Core-UUID>
? ? <FreeSWITCH-Hostname>PBX</FreeSWITCH-Hostname>
? ? <FreeSWITCH-Switchname>PBX</FreeSWITCH-Switchname>
? ? <FreeSWITCH-IPv4>192.168.10.89</FreeSWITCH-IPv4>
? ? <FreeSWITCH-IPv6>%3A%3A1</FreeSWITCH-IPv6>
? ? <Event-Date-Local>2012-06-24%2018%3A13%3A48</Event-Date-Local>
? ? <Event-Date-GMT>Sun,%2024%20Jun%202012%2013%3A43%3A48%20GMT</Event-Date-GMT>
? ? <Event-Date-Timestamp>1340545428075826</Event-Date-Timestamp>
? ? <Event-Calling-File>sofia_reg.c</Event-Calling-File>
? ? <Event-Calling-Function>sofia_reg_parse_auth</Event-Calling-Function>
? ? <Event-Calling-Line-Number>2364</Event-Calling-Line-Number>
? ? <Event-Sequence>1240</Event-Sequence>
? ? <action>sip_auth</action>
? ? <sip_profile>internal</sip_profile>
? ? <sip_user_agent>eyeBeam%20release%201102q%20stamp%2051814</sip_user_agent>
? ? <sip_auth_username>1000</sip_auth_username>
? ? <sip_auth_realm>192.168.10.89</sip_auth_realm>
? ? <sip_auth_nonce>5fef3276-be02-11e1-b2fd-4109894802cc</sip_auth_nonce>
? ? <sip_auth_uri>sip%3A192.168.10.89</sip_auth_uri>
? ? <sip_contact_user>1000</sip_contact_user>
? ? <sip_contact_host>192.168.18.120</sip_contact_host>
? ? <sip_to_user>1000</sip_to_user>
? ? <sip_to_host>192.168.10.89</sip_to_host>
? ? <sip_from_user>1000</sip_from_user>
? ? <sip_from_host>192.168.10.89</sip_from_host>
? ? <sip_request_host>192.168.10.89</sip_request_host>
? ? <sip_auth_qop>auth</sip_auth_qop>
? ? <sip_auth_cnonce>8725f832cb29a466ba073ef5d63de959</sip_auth_cnonce>
? ? <sip_auth_nc>00000010</sip_auth_nc>
? ? <sip_auth_response>ee0155cd5b41536e56898ff585502e75</sip_auth_response>
? ? <sip_auth_method>REGISTER</sip_auth_method>
? ? <key>id</key>
? ? <user>1000</user>
? ? <domain>192.168.10.89</domain>
? ? <ip>192.168.18.120</ip>
? </headers>
</event>

sip profile is?:internal

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

Now i am going to check ?using reltime every user that registers on sip phone -like 3CXphone - instead of in-memory method (flat file on freeswitch), now?
what is the next step for implementing??

Now ?need to define mysql and insert query ?in lua script and compare the query with the value of?params:getHeader ?
the resource for mod_lua serving configuration is a bit


plz help,?
thanks so much
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From jpablolorenzetti at hotmail.com  Sun Jun 24 18:56:19 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Sun, 24 Jun 2012 14:56:19 +0000
Subject: [Freeswitch-users] internal extension is unregistered when
 registering from remote location
In-Reply-To: <90E5CD1D-4124-4DE1-9D60-BC30B3CBAA9A@thewinelake.com>
References: <DUB109-W8194139EEDA5F62A96753DD9E20@phx.gbl>,
	<90E5CD1D-4124-4DE1-9D60-BC30B3CBAA9A@thewinelake.com>
Message-ID: <DUB109-W26DC086CB692D08E936014D9E20@phx.gbl>


thanks for replying, no that command returns user not registers. i even go to the database and check the table contents

and 2000 is effectivly not there.

----------------------------------------
> From: alex at thewinelake.com
> Date: Sun, 24 Jun 2012 09:12:56 +0100
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] internal extension is unregistered when registering from remote location
>
> ...can you confirm that the registration sticks with this from fs_cl:
>
> sofia_contact 2000
>
> We sometimes get similar results when there are client LAN problems.
>
> On 24 Jun 2012, at 02:52, Juan Pablo L. wrote:
>
> >
> >
> > Hi, i have an internal extension 2000 registering from a remote location, extension 2000 can make calls
> > to internal extension and through gateways but nobody can call the 2000 extension, when someone tries to reach extension
> > 2000 there is this error:
> > "switch_ivr_originate.c:2492 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]"
> >
> > and voicemail invites to leave a message.
> >
> > when extension 2000 registers i get this in the FS logs:
> >
> > sofia_reg.c:1445 SIP auth challenge (REGISTER) on sofia profile 'internal-hq' for [2000 at 10.19.0.8] from ip 190.242.32.118
> >
> > so it is registering, when i check in the database this extension is not there. i would appreciate any hints.
> >
> > thanks!
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  

From darcyp at voice2net.ca  Sun Jun 24 19:13:46 2012
From: darcyp at voice2net.ca (Darcy Primrose)
Date: Sun, 24 Jun 2012 11:13:46 -0400
Subject: [Freeswitch-users] persistent registration
Message-ID: <5BC3500296F64506B7C9C8F1E3D8287C@owner397fa27d2>

I have mutliple_registrations set to 'true", if a user on a dsl gets rebooted and receives a new IP address, the old registration remains persistent and gets a value of
EXPSECS(-xxxx) and this continues unless it is flushed or if the command 'RELOADXML" is executed.  Is there anyway to clear these negative (persistent) registrations with intervention.

Darcy Primrose
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From paul at cupis.co.uk  Sun Jun 24 19:39:09 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Sun, 24 Jun 2012 16:39:09 +0100
Subject: [Freeswitch-users] Changing FROM header
In-Reply-To: <CAHT_yt-n7YsZw=pC0KSPFz0iMKjp5CxAsPeQJUYY+F70WTrcYw@mail.gmail.com>
References: <CAHT_yt-n7YsZw=pC0KSPFz0iMKjp5CxAsPeQJUYY+F70WTrcYw@mail.gmail.com>
Message-ID: <4FE7349D.8020508@cupis.co.uk>

On 23/06/12 16:03, Shishko wrote:
> is there a way I can change From header in bridge application line?
> In a-leg I get something as "xxxxxx<sip:xxxxxx at domain.tld>", and I'd
> like to somehow change it to "yyyyyy<sip:yyyyyy at domain.tld>" in
> b-leg.
> I tried with sip_h_From and sip_from_uri, but no success...

Have you tried setting effective_caller_id_name and 
effective_caller_id_number before your bridge?

Regards,


From jpablolorenzetti at hotmail.com  Sun Jun 24 20:50:32 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Sun, 24 Jun 2012 16:50:32 +0000
Subject: [Freeswitch-users] internal extension is unregistered when
 registering from remote location
In-Reply-To: <DUB109-W26DC086CB692D08E936014D9E20@phx.gbl>
References: <DUB109-W8194139EEDA5F62A96753DD9E20@phx.gbl>,
	<90E5CD1D-4124-4DE1-9D60-BC30B3CBAA9A@thewinelake.com>,
	<DUB109-W26DC086CB692D08E936014D9E20@phx.gbl>
Message-ID: <DUB109-W42987F7E51AD29601226F4D9E20@phx.gbl>


Hi, just an update, it seems to be an issue with the sip softphone and not FS configurations issues.
i m testing with ekiga on my linux machine and i get this unregistered user error, but then
i downloaded a client for my android phone and it works like a charm, i can call out from extension 2000
and be called from any internal extension or DID.

----------------------------------------
> From: jpablolorenzetti at hotmail.com
> To: freeswitch-users at lists.freeswitch.org; alex at thewinelake.com
> Subject: RE: [Freeswitch-users] internal extension is unregistered when registering from remote location
> Date: Sun, 24 Jun 2012 14:56:19 +0000
>
>
> thanks for replying, no that command returns user not registers. i even go to the database and check the table contents
>
> and 2000 is effectivly not there.
>
> ----------------------------------------
> > From: alex at thewinelake.com
> > Date: Sun, 24 Jun 2012 09:12:56 +0100
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] internal extension is unregistered when registering from remote location
> >
> > ...can you confirm that the registration sticks with this from fs_cl:
> >
> > sofia_contact 2000
> >
> > We sometimes get similar results when there are client LAN problems.
> >
> > On 24 Jun 2012, at 02:52, Juan Pablo L. wrote:
> >
> > >
> > >
> > > Hi, i have an internal extension 2000 registering from a remote location, extension 2000 can make calls
> > > to internal extension and through gateways but nobody can call the 2000 extension, when someone tries to reach extension
> > > 2000 there is this error:
> > > "switch_ivr_originate.c:2492 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]"
> > >
> > > and voicemail invites to leave a message.
> > >
> > > when extension 2000 registers i get this in the FS logs:
> > >
> > > sofia_reg.c:1445 SIP auth challenge (REGISTER) on sofia profile 'internal-hq' for [2000 at 10.19.0.8] from ip 190.242.32.118
> > >
> > > so it is registering, when i check in the database this extension is not there. i would appreciate any hints.
> > >
> > > thanks!
> > >
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > Join Us At ClueCon - Aug 7-9, 2012
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
 		 	   		  

From drk at drkngs.net  Sun Jun 24 23:02:09 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Sun, 24 Jun 2012 12:02:09 -0700
Subject: [Freeswitch-users]
	=?iso-8859-1?q?question_on_mod=5Flua_for=09ser?=
	=?iso-8859-1?q?ving=5Fconfigurations=28implement_realtime_in_frees?=
	=?iso-8859-1?q?witch=29=3F?=
In-Reply-To: <1340545809.7895.YahooMailNeo@web120105.mail.ne1.yahoo.com>
Message-ID: <20120624190209.f942184d@mail.tritonwest.net>

You want to return a string that is the XML of what the users directory entry would look like. Do lookup in database, etc..  
   
If the user is not found you either want to return NULL, which will cause it to continue and check the XML, or return the 'not found document' just like in XML_CURL that will end the search.   
   
--Dave
      _____  

  From: Samira Mh [mailto:saami_mh at ymail.com]
To: Free SWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Sun, 24 Jun 2012 06:50:09 -0700
Subject: [Freeswitch-users] question on mod_lua for serving_configurations(implement realtime in freeswitch)?

  
  
hi,  
in mod_lua for Serving_Configuration ( http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration ) i have defined the settings as follows:  
1-  

  
 <configuration name="lua.conf" description="LUA Configuration">
  
  <settings>  

  
       <param name="xml-handler-script" value="dp.lua"/>  
    <param name="xml-handler-bindings" value="directory"/>  
         
  </settings>  
</configuration>  

  
2- /usr/local/freeswitch/scripts/dp.lua  
  
freeswitch.consoleLog("INFO", "SECTION    " .. XML_REQUEST["section"] .. "\n")  
freeswitch.consoleLog("INFO", "TAG_NAME   " .. XML_REQUEST["tag_name"] .. "\n")  
freeswitch.consoleLog("INFO", "KEY_NAME   " .. XML_REQUEST["key_name"] .. "\n")  
freeswitch.consoleLog("INFO", "KEY_VALUE  " .. XML_REQUEST["key_value"] .. "\n")  

  
freeswitch.consoleLog("INFO",params:serialize())  
io.write("TEST \n" .. params:serialize("xml") .. "\n");  
--io.write("SIP_AUTH_METHOD\n" .. params:getHeader("sip_auth_method") .. "\n");  
--io.write("sip_profile\n" .. params:getHeader("sip_profile") .. "\n");  
local sipprofile = params:getHeader("sip_profile");  
print ("sip profile is:" .. sipprofile);  

  
3- the output of the above lua file is :  

  

  

  
  
  
TEST  
<event>  
  <headers>  
    <Event-Name>REQUEST_PARAMS</Event-Name>  
    <Core-UUID>1d640ebe-be01-11e1-b2cd-4109894802cc</Core-UUID>  
    <FreeSWITCH-Hostname>PBX</FreeSWITCH-Hostname>  
    <FreeSWITCH-Switchname>PBX</FreeSWITCH-Switchname>  
    <FreeSWITCH-IPv4>192.168.10.89</FreeSWITCH-IPv4>  
    <FreeSWITCH-IPv6>%3A%3A1</FreeSWITCH-IPv6>  
    <Event-Date-Local>2012-06-24%2018%3A13%3A48</Event-Date-Local>  
    <Event-Date-GMT>Sun,%2024%20Jun%202012%2013%3A43%3A48%20GMT</Event-Date-GMT>  
    <Event-Date-Timestamp>1340545428075826</Event-Date-Timestamp>  
    <Event-Calling-File>sofia_reg.c</Event-Calling-File>  
    <Event-Calling-Function>sofia_reg_parse_auth</Event-Calling-Function>  
    <Event-Calling-Line-Number>2364</Event-Calling-Line-Number>  
    <Event-Sequence>1240</Event-Sequence>  
    <action>sip_auth</action>  
    <sip_profile>internal</sip_profile>  
    <sip_user_agent>eyeBeam%20release%201102q%20stamp%2051814</sip_user_agent>  
    <sip_auth_username>1000</sip_auth_username>  
    <sip_auth_realm>192.168.10.89</sip_auth_realm>  
    <sip_auth_nonce>5fef3276-be02-11e1-b2fd-4109894802cc</sip_auth_nonce>  
    <sip_auth_uri>sip%3A192.168.10.89</sip_auth_uri>  
    <sip_contact_user>1000</sip_contact_user>  
    <sip_contact_host>192.168.18.120</sip_contact_host>  
    <sip_to_user>1000</sip_to_user>  
    <sip_to_host>192.168.10.89</sip_to_host>  
    <sip_from_user>1000</sip_from_user>  
    <sip_from_host>192.168.10.89</sip_from_host>  
    <sip_request_host>192.168.10.89</sip_request_host>  
    <sip_auth_qop>auth</sip_auth_qop>  
    <sip_auth_cnonce>8725f832cb29a466ba073ef5d63de959</sip_auth_cnonce>  
    <sip_auth_nc>00000010</sip_auth_nc>  
    <sip_auth_response>ee0155cd5b41536e56898ff585502e75</sip_auth_response>  
    <sip_auth_method>REGISTER</sip_auth_method>  
    <key>id</key>  
    <user>1000</user>  
    <domain>192.168.10.89</domain>  
    <ip>192.168.18.120</ip>  
  </headers>  
</event>  

  
sip profile is :internal
  
---------------------------------------------------  

  
Now i am going to check  using reltime every user that registers on sip phone -like 3CXphone - instead of in-memory method (flat file on freeswitch), now   
what is the next step for implementing? 
  
Now  need to define mysql and insert query  in lua script and compare the query with the value of params:getHeader ?  
the resource for mod_lua serving configuration is a bit
  

  
plz help,   
thanks so much  

  

      
   
 
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From vallimamod.abdullah at imtelecom.fr  Sun Jun 24 23:41:50 2012
From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH)
Date: Sun, 24 Jun 2012 21:41:50 +0200
Subject: [Freeswitch-users] Sending NOTIFY with perl ESL
Message-ID: <02C66638-CD73-49F6-B200-CA76C3733EAF@imtelecom.fr>

Hello List,

I am trying to change BLF lamps status by sending PRESENCE_IN event via perl ESL. My event is sent correctly but corresponding notify always have an empty <dialog-info> wether Answer-State is confirmed or early.

My script is like this:

my $e = ESL::ESLevent->new("PRESENCE_IN");
$e->addHeader("proto", "sip");
$e->addHeader("from", "51 at 10.0.0.1");
$e->addHeader("login", "51 at 10.0.0.1");
$e->addHeader("event_type", "presence");
$e->addHeader("alt_event_type", "dialog");
$e->addHeader("event_count", "1");
$e->addHeader("Presence-Call-Direction", "outbound");
$e->addHeader("Answer-State", "early");
$con->sendEvent($e);


The corresponding ESL event as received by freeswitch is:

sendevent PRESENCE_IN
Event-Name: PRESENCE_IN
proto: sip
from: 51 at 10.0.0.1
login: 51 at 10.0.0.1
event_type: presence
alt_event_type: dialog
event_count: 1
Presence-Call-Direction: outbound
Answer-State: early

But the SIP NOTIFY sent to the subscribers is: 

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="563" state="partial" entity="sip:51 at 10.0.0.1">
</dialog-info>

i.e. it does not contain any <dialog> tag. Is there something wrong with my script ?

Thank you!

-- 
Best Regards,
- vma
.





From abaci64 at gmail.com  Mon Jun 25 01:05:18 2012
From: abaci64 at gmail.com (Abaci)
Date: Sun, 24 Jun 2012 17:05:18 -0400
Subject: [Freeswitch-users] Playback loop
In-Reply-To: <0B02E756F603CC409EB553879B090CC80A1B72BA@HPEXCHVS01.exchange.airg>
References: <0B02E756F603CC409EB553879B090CC80A1B6DC2@HPEXCHVS01.exchange.airg><87f2f3b90901291334i27052568yc90a83d250cc1b76@mail.gmail.com>
	<191c3a030901291354t5cc16a22h32b5defb755c1d3c@mail.gmail.com>
	<0B02E756F603CC409EB553879B090CC80A1B72BA@HPEXCHVS01.exchange.airg>
Message-ID: <4FE7810E.7050406@gmail.com>

is thee a way to loop a file when using playback in xml dialplan? I see 
http://wiki.freeswitch.org/wiki/Channel_Variables#loops but no 
indication on how to use it. can someone please advise how it's supposed 
to work.


From shishko69 at gmail.com  Mon Jun 25 00:20:00 2012
From: shishko69 at gmail.com (Shishko)
Date: Sun, 24 Jun 2012 22:20:00 +0200
Subject: [Freeswitch-users] Changing FROM header
Message-ID: <CAHT_yt93xQSTp-hVhWw3pj_xYXMf0QQ7XJAzv_RDaaAN7bbU6Q@mail.gmail.com>

Yes Paul, I tried, and it works now. Thanks!


From rnbrady at gmail.com  Mon Jun 25 00:47:30 2012
From: rnbrady at gmail.com (Richard Brady)
Date: Sun, 24 Jun 2012 21:47:30 +0100
Subject: [Freeswitch-users] Transport parameters when using RFC3327 Path and
	fs_path
Message-ID: <CAN5esQLRUie3nqwhjtC_+a1S=h5sXLHHJEA0CYb_Jug1_ao-HA@mail.gmail.com>

Hi folks

I have FreeSWITCH running behind Kamailio, which acts as a TCP (client
side) to UDP (FreeSWITCH side) bridge.

The client registers with a binding like:

Contact: <sip:1010 at 192.168.1.228:32204;transport=tcp>

and the Kamailio server is adding a Path header as follows:

Path: <sip:10.1.1.1;lr>

FreeSWITCH stores this as:

Contact:<sip:1010 at 192.168.1.228:32204;transport=tcp;fs_path=%3Csip%3A10.1.1.1%3Blr%3E>

But then when originating /bridging a call to this user, I get:

[DEBUG] sofia_glue.c:2055 sip:10.1.1.1;lr;transport=tcp Setting proxy
route to sofia/internal/sip:1010 at 192.168.1.228:32204

which indicates that the wrong transport is being used.

If I add transport=udp to the Path header generated by Kamailio, the
correct transport is used. However the Path URI, by not including a
transport parameter, implicitly specifies UDP.

Am I doing something wrong or could this possibly be a bug?

Regards,
Richard


From abaci64 at gmail.com  Mon Jun 25 01:34:12 2012
From: abaci64 at gmail.com (Abaci)
Date: Sun, 24 Jun 2012 17:34:12 -0400
Subject: [Freeswitch-users] Bind extension to an IP address
In-Reply-To: <4FE7175F.90209@the800group.com>
References: <4FE7175F.90209@the800group.com>
Message-ID: <4FE787D4.5080603@gmail.com>

if you serve configs xml_curl you can check that it you web app and 
return not found 
<http://wiki.freeswitch.org/wiki/Mod_xml_curl#General_.22Not_found.22_reply>if 
it's not coming from the correct ip.

On 6/24/2012 9:34 AM, ocset wrote:
> Hi
>
> Is it possible to bind an user/extension to a specific IP address? (e.g.
> extension 1001 can only log in from IP address 192.168.0.101)
>
> Thanks
> O.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From cmrienzo at gmail.com  Mon Jun 25 01:37:12 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Sun, 24 Jun 2012 17:37:12 -0400
Subject: [Freeswitch-users] Playback loop
In-Reply-To: <4FE7810E.7050406@gmail.com>
References: <0B02E756F603CC409EB553879B090CC80A1B6DC2@HPEXCHVS01.exchange.airg>
	<87f2f3b90901291334i27052568yc90a83d250cc1b76@mail.gmail.com>
	<191c3a030901291354t5cc16a22h32b5defb755c1d3c@mail.gmail.com>
	<0B02E756F603CC409EB553879B090CC80A1B72BA@HPEXCHVS01.exchange.airg>
	<4FE7810E.7050406@gmail.com>
Message-ID: <CAD-qRkibgFUigsDVZvAcs4tH7K1zoSWYRDPbxDseLXXQBm24bw@mail.gmail.com>

I couldn't find any such channel variable in the source, so I removed that
recent change to the wiki.


On Sun, Jun 24, 2012 at 5:05 PM, Abaci <abaci64 at gmail.com> wrote:

> is thee a way to loop a file when using playback in xml dialplan? I see
> http://wiki.freeswitch.org/wiki/Channel_Variables#loops but no
> indication on how to use it. can someone please advise how it's supposed
> to work.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From avi at avimarcus.net  Mon Jun 25 02:24:50 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 25 Jun 2012 01:24:50 +0300
Subject: [Freeswitch-users] Playback loop
In-Reply-To: <CAD-qRkibgFUigsDVZvAcs4tH7K1zoSWYRDPbxDseLXXQBm24bw@mail.gmail.com>
References: <0B02E756F603CC409EB553879B090CC80A1B6DC2@HPEXCHVS01.exchange.airg>
	<87f2f3b90901291334i27052568yc90a83d250cc1b76@mail.gmail.com>
	<191c3a030901291354t5cc16a22h32b5defb755c1d3c@mail.gmail.com>
	<0B02E756F603CC409EB553879B090CC80A1B72BA@HPEXCHVS01.exchange.airg>
	<4FE7810E.7050406@gmail.com>
	<CAD-qRkibgFUigsDVZvAcs4tH7K1zoSWYRDPbxDseLXXQBm24bw@mail.gmail.com>
Message-ID: <CALNZuwn4OBhYdhnh1w+XyR42WzM1Kusx4OLx_eQU7N8TDisRWw@mail.gmail.com>

Do you want this..?
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_endless_playback

-Avi


On Mon, Jun 25, 2012 at 12:37 AM, Christopher Rienzo <cmrienzo at gmail.com>wrote:

> I couldn't find any such channel variable in the source, so I removed that
> recent change to the wiki.
>
>
>
> On Sun, Jun 24, 2012 at 5:05 PM, Abaci <abaci64 at gmail.com> wrote:
>
>> is thee a way to loop a file when using playback in xml dialplan? I see
>> http://wiki.freeswitch.org/wiki/Channel_Variables#loops but no
>> indication on how to use it. can someone please advise how it's supposed
>> to work.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Mon Jun 25 02:25:42 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 25 Jun 2012 01:25:42 +0300
Subject: [Freeswitch-users] Bind extension to an IP address
In-Reply-To: <4FE787D4.5080603@gmail.com>
References: <4FE7175F.90209@the800group.com> <4FE787D4.5080603@gmail.com>
Message-ID: <CALNZuwm-RPQRBLMRq65fb9eVdUS2W9HURZC5y+=gQ5a2GWhtqA@mail.gmail.com>

Or do a check in the dialplan on that extension's outbound calls.
Or only do IP auth for that account and not digest auth.
-Avi


On Mon, Jun 25, 2012 at 12:34 AM, Abaci <abaci64 at gmail.com> wrote:

>  if you serve configs xml_curl you can check that it you web app and
> return not found
> <http://wiki.freeswitch.org/wiki/Mod_xml_curl#General_.22Not_found.22_reply>if
> it's not coming from the correct ip.
>
>
> On 6/24/2012 9:34 AM, ocset wrote:
>
> Hi
>
> Is it possible to bind an user/extension to a specific IP address? (e.g.
> extension 1001 can only log in from IP address 192.168.0.101)
>
> Thanks
> O.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Mon Jun 25 03:04:26 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 25 Jun 2012 00:04:26 +0100
Subject: [Freeswitch-users] is it possible to use lua in mod_xml_curl ?
In-Reply-To: <CALNZuw=d+aqku6kbwGe0fhTad8V8xnP50bFOZtXnSY1UD2FsCQ@mail.gmail.com>
References: <1340429990.6414.YahooMailNeo@web120105.mail.ne1.yahoo.com>
	<073BDA71-2741-42A4-A85E-13A4A4846C68@gmail.com>
	<CALNZuw=d+aqku6kbwGe0fhTad8V8xnP50bFOZtXnSY1UD2FsCQ@mail.gmail.com>
Message-ID: <8D32F7E0-71FC-4CEB-9685-916CFD02ED0C@gmail.com>

Indeed, although in setup this case mod_lua is registering the same hook that mod_xml_curl does - mod_xml_curl doesn't get used at all.

Sent from my iPad


On 23 Jun 2012, at 21:01, Avi Marcus <avi at avimarcus.net> wrote:

> Lua can serve xml_curl directly without a webserver.
> See: http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration
> 
> -Avi
> 
> 
> 
> On Sat, Jun 23, 2012 at 3:12 PM, Steven Ayre <steveayre at gmail.com> wrote:
> This isn't a FS question.
> 
> You can use any language your webserver allows you to use. FS only cares about the XML output.
> 
> If your webserver supports lua, then yes you can use lua.
> 
> Steve on iPhone
> 
> 
> 
> On 23 Jun 2012, at 06:39, Samira Mh <saami_mh at ymail.com> wrote:
> 
>> the supported languages in mod_xml_cur include:
>> perl,python,php,c++,c# , but can i use lua ?
>> lua is meet our requirements for implementing mod_xml_curl
>> 
>>  
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing 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  Mon Jun 25 05:54:06 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 24 Jun 2012 20:54:06 -0500
Subject: [Freeswitch-users] Sharing: Call Intercept
In-Reply-To: <C066384B0C1F1E4B9B870F050D7F1A9D402150F5@ADAMA.mcpeekdodge.local>
References: <C066384B0C1F1E4B9B870F050D7F1A9D402150F5@ADAMA.mcpeekdodge.local>
Message-ID: <CAKbxfG9qBAQnNB0wfMR3Bkm7uUK83QgiLHSUAVbAeDiFuhD62A@mail.gmail.com>

Thanks for the positive comment. =D

BTW, that reminds me.  I added a new pickup endpoint that you might
want to have a look at as well.
It lets you add a special pickup endpoint to a forked dial so say you
are calling ext 1000, you can do

<action application="bridge" data="user/1000,pickup/1000"/>

Then from another call you can execute:

<action application="pickup" data="1000"/>
And your channel will be returned to that unanswered originate.

We should probably get it documented.


On Sat, Jun 23, 2012 at 9:29 AM, Frank Busalacchi Jr
<frankjr at mcpeekdodge.com> wrote:
> Over the last 6 months, I have received all kinds of support and help from
> the freeswitch community (both IRC and this list) in learning to control
> this incredible piece of software.? Im truly grateful, thanks everyone.
>
>
>
> I run freeswitch at my car dealership.? We have approximately 40 sip
> extensions, an?ATT voip?connection,?and?6 analog outbound?fax lines provided
> by freetdm.? Prior to freeswitch, I was running an old Lucent/Avaya Merlin
> "key" system.? One of the features of the Merlin was that it was pretty easy
> to do "directed callpick".? In essence, you would dial *7 + extension you
> wanted to pickup.? Provided the call was either on hold or ringing at that
> extension, you would pick it up.? When I first started using Freeswitch, I
> was very disappointed in the directed callpick.? Before I realized that the
> call pickup that was built into the sample dialplan was really just an
> example to?show a simple possibility for using intercept.? I was convinced
> that it was a totally useful feature.? I couldn't believe that was as good
> as it got...? My problem with the directed pickup that is in the sample
> dialplan is that it can steal a live call right from another extension, as
> well as it really only works with the most recent call on that extension.
>
>
>
> Then?I woke up, and rolled my own 8-) .? In an effort to provide just
> another example, I wanted to share the piece of code I've developed.? I will
> warn that this was the first perl script I have ever written, and probably
> is pretty ugly,?but it seems to work perfectly in our environment.? Bottom
> line, do what you want with it...I'm just trying to help other beginner's
> like myself:
>
>
>
> To all beginners:? You will need to build and enable mod_perl for this to
> work...
>
>
>
> First, we need an entry in your XML dialplan to see the user's request to
> interecept a "held" or "ringing" extension.? The only 4 digit extensions at
> my company are internal ones so I match on ** + 4-digit-extension.
>
>
>
> In the xML dialplan:
>
>
>
> ??? <extension name="intercept-ext">
> ????? <condition field="destination_number" expression="^\*\*(\d{4})$">
> ??????? <action application="answer"/>
> ??????? <action application="perl"
> data="/usr/local/freeswitch/scripts/myintercept.pl $1"/>
> ??????? <action application="sleep" data="2000"/>
> ????? </condition>
> ??? </extension>
>
> Then, we need the actual perl script:
>
> /usr/local/freeswitch/scripts/myintercept.pl:
>
>
> # This is code to allow us to use intercept?to grab?only calls that are on
> hold or ringing.
> #
> # by Frank Busalacchi 2012-06-13
>
> use strict;
> use Switch;
> use Data::Dumper;
> use XML::Simple;
> use POSIX qw(strftime);
> our $session;
>
> {
> ??? my $dest_target = $ARGV[0];
>
> ??? my $api = new freeswitch::API();
> ??? my $xml = new XML::Simple;
> ??? my $chan_xml = $xml->XMLin($api->executeString("show detailed_calls as
> xml"),KeyAttr => 'row', ForceArray => ['row']);
>
> ??? my %candidates;
> ??? my $result;
>
> ??? foreach my $row (@{$chan_xml->{row}}) {
> ??????? if ($row->{callstate} eq "RINGING" && $row->{dest} eq $dest_target)
> {
> ??????????? $candidates{ $row->{created_epoch} - 1000 }? =
> $row->{call_uuid};?????#In the case of a tie, we want?to pickup ringing
> calls before held calls...
> ??????????? next;
> ??????? }
>
> ??????? elsif ($row->{callstate} eq "HELD" && $row->{b_sent_callee_num} eq
> $dest_target ) {
> ??????????? $candidates{ $row->{created_epoch} } = $row->{b_uuid};
> ??????????? next;
> ??????? }
>
> ??????? elsif ($row->{b_callstate} eq "HELD" && $row->{b_callee_num} eq
> $dest_target ) {
> ??????????? $candidates{ $row->{b_created_epoch} } = $row->{uuid};
> ??????????? next;
> ??????? }
> ??? }
>
> ??? if( keys( %candidates ) == 0) {
> ??????? $result = "[Intercept] No suitable channel to intercept on extension
> $dest_target.";
> ??????? freeswitch::consoleLog("INFO","$result\n");
> ??? }
>
> ??? else {
> ??????? my $user = $session->getVariable('caller_id_name');
> ??????? my $cid_num = $session->getVariable('caller_id_number');
>
> ??????? for my $winner ( sort keys %candidates ) {
>
> ??????????? my $uuid= $candidates{$winner};
>
> ??????????? $result = "[Intercept]User:$user" . "[" . "$cid_num" . "]
> intercepting Extension:$dest_target" . ".? UUID:" . "$uuid" . "\n";
> ??????????? freeswitch::consoleLog("INFO","$result\n");
> ??????????? $session->execute("intercept",$uuid);
> ??????????? last;
> ??????? }
> ??? }
> ??? 1;
> }
>
>
>
> Enjoy!
>
>
>
> -Frank
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Mon Jun 25 06:05:51 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 24 Jun 2012 21:05:51 -0500
Subject: [Freeswitch-users] Bind extension to an IP address
In-Reply-To: <CALNZuwm-RPQRBLMRq65fb9eVdUS2W9HURZC5y+=gQ5a2GWhtqA@mail.gmail.com>
References: <4FE7175F.90209@the800group.com> <4FE787D4.5080603@gmail.com>
	<CALNZuwm-RPQRBLMRq65fb9eVdUS2W9HURZC5y+=gQ5a2GWhtqA@mail.gmail.com>
Message-ID: <CAKbxfG_iUqoU0VFAsKu5a=cwV8fh02WzrEx1R70cMgXXcFy6iQ@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Acl#Users

On Sun, Jun 24, 2012 at 5:25 PM, Avi Marcus <avi at avimarcus.net> wrote:
> Or do a check in the dialplan on that extension's outbound calls.
> Or only do IP auth for that account and not digest auth.
> -Avi
>
>
> On Mon, Jun 25, 2012 at 12:34 AM, Abaci <abaci64 at gmail.com> wrote:
>>
>> if you serve configs xml_curl you can check that it you web app and return
>> not found if it's not coming from the correct ip.
>>
>>
>> On 6/24/2012 9:34 AM, ocset wrote:
>>
>> Hi
>>
>> Is it possible to bind an user/extension to a specific IP address? (e.g.
>> extension 1001 can only log in from IP address 192.168.0.101)
>>
>> Thanks
>> O.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Mon Jun 25 06:40:22 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 24 Jun 2012 19:40:22 -0700
Subject: [Freeswitch-users] Are you using NDLB-force-rport = "safe" ?
Message-ID: <CAKzWOxWvgHW0cR56D26=hxbey2HT=k2cdibPkArA66y7jKM4WA@mail.gmail.com>

Hi gang,

A friend of mine asked how many people are using this. Are you? It's mostly
for Polycoms and non-polys to have force rport in a typical hosted
environment. If you have been using this setting in production please give
us some feedback.

Thanks!

-Michael
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From dujinfang at gmail.com  Mon Jun 25 09:51:30 2012
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 25 Jun 2012 13:51:30 +0800
Subject: [Freeswitch-users] show channels, comma vs TAB ?
Message-ID: <C10DA97B139149F19DBFC99DDEC23543@gmail.com>

Hi,

I use originate user/1000 &bridge({aa=bb,cc=dd}user/1002)

show channels gives a comma separated list, however, as some app data contains commas, this will be hard to parse, could we change to TAB? I understand query the db directly would be fine but sometimes it's just easy to parse the cli output. ideas? thanks.


uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
d412fdda-5fbc-4754-9803-2db06898a8b0,outbound,2012-06-25 13:47:03,1340603223,sofia/internal/sip:1000 at 192.168.2.102:26570,CS_EXECUTE,Outbound Call,1000,,1000,bridge,{aa=bb,cc=dd}user/1002,,default,PCMA,8000,64000,PCMA,8000,64000,,seven-macpro.local,1000 at 192.168.2.102,,ACTIVE,,,RECV,d412fdda-5fbc-4754-9803-2db06898a8b0,,
b5ca402e-d72e-4109-a8ee-634e873f3c31,outbound,2012-06-25 13:47:30,1340603250,sofia/internal/sip:1002 at 192.168.2.102:20676,CS_CONSUME_MEDIA,Extension 1000,1000,192.168.2.102,1002,,,,default,,,,,,,,seven-macpro.local,1002 at 192.168.2.102,,RINGING,Outbound Call,1002,,d412fdda-5fbc-4754-9803-2db06898a8b0,,


-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn

Sent with Sparrow (http://www.sparrowmailapp.com)

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From jmesquita at freeswitch.org  Mon Jun 25 09:55:10 2012
From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Mon, 25 Jun 2012 02:55:10 -0300
Subject: [Freeswitch-users] Sharing: Call Intercept
In-Reply-To: <CAKbxfG9qBAQnNB0wfMR3Bkm7uUK83QgiLHSUAVbAeDiFuhD62A@mail.gmail.com>
References: <C066384B0C1F1E4B9B870F050D7F1A9D402150F5@ADAMA.mcpeekdodge.local>
	<CAKbxfG9qBAQnNB0wfMR3Bkm7uUK83QgiLHSUAVbAeDiFuhD62A@mail.gmail.com>
Message-ID: <62BB9D5255424985932A96663192F254@freeswitch.org>

Tony, I will get it documented tomorrow. Will create a page just for it.

Regards,  

--  
Jo?o Mesquita
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Sunday, June 24, 2012 at 10:54 PM, Anthony Minessale wrote:

> Thanks for the positive comment. =D
>  
> BTW, that reminds me. I added a new pickup endpoint that you might
> want to have a look at as well.
> It lets you add a special pickup endpoint to a forked dial so say you
> are calling ext 1000, you can do
>  
> <action application="bridge" data="user/1000,pickup/1000"/>
>  
> Then from another call you can execute:
>  
> <action application="pickup" data="1000"/>
> And your channel will be returned to that unanswered originate.
>  
> We should probably get it documented.
>  
>  
> On Sat, Jun 23, 2012 at 9:29 AM, Frank Busalacchi Jr
> <frankjr at mcpeekdodge.com (mailto:frankjr at mcpeekdodge.com)> wrote:
> > Over the last 6 months, I have received all kinds of support and help from
> > the freeswitch community (both IRC and this list) in learning to control
> > this incredible piece of software.  Im truly grateful, thanks everyone.
> >  
> >  
> >  
> > I run freeswitch at my car dealership.  We have approximately 40 sip
> > extensions, an ATT voip connection, and 6 analog outbound fax lines provided
> > by freetdm.  Prior to freeswitch, I was running an old Lucent/Avaya Merlin
> > "key" system.  One of the features of the Merlin was that it was pretty easy
> > to do "directed callpick".  In essence, you would dial *7 + extension you
> > wanted to pickup.  Provided the call was either on hold or ringing at that
> > extension, you would pick it up.  When I first started using Freeswitch, I
> > was very disappointed in the directed callpick.  Before I realized that the
> > call pickup that was built into the sample dialplan was really just an
> > example to show a simple possibility for using intercept.  I was convinced
> > that it was a totally useful feature.  I couldn't believe that was as good
> > as it got...  My problem with the directed pickup that is in the sample
> > dialplan is that it can steal a live call right from another extension, as
> > well as it really only works with the most recent call on that extension.
> >  
> >  
> >  
> > Then I woke up, and rolled my own 8-) .  In an effort to provide just
> > another example, I wanted to share the piece of code I've developed.  I will
> > warn that this was the first perl script I have ever written, and probably
> > is pretty ugly, but it seems to work perfectly in our environment.  Bottom
> > line, do what you want with it...I'm just trying to help other beginner's
> > like myself:
> >  
> >  
> >  
> > To all beginners:  You will need to build and enable mod_perl for this to
> > work...
> >  
> >  
> >  
> > First, we need an entry in your XML dialplan to see the user's request to
> > interecept a "held" or "ringing" extension.  The only 4 digit extensions at
> > my company are internal ones so I match on ** + 4-digit-extension.
> >  
> >  
> >  
> > In the xML dialplan:
> >  
> >  
> >  
> >     <extension name="intercept-ext">
> >       <condition field="destination_number" expression="^\*\*(\d{4})$">
> >         <action application="answer"/>
> >         <action application="perl"
> > data="/usr/local/freeswitch/scripts/myintercept.pl (http://myintercept.pl) $1"/>
> >         <action application="sleep" data="2000"/>
> >       </condition>
> >     </extension>
> >  
> > Then, we need the actual perl script:
> >  
> > /usr/local/freeswitch/scripts/myintercept.pl (http://myintercept.pl):
> >  
> >  
> > # This is code to allow us to use intercept to grab only calls that are on
> > hold or ringing.
> > #
> > # by Frank Busalacchi 2012-06-13
> >  
> > use strict;
> > use Switch;
> > use Data::Dumper;
> > use XML::Simple;
> > use POSIX qw(strftime);
> > our $session;
> >  
> > {
> >     my $dest_target = $ARGV[0];
> >  
> >     my $api = new freeswitch::API();
> >     my $xml = new XML::Simple;
> >     my $chan_xml = $xml->XMLin($api->executeString("show detailed_calls as
> > xml"),KeyAttr => 'row', ForceArray => ['row']);
> >  
> >     my %candidates;
> >     my $result;
> >  
> >     foreach my $row (@{$chan_xml->{row}}) {
> >         if ($row->{callstate} eq "RINGING" && $row->{dest} eq $dest_target)
> > {
> >             $candidates{ $row->{created_epoch} - 1000 }  =
> > $row->{call_uuid};     #In the case of a tie, we want to pickup ringing
> > calls before held calls...
> >             next;
> >         }
> >  
> >         elsif ($row->{callstate} eq "HELD" && $row->{b_sent_callee_num} eq
> > $dest_target ) {
> >             $candidates{ $row->{created_epoch} } = $row->{b_uuid};
> >             next;
> >         }
> >  
> >         elsif ($row->{b_callstate} eq "HELD" && $row->{b_callee_num} eq
> > $dest_target ) {
> >             $candidates{ $row->{b_created_epoch} } = $row->{uuid};
> >             next;
> >         }
> >     }
> >  
> >     if( keys( %candidates ) == 0) {
> >         $result = "[Intercept] No suitable channel to intercept on extension
> > $dest_target.";
> >         freeswitch::consoleLog("INFO","$result\n");
> >     }
> >  
> >     else {
> >         my $user = $session->getVariable('caller_id_name');
> >         my $cid_num = $session->getVariable('caller_id_number');
> >  
> >         for my $winner ( sort keys %candidates ) {
> >  
> >             my $uuid= $candidates{$winner};
> >  
> >             $result = "[Intercept]User:$user" . "[" . "$cid_num" . "]
> > intercepting Extension:$dest_target" . ".  UUID:" . "$uuid" . "\n";
> >             freeswitch::consoleLog("INFO","$result\n");
> >             $session->execute("intercept",$uuid);
> >             last;
> >         }
> >     }
> >     1;
> > }
> >  
> >  
> >  
> > Enjoy!
> >  
> >  
> >  
> > -Frank
> >  
> >  
> >  
> >  
> >  
> >  
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > http://www.freeswitchsolutions.com
> >  
> > 
> > 
> >  
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >  
> > Join Us At ClueCon - Aug 7-9, 2012
> >  
> > 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:anthony_minessale at hotmail.com)
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:anthony.minessale at gmail.com)
> IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch
>  
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org (mailto:888 at conference.freeswitch.org)
> googletalk:conf+888 at conference.freeswitch.org (mailto:conf+888 at conference.freeswitch.org)
> pstn:+19193869900
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> Join Us At ClueCon - Aug 7-9, 2012
>  
> 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
>  
>  


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From peter.olsson at visionutveckling.se  Mon Jun 25 10:07:33 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 25 Jun 2012 06:07:33 +0000
Subject: [Freeswitch-users] show channels, comma vs TAB ?
In-Reply-To: <C10DA97B139149F19DBFC99DDEC23543@gmail.com>
References: <C10DA97B139149F19DBFC99DDEC23543@gmail.com>
Message-ID: <4433A21B-E9D8-4D6B-A1EE-40769B0F90EF@visionutveckling.se>

I think you can also use "show channels as xml", which will give you better output for parsing.

/Peter

25 jun 2012 kl. 07:59 skrev "Seven Du" <dujinfang at gmail.com<mailto:dujinfang at gmail.com>>:

Hi,

I use originate user/1000 &bridge({aa=bb,cc=dd}user/1002)

show channels gives a comma separated list, however, as some app data contains commas, this will be hard to parse, could we change to TAB? I understand query the db directly would be fine but sometimes it's just easy to parse the cli output. ideas? thanks.


uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
d412fdda-5fbc-4754-9803-2db06898a8b0,outbound,2012-06-25 13:47:03,1340603223,sofia/internal/sip:1000 at 192.168.2.102:26570,CS_EXECUTE,Outbound Call,1000,,1000,bridge,{aa=bb,cc=dd}user/1002,,default,PCMA,8000,64000,PCMA,8000,64000,,seven-macpro.local,1000 at 192.168.2.102<mailto:1000 at 192.168.2.102>,,ACTIVE,,,RECV,d412fdda-5fbc-4754-9803-2db06898a8b0,,
b5ca402e-d72e-4109-a8ee-634e873f3c31,outbound,2012-06-25 13:47:30,1340603250,sofia/internal/sip:1002 at 192.168.2.102:20676,CS_CONSUME_MEDIA,Extension 1000,1000,192.168.2.102,1002,,,,default,,,,,,,,seven-macpro.local,1002 at 192.168.2.102<mailto:1002 at 192.168.2.102>,,RINGING,Outbound Call,1002,,d412fdda-5fbc-4754-9803-2db06898a8b0,,


--
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn

Sent with Sparrow<http://www.sparrowmailapp.com>

!DSPAM:4fe7fb0c32761667990988!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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!DSPAM:4fe7fb0c32761667990988!


From avi at avimarcus.net  Mon Jun 25 10:30:49 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 25 Jun 2012 09:30:49 +0300
Subject: [Freeswitch-users] show channels, comma vs TAB ?
In-Reply-To: <4433A21B-E9D8-4D6B-A1EE-40769B0F90EF@visionutveckling.se>
References: <C10DA97B139149F19DBFC99DDEC23543@gmail.com>
	<4433A21B-E9D8-4D6B-A1EE-40769B0F90EF@visionutveckling.se>
Message-ID: <CALNZuwnLfCEmiL7+bzmnaMGotemUk5jtuRssA-WU2j3nug-3Cg@mail.gmail.com>

There's no json format, unfortunately...
-Avi


On Mon, Jun 25, 2012 at 9:07 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> I think you can also use "show channels as xml", which will give you
> better output for parsing.
>
> /Peter
>
> 25 jun 2012 kl. 07:59 skrev "Seven Du" <dujinfang at gmail.com<mailto:
> dujinfang at gmail.com>>:
>
> Hi,
>
> I use originate user/1000 &bridge({aa=bb,cc=dd}user/1002)
>
> show channels gives a comma separated list, however, as some app data
> contains commas, this will be hard to parse, could we change to TAB? I
> understand query the db directly would be fine but sometimes it's just easy
> to parse the cli output. ideas? thanks.
>
>
>
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
> d412fdda-5fbc-4754-9803-2db06898a8b0,outbound,2012-06-25
> 13:47:03,1340603223,sofia/internal/sip:1000 at 192.168.2.102:26570,CS_EXECUTE,Outbound
> Call,1000,,1000,bridge,{aa=bb,cc=dd}user/1002,,default,PCMA,8000,64000,PCMA,8000,64000,,seven-macpro.local,
> 1000 at 192.168.2.102<mailto:1000 at 192.168.2.102
> >,,ACTIVE,,,RECV,d412fdda-5fbc-4754-9803-2db06898a8b0,,
> b5ca402e-d72e-4109-a8ee-634e873f3c31,outbound,2012-06-25
> 13:47:30,1340603250,sofia/internal/sip:1002 at 192.168.2.102:20676,CS_CONSUME_MEDIA,Extension
> 1000,1000,192.168.2.102,1002,,,,default,,,,,,,,seven-macpro.local,
> 1002 at 192.168.2.102<mailto:1002 at 192.168.2.102>,,RINGING,Outbound
> Call,1002,,d412fdda-5fbc-4754-9803-2db06898a8b0,,
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> Sent with Sparrow<http://www.sparrowmailapp.com>
>
> !DSPAM:4fe7fb0c32761667990988!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>
> !DSPAM:4fe7fb0c32761667990988!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Jun 25 11:27:11 2012
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 25 Jun 2012 15:27:11 +0800
Subject: [Freeswitch-users] show channels, comma vs TAB ?
In-Reply-To: <CALNZuwnLfCEmiL7+bzmnaMGotemUk5jtuRssA-WU2j3nug-3Cg@mail.gmail.com>
References: <C10DA97B139149F19DBFC99DDEC23543@gmail.com>
	<4433A21B-E9D8-4D6B-A1EE-40769B0F90EF@visionutveckling.se>
	<CALNZuwnLfCEmiL7+bzmnaMGotemUk5jtuRssA-WU2j3nug-3Cg@mail.gmail.com>
Message-ID: <8A9D25F4E82046D68A56F2CF0F8E36EB@gmail.com>

as xml is nice, I just forgot it.

And we could make a json format :), let me take a look.

Thanks.


On Monday, June 25, 2012 at 2:30 PM, Avi Marcus wrote:

> There's no json format, unfortunately...
> -Avi 
> 
> On Mon, Jun 25, 2012 at 9:07 AM, Peter Olsson <peter.olsson at visionutveckling.se (mailto:peter.olsson at visionutveckling.se)> wrote:
> > I think you can also use "show channels as xml", which will give you better output for parsing.
> > 
> > /Peter
> > 
> > 25 jun 2012 kl. 07:59 skrev "Seven Du" <dujinfang at gmail.com (mailto:dujinfang at gmail.com)<mailto:dujinfang at gmail.com>>:
> > 
> > Hi,
> > 
> > I use originate user/1000 &bridge({aa=bb,cc=dd}user/1002)
> > 
> > show channels gives a comma separated list, however, as some app data contains commas, this will be hard to parse, could we change to TAB? I understand query the db directly would be fine but sometimes it's just easy to parse the cli output. ideas? thanks.
> > 
> > 
> > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
> > d412fdda-5fbc-4754-9803-2db06898a8b0,outbound,2012-06-25 13:47:03,1340603223,sofia/internal/sip:1000 at 192.168.2.102:26570 (http://sip:1000 at 192.168.2.102:26570),CS_EXECUTE,Outbound Call,1000,,1000,bridge,{aa=bb,cc=dd}user/1002,,default,PCMA,8000,64000,PCMA,8000,64000,,seven-macpro.local,1000 at 192.168.2.102 (mailto:1000 at 192.168.2.102)<mailto:1000 at 192.168.2.102>,,ACTIVE,,,RECV,d412fdda-5fbc-4754-9803-2db06898a8b0,,
> > b5ca402e-d72e-4109-a8ee-634e873f3c31,outbound,2012-06-25 13:47:30,1340603250,sofia/internal/sip:1002 at 192.168.2.102:20676 (http://sip:1002 at 192.168.2.102:20676),CS_CONSUME_MEDIA,Extension 1000,1000,192.168.2.102,1002,,,,default,,,,,,,,seven-macpro.local,1002 at 192.168.2.102 (mailto:1002 at 192.168.2.102)<mailto:1002 at 192.168.2.102>,,RINGING,Outbound Call,1002,,d412fdda-5fbc-4754-9803-2db06898a8b0,,
> > 
> > 
> > --
> > About: http://about.me/dujinfang
> > Blog: http://www.dujinfang.com
> > Proj:  http://www.freeswitch.org.cn
> > 
> > Sent with Sparrow<http://www.sparrowmailapp.com>
> > 
> > !DSPAM:4fe7fb0c32761667990988!
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)<mailto:consulting at freeswitch.org>
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > Join Us At ClueCon - Aug 7-9, 2012
> > 
> > FreeSWITCH-users mailing list
> > 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
> > 
> > 
> > !DSPAM:4fe7fb0c32761667990988!
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > Join Us At ClueCon - Aug 7-9, 2012
> > 
> > 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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> 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
> 
> 


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From shadowym at hotmail.com  Mon Jun 25 04:32:47 2012
From: shadowym at hotmail.com (shadowym)
Date: Sun, 24 Jun 2012 17:32:47 -0700
Subject: [Freeswitch-users] http_cache not working with latest RPM
Message-ID: <COL124-DS18C9142F57F43CE265E72FDDE10@phx.gbl>

I'm running Whistle or Kazoo or whatever they are changing the name to.  The
2600hz cloud thingy.  

 

Anways, it uses http_cache for sound files.  Using latest Freeswitch RPM it
doesn't work.  I found out that I needed to create
/usr/share/freeswitch/http_cache directory. Without that it won't work
because that is where /etc/freeswitch/autoload_configs/http_cache.conf.xml
points to by default.  Didn't have to do that before so I am assuming the
older RPM's created that before.

 

Not sure where I am supposed to post this bug so I am posting it here.  The
2600hz guys says it's a problem with the FS RPM SPEC file.

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From dujinfang at gmail.com  Mon Jun 25 11:41:25 2012
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 25 Jun 2012 15:41:25 +0800
Subject: [Freeswitch-users] show channels, comma vs TAB ?
In-Reply-To: <8A9D25F4E82046D68A56F2CF0F8E36EB@gmail.com>
References: <C10DA97B139149F19DBFC99DDEC23543@gmail.com>
	<4433A21B-E9D8-4D6B-A1EE-40769B0F90EF@visionutveckling.se>
	<CALNZuwnLfCEmiL7+bzmnaMGotemUk5jtuRssA-WU2j3nug-3Cg@mail.gmail.com>
	<8A9D25F4E82046D68A56F2CF0F8E36EB@gmail.com>
Message-ID: <57A3B7695BD048C5BF17D6AB291C7821@gmail.com>

I just found I can run

show channels as delim \t

Cool! 

On Monday, June 25, 2012 at 3:27 PM, Seven Du wrote:

> as xml is nice, I just forgot it.
> 
> And we could make a json format :), let me take a look.
> 
> Thanks.
> 
> 
> On Monday, June 25, 2012 at 2:30 PM, Avi Marcus wrote:
> 
> > There's no json format, unfortunately...
> > -Avi 
> > 
> > On Mon, Jun 25, 2012 at 9:07 AM, Peter Olsson <peter.olsson at visionutveckling.se (mailto:peter.olsson at visionutveckling.se)> wrote:
> > > I think you can also use "show channels as xml", which will give you better output for parsing.
> > > 
> > > /Peter
> > > 
> > > 25 jun 2012 kl. 07:59 skrev "Seven Du" <dujinfang at gmail.com (mailto:dujinfang at gmail.com)<mailto:dujinfang at gmail.com>>:
> > > 
> > > Hi,
> > > 
> > > I use originate user/1000 &bridge({aa=bb,cc=dd}user/1002)
> > > 
> > > show channels gives a comma separated list, however, as some app data contains commas, this will be hard to parse, could we change to TAB? I understand query the db directly would be fine but sometimes it's just easy to parse the cli output. ideas? thanks.
> > > 
> > > 
> > > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
> > > d412fdda-5fbc-4754-9803-2db06898a8b0,outbound,2012-06-25 13:47:03,1340603223,sofia/internal/sip:1000 at 192.168.2.102:26570 (http://sip:1000 at 192.168.2.102:26570),CS_EXECUTE,Outbound Call,1000,,1000,bridge,{aa=bb,cc=dd}user/1002,,default,PCMA,8000,64000,PCMA,8000,64000,,seven-macpro.local,1000 at 192.168.2.102 (mailto:1000 at 192.168.2.102)<mailto:1000 at 192.168.2.102>,,ACTIVE,,,RECV,d412fdda-5fbc-4754-9803-2db06898a8b0,,
> > > b5ca402e-d72e-4109-a8ee-634e873f3c31,outbound,2012-06-25 13:47:30,1340603250,sofia/internal/sip:1002 at 192.168.2.102:20676 (http://sip:1002 at 192.168.2.102:20676),CS_CONSUME_MEDIA,Extension 1000,1000,192.168.2.102,1002,,,,default,,,,,,,,seven-macpro.local,1002 at 192.168.2.102 (mailto:1002 at 192.168.2.102)<mailto:1002 at 192.168.2.102>,,RINGING,Outbound Call,1002,,d412fdda-5fbc-4754-9803-2db06898a8b0,,
> > > 
> > > 
> > > --
> > > About: http://about.me/dujinfang
> > > Blog: http://www.dujinfang.com
> > > Proj:  http://www.freeswitch.org.cn
> > > 
> > > Sent with Sparrow<http://www.sparrowmailapp.com>
> > > 
> > > !DSPAM:4fe7fb0c32761667990988!
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org (mailto:consulting at freeswitch.org)<mailto:consulting at freeswitch.org>
> > > http://www.freeswitchsolutions.com
> > > 
> > > 
> > > 
> > > 
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > > 
> > > Join Us At ClueCon - Aug 7-9, 2012
> > > 
> > > FreeSWITCH-users mailing list
> > > 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
> > > 
> > > 
> > > !DSPAM:4fe7fb0c32761667990988!
> > > 
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > > http://www.freeswitchsolutions.com
> > > 
> > > 
> > > 
> > > 
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > > 
> > > Join Us At ClueCon - Aug 7-9, 2012
> > > 
> > > 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
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > Join Us At ClueCon - Aug 7-9, 2012
> > 
> > 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
> > 
> > 
> > 
> 
> 

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From avi at avimarcus.net  Mon Jun 25 11:43:49 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 25 Jun 2012 10:43:49 +0300
Subject: [Freeswitch-users] show channels, comma vs TAB ?
In-Reply-To: <8A9D25F4E82046D68A56F2CF0F8E36EB@gmail.com>
References: <C10DA97B139149F19DBFC99DDEC23543@gmail.com>
	<4433A21B-E9D8-4D6B-A1EE-40769B0F90EF@visionutveckling.se>
	<CALNZuwnLfCEmiL7+bzmnaMGotemUk5jtuRssA-WU2j3nug-3Cg@mail.gmail.com>
	<8A9D25F4E82046D68A56F2CF0F8E36EB@gmail.com>
Message-ID: <CALNZuwmr31j3xtWBuu164uq8ymMeo3q8N7HoJxWyHz=8bpy8EA@mail.gmail.com>

There's a core function for json export I think.
-Avi



On Mon, Jun 25, 2012 at 10:27 AM, Seven Du <dujinfang at gmail.com> wrote:

> as xml is nice, I just forgot it.
>
> And we could make a json format :), let me take a look.
>
> Thanks.
>
> On Monday, June 25, 2012 at 2:30 PM, Avi Marcus wrote:
>
> There's no json format, unfortunately...
> -Avi
>
>
> On Mon, Jun 25, 2012 at 9:07 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> I think you can also use "show channels as xml", which will give you
> better output for parsing.
>
> /Peter
>
> 25 jun 2012 kl. 07:59 skrev "Seven Du" <dujinfang at gmail.com<mailto:
> dujinfang at gmail.com>>:
>
> Hi,
>
> I use originate user/1000 &bridge({aa=bb,cc=dd}user/1002)
>
> show channels gives a comma separated list, however, as some app data
> contains commas, this will be hard to parse, could we change to TAB? I
> understand query the db directly would be fine but sometimes it's just easy
> to parse the cli output. ideas? thanks.
>
>
>
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
> d412fdda-5fbc-4754-9803-2db06898a8b0,outbound,2012-06-25
> 13:47:03,1340603223,sofia/internal/sip:1000 at 192.168.2.102:26570,CS_EXECUTE,Outbound
> Call,1000,,1000,bridge,{aa=bb,cc=dd}user/1002,,default,PCMA,8000,64000,PCMA,8000,64000,,seven-macpro.local,
> 1000 at 192.168.2.102<mailto:1000 at 192.168.2.102
> >,,ACTIVE,,,RECV,d412fdda-5fbc-4754-9803-2db06898a8b0,,
> b5ca402e-d72e-4109-a8ee-634e873f3c31,outbound,2012-06-25
> 13:47:30,1340603250,sofia/internal/sip:1002 at 192.168.2.102:20676,CS_CONSUME_MEDIA,Extension
> 1000,1000,192.168.2.102,1002,,,,default,,,,,,,,seven-macpro.local,
> 1002 at 192.168.2.102<mailto:1002 at 192.168.2.102>,,RINGING,Outbound
> Call,1002,,d412fdda-5fbc-4754-9803-2db06898a8b0,,
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> Sent with Sparrow<http://www.sparrowmailapp.com>
>
> !DSPAM:4fe7fb0c32761667990988!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>
> !DSPAM:4fe7fb0c32761667990988!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Mon Jun 25 11:44:22 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 25 Jun 2012 10:44:22 +0300
Subject: [Freeswitch-users] http_cache not working with latest RPM
In-Reply-To: <COL124-DS18C9142F57F43CE265E72FDDE10@phx.gbl>
References: <COL124-DS18C9142F57F43CE265E72FDDE10@phx.gbl>
Message-ID: <CALNZuwmTut=ZFWpzp49Q52cFzTYDQZkTAOTVG5mnnSdxyB0NiA@mail.gmail.com>

Please post this on http://jira.freeswitch.org to ensure it gets followed
up on.
Thanks,
-Avi
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From dujinfang at gmail.com  Mon Jun 25 14:09:27 2012
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 25 Jun 2012 18:09:27 +0800
Subject: [Freeswitch-users] show channels, comma vs TAB ?
In-Reply-To: <CALNZuwmr31j3xtWBuu164uq8ymMeo3q8N7HoJxWyHz=8bpy8EA@mail.gmail.com>
References: <C10DA97B139149F19DBFC99DDEC23543@gmail.com>
	<4433A21B-E9D8-4D6B-A1EE-40769B0F90EF@visionutveckling.se>
	<CALNZuwnLfCEmiL7+bzmnaMGotemUk5jtuRssA-WU2j3nug-3Cg@mail.gmail.com>
	<8A9D25F4E82046D68A56F2CF0F8E36EB@gmail.com>
	<CALNZuwmr31j3xtWBuu164uq8ymMeo3q8N7HoJxWyHz=8bpy8EA@mail.gmail.com>
Message-ID: <BD5D22B9A955420095774EF23CFA2A3D@gmail.com>

Yes, it's simple

http://jira.freeswitch.org/browse/FS-4348


On Monday, June 25, 2012 at 3:43 PM, Avi Marcus wrote:

> There's a core function for json export I think.
> -Avi
> 
> 
> On Mon, Jun 25, 2012 at 10:27 AM, Seven Du <dujinfang at gmail.com (mailto:dujinfang at gmail.com)> wrote:
> > as xml is nice, I just forgot it.
> > 
> > And we could make a json format :), let me take a look.
> > 
> > Thanks.
> > 
> > 
> > On Monday, June 25, 2012 at 2:30 PM, Avi Marcus wrote:
> > 
> > > There's no json format, unfortunately...
> > > -Avi 
> > > 
> > > On Mon, Jun 25, 2012 at 9:07 AM, Peter Olsson <peter.olsson at visionutveckling.se (mailto:peter.olsson at visionutveckling.se)> wrote:
> > > > I think you can also use "show channels as xml", which will give you better output for parsing.
> > > > 
> > > > /Peter
> > > > 
> > > > 25 jun 2012 kl. 07:59 skrev "Seven Du" <dujinfang at gmail.com (mailto:dujinfang at gmail.com)<mailto:dujinfang at gmail.com>>:
> > > > 
> > > > Hi,
> > > > 
> > > > I use originate user/1000 &bridge({aa=bb,cc=dd}user/1002)
> > > > 
> > > > show channels gives a comma separated list, however, as some app data contains commas, this will be hard to parse, could we change to TAB? I understand query the db directly would be fine but sometimes it's just easy to parse the cli output. ideas? thanks.
> > > > 
> > > > 
> > > > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
> > > > d412fdda-5fbc-4754-9803-2db06898a8b0,outbound,2012-06-25 13:47:03,1340603223,sofia/internal/sip:1000 at 192.168.2.102:26570 (http://sip:1000 at 192.168.2.102:26570),CS_EXECUTE,Outbound Call,1000,,1000,bridge,{aa=bb,cc=dd}user/1002,,default,PCMA,8000,64000,PCMA,8000,64000,,seven-macpro.local,1000 at 192.168.2.102 (mailto:1000 at 192.168.2.102)<mailto:1000 at 192.168.2.102>,,ACTIVE,,,RECV,d412fdda-5fbc-4754-9803-2db06898a8b0,,
> > > > b5ca402e-d72e-4109-a8ee-634e873f3c31,outbound,2012-06-25 13:47:30,1340603250,sofia/internal/sip:1002 at 192.168.2.102:20676 (http://sip:1002 at 192.168.2.102:20676),CS_CONSUME_MEDIA,Extension 1000,1000,192.168.2.102,1002,,,,default,,,,,,,,seven-macpro.local,1002 at 192.168.2.102 (mailto:1002 at 192.168.2.102)<mailto:1002 at 192.168.2.102>,,RINGING,Outbound Call,1002,,d412fdda-5fbc-4754-9803-2db06898a8b0,,
> > > > 
> > > > 
> > > > --
> > > > About: http://about.me/dujinfang
> > > > Blog: http://www.dujinfang.com
> > > > Proj:  http://www.freeswitch.org.cn
> > > > 
> > > > Sent with Sparrow<http://www.sparrowmailapp.com>
> > > > 
> > > > !DSPAM:4fe7fb0c32761667990988!
> > > > _________________________________________________________________________
> > > > Professional FreeSWITCH Consulting Services:
> > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org)<mailto:consulting at freeswitch.org>
> > > > http://www.freeswitchsolutions.com
> > > > 
> > > > 
> > > > 
> > > > 
> > > > Official FreeSWITCH Sites
> > > > http://www.freeswitch.org
> > > > http://wiki.freeswitch.org
> > > > http://www.cluecon.com
> > > > 
> > > > Join Us At ClueCon - Aug 7-9, 2012
> > > > 
> > > > FreeSWITCH-users mailing list
> > > > 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
> > > > 
> > > > 
> > > > !DSPAM:4fe7fb0c32761667990988!
> > > > 
> > > > _________________________________________________________________________
> > > > Professional FreeSWITCH Consulting Services:
> > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > > > http://www.freeswitchsolutions.com
> > > > 
> > > > 
> > > > 
> > > > 
> > > > Official FreeSWITCH Sites
> > > > http://www.freeswitch.org
> > > > http://wiki.freeswitch.org
> > > > http://www.cluecon.com
> > > > 
> > > > Join Us At ClueCon - Aug 7-9, 2012
> > > > 
> > > > 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
> > > 
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > > http://www.freeswitchsolutions.com
> > > 
> > > 
> > > 
> > > 
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > > 
> > > Join Us At ClueCon - Aug 7-9, 2012
> > > 
> > > 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
> > > 
> > > 
> > > 
> > 
> > 
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > Join Us At ClueCon - Aug 7-9, 2012
> > 
> > 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
> > 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> 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
> 
> 


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From vipkilla at gmail.com  Mon Jun 25 16:00:59 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 25 Jun 2012 08:00:59 -0400
Subject: [Freeswitch-users] Sharing: Call Intercept
In-Reply-To: <62BB9D5255424985932A96663192F254@freeswitch.org>
References: <C066384B0C1F1E4B9B870F050D7F1A9D402150F5@ADAMA.mcpeekdodge.local>
	<CAKbxfG9qBAQnNB0wfMR3Bkm7uUK83QgiLHSUAVbAeDiFuhD62A@mail.gmail.com>
	<62BB9D5255424985932A96663192F254@freeswitch.org>
Message-ID: <CAC-LwPMkan5urY_LkmcwQ=mas33fJfpdQMn5kt6jc98H9izh4w@mail.gmail.com>

I recently added some more details to this page:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pickup


From a.villa at seletech.com  Mon Jun 25 17:17:35 2012
From: a.villa at seletech.com (alberto Villa)
Date: Mon, 25 Jun 2012 15:17:35 +0200
Subject: [Freeswitch-users] FreeSWITCH bind to hostname
In-Reply-To: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
References: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
Message-ID: <4FE864EF.40703@seletech.com>

Hello,

how can I set FreeSWITCH's configuration in order to bind it to a 
hostname like freeswitch.mydomain.com?

Than you in advance,

Alberto


From rnbrady at gmail.com  Mon Jun 25 13:26:36 2012
From: rnbrady at gmail.com (Richard Brady)
Date: Mon, 25 Jun 2012 10:26:36 +0100
Subject: [Freeswitch-users] Bind extension to an IP address
In-Reply-To: <CAKbxfG_iUqoU0VFAsKu5a=cwV8fh02WzrEx1R70cMgXXcFy6iQ@mail.gmail.com>
References: <4FE7175F.90209@the800group.com> <4FE787D4.5080603@gmail.com>
	<CALNZuwm-RPQRBLMRq65fb9eVdUS2W9HURZC5y+=gQ5a2GWhtqA@mail.gmail.com>
	<CAKbxfG_iUqoU0VFAsKu5a=cwV8fh02WzrEx1R70cMgXXcFy6iQ@mail.gmail.com>
Message-ID: <CAN5esQ+ZQ4v6U-EB9+6rdQ1uBr84c6sZL4S6oi2R=1wdah3k-Q@mail.gmail.com>

Is there an equivalent for gateways?

I can't find a way to bind incoming calls from a CIDR to a particular gateway.


On 25 June 2012 03:05, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> http://wiki.freeswitch.org/wiki/Acl#Users
>
> On Sun, Jun 24, 2012 at 5:25 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> Or do a check in the dialplan on that extension's outbound calls.
>> Or only do IP auth for that account and not digest auth.
>> -Avi
>>
>>
>> On Mon, Jun 25, 2012 at 12:34 AM, Abaci <abaci64 at gmail.com> wrote:
>>>
>>> if you serve configs xml_curl you can check that it you web app and return
>>> not found if it's not coming from the correct ip.
>>>
>>>
>>> On 6/24/2012 9:34 AM, ocset wrote:
>>>
>>> Hi
>>>
>>> Is it possible to bind an user/extension to a specific IP address? (e.g.
>>> extension 1001 can only log in from IP address 192.168.0.101)
>>>
>>> Thanks
>>> O.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From grcamauer at gmail.com  Mon Jun 25 18:19:42 2012
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Mon, 25 Jun 2012 11:19:42 -0300
Subject: [Freeswitch-users] Call State Diagram
Message-ID: <CAPNTnnpCtjRVoFa3qFbqfmLiRJBJ3--MZ6zwE-dmOaNZJV3C4w@mail.gmail.com>

Does a Call-State Diagram exist for FreeSwitch?  I am looking for something
similar to this:
http://threebit.net/documentation/dialogic/resource.intel.com/telecom/support/releases/winnt/SR511/docs/htmlfiles/gcapdvgd/1526-002-3.gif.

I have found a list of all the different states of a call (
http://wiki.freeswitch.org/wiki/Channel_States), but would like to see what
events/actions trigger states changes to other states, etc.

Thanks in advance,

Guillermo Ruiz Camauer
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From msc at freeswitch.org  Mon Jun 25 18:27:04 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Jun 2012 07:27:04 -0700
Subject: [Freeswitch-users] Bind extension to an IP address
In-Reply-To: <CAN5esQ+ZQ4v6U-EB9+6rdQ1uBr84c6sZL4S6oi2R=1wdah3k-Q@mail.gmail.com>
References: <4FE7175F.90209@the800group.com> <4FE787D4.5080603@gmail.com>
	<CALNZuwm-RPQRBLMRq65fb9eVdUS2W9HURZC5y+=gQ5a2GWhtqA@mail.gmail.com>
	<CAKbxfG_iUqoU0VFAsKu5a=cwV8fh02WzrEx1R70cMgXXcFy6iQ@mail.gmail.com>
	<CAN5esQ+ZQ4v6U-EB9+6rdQ1uBr84c6sZL4S6oi2R=1wdah3k-Q@mail.gmail.com>
Message-ID: <CAKzWOxWJuxd+PjBxDmRPa3UNJ6L68xEZotfCKc7HXccg7Jxtvw@mail.gmail.com>

On Mon, Jun 25, 2012 at 2:26 AM, Richard Brady <rnbrady at gmail.com> wrote:

> Is there an equivalent for gateways?
>
> I can't find a way to bind incoming calls from a CIDR to a particular
> gateway.
>
> Just curious - what functionality do you need this for?
-MC
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From marketing at cluecon.com  Mon Jun 25 20:09:54 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 25 Jun 2012 09:09:54 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxVD9tMHg12mV5S44pHewJWVjWWFJgvfa_Guiz1bhpv_WQ@mail.gmail.com>

Welcome to the last week of June!

Things quieted down a bit last week, but we still had some interesting
items for your consideration. Darren Schreiber from the
2600hz<http://www.2600hz.com?cc12-0625>project joined us last week on
our conference
call <http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12-0625> to
present "SIP 101" - a gentle introduction to the SIP protocol for those
doing VoIP. It was a nice discussion and we plan to do a follow up,
tentatively scheduled for July 18. The audio is available in the usual
place<http://torrents.freeswitch.org/conf_call_2012-06-20.torrent?cc12-0625>and
the slides are available
here <http://wiki.freeswitch.org/files/SIP_101_Presentation.ppt?cc12-0625>.
For this week's<http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_27?cc12-0625>conference
call we are going to have an open discussion about topics of
interest to our community members.

Our ClueCon schedule <http://www.cluecon.com/schedule?cc12-0625> is now
full! Please visit the schedule to see a list of speakers and talk titles.
We have a few presentations whose titles are yet to be announced, so check
back often to stay updated. We are pleased to have Brad Pitt joining us
this year! (No, not that one <http://en.wikipedia.org/wiki/Brad_pitt> - we
have this one <http://www.cluecon.com/speaker/108?cc12-0625>.) Bradley J
Pitt is a long-time veteran of the technology and telecom industries and
recently joined Barracuda Networks <?cc12-0625>. His
discussion will touch upon both technology and business aspects of the IP
PBX world.

As a reminder, you still have about one week before the "bits shift" and
you go from 16 down to 8 tickets for your ClueCon
registration<http://www.cluecon.com/register?cc12-0625>.
Register right away so that you have the best chance possible to win one of
the 10 Android tablets that will be given away this year!

See you in Chicago!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
cc12-0625
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From peter.olsson at visionutveckling.se  Mon Jun 25 20:13:55 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 25 Jun 2012 16:13:55 +0000
Subject: [Freeswitch-users] Bind extension to an IP address
In-Reply-To: <CAN5esQ+ZQ4v6U-EB9+6rdQ1uBr84c6sZL4S6oi2R=1wdah3k-Q@mail.gmail.com>
References: <4FE7175F.90209@the800group.com> <4FE787D4.5080603@gmail.com>
	<CALNZuwm-RPQRBLMRq65fb9eVdUS2W9HURZC5y+=gQ5a2GWhtqA@mail.gmail.com>
	<CAKbxfG_iUqoU0VFAsKu5a=cwV8fh02WzrEx1R70cMgXXcFy6iQ@mail.gmail.com>,
	<CAN5esQ+ZQ4v6U-EB9+6rdQ1uBr84c6sZL4S6oi2R=1wdah3k-Q@mail.gmail.com>
Message-ID: <FF24894F-D8D7-4330-A93E-62AFB7CEA3D0@visionutveckling.se>

There is no setting for it (as far as I know), but you can compare the network_addr in the dialplan, and then do whatever you want to do with it, ie transfer to a specific context.

/Peter

25 jun 2012 kl. 15:29 skrev "Richard Brady" <rnbrady at gmail.com>:

> Is there an equivalent for gateways?
> 
> I can't find a way to bind incoming calls from a CIDR to a particular gateway.
> 
> 
> On 25 June 2012 03:05, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> http://wiki.freeswitch.org/wiki/Acl#Users
>> 
>> On Sun, Jun 24, 2012 at 5:25 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>> Or do a check in the dialplan on that extension's outbound calls.
>>> Or only do IP auth for that account and not digest auth.
>>> -Avi
>>> 
>>> 
>>> On Mon, Jun 25, 2012 at 12:34 AM, Abaci <abaci64 at gmail.com> wrote:
>>>> 
>>>> if you serve configs xml_curl you can check that it you web app and return
>>>> not found if it's not coming from the correct ip.
>>>> 
>>>> 
>>>> On 6/24/2012 9:34 AM, ocset wrote:
>>>> 
>>>> Hi
>>>> 
>>>> Is it possible to bind an user/extension to a specific IP address? (e.g.
>>>> extension 1001 can only log in from IP address 192.168.0.101)
>>>> 
>>>> Thanks
>>>> O.
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> Join Us At ClueCon - Aug 7-9, 2012
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> 
>> 
>> --
>> Anthony Minessale II
>> 
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> !DSPAM:4fe8646232762482116809!
> 


From jpablolorenzetti at hotmail.com  Mon Jun 25 20:19:49 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Mon, 25 Jun 2012 16:19:49 +0000
Subject: [Freeswitch-users] softphones and background noise
Message-ID: <DUB109-W124EB7C5B9B0D5EEA293CA8D9E10@phx.gbl>



Good morning, i m having lots of issues with softphones, when a call is initiated/terminated to a softphone 
the background noise is too high that you can hardly hear the actual voice.
i have played around with VAD and CNG turning them on and off in the profile and/or the dialplan but i have not had any luck.
it does not matter if the call is internal or goes through a gateway, it is however, worse when it goes through a gateway, those are the
cases you can not even hear the voice for so much noise and when the call is to a next extension the background noise is high
but not as high.
But, if you initiate/terminate the call to a hardphone (like polycom phones) the call is perfectly clear in all cases 
(except when the other end is an internal softphone).
can someone please give me a hint. thanks a lot!
 		 	   		  

From mitch.capper at gmail.com  Mon Jun 25 20:27:17 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 25 Jun 2012 09:27:17 -0700
Subject: [Freeswitch-users] softphones and background noise
In-Reply-To: <DUB109-W124EB7C5B9B0D5EEA293CA8D9E10@phx.gbl>
References: <DUB109-W124EB7C5B9B0D5EEA293CA8D9E10@phx.gbl>
Message-ID: <CALn233qL-ML74pany_rH221RLom0h1GKqoCHk2pwPUy3ceTDxw@mail.gmail.com>

Try Other softphones?  Are you using a headset?

~Mitch

On Mon, Jun 25, 2012 at 9:19 AM, Juan Pablo L.
<jpablolorenzetti at hotmail.com> wrote:
>
>
> Good morning, i m having lots of issues with softphones, when a call is initiated/terminated to a softphone
> the background noise is too high that you can hardly hear the actual voice.
> i have played around with VAD and CNG turning them on and off in the profile and/or the dialplan but i have not had any luck.
> it does not matter if the call is internal or goes through a gateway, it is however, worse when it goes through a gateway, those are the
> cases you can not even hear the voice for so much noise and when the call is to a next extension the background noise is high
> but not as high.
> But, if you initiate/terminate the call to a hardphone (like polycom phones) the call is perfectly clear in all cases
> (except when the other end is an internal softphone).
> can someone please give me a hint. thanks a lot!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From peter.olsson at visionutveckling.se  Mon Jun 25 20:31:43 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 25 Jun 2012 16:31:43 +0000
Subject: [Freeswitch-users] softphones and background noise
In-Reply-To: <DUB109-W124EB7C5B9B0D5EEA293CA8D9E10@phx.gbl>
References: <DUB109-W124EB7C5B9B0D5EEA293CA8D9E10@phx.gbl>
Message-ID: <85828F65-E5D2-4642-8FF2-6479D0E7784D@visionutveckling.se>

I've noticed issues sometimes when I had zrtp enabled, but I guess the reason for that was because of a bad SDP sent in the INVITE (since the endpoint didn't really support zrtp). This is surely a codec issue of some kind, start by checking the logs, and see if you find something suspicious there.

/Peter

25 jun 2012 kl. 18:24 skrev "Juan Pablo L." <jpablolorenzetti at hotmail.com>:

> 
> 
> Good morning, i m having lots of issues with softphones, when a call is initiated/terminated to a softphone 
> the background noise is too high that you can hardly hear the actual voice.
> i have played around with VAD and CNG turning them on and off in the profile and/or the dialplan but i have not had any luck.
> it does not matter if the call is internal or goes through a gateway, it is however, worse when it goes through a gateway, those are the
> cases you can not even hear the voice for so much noise and when the call is to a next extension the background noise is high
> but not as high.
> But, if you initiate/terminate the call to a hardphone (like polycom phones) the call is perfectly clear in all cases 
> (except when the other end is an internal softphone).
> can someone please give me a hint. thanks a lot!
>                         
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> !DSPAM:4fe88d8932761983755573!
> 


From jpablolorenzetti at hotmail.com  Mon Jun 25 20:33:03 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Mon, 25 Jun 2012 10:33:03 -0600
Subject: [Freeswitch-users] softphones and background noise
Message-ID: <col0-p4-eas433A62F51B0F7504A32F9FD9E10@phx.gbl>

Hi, thanks, Yes, i have tried many on android, linux and wndows, the same problem with all of them.
Yes i have tested with a hardphone, a polycom, and its fine.

-----Original Message-----

From: Mitch Capper
Sent: 25 Jun 2012 16:28:40 GMT
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] softphones and background noise

Try Other softphones?  Are you using a headset?

~Mitch

On Mon, Jun 25, 2012 at 9:19 AM, Juan Pablo L.
<jpablolorenzetti at hotmail.com> wrote:
>
>
> Good morning, i m having lots of issues with softphones, when a call is initiated/terminated to a softphone
> the background noise is too high that you can hardly hear the actual voice.
> i have played around with VAD and CNG turning them on and off in the profile and/or the dialplan but i have not had any luck.
> it does not matter if the call is internal or goes through a gateway, it is however, worse when it goes through a gateway, those are the
> cases you can not even hear the voice for so much noise and when the call is to a next extension the background noise is high
> but not as high.
> But, if you initiate/terminate the call to a hardphone (like polycom phones) the call is perfectly clear in all cases
> (except when the other end is an internal softphone).
> can someone please give me a hint. thanks a lot!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


From jpablolorenzetti at hotmail.com  Mon Jun 25 20:35:02 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Mon, 25 Jun 2012 10:35:02 -0600
Subject: [Freeswitch-users] softphones and background noise
Message-ID: <col0-p5-eas49525BFBD9CDB5C890F901D9E10@phx.gbl>

Hi, thanks, i m looking in the logs at the moment and playing with codecs too but it gets worse and not better thou.

-----Original Message-----

From: Peter Olsson
Sent: 25 Jun 2012 16:32:51 GMT
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] softphones and background noise

I've noticed issues sometimes when I had zrtp enabled, but I guess the reason for that was because of a bad SDP sent in the INVITE (since the endpoint didn't really support zrtp). This is surely a codec issue of some kind, start by checking the logs, and see if you find something suspicious there.

/Peter

25 jun 2012 kl. 18:24 skrev "Juan Pablo L." <jpablolorenzetti at hotmail.com>:

>
>
> Good morning, i m having lots of issues with softphones, when a call is initiated/terminated to a softphone
> the background noise is too high that you can hardly hear the actual voice.
> i have played around with VAD and CNG turning them on and off in the profile and/or the dialplan but i have not had any luck.
> it does not matter if the call is internal or goes through a gateway, it is however, worse when it goes through a gateway, those are the
> cases you can not even hear the voice for so much noise and when the call is to a next extension the background noise is high
> but not as high.
> But, if you initiate/terminate the call to a hardphone (like polycom phones) the call is perfectly clear in all cases
> (except when the other end is an internal softphone).
> can someone please give me a hint. thanks a lot!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4fe88d8932761983755573!
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


From mitch.capper at gmail.com  Mon Jun 25 20:36:11 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 25 Jun 2012 09:36:11 -0700
Subject: [Freeswitch-users] softphones and background noise
In-Reply-To: <col0-p4-eas433A62F51B0F7504A32F9FD9E10@phx.gbl>
References: <col0-p4-eas433A62F51B0F7504A32F9FD9E10@phx.gbl>
Message-ID: <CALn233oDK=54WzMk308bCkJ5zafWsb4e9aVOMymL4YbDJbp-cw@mail.gmail.com>

headset ?

~Mitch

On Mon, Jun 25, 2012 at 9:33 AM, Juan Pablo L.
<jpablolorenzetti at hotmail.com> wrote:
> Hi, thanks, Yes, i have tried many on android, linux and wndows, the same problem with all of them.
> Yes i have tested with a hardphone, a polycom, and its fine.
>
> -----Original Message-----
>
> From: Mitch Capper
> Sent: 25 Jun 2012 16:28:40 GMT
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] softphones and background noise
>
> Try Other softphones? ?Are you using a headset?
>
> ~Mitch
>
> On Mon, Jun 25, 2012 at 9:19 AM, Juan Pablo L.
> <jpablolorenzetti at hotmail.com> wrote:
>>
>>
>> Good morning, i m having lots of issues with softphones, when a call is initiated/terminated to a softphone
>> the background noise is too high that you can hardly hear the actual voice.
>> i have played around with VAD and CNG turning them on and off in the profile and/or the dialplan but i have not had any luck.
>> it does not matter if the call is internal or goes through a gateway, it is however, worse when it goes through a gateway, those are the
>> cases you can not even hear the voice for so much noise and when the call is to a next extension the background noise is high
>> but not as high.
>> But, if you initiate/terminate the call to a hardphone (like polycom phones) the call is perfectly clear in all cases
>> (except when the other end is an internal softphone).
>> can someone please give me a hint. thanks a lot!
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Mon Jun 25 20:48:00 2012
From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Mon, 25 Jun 2012 13:48:00 -0300
Subject: [Freeswitch-users] Call State Diagram
In-Reply-To: <CAPNTnnpCtjRVoFa3qFbqfmLiRJBJ3--MZ6zwE-dmOaNZJV3C4w@mail.gmail.com>
References: <CAPNTnnpCtjRVoFa3qFbqfmLiRJBJ3--MZ6zwE-dmOaNZJV3C4w@mail.gmail.com>
Message-ID: <315ADFC93BC3472AA94965F8AC69E2EB@freeswitch.org>

I am not aware of the existence of such a documentation Guillermo. I believe that if you can read C and are up to the task of making the documentation, many of us would benefit from it.

Regards,  

--  
Jo?o Mesquita
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Monday, June 25, 2012 at 11:19 AM, Guillermo Ruiz Camauer wrote:

> Does a Call-State Diagram exist for FreeSwitch?  I am looking for something similar to this: http://threebit.net/documentation/dialogic/resource.intel.com/telecom/support/releases/winnt/SR511/docs/htmlfiles/gcapdvgd/1526-002-3.gif.   
> I have found a list of all the different states of a call (http://wiki.freeswitch.org/wiki/Channel_States), but would like to see what events/actions trigger states changes to other states, etc.
>  
> Thanks in advance,
>  
> Guillermo Ruiz Camauer
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> Join Us At ClueCon - Aug 7-9, 2012
>  
> 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
>  
>  


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From jpablolorenzetti at hotmail.com  Mon Jun 25 20:51:53 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Mon, 25 Jun 2012 16:51:53 +0000
Subject: [Freeswitch-users] softphones and background noise
In-Reply-To: <CALn233oDK=54WzMk308bCkJ5zafWsb4e9aVOMymL4YbDJbp-cw@mail.gmail.com>
References: <col0-p4-eas433A62F51B0F7504A32F9FD9E10@phx.gbl>,
	<CALn233oDK=54WzMk308bCkJ5zafWsb4e9aVOMymL4YbDJbp-cw@mail.gmail.com>
Message-ID: <DUB109-W82BB9C720145EE2D7CC859D9E10@phx.gbl>


no i have not tested with a headset .. i dont have one? :( ..

----------------------------------------
> From: mitch.capper at gmail.com
> Date: Mon, 25 Jun 2012 09:36:11 -0700
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] softphones and background noise
>
> headset ?
>
> ~Mitch
>
> On Mon, Jun 25, 2012 at 9:33 AM, Juan Pablo L.
> <jpablolorenzetti at hotmail.com> wrote:
> > Hi, thanks, Yes, i have tried many on android, linux and wndows, the same problem with all of them.
> > Yes i have tested with a hardphone, a polycom, and its fine.
> >
> > -----Original Message-----
> >
> > From: Mitch Capper
> > Sent: 25 Jun 2012 16:28:40 GMT
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] softphones and background noise
> >
> > Try Other softphones?  Are you using a headset?
> >
> > ~Mitch
> >
> > On Mon, Jun 25, 2012 at 9:19 AM, Juan Pablo L.
> > <jpablolorenzetti at hotmail.com> wrote:
> >>
> >>
> >> Good morning, i m having lots of issues with softphones, when a call is initiated/terminated to a softphone
> >> the background noise is too high that you can hardly hear the actual voice.
> >> i have played around with VAD and CNG turning them on and off in the profile and/or the dialplan but i have not had any luck.
> >> it does not matter if the call is internal or goes through a gateway, it is however, worse when it goes through a gateway, those are the
> >> cases you can not even hear the voice for so much noise and when the call is to a next extension the background noise is high
> >> but not as high.
> >> But, if you initiate/terminate the call to a hardphone (like polycom phones) the call is perfectly clear in all cases
> >> (except when the other end is an internal softphone).
> >> can someone please give me a hint. thanks a lot!
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  

From jpablolorenzetti at hotmail.com  Tue Jun 26 00:17:36 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Mon, 25 Jun 2012 20:17:36 +0000
Subject: [Freeswitch-users] softphones and background noise
In-Reply-To: <CALn233oDK=54WzMk308bCkJ5zafWsb4e9aVOMymL4YbDJbp-cw@mail.gmail.com>
References: <col0-p4-eas433A62F51B0F7504A32F9FD9E10@phx.gbl>,
	<CALn233oDK=54WzMk308bCkJ5zafWsb4e9aVOMymL4YbDJbp-cw@mail.gmail.com>
Message-ID: <DUB109-W59C4F1A5E5E3B377C8CBF6D9E10@phx.gbl>



just an update, i tested with linphone on linux and it does not produce the noise.
i m still investigating though.
----------------------------------------
> From: mitch.capper at gmail.com
> Date: Mon, 25 Jun 2012 09:36:11 -0700
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] softphones and background noise
>
> headset ?
>
> ~Mitch
>
> On Mon, Jun 25, 2012 at 9:33 AM, Juan Pablo L.
> <jpablolorenzetti at hotmail.com> wrote:
> > Hi, thanks, Yes, i have tried many on android, linux and wndows, the same problem with all of them.
> > Yes i have tested with a hardphone, a polycom, and its fine.
> >
> > -----Original Message-----
> >
> > From: Mitch Capper
> > Sent: 25 Jun 2012 16:28:40 GMT
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] softphones and background noise
> >
> > Try Other softphones?  Are you using a headset?
> >
> > ~Mitch
> >
> > On Mon, Jun 25, 2012 at 9:19 AM, Juan Pablo L.
> > <jpablolorenzetti at hotmail.com> wrote:
> >>
> >>
> >> Good morning, i m having lots of issues with softphones, when a call is initiated/terminated to a softphone
> >> the background noise is too high that you can hardly hear the actual voice.
> >> i have played around with VAD and CNG turning them on and off in the profile and/or the dialplan but i have not had any luck.
> >> it does not matter if the call is internal or goes through a gateway, it is however, worse when it goes through a gateway, those are the
> >> cases you can not even hear the voice for so much noise and when the call is to a next extension the background noise is high
> >> but not as high.
> >> But, if you initiate/terminate the call to a hardphone (like polycom phones) the call is perfectly clear in all cases
> >> (except when the other end is an internal softphone).
> >> can someone please give me a hint. thanks a lot!
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 cassidywebservices.co.uk  Tue Jun 26 00:53:15 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Mon, 25 Jun 2012 21:53:15 +0100
Subject: [Freeswitch-users] softphones and background noise
In-Reply-To: <DUB109-W59C4F1A5E5E3B377C8CBF6D9E10@phx.gbl>
References: <col0-p4-eas433A62F51B0F7504A32F9FD9E10@phx.gbl>
	<CALn233oDK=54WzMk308bCkJ5zafWsb4e9aVOMymL4YbDJbp-cw@mail.gmail.com>
	<DUB109-W59C4F1A5E5E3B377C8CBF6D9E10@phx.gbl>
Message-ID: <CAMbW9oLzWZf4MapYVi68OdvNYzXuVKogkEDQWvDH53duO4HQGA@mail.gmail.com>

Certain sound cards are known to be susceptible to interference. The
AC'97's were horrid for it in my experience for example.

On 25 June 2012 21:17, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:

>
>
> just an update, i tested with linphone on linux and it does not produce
> the noise.
> i m still investigating though.
> ----------------------------------------
> > From: mitch.capper at gmail.com
> > Date: Mon, 25 Jun 2012 09:36:11 -0700
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] softphones and background noise
> >
> > headset ?
> >
> > ~Mitch
> >
> > On Mon, Jun 25, 2012 at 9:33 AM, Juan Pablo L.
> > <jpablolorenzetti at hotmail.com> wrote:
> > > Hi, thanks, Yes, i have tried many on android, linux and wndows, the
> same problem with all of them.
> > > Yes i have tested with a hardphone, a polycom, and its fine.
> > >
> > > -----Original Message-----
> > >
> > > From: Mitch Capper
> > > Sent: 25 Jun 2012 16:28:40 GMT
> > > To: FreeSWITCH Users Help
> > > Subject: Re: [Freeswitch-users] softphones and background noise
> > >
> > > Try Other softphones?  Are you using a headset?
> > >
> > > ~Mitch
> > >
> > > On Mon, Jun 25, 2012 at 9:19 AM, Juan Pablo L.
> > > <jpablolorenzetti at hotmail.com> wrote:
> > >>
> > >>
> > >> Good morning, i m having lots of issues with softphones, when a call
> is initiated/terminated to a softphone
> > >> the background noise is too high that you can hardly hear the actual
> voice.
> > >> i have played around with VAD and CNG turning them on and off in the
> profile and/or the dialplan but i have not had any luck.
> > >> it does not matter if the call is internal or goes through a gateway,
> it is however, worse when it goes through a gateway, those are the
> > >> cases you can not even hear the voice for so much noise and when the
> call is to a next extension the background noise is high
> > >> but not as high.
> > >> But, if you initiate/terminate the call to a hardphone (like polycom
> phones) the call is perfectly clear in all cases
> > >> (except when the other end is an internal softphone).
> > >> can someone please give me a hint. thanks a lot!
> > >>
> > >>
> _________________________________________________________________________
> > >> Professional FreeSWITCH Consulting Services:
> > >> consulting at freeswitch.org
> > >> http://www.freeswitchsolutions.com
> > >>
> > >> 
> > >> 
> > >>
> > >> Official FreeSWITCH Sites
> > >> http://www.freeswitch.org
> > >> http://wiki.freeswitch.org
> > >> http://www.cluecon.com
> > >>
> > >> Join Us At ClueCon - Aug 7-9, 2012
> > >>
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > Join Us At ClueCon - Aug 7-9, 2012
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > Join Us At ClueCon - Aug 7-9, 2012
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From jpablolorenzetti at hotmail.com  Tue Jun 26 01:58:02 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Mon, 25 Jun 2012 21:58:02 +0000
Subject: [Freeswitch-users] softphones and background noise
In-Reply-To: <CAMbW9oLzWZf4MapYVi68OdvNYzXuVKogkEDQWvDH53duO4HQGA@mail.gmail.com>
References: <col0-p4-eas433A62F51B0F7504A32F9FD9E10@phx.gbl>,
	<CALn233oDK=54WzMk308bCkJ5zafWsb4e9aVOMymL4YbDJbp-cw@mail.gmail.com>,
	<DUB109-W59C4F1A5E5E3B377C8CBF6D9E10@phx.gbl>,
	<CAMbW9oLzWZf4MapYVi68OdvNYzXuVKogkEDQWvDH53duO4HQGA@mail.gmail.com>
Message-ID: <DUB109-W21FA5C84C7C3CA014D00ED9E10@phx.gbl>


thanks for the info. it is very weired though because i m using the linphone on the same machine.
also on the same android phone.

________________________________
> Date: Mon, 25 Jun 2012 21:53:15 +0100 
> From: andrew at cassidywebservices.co.uk 
> To: freeswitch-users at lists.freeswitch.org 
> Subject: Re: [Freeswitch-users] softphones and background noise 
>  
> Certain sound cards are known to be susceptible to interference. The  
> AC'97's were horrid for it in my experience for example. 
>  
> On 25 June 2012 21:17, Juan Pablo L.  
> <jpablolorenzetti at hotmail.com<mailto:jpablolorenzetti at hotmail.com>>  
> wrote: 
>  
>  
> just an update, i tested with linphone on linux and it does not produce  
> the noise. 
> i m still investigating though. 
> ---------------------------------------- 
>  > From: mitch.capper at gmail.com<mailto:mitch.capper at gmail.com> 
>  > Date: Mon, 25 Jun 2012 09:36:11 -0700 
>  > To:  
> freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org> 
>  > Subject: Re: [Freeswitch-users] softphones and background noise 
>  > 
>  > headset ? 
>  > 
>  > ~Mitch 
>  > 
>  > On Mon, Jun 25, 2012 at 9:33 AM, Juan Pablo L. 
>  > <jpablolorenzetti at hotmail.com<mailto:jpablolorenzetti at hotmail.com>> wrote: 
>  > > Hi, thanks, Yes, i have tried many on android, linux and wndows,  
> the same problem with all of them. 
>  > > Yes i have tested with a hardphone, a polycom, and its fine. 
>  > > 
>  > > -----Original Message----- 
>  > > 
>  > > From: Mitch Capper 
>  > > Sent: 25 Jun 2012 16:28:40 GMT 
>  > > To: FreeSWITCH Users Help 
>  > > Subject: Re: [Freeswitch-users] softphones and background noise 
>  > > 
>  > > Try Other softphones?  Are you using a headset? 
>  > > 
>  > > ~Mitch 
>  > > 
>  > > On Mon, Jun 25, 2012 at 9:19 AM, Juan Pablo L. 
>  > > <jpablolorenzetti at hotmail.com<mailto:jpablolorenzetti at hotmail.com>>  
> wrote: 
>  > >> 
>  > >> 
>  > >> Good morning, i m having lots of issues with softphones, when a  
> call is initiated/terminated to a softphone 
>  > >> the background noise is too high that you can hardly hear the  
> actual voice. 
>  > >> i have played around with VAD and CNG turning them on and off in  
> the profile and/or the dialplan but i have not had any luck. 
>  > >> it does not matter if the call is internal or goes through a  
> gateway, it is however, worse when it goes through a gateway, those are  
> the 
>  > >> cases you can not even hear the voice for so much noise and when  
> the call is to a next extension the background noise is high 
>  > >> but not as high. 
>  > >> But, if you initiate/terminate the call to a hardphone (like  
> polycom phones) the call is perfectly clear in all cases 
>  > >> (except when the other end is an internal softphone). 
>  > >> can someone please give me a hint. thanks a lot! 
>  > >> 
>  > >>  
> _________________________________________________________________________ 
>  > >> Professional FreeSWITCH Consulting Services: 
>  > >> consulting at freeswitch.org<mailto:consulting at freeswitch.org> 
>  > >> http://www.freeswitchsolutions.com 
>  > >> 
>  > >>  
>  > >>  
>  > >> 
>  > >> Official FreeSWITCH Sites 
>  > >> http://www.freeswitch.org 
>  > >> http://wiki.freeswitch.org 
>  > >> http://www.cluecon.com 
>  > >> 
>  > >> Join Us At ClueCon - Aug 7-9, 2012 
>  > >> 
>  > >> 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 
>  > > 
>  > > _________________________________________________________________________ 
>  > > Professional FreeSWITCH Consulting Services: 
>  > > consulting at freeswitch.org<mailto:consulting at freeswitch.org> 
>  > > http://www.freeswitchsolutions.com 
>  > > 
>  > >  
>  > >  
>  > > 
>  > > Official FreeSWITCH Sites 
>  > > http://www.freeswitch.org 
>  > > http://wiki.freeswitch.org 
>  > > http://www.cluecon.com 
>  > > 
>  > > Join Us At ClueCon - Aug 7-9, 2012 
>  > > 
>  > > 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 
>  > > 
>  > > _________________________________________________________________________ 
>  > > Professional FreeSWITCH Consulting Services: 
>  > > consulting at freeswitch.org<mailto:consulting at freeswitch.org> 
>  > > http://www.freeswitchsolutions.com 
>  > > 
>  > >  
>  > >  
>  > > 
>  > > Official FreeSWITCH Sites 
>  > > http://www.freeswitch.org 
>  > > http://wiki.freeswitch.org 
>  > > http://www.cluecon.com 
>  > > 
>  > > Join Us At ClueCon - Aug 7-9, 2012 
>  > > 
>  > > 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 
>  > 
>  > _________________________________________________________________________ 
>  > Professional FreeSWITCH Consulting Services: 
>  > consulting at freeswitch.org<mailto:consulting at freeswitch.org> 
>  > http://www.freeswitchsolutions.com 
>  > 
>  >  
>  >  
>  > 
>  > Official FreeSWITCH Sites 
>  > http://www.freeswitch.org 
>  > http://wiki.freeswitch.org 
>  > http://www.cluecon.com 
>  > 
>  > Join Us At ClueCon - Aug 7-9, 2012 
>  > 
>  > 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 
>  
> _________________________________________________________________________ 
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org<mailto:consulting at freeswitch.org> 
> http://www.freeswitchsolutions.com 
>  
>  
>  
>  
> Official FreeSWITCH Sites 
> http://www.freeswitch.org 
> http://wiki.freeswitch.org 
> http://www.cluecon.com 
>  
> Join Us At ClueCon - Aug 7-9, 2012 
>  
> 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 
>  
>  
>  
> --  
> Andrew Cassidy BSc (Hons) MBCS SSCA 
> Managing Director 
> [http://c1170247.r47.cf3.rackcdn.com/emailsig.png] 
>  
> T<mailto:info at cassidywebservices.co.uk> 03300 100 960   
> F<mailto:info at cassidywebservices.co.uk> 03300 100 961 
> E<mailto:info at cassidywebservices.co.uk> andrew at cassidywebservices.co.uk<mailto:andrew at cassidywebservices.co.uk> 
> W<mailto:info at cassidywebservices.co.uk> www.cassidywebservices.co.uk<http://www.cassidywebservices.co.uk> 
>  
>  
> _________________________________________________________________________  
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org  
> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The  
> CudaTel Communication Server  Official FreeSWITCH  
> Sites http://www.freeswitch.org http://wiki.freeswitch.org  
> http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012  
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org  
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users  
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users  
> http://www.freeswitch.org 
 		 	   		  

From admin at blindi.net  Tue Jun 26 03:00:07 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 26 Jun 2012 01:00:07 +0200 (CEST)
Subject: [Freeswitch-users] Lua help to generate a existing uuid channel list
In-Reply-To: <DUB109-W21FA5C84C7C3CA014D00ED9E10@phx.gbl>
References: <col0-p4-eas433A62F51B0F7504A32F9FD9E10@phx.gbl>,
	<CALn233oDK=54WzMk308bCkJ5zafWsb4e9aVOMymL4YbDJbp-cw@mail.gmail.com>,
	<DUB109-W59C4F1A5E5E3B377C8CBF6D9E10@phx.gbl>,
	<CAMbW9oLzWZf4MapYVi68OdvNYzXuVKogkEDQWvDH53duO4HQGA@mail.gmail.com>
	<DUB109-W21FA5C84C7C3CA014D00ED9E10@phx.gbl>
Message-ID: <Pine.LNX.4.64.1206260054100.14022@tomback.antagus.de>

Hi all, i have a Problem
I setup a conference system.
I like to ceate a channel list: for example
i press:
9 i heared:
  There are 5 people attended.
Press 1 or 3 to hear the channelname.
1 backword 3 forward, 2 hear the current user.
press 4 to kick the user.

I don.t find a list funtion.

Can your help please?
Thanks


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From admin at blindi.net  Tue Jun 26 03:01:55 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 26 Jun 2012 01:01:55 +0200 (CEST)
Subject: [Freeswitch-users] Extension limit per session?
In-Reply-To: <CAD-qRkhDRdnydoEHOykSVfTW7Ob5KMW7K2bGqobgn7AGKMzVQQ@mail.gmail.com>
References: <CAP2fbbeVmMz8NMLkro88ffAkaKTtVp4gDd2-q2NACUb_qejCjA@mail.gmail.com>
	<1340357943.10949.YahooMailClassic@web110804.mail.gq1.yahoo.com>
	<F8AE005E-E72A-437A-9F44-4A54BF0568B7@visionutveckling.se>
	<Pine.LNX.4.64.1206221516050.22290@tomback.antagus.de>
	<CAD-qRkhDRdnydoEHOykSVfTW7Ob5KMW7K2bGqobgn7AGKMzVQQ@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1206260100380.14022@tomback.antagus.de>

Hi Christopher,
Thanks for your nice help!!
It.s eotkd find.

I have more than 16 extesnions.
I set it to 900

Thanks.


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From freeswitch at zoho.com  Tue Jun 26 03:32:38 2012
From: freeswitch at zoho.com (dingdong)
Date: Mon, 25 Jun 2012 16:32:38 -0700
Subject: [Freeswitch-users] FS upgrade succeeded but failed to recover
	voicemails
Message-ID: <13825fd601c.-7467766545402872296.-2053669043336472291@zoho.com>


Hello all, i have a box that has been running for like 8 months now in production,its setup has been using a odbc in core.  so i upgraded to a newer FS version,and honestly didnt know that it resets the config file to the defaults too,(without the odbc configurations). tried to reconnect the whole odbc thing. and it erased my old voicemail settings including ALL my users greetings and passwords.


how can i prevent this from happening the next time i upgrade to newer versions?




thanks

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From steveayre at gmail.com  Tue Jun 26 05:36:44 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 26 Jun 2012 02:36:44 +0100
Subject: [Freeswitch-users] FS upgrade succeeded but failed to recover
	voicemails
In-Reply-To: <13825fd601c.-7467766545402872296.-2053669043336472291@zoho.com>
References: <13825fd601c.-7467766545402872296.-2053669043336472291@zoho.com>
Message-ID: <C0E2F62F-EA3C-4400-B572-2CEAE75DEFBB@gmail.com>

How did you upgrade it? Afaik it normally shouldn't overwrite the existing config.

In future though it's probably a good idea to backup the config before an upgrade, just in case.

Sent from my iPad



On 26 Jun 2012, at 00:32, dingdong <freeswitch at zoho.com> wrote:

> 
> Hello all,
>  i have a box that has been running for like 8 months now in production,its setup has been using a odbc in core.  so i upgraded to a newer FS version,and honestly didnt know that it resets the config file to the defaults too,(without the odbc configurations). tried to reconnect the whole odbc thing. and it erased my old voicemail settings including ALL my users greetings and passwords.
> 
> how can i prevent this from happening the next time i upgrade to newer versions?
> 
> 
> thanks
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mitch.capper at gmail.com  Tue Jun 26 05:38:17 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 25 Jun 2012 18:38:17 -0700
Subject: [Freeswitch-users] FS upgrade succeeded but failed to recover
	voicemails
In-Reply-To: <13825fd601c.-7467766545402872296.-2053669043336472291@zoho.com>
References: <13825fd601c.-7467766545402872296.-2053669043336472291@zoho.com>
Message-ID: <CALn233q8Nmq7ec6Vp-7f0Q7mifF+phnDceSNjNePiCDNrFWX5Q@mail.gmail.com>

In any upgrade always backup your database and FS before upgrading so
you can rollback if needed. In addition depending on your file setup
there is a good chance the greetings and voicemails are still there
just not in the database and could be re-inserted and work again.

~mitch

On Mon, Jun 25, 2012 at 4:32 PM, dingdong <freeswitch at zoho.com> wrote:
>
> Hello all,
> ?i have a box that has been running for like 8 months now in production,its
> setup has been using a odbc in core. ?so i upgraded to a newer FS
> version,and honestly didnt know that it resets the config file to the
> defaults too,(without the odbc configurations). tried to reconnect the whole
> odbc thing. and it erased my old voicemail settings including ALL my users
> greetings and passwords.
>
> how can i prevent this from happening the next time i upgrade to newer
> versions?
>
>
> thanks
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Tue Jun 26 05:40:24 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 26 Jun 2012 02:40:24 +0100
Subject: [Freeswitch-users] Call State Diagram
In-Reply-To: <CAPNTnnpCtjRVoFa3qFbqfmLiRJBJ3--MZ6zwE-dmOaNZJV3C4w@mail.gmail.com>
References: <CAPNTnnpCtjRVoFa3qFbqfmLiRJBJ3--MZ6zwE-dmOaNZJV3C4w@mail.gmail.com>
Message-ID: <5E0D5F35-A1FE-474B-996B-0003623DD3EF@gmail.com>

I think http://wiki.freeswitch.org/wiki/Life_Cycle_of_a_Call is the closest, but it's wildly out of date.

Sent from my iPad



On 25 Jun 2012, at 15:19, Guillermo Ruiz Camauer <grcamauer at gmail.com> wrote:

> Does a Call-State Diagram exist for FreeSwitch?  I am looking for something similar to this: http://threebit.net/documentation/dialogic/resource.intel.com/telecom/support/releases/winnt/SR511/docs/htmlfiles/gcapdvgd/1526-002-3.gif.  
> I have found a list of all the different states of a call (http://wiki.freeswitch.org/wiki/Channel_States), but would like to see what events/actions trigger states changes to other states, etc.
> 
> Thanks in advance,
> 
> Guillermo Ruiz Camauer
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 zoho.com  Tue Jun 26 05:41:30 2012
From: freeswitch at zoho.com (dingdong)
Date: Mon, 25 Jun 2012 18:41:30 -0700
Subject: [Freeswitch-users]
 =?utf-8?q?FS_upgrade_succeeded_but_failed_to_r?=
 =?utf-8?b?ZWNvdmVywqDCoMKgwqB2b2ljZW1haWxz?=
In-Reply-To: <C0E2F62F-EA3C-4400-B572-2CEAE75DEFBB@gmail.com>
References: <13825fd601c.-7467766545402872296.-2053669043336472291@zoho.com>
	<C0E2F62F-EA3C-4400-B572-2CEAE75DEFBB@gmail.com>
Message-ID: <13826735ad5.-8846034897575291118.-4363218593108213977@zoho.com>

make current?

btw
i had some lua scripts that when executed freezes everything. was thinking i had to upgrade as it was live that moment only to see that i still have that same issue.





---- On Mon, 25 Jun 2012 18:36:44 -0700 Steven Ayre&lt;steveayre at gmail.com&gt; wrote ---- 


How did you upgrade it? Afaik it normally shouldn't overwrite the existing config.


In future though it's probably a good idea to backup the config before an upgrade, just in case.

Sent from my iPad





On 26 Jun 2012, at 00:32, dingdong &lt;freeswitch at zoho.com&gt; wrote:




Hello all, i have a box that has been running for like 8 months now in production,its setup has been using a odbc in core.  so i upgraded to a newer FS version,and honestly didnt know that it resets the config file to the defaults too,(without the odbc configurations). tried to reconnect the whole odbc thing. and it erased my old voicemail settings including ALL my users greetings and passwords.


how can i prevent this from happening the next time i upgrade to newer versions?




thanks


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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

_________________________________________________________________________ 
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org 
http://www.freeswitchsolutions.com 
 
 
 
 
Official FreeSWITCH Sites 
http://www.freeswitch.org 
http://wiki.freeswitch.org 
http://www.cluecon.com 
 
Join Us At ClueCon - Aug 7-9, 2012 
 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.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 zoho.com  Tue Jun 26 05:44:31 2012
From: freeswitch at zoho.com (dingdong)
Date: Mon, 25 Jun 2012 18:44:31 -0700
Subject: [Freeswitch-users]
 =?utf-8?q?FS_upgrade_succeeded_but_failed_to_r?=
 =?utf-8?b?ZWNvdmVywqDCoMKgwqB2b2ljZW1haWxz?=
In-Reply-To: <CALn233q8Nmq7ec6Vp-7f0Q7mifF+phnDceSNjNePiCDNrFWX5Q@mail.gmail.com>
References: <13825fd601c.-7467766545402872296.-2053669043336472291@zoho.com>
	<CALn233q8Nmq7ec6Vp-7f0Q7mifF+phnDceSNjNePiCDNrFWX5Q@mail.gmail.com>
Message-ID: <13826761e1d.9106865966828817104.-2630298262713724873@zoho.com>

mitch, id love to see where these greetings are usually saved and be able to re insert it,i may be paranoid but i think i can see a 100 or so people rather looking at me furiously and about to stab me to death as i also lost their saved voicemails :)




---- On Mon, 25 Jun 2012 18:38:17 -0700 Mitch Capper&lt;mitch.capper at gmail.com&gt; wrote ---- 


In any upgrade always backup your database and FS before upgrading so 
you can rollback if needed. In addition depending on your file setup 
there is a good chance the greetings and voicemails are still there 
just not in the database and could be re-inserted and work again. 
 
~mitch 
 
On Mon, Jun 25, 2012 at 4:32 PM, dingdong &lt;freeswitch at zoho.com&gt; wrote: 
&gt; 
&gt; Hello all, 
&gt;  i have a box that has been running for like 8 months now in production,its 
&gt; setup has been using a odbc in core.  so i upgraded to a newer FS 
&gt; version,and honestly didnt know that it resets the config file to the 
&gt; defaults too,(without the odbc configurations). tried to reconnect the whole 
&gt; odbc thing. and it erased my old voicemail settings including ALL my users 
&gt; greetings and passwords. 
&gt; 
&gt; how can i prevent this from happening the next time i upgrade to newer 
&gt; versions? 
&gt; 
&gt; 
&gt; thanks 
&gt; 
&gt; _________________________________________________________________________ 
&gt; Professional FreeSWITCH Consulting Services: 
&gt; consulting at freeswitch.org 
&gt; http://www.freeswitchsolutions.com 
&gt; 
&gt;  
&gt;  
&gt; 
&gt; Official FreeSWITCH Sites 
&gt; http://www.freeswitch.org 
&gt; http://wiki.freeswitch.org 
&gt; http://www.cluecon.com 
&gt; 
&gt; Join Us At ClueCon - Aug 7-9, 2012 
&gt; 
&gt; FreeSWITCH-users mailing list 
&gt; FreeSWITCH-users at lists.freeswitch.org 
&gt; http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
&gt; UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
&gt; http://www.freeswitch.org 
&gt; 
 
_________________________________________________________________________ 
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org 
http://www.freeswitchsolutions.com 
 
 
 
 
Official FreeSWITCH Sites 
http://www.freeswitch.org 
http://wiki.freeswitch.org 
http://www.cluecon.com 
 
Join Us At ClueCon - Aug 7-9, 2012 
 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 



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From jlyons at quikvoip.net  Tue Jun 26 04:25:52 2012
From: jlyons at quikvoip.net (James Lyons)
Date: Mon, 25 Jun 2012 20:25:52 -0400
Subject: [Freeswitch-users] Issue with sending fax with t.38
Message-ID: <4FE90190.8050805@quikvoip.net>

I'm attempting to use txfax in mod_spandsp to send a fax over t.38 to a 
fax machine at a PSTN number.

The receiving side hangs up, after appearing to send a number of 
malformed RTP packets.

"Ignoring invalid RTP packet size of 8 bytes."

Can anyone explain this?

opensips proxy is 1.2.3.4
fs gw is 1.2.3.5
fs originator is 1.2.3.6

Thanks,
James
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From msc at freeswitch.org  Tue Jun 26 08:53:19 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Jun 2012 21:53:19 -0700
Subject: [Freeswitch-users] Lua help to generate a existing uuid channel
	list
In-Reply-To: <Pine.LNX.4.64.1206260054100.14022@tomback.antagus.de>
References: <col0-p4-eas433A62F51B0F7504A32F9FD9E10@phx.gbl>
	<CALn233oDK=54WzMk308bCkJ5zafWsb4e9aVOMymL4YbDJbp-cw@mail.gmail.com>
	<DUB109-W59C4F1A5E5E3B377C8CBF6D9E10@phx.gbl>
	<CAMbW9oLzWZf4MapYVi68OdvNYzXuVKogkEDQWvDH53duO4HQGA@mail.gmail.com>
	<DUB109-W21FA5C84C7C3CA014D00ED9E10@phx.gbl>
	<Pine.LNX.4.64.1206260054100.14022@tomback.antagus.de>
Message-ID: <CAKzWOxVS=3NGsoLc1FNmjL+ze9XdQfEpfkgVgVTzzrPQhbRK_w@mail.gmail.com>

This is totally doable with Lua. I've done something similar for conference
management. However, it's not something you can just whip up in 5 minutes.
How's your Lua fu?

-MC

On Mon, Jun 25, 2012 at 4:00 PM, Thomas Hoellriegel <admin at blindi.net>wrote:

> Hi all, i have a Problem
> I setup a conference system.
> I like to ceate a channel list: for example
> i press:
> 9 i heared:
>  There are 5 people attended.
> Press 1 or 3 to hear the channelname.
> 1 backword 3 forward, 2 hear the current user.
> press 4 to kick the user.
>
> I don.t find a list funtion.
>
> Can your help please?
> Thanks
>
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/**listinfo/blinde-misc<http://www.blindi.net/mailman/listinfo/blinde-misc>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shcherbatyuk at belrosbank.by  Tue Jun 26 10:04:41 2012
From: shcherbatyuk at belrosbank.by (Eugene Shcherbatyuk)
Date: Tue, 26 Jun 2012 09:04:41 +0300
Subject: [Freeswitch-users] Bind extension to an IP address
In-Reply-To: <CAN5esQ+ZQ4v6U-EB9+6rdQ1uBr84c6sZL4S6oi2R=1wdah3k-Q@mail.gmail.com>
References: <4FE7175F.90209@the800group.com> <4FE787D4.5080603@gmail.com>
	<CALNZuwm-RPQRBLMRq65fb9eVdUS2W9HURZC5y+=gQ5a2GWhtqA@mail.gmail.com>
	<CAKbxfG_iUqoU0VFAsKu5a=cwV8fh02WzrEx1R70cMgXXcFy6iQ@mail.gmail.com>
	<CAN5esQ+ZQ4v6U-EB9+6rdQ1uBr84c6sZL4S6oi2R=1wdah3k-Q@mail.gmail.com>
Message-ID: <4FE950F9.3040903@belrosbank.by>


But why? Gateways are distinguished by their addr/port. One can
configure particular incoming CIDR/trunk to connect to particular
gateway. Does not that solve the problem?

On 25/06/12 12:26, Richard Brady wrote:
> Is there an equivalent for gateways?
>
> I can't find a way to bind incoming calls from a CIDR to a particular gateway.
>
>


=========================================================

?????? ????????? ? ????? ???????? (???????????) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????.

=========================================================

This message and any attachments (the ?message?) are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC ?BELROSBANK? nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified.

=========================================================


From avi at avimarcus.net  Tue Jun 26 10:19:13 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 26 Jun 2012 09:19:13 +0300
Subject: [Freeswitch-users] FS upgrade succeeded but failed to recover
	voicemails
In-Reply-To: <13826761e1d.9106865966828817104.-2630298262713724873@zoho.com>
References: <13825fd601c.-7467766545402872296.-2053669043336472291@zoho.com>
	<CALn233q8Nmq7ec6Vp-7f0Q7mifF+phnDceSNjNePiCDNrFWX5Q@mail.gmail.com>
	<13826761e1d.9106865966828817104.-2630298262713724873@zoho.com>
Message-ID: <CALNZuw=XB4rxTDmLa1cZ=AQQBZzFwc_DdBmn0RPUPT6Q7o_WHw@mail.gmail.com>

Inside the freeswitch folder (/opt/freeswitch, /usr/local/freeswitch, etc)
is: storage/voicemail/ etc.
It doesn't have the meta-data of who the messages are from or when, but it
does have the greetings and all the messages there.

-Avi


On Tue, Jun 26, 2012 at 4:44 AM, dingdong <freeswitch at zoho.com> wrote:

> **
> mitch,
>  id love to see where these greetings are usually saved and be able to re
> insert it,i may be paranoid but i think i can see a 100 or so people rather
> looking at me furiously and about to stab me to death as i also lost their
> saved voicemails :)
>
>
>
> ---- On Mon, 25 Jun 2012 18:38:17 -0700 *Mitch Capper<
> mitch.capper at gmail.com>* wrote ----
>
> In any upgrade always backup your database and FS before upgrading so
> you can rollback if needed. In addition depending on your file setup
> there is a good chance the greetings and voicemails are still there
> just not in the database and could be re-inserted and work again.
>
> ~mitch
>
> On Mon, Jun 25, 2012 at 4:32 PM, dingdong <freeswitch at zoho.com> wrote:
> >
> > Hello all,
> >  i have a box that has been running for like 8 months now in
> production,its
> > setup has been using a odbc in core.  so i upgraded to a newer FS
> > version,and honestly didnt know that it resets the config file to the
> > defaults too,(without the odbc configurations). tried to reconnect the
> whole
> > odbc thing. and it erased my old voicemail settings including ALL my
> users
> > greetings and passwords.
> >
> > how can i prevent this from happening the next time i upgrade to newer
> > versions?
> >
> >
> > thanks
> >
> >
> _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shcherbatyuk at belrosbank.by  Tue Jun 26 10:32:57 2012
From: shcherbatyuk at belrosbank.by (Eugene Shcherbatyuk)
Date: Tue, 26 Jun 2012 09:32:57 +0300
Subject: [Freeswitch-users] softphones and background noise
In-Reply-To: <DUB109-W124EB7C5B9B0D5EEA293CA8D9E10@phx.gbl>
References: <DUB109-W124EB7C5B9B0D5EEA293CA8D9E10@phx.gbl>
Message-ID: <4FE95799.6030507@belrosbank.by>

Hello,

We had a number of sound quality issues with our softphones. We could
decuce some causes for that. Often quality of sound was affected by a
combination of them. Here is a something like a checklist:

-- Softphone codec implementation is not that great. (Try different
codecs and/or softphones)
-- Softphone echo cancellation introduces noise (turn off soft echo
cancellation)
-- Transcoding issues at the softphone side.
-- Number and order of softphone codecs are different from these of VOIP
server (make them the same);
-- Noise is generated due to sound card/OS sound equalizer(mixer)
misconfiguration (turn off any effects, see that mic is not echoed to
speakers, etc., etc., etc.)
-- User PC is overloaded ('weak' CPU, little of RAM, RAM spooling, too
many open windows/processes, greedy antiviral software etc.)
-- Weared out and/or buggy headsets.

Hope this will help.


On 25/06/12 19:19, Juan Pablo L. wrote:
>
>
> Good morning, i m having lots of issues with softphones, when a call
> is initiated/terminated to a softphone the background noise is too
> high that you can hardly hear the actual voice. i have played around
> with VAD and CNG turning them on and off in the profile and/or the
> dialplan but i have not had any luck. it does not matter if the call
> is internal or goes through a gateway, it is however, worse when it
> goes through a gateway, those are the cases you can not even hear the
> voice for so much noise and when the call is to a next extension the
> background noise is high but not as high. But, if you
> initiate/terminate the call to a hardphone (like polycom phones) the
> call is perfectly clear in all cases (except when the other end is an
> internal softphone). can someone please give me a hint. thanks a
> lot!


=========================================================

?????? ????????? ? ????? ???????? (???????????) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????.

=========================================================

This message and any attachments (the ?message?) are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC ?BELROSBANK? nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified.

=========================================================


From kawarod at laposte.net  Tue Jun 26 11:53:37 2012
From: kawarod at laposte.net (kawarod)
Date: Tue, 26 Jun 2012 11:53:37 +0400
Subject: [Freeswitch-users] Bypass Media status and Can't re-establsh media
	error message
Message-ID: <4FE96A81.1010402@laposte.net>

Hi List,

I have the following error message in the console:
     2012-06-26 11:29:33.399845 [ERR] switch_ivr.c:1414 Can't 
re-establsh media on sofia/INSIDE/1000 at 10.10.10.10:5060

My dialplan is the following:
     - set bypass_media = true (I don't want to process RTP on freeswitch)
     - bridge the call
     - if bridge is unsuccessful, I execute a perl script

The Perl script catch the sip_error_code that is returned and I play an 
annoucement message depending on the SIP error.

In the perl script I do the following:
     - set bypass_media = false
     - use pre_answer to trigger a SIP 183 cause I don't want the call 
to be billed
     - playback error message
     - hangup with error code

Everything is working fine, I have a SIP 183, the message is played and 
bill_duration = 0; but I have the error message above in console.

If I use answer, instead of pre_answer, the message is played and the 
error message disapear but bill_duration is not null.

I tried to use uuid_media in my perl script but this is not working. I 
got an error message, I assume this is because uuid_media should be used 
when call is bridged.

If somebody as an idea to share :p

regards,
Rod.



From ocset at the800group.com  Tue Jun 26 13:52:42 2012
From: ocset at the800group.com (ocset)
Date: Tue, 26 Jun 2012 17:52:42 +0800
Subject: [Freeswitch-users] Bind extension to an IP address
In-Reply-To: <CAKzWOxWJuxd+PjBxDmRPa3UNJ6L68xEZotfCKc7HXccg7Jxtvw@mail.gmail.com>
References: <4FE7175F.90209@the800group.com> <4FE787D4.5080603@gmail.com>
	<CALNZuwm-RPQRBLMRq65fb9eVdUS2W9HURZC5y+=gQ5a2GWhtqA@mail.gmail.com>
	<CAKbxfG_iUqoU0VFAsKu5a=cwV8fh02WzrEx1R70cMgXXcFy6iQ@mail.gmail.com>
	<CAN5esQ+ZQ4v6U-EB9+6rdQ1uBr84c6sZL4S6oi2R=1wdah3k-Q@mail.gmail.com>
	<CAKzWOxWJuxd+PjBxDmRPa3UNJ6L68xEZotfCKc7HXccg7Jxtvw@mail.gmail.com>
Message-ID: <4FE9866A.3030207@the800group.com>

My thought were as follows

If I could bind all of the internal extensions to an IP address, then I 
can limit the number of extension that are exposed to outside attack. So 
if I open up port 5090 and allow registration from internet based 
phones, I can limit it to only a couple of extensions rather than having 
all extensions open to the outside world.

The purpose of these few extensions is to allow someone in the office to 
transfer a call to someone on the road. They would not be given 
permission to use any of the gateways so even if someone did manage to 
hack in, they  would not be able to make long distance calls - limit the 
damage.

Does that make sense?

Regards
O

On 25/06/12 22:27, Michael Collins wrote:
>
>
> On Mon, Jun 25, 2012 at 2:26 AM, Richard Brady <rnbrady at gmail.com 
> <mailto:rnbrady at gmail.com>> wrote:
>
>     Is there an equivalent for gateways?
>
>     I can't find a way to bind incoming calls from a CIDR to a
>     particular gateway.
>
> Just curious - what functionality do you need this for?
> -MC
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From admin at blindi.net  Tue Jun 26 15:03:58 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 26 Jun 2012 13:03:58 +0200 (CEST)
Subject: [Freeswitch-users] Lua help to generate a existing uuid channel
 list
In-Reply-To: <CAKzWOxVS=3NGsoLc1FNmjL+ze9XdQfEpfkgVgVTzzrPQhbRK_w@mail.gmail.com>
References: <col0-p4-eas433A62F51B0F7504A32F9FD9E10@phx.gbl>
	<CALn233oDK=54WzMk308bCkJ5zafWsb4e9aVOMymL4YbDJbp-cw@mail.gmail.com>
	<DUB109-W59C4F1A5E5E3B377C8CBF6D9E10@phx.gbl>
	<CAMbW9oLzWZf4MapYVi68OdvNYzXuVKogkEDQWvDH53duO4HQGA@mail.gmail.com>
	<DUB109-W21FA5C84C7C3CA014D00ED9E10@phx.gbl>
	<Pine.LNX.4.64.1206260054100.14022@tomback.antagus.de>
	<CAKzWOxVS=3NGsoLc1FNmjL+ze9XdQfEpfkgVgVTzzrPQhbRK_w@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1206261255550.18174@tomback.antagus.de>

Hi Michael, 
I.m not a lua guru.-) Yes i search a conferencemanagement.
I don.t find these.
Can you help please?
Thanks

---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From jpablolorenzetti at hotmail.com  Tue Jun 26 18:27:59 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Tue, 26 Jun 2012 14:27:59 +0000
Subject: [Freeswitch-users] softphones and background noise
In-Reply-To: <4FE95799.6030507@belrosbank.by>
References: <DUB109-W124EB7C5B9B0D5EEA293CA8D9E10@phx.gbl>,
	<4FE95799.6030507@belrosbank.by>
Message-ID: <DUB109-W114F2A317413DA653D24D04D9E00@phx.gbl>


thanks Eugene for all these suggestions, i followed them and they helped a lot.
the situation has improved and i think i found the root cause of this issue in the softphones.
thanks everybody for all your help.

----------------------------------------
> Date: Tue, 26 Jun 2012 09:32:57 +0300
> From: shcherbatyuk at belrosbank.by
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] softphones and background noise
>
> Hello,
>
> We had a number of sound quality issues with our softphones. We could
> decuce some causes for that. Often quality of sound was affected by a
> combination of them. Here is a something like a checklist:
>
> -- Softphone codec implementation is not that great. (Try different
> codecs and/or softphones)
> -- Softphone echo cancellation introduces noise (turn off soft echo
> cancellation)
> -- Transcoding issues at the softphone side.
> -- Number and order of softphone codecs are different from these of VOIP
> server (make them the same);
> -- Noise is generated due to sound card/OS sound equalizer(mixer)
> misconfiguration (turn off any effects, see that mic is not echoed to
> speakers, etc., etc., etc.)
> -- User PC is overloaded ('weak' CPU, little of RAM, RAM spooling, too
> many open windows/processes, greedy antiviral software etc.)
> -- Weared out and/or buggy headsets.
>
> Hope this will help.
>
>
> On 25/06/12 19:19, Juan Pablo L. wrote:
>>
>>
>> Good morning, i m having lots of issues with softphones, when a call
>> is initiated/terminated to a softphone the background noise is too
>> high that you can hardly hear the actual voice. i have played around
>> with VAD and CNG turning them on and off in the profile and/or the
>> dialplan but i have not had any luck. it does not matter if the call
>> is internal or goes through a gateway, it is however, worse when it
>> goes through a gateway, those are the cases you can not even hear the
>> voice for so much noise and when the call is to a next extension the
>> background noise is high but not as high. But, if you
>> initiate/terminate the call to a hardphone (like polycom phones) the
>> call is perfectly clear in all cases (except when the other end is an
>> internal softphone). can someone please give me a hint. thanks a
>> lot!
>
>
> =========================================================
>
> ?????? ????????? ? ????? ???????? (???????????) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????.
>
> =========================================================
>
> This message and any attachments (the ?message?) are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC ?BELROSBANK? nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified.
>
> =========================================================
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  

From steveu at coppice.org  Tue Jun 26 18:31:34 2012
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 26 Jun 2012 22:31:34 +0800
Subject: [Freeswitch-users] Issue with sending fax with t.38
In-Reply-To: <4FE90190.8050805@quikvoip.net>
References: <4FE90190.8050805@quikvoip.net>
Message-ID: <4FE9C7C6.9030907@coppice.org>

On 06/26/2012 08:25 AM, James Lyons wrote:
> ring invalid RTP packet size of 8 bytes.
is the T.38 box at the far end an old Cisco which has not been updated 
for a long time? It looks like you are getting T.38 packets on the RTP 
port, and Cisco used to do this by mistake. In fact, until they ripped 
out and replaced their T.38 engine, Cisco's was one of the buggiest.

Steve




From andrew at cassidywebservices.co.uk  Tue Jun 26 19:10:47 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Tue, 26 Jun 2012 16:10:47 +0100
Subject: [Freeswitch-users] softphones and background noise
In-Reply-To: <DUB109-W114F2A317413DA653D24D04D9E00@phx.gbl>
References: <DUB109-W124EB7C5B9B0D5EEA293CA8D9E10@phx.gbl>
	<4FE95799.6030507@belrosbank.by>
	<DUB109-W114F2A317413DA653D24D04D9E00@phx.gbl>
Message-ID: <CAMbW9oKKtQBs4G1Tp=emHcVKfPAT0CoHLDnK_B3hFe=JDtED=g@mail.gmail.com>

Hi Juan,

For the benefit of anyone else that has the same problem in the future, do
you know which settings helped the most?

On 26 June 2012 15:27, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:

>
> thanks Eugene for all these suggestions, i followed them and they helped a
> lot.
> the situation has improved and i think i found the root cause of this
> issue in the softphones.
> thanks everybody for all your help.
>
> ----------------------------------------
> > Date: Tue, 26 Jun 2012 09:32:57 +0300
> > From: shcherbatyuk at belrosbank.by
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] softphones and background noise
> >
> > Hello,
> >
> > We had a number of sound quality issues with our softphones. We could
> > decuce some causes for that. Often quality of sound was affected by a
> > combination of them. Here is a something like a checklist:
> >
> > -- Softphone codec implementation is not that great. (Try different
> > codecs and/or softphones)
> > -- Softphone echo cancellation introduces noise (turn off soft echo
> > cancellation)
> > -- Transcoding issues at the softphone side.
> > -- Number and order of softphone codecs are different from these of VOIP
> > server (make them the same);
> > -- Noise is generated due to sound card/OS sound equalizer(mixer)
> > misconfiguration (turn off any effects, see that mic is not echoed to
> > speakers, etc., etc., etc.)
> > -- User PC is overloaded ('weak' CPU, little of RAM, RAM spooling, too
> > many open windows/processes, greedy antiviral software etc.)
> > -- Weared out and/or buggy headsets.
> >
> > Hope this will help.
> >
> >
> > On 25/06/12 19:19, Juan Pablo L. wrote:
> >>
> >>
> >> Good morning, i m having lots of issues with softphones, when a call
> >> is initiated/terminated to a softphone the background noise is too
> >> high that you can hardly hear the actual voice. i have played around
> >> with VAD and CNG turning them on and off in the profile and/or the
> >> dialplan but i have not had any luck. it does not matter if the call
> >> is internal or goes through a gateway, it is however, worse when it
> >> goes through a gateway, those are the cases you can not even hear the
> >> voice for so much noise and when the call is to a next extension the
> >> background noise is high but not as high. But, if you
> >> initiate/terminate the call to a hardphone (like polycom phones) the
> >> call is perfectly clear in all cases (except when the other end is an
> >> internal softphone). can someone please give me a hint. thanks a
> >> lot!
> >
> >
> > =========================================================
> >
> > ?????? ????????? ? ????? ???????? (<<?????????>>) ????????
> ?????????????????, ???????????????? ????????????? ??? ?????????, ? ?????
> ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ?????
> ??????????????????? ????????????? ??? ??????????????? ????????? ?????????.
> ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "???
> "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ?????
> ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ????
> ?????????????.
> >
> > =========================================================
> >
> > This message and any attachments (the "message") are confidential,
> intended solely for the addressees, and may contain legally privileged
> information. Any unauthorized use or dissemination is prohibited. E-mails
> are susceptible to alteration. Neither JSC "BELROSBANK" nor any of its
> subsidiaries or affiliates shall be liable for the message if altered,
> changed or falsified.
> >
> > =========================================================
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From jpablolorenzetti at hotmail.com  Tue Jun 26 19:26:19 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Tue, 26 Jun 2012 15:26:19 +0000
Subject: [Freeswitch-users] softphones and background noise
In-Reply-To: <CAMbW9oKKtQBs4G1Tp=emHcVKfPAT0CoHLDnK_B3hFe=JDtED=g@mail.gmail.com>
References: <DUB109-W124EB7C5B9B0D5EEA293CA8D9E10@phx.gbl>,
	<4FE95799.6030507@belrosbank.by>,
	<DUB109-W114F2A317413DA653D24D04D9E00@phx.gbl>,
	<CAMbW9oKKtQBs4G1Tp=emHcVKfPAT0CoHLDnK_B3hFe=JDtED=g@mail.gmail.com>
Message-ID: <DUB109-W5753E42352290CE266EF1DD9E00@phx.gbl>


Hi , i think it was a combination of factors:
1. reordering the codecs in the softphones, so for example, i noticed my gateway
always uses GSM 8K so i made the softphone send that codec at the top of its list
2. removing echo cancelling (and/or reduce background noise) on the softphone helped too
3. lowering the mic volume helped too.

i hope this helps. regards and thanks for all your help.

________________________________
> Date: Tue, 26 Jun 2012 16:10:47 +0100 
> From: andrew at cassidywebservices.co.uk 
> To: freeswitch-users at lists.freeswitch.org 
> Subject: Re: [Freeswitch-users] softphones and background noise 
>  
> Hi Juan, 
>  
> For the benefit of anyone else that has the same problem in the future,  
> do you know which settings helped the most? 
>  
> On 26 June 2012 15:27, Juan Pablo L.  
> <jpablolorenzetti at hotmail.com<mailto:jpablolorenzetti at hotmail.com>>  
> wrote: 
>  
> thanks Eugene for all these suggestions, i followed them and they  
> helped a lot. 
> the situation has improved and i think i found the root cause of this  
> issue in the softphones. 
> thanks everybody for all your help. 
>  
> ---------------------------------------- 
>  > Date: Tue, 26 Jun 2012 09:32:57 +0300 
>  > From: shcherbatyuk at belrosbank.by<mailto:shcherbatyuk at belrosbank.by> 
>  > To:  
> freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org> 
>  > Subject: Re: [Freeswitch-users] softphones and background noise 
>  > 
>  > Hello, 
>  > 
>  > We had a number of sound quality issues with our softphones. We could 
>  > decuce some causes for that. Often quality of sound was affected by a 
>  > combination of them. Here is a something like a checklist: 
>  > 
>  > -- Softphone codec implementation is not that great. (Try different 
>  > codecs and/or softphones) 
>  > -- Softphone echo cancellation introduces noise (turn off soft echo 
>  > cancellation) 
>  > -- Transcoding issues at the softphone side. 
>  > -- Number and order of softphone codecs are different from these of VOIP 
>  > server (make them the same); 
>  > -- Noise is generated due to sound card/OS sound equalizer(mixer) 
>  > misconfiguration (turn off any effects, see that mic is not echoed to 
>  > speakers, etc., etc., etc.) 
>  > -- User PC is overloaded ('weak' CPU, little of RAM, RAM spooling, too 
>  > many open windows/processes, greedy antiviral software etc.) 
>  > -- Weared out and/or buggy headsets. 
>  > 
>  > Hope this will help. 
>  > 
>  > 
>  > On 25/06/12 19:19, Juan Pablo L. wrote: 
>  >> 
>  >> 
>  >> Good morning, i m having lots of issues with softphones, when a call 
>  >> is initiated/terminated to a softphone the background noise is too 
>  >> high that you can hardly hear the actual voice. i have played around 
>  >> with VAD and CNG turning them on and off in the profile and/or the 
>  >> dialplan but i have not had any luck. it does not matter if the call 
>  >> is internal or goes through a gateway, it is however, worse when it 
>  >> goes through a gateway, those are the cases you can not even hear the 
>  >> voice for so much noise and when the call is to a next extension the 
>  >> background noise is high but not as high. But, if you 
>  >> initiate/terminate the call to a hardphone (like polycom phones) the 
>  >> call is perfectly clear in all cases (except when the other end is an 
>  >> internal softphone). can someone please give me a hint. thanks a 
>  >> lot! 
>  > 
>  > 
>  > ========================================================= 
>  > 
>  > ?????? ????????? ? ????? ???????? (???????????) ????????  
> ?????????????????, ???????????????? ????????????? ??? ?????????, ?  
> ????? ????????? ????????, ?????????? ? ???????????? ?  
> ?????????????????. ????? ??????????????????? ????????????? ???  
> ??????????????? ????????? ?????????. ??????????? ????? ?? ???????????  
> ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ???  
> ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ??  
> ????????? ? ?????? ??? ???????, ????????? ???? ?????????????. 
>  > 
>  > ========================================================= 
>  > 
>  > This message and any attachments (the ?message?) are confidential,  
> intended solely for the addressees, and may contain legally privileged  
> information. Any unauthorized use or dissemination is prohibited.  
> E-mails are susceptible to alteration. Neither JSC ?BELROSBANK? nor any  
> of its subsidiaries or affiliates shall be liable for the message if  
> altered, changed or falsified. 
>  > 
>  > ========================================================= 
>  > 
>  > _________________________________________________________________________ 
>  > Professional FreeSWITCH Consulting Services: 
>  > consulting at freeswitch.org<mailto:consulting at freeswitch.org> 
>  > http://www.freeswitchsolutions.com 
>  > 
>  >  
>  >  
>  > 
>  > Official FreeSWITCH Sites 
>  > http://www.freeswitch.org 
>  > http://wiki.freeswitch.org 
>  > http://www.cluecon.com 
>  > 
>  > Join Us At ClueCon - Aug 7-9, 2012 
>  > 
>  > 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 
>  
> _________________________________________________________________________ 
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org<mailto:consulting at freeswitch.org> 
> http://www.freeswitchsolutions.com 
>  
>  
>  
>  
> Official FreeSWITCH Sites 
> http://www.freeswitch.org 
> http://wiki.freeswitch.org 
> http://www.cluecon.com 
>  
> Join Us At ClueCon - Aug 7-9, 2012 
>  
> 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 
>  
>  
>  
> --  
> Andrew Cassidy BSc (Hons) MBCS SSCA 
> Managing Director 
> [http://c1170247.r47.cf3.rackcdn.com/emailsig.png] 
>  
> T<mailto:info at cassidywebservices.co.uk> 03300 100 960   
> F<mailto:info at cassidywebservices.co.uk> 03300 100 961 
> E<mailto:info at cassidywebservices.co.uk> andrew at cassidywebservices.co.uk<mailto:andrew at cassidywebservices.co.uk> 
> W<mailto:info at cassidywebservices.co.uk> www.cassidywebservices.co.uk<http://www.cassidywebservices.co.uk> 
>  
>  
> _________________________________________________________________________  
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org  
> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The  
> CudaTel Communication Server  Official FreeSWITCH  
> Sites http://www.freeswitch.org http://wiki.freeswitch.org  
> http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012  
> FreeSWITCH-users mailing 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 s.lam at keensystems.eu  Tue Jun 26 19:36:42 2012
From: s.lam at keensystems.eu (Steven Lam, KeenSystems B.V.)
Date: Tue, 26 Jun 2012 15:36:42 +0000
Subject: [Freeswitch-users] T.38 rtp auto adjust
Message-ID: <2AB1E3C03738764094BDD615FEE595AD097FC9C2@DB3PRD0410MB369.eurprd04.prod.outlook.com>

Hi all,

When FS has a public IP and clients from behind nat connect to this machine the "rtp auto adjust" function is really very handy!

Now when using T.38 (T38 Passthru), it seems FS is no longer doing "rtp auto adjust".

Is there a way to make FS do "rtp auto adjust" for a T.38 fax session?

Steven.



From msc at freeswitch.org  Tue Jun 26 19:46:21 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 26 Jun 2012 08:46:21 -0700
Subject: [Freeswitch-users] Strange behaviour when sending to voicemail
In-Reply-To: <4FE7137C.4000905@the800group.com>
References: <4FE7137C.4000905@the800group.com>
Message-ID: <CAKzWOxUivV+zaz45RxmwuZLT0durUhR4RbO+c7Qw3Sa08ByN7A@mail.gmail.com>

Get a console debug log of this behavior and put it on
pastebin.freeswitch.org, then reply to this thread with the PB URL. We'll
take a peek.
-MC

On Sun, Jun 24, 2012 at 6:17 AM, ocset <ocset at the800group.com> wrote:

> Hi
>
> I have a FreeSWITCH install that uses a Faktortel SIP gateway for
> incoming/outgoing calls. The gateway codec is g.722 and everything works
> fine except for redirection to voicemail. (I have hard coded the codec
> for the gateway since it would not work otherwise.)
>
> Here is the conf/dialplan/public/01_line1.xml file with the dialplan for
> incoming calls
>
> <extension name="line1">
> <condition field="destination_number" expression="^(5555555555)$">
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set" data="continue_on_fail=true"/>
> <action application="set" data="call_timeout=20"/>
> <action application="bridge" data="group/sales@${domain_name}"/>
> <action application="answer"/>
> <action application="sleep" data="1000"/>
> <action application="voicemail" data="default ${domain} 1000"/>
> </condition>
> </extension>
>
> When the call is redirected to voicemail, half of the playback message
> is lost and the first thing the user hears is "to listen to the message
> press 1..."
>
> If I change the dialplan to transfer to the "1000" extension instead,
> allowing FreeSWITCH to redirect to voicemail automatically when ext 1001
> does not answer, then voicemail works correctly.
>
> Any suggestions?
>
> Thanks
> O.
>
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From jlyons at quikvoip.net  Tue Jun 26 18:37:41 2012
From: jlyons at quikvoip.net (James Lyons)
Date: Tue, 26 Jun 2012 10:37:41 -0400
Subject: [Freeswitch-users] Issue with sending fax with t.38
In-Reply-To: <4FE9C7C6.9030907@coppice.org>
References: <4FE90190.8050805@quikvoip.net> <4FE9C7C6.9030907@coppice.org>
Message-ID: <4FE9C935.4000208@quikvoip.net>

The end device is a samsung fax machine off of the PSTN. It's an AT&T 
line, off of what appears to be a sonus switch. The originating device 
is a FS server which is using txfax.

On 06/26/2012 10:31 AM, Steve Underwood wrote:
> On 06/26/2012 08:25 AM, James Lyons wrote:
>> ring invalid RTP packet size of 8 bytes.
> is the T.38 box at the far end an old Cisco which has not been updated
> for a long time? It looks like you are getting T.38 packets on the RTP
> port, and Cisco used to do this by mistake. In fact, until they ripped
> out and replaced their T.38 engine, Cisco's was one of the buggiest.
>
> Steve
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mbodbg at gmx.net  Tue Jun 26 22:27:09 2012
From: mbodbg at gmx.net (mbo)
Date: Tue, 26 Jun 2012 20:27:09 +0200
Subject: [Freeswitch-users] user variables on outbound calls
Message-ID: <CC0B0AB6-0745-4296-8199-CED3805F8099@gmx.net>

Hello,

I've configured the following sip extension :

 <user id="1005">
    <params>
      <param name="password" value="$${default_password}"/>
      <param name="vm-password" value="1005"/>
    </params>
    <variables>
      <variable name="effective_caller_id_name" value="0221123456789"/>
      <variable name="effective_caller_id_number" value="02212345679" />
      <variable name="outbound_caller_id_name" value="02212345679"/>
      <variable name="outbound_caller_id_number" value="02212345679"/>
      <variable name="user_context" value="sipoutbound"/>
    </variables>
  </user>

When I'm dialing out from my sip client, the variables effective_caller_id_number and outbound_caller_id_number are not set. 

 <context name="sipoutbound">
         <extension name="socket">
                <condition field="destination_number" expression="^(\d{5,20})$">
			<action application="set" data="calltype=OUTBOUND" />
                        <action application="log" data="CRIT effective_caller_id_number ${effective_caller_id_number}"/>                        
                        <action application="log" data="CRIT outbound_caller_id_number ${outbound_caller_id_number}"/>                  
                        <action application="socket" data="192.168.1.185:8025 async full" />
                        <action application="log" data="DEBUG Back from Socket"/>
                </condition>
        </extension>
  </context>


should't the variables be see with that configuration?

Thanks

Markus

From msc at freeswitch.org  Tue Jun 26 23:22:49 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 26 Jun 2012 12:22:49 -0700
Subject: [Freeswitch-users] user variables on outbound calls
In-Reply-To: <CC0B0AB6-0745-4296-8199-CED3805F8099@gmx.net>
References: <CC0B0AB6-0745-4296-8199-CED3805F8099@gmx.net>
Message-ID: <CAKzWOxVZdHcDRgKtj1ZskU8cOA9zy-0ydV=23nNVTrs0LwpyAw@mail.gmail.com>

Are you using digest authentication or ACL? Get a console debug log and
drop it onto pastebin.freeswitch.org.
-MC

On Tue, Jun 26, 2012 at 11:27 AM, mbo <mbodbg at gmx.net> wrote:

> Hello,
>
> I've configured the following sip extension :
>
>  <user id="1005">
>    <params>
>      <param name="password" value="$${default_password}"/>
>      <param name="vm-password" value="1005"/>
>    </params>
>    <variables>
>      <variable name="effective_caller_id_name" value="0221123456789"/>
>      <variable name="effective_caller_id_number" value="02212345679" />
>      <variable name="outbound_caller_id_name" value="02212345679"/>
>      <variable name="outbound_caller_id_number" value="02212345679"/>
>      <variable name="user_context" value="sipoutbound"/>
>    </variables>
>  </user>
>
> When I'm dialing out from my sip client, the variables
> effective_caller_id_number and outbound_caller_id_number are not set.
>
>  <context name="sipoutbound">
>         <extension name="socket">
>                <condition field="destination_number"
> expression="^(\d{5,20})$">
>                        <action application="set" data="calltype=OUTBOUND"
> />
>                        <action application="log" data="CRIT
> effective_caller_id_number ${effective_caller_id_number}"/>
>                        <action application="log" data="CRIT
> outbound_caller_id_number ${outbound_caller_id_number}"/>
>                        <action application="socket" data="
> 192.168.1.185:8025 async full" />
>                        <action application="log" data="DEBUG Back from
> Socket"/>
>                </condition>
>        </extension>
>  </context>
>
>
> should't the variables be see with that configuration?
>
> Thanks
>
> Markus
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mark at callassistants.com  Wed Jun 27 02:00:29 2012
From: mark at callassistants.com (marksly)
Date: Tue, 26 Jun 2012 15:00:29 -0700 (PDT)
Subject: [Freeswitch-users] Strange "timer_test" result
In-Reply-To: <F615FA10-493C-4AF5-ABE1-DB7E3AE8CA67@gmail.com>
References: <CAO4LeU18R_MgkKU=UL1t=X2qePyuKRQ9FZTssVTkYyvHnXDRkQ@mail.gmail.com>
	<CAKbxfG9rNPkqhDsB89bMydEayfmskDUhxnu9-n7JoT9ELnWyRA@mail.gmail.com>
	<CAO4LeU3Te+xPZo0BVUDX2k3pN2aW7x-o3XvBvvqtHg8ao+CCjg@mail.gmail.com>
	<CAD-qRkiV4qn3B4NNNdqXrR=mSSpbeNx=MeN8Ty8ChL=REwoCrg@mail.gmail.com>
	<CAO4LeU2a8-417E6aPgb5=_tc1ZZ-Xmai1Fyn=WZ+3pgrYv4cYQ@mail.gmail.com>
	<CAD-qRkhizVe+QsRHcgEseYbPhe2h7=VvtkMj1Nu0vLa00imwSw@mail.gmail.com>
	<CAO4LeU1ppjAo5dTgPywBOeGH0mn9pZC+ov+uu2LNNc6VxyDQcA@mail.gmail.com>
	<0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com>
	<CAO4LeU0F6kcH0tgpEx9039Xf45we2ZFLD_BHU_puhaEyY5mH7Q@mail.gmail.com>
	<F615FA10-493C-4AF5-ABE1-DB7E3AE8CA67@gmail.com>
Message-ID: <1340748029314-7580297.post@n2.nabble.com>

I am having the same issue as Stephen Wilde regarding the strange
"timer_test" results. Was it determined that the mod_posix timer is the best
solution? Likewise with Stephen, any timer except 20ms works great.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Strange-timer-test-result-tp7399299p7580297.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mitch.capper at gmail.com  Wed Jun 27 02:39:38 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 26 Jun 2012 15:39:38 -0700
Subject: [Freeswitch-users] FS upgrade succeeded but failed to recover
	voicemails
In-Reply-To: <CALNZuw=XB4rxTDmLa1cZ=AQQBZzFwc_DdBmn0RPUPT6Q7o_WHw@mail.gmail.com>
References: <13825fd601c.-7467766545402872296.-2053669043336472291@zoho.com>
	<CALn233q8Nmq7ec6Vp-7f0Q7mifF+phnDceSNjNePiCDNrFWX5Q@mail.gmail.com>
	<13826761e1d.9106865966828817104.-2630298262713724873@zoho.com>
	<CALNZuw=XB4rxTDmLa1cZ=AQQBZzFwc_DdBmn0RPUPT6Q7o_WHw@mail.gmail.com>
Message-ID: <CALn233qY9sv944OZVBTTh1jFDPJay=EGaofdkDyJUtYHBr8_pg@mail.gmail.com>

Don't forget about file created time and numbers embeded in your
voicemail files so it may be possible to reconstruct most:)

~Mitch

On Mon, Jun 25, 2012 at 11:19 PM, Avi Marcus <avi at avimarcus.net> wrote:
> Inside the freeswitch folder (/opt/freeswitch, /usr/local/freeswitch, etc)
> is: storage/voicemail/ etc.
> It doesn't have the meta-data of who the messages are from or when, but it
> does have the greetings and all the messages there.
>
> -Avi
>
>
> On Tue, Jun 26, 2012 at 4:44 AM, dingdong <freeswitch at zoho.com> wrote:
>>
>> mitch,
>> ?id love to see where these greetings are usually saved and be able to re
>> insert it,i may be paranoid but i think i can see a 100 or so people rather
>> looking at me furiously and about to stab me to death as i also lost their
>> saved voicemails :)
>>
>>
>>
>> ---- On Mon, 25 Jun 2012 18:38:17 -0700 Mitch
>> Capper<mitch.capper at gmail.com> wrote ----
>>
>> In any upgrade always backup your database and FS before upgrading so
>> you can rollback if needed. In addition depending on your file setup
>> there is a good chance the greetings and voicemails are still there
>> just not in the database and could be re-inserted and work again.
>>
>> ~mitch
>>
>> On Mon, Jun 25, 2012 at 4:32 PM, dingdong <freeswitch at zoho.com> wrote:
>> >
>> > Hello all,
>> > ?i have a box that has been running for like 8 months now in
>> > production,its
>> > setup has been using a odbc in core. ?so i upgraded to a newer FS
>> > version,and honestly didnt know that it resets the config file to the
>> > defaults too,(without the odbc configurations). tried to reconnect the
>> > whole
>> > odbc thing. and it erased my old voicemail settings including ALL my
>> > users
>> > greetings and passwords.
>> >
>> > how can i prevent this from happening the next time i upgrade to newer
>> > versions?
>> >
>> >
>> > thanks
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > Join Us At ClueCon - Aug 7-9, 2012
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From cmrienzo at gmail.com  Wed Jun 27 03:20:03 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Tue, 26 Jun 2012 19:20:03 -0400
Subject: [Freeswitch-users] Strange "timer_test" result
In-Reply-To: <1340748029314-7580297.post@n2.nabble.com>
References: <CAO4LeU18R_MgkKU=UL1t=X2qePyuKRQ9FZTssVTkYyvHnXDRkQ@mail.gmail.com>
	<CAKbxfG9rNPkqhDsB89bMydEayfmskDUhxnu9-n7JoT9ELnWyRA@mail.gmail.com>
	<CAO4LeU3Te+xPZo0BVUDX2k3pN2aW7x-o3XvBvvqtHg8ao+CCjg@mail.gmail.com>
	<CAD-qRkiV4qn3B4NNNdqXrR=mSSpbeNx=MeN8Ty8ChL=REwoCrg@mail.gmail.com>
	<CAO4LeU2a8-417E6aPgb5=_tc1ZZ-Xmai1Fyn=WZ+3pgrYv4cYQ@mail.gmail.com>
	<CAD-qRkhizVe+QsRHcgEseYbPhe2h7=VvtkMj1Nu0vLa00imwSw@mail.gmail.com>
	<CAO4LeU1ppjAo5dTgPywBOeGH0mn9pZC+ov+uu2LNNc6VxyDQcA@mail.gmail.com>
	<0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com>
	<CAO4LeU0F6kcH0tgpEx9039Xf45we2ZFLD_BHU_puhaEyY5mH7Q@mail.gmail.com>
	<F615FA10-493C-4AF5-ABE1-DB7E3AE8CA67@gmail.com>
	<1340748029314-7580297.post@n2.nabble.com>
Message-ID: <CAD-qRkjYe4onadmWrf4xv8Z+QfY9o1+Q+-FrG+-k8aJAq+5wBw@mail.gmail.com>

If you are running an older linux kernel (like the kernels in CentOS 5),
try it and see if it works for you.  There was a recent report of memory
leak using it, but I couldn't find anything when I ran valgrind.  I
currently have it set up on one of my production servers and am monitoring
it.

Chris



On Tue, Jun 26, 2012 at 6:00 PM, marksly <mark at callassistants.com> wrote:

> I am having the same issue as Stephen Wilde regarding the strange
> "timer_test" results. Was it determined that the mod_posix timer is the
> best
> solution? Likewise with Stephen, any timer except 20ms works great.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Strange-timer-test-result-tp7399299p7580297.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lloyd.aloysius at gmail.com  Wed Jun 27 04:21:27 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Tue, 26 Jun 2012 20:21:27 -0400
Subject: [Freeswitch-users] xml_curl - group - URL Request
Message-ID: <CAGvuWYa2qnXhm1UsOstJpBUpjBxu=rM=gHv5=2mt16_3wXzsog@mail.gmail.com>

Hi All,

How xml_curl works with the group.

For Example in the default configuration

 <group name="billing">
        <users>
          <user id="1005" type="pointer"/>
          <user id="1006" type="pointer"/>
          <user id="1007" type="pointer"/>
          <user id="1008" type="pointer"/>
          <user id="1009" type="pointer"/>
        </users>
      </group>


=======

User Information FreeSWITCH URL Request => *
section=directory&user=mike&domain=abc.mydomain.com*

For Group  what is the format of the request  url?

I could not find any reference in wiki

Thanks
Lloyd
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From mitch.capper at gmail.com  Wed Jun 27 05:01:48 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 26 Jun 2012 18:01:48 -0700
Subject: [Freeswitch-users] Successful faxing with spandsp / T38 and stats
Message-ID: <CALn233o+3wG0WUY1Qw3rfxdnzA7UCuv2DEobFmKVyNYxS04_CA@mail.gmail.com>

Hello everyone,
I have been working on faxing for awhile now and figure I would post
back with some results and method.  Much of my method I have
documented on the wiki about faxing but some more details on my method
are also below.  Faxing overall is a horrific thing and when I first
looked into VOIP faxing I did not think I would be able to achieve
nearly the success rate I have been able to with FreeSWITCH.

These stats were collected over 1000 individual faxes and over 4000
pages sent to over 500 unique fax numbers.   I would guess a lot of
these numbers are geographically clustered around the northwest of the
US but I doubt that should matter a whole lot.
Up until last week I had but 1 fax that didn't complete (it was an 80+
page fax that failed twice mid-fax before we just emailed it so it
probably would have completed with some extra time).

In the last week I have had two I have yet to get through (not sure
why or if I have finally found fax machines I just can't send to or
what have you).   I was blown away with the current success rate.  I
did not collect as detailed stats the entire time I was logging data
(probably only 3/4's+ of the total faxes have full detailed stats).
With all that said the below numbers should be decently reflective of
real world use.

First on the method:

I try a progressive retry system,  I try one method if it fails I wait
60 seconds then try the next.  I use two providers, flowroute and
t38faxing.com.   Flowroute I highly recommend they are a great
provider and have worked very well for faxing.  T38Faxing.com / EZCALL
are quite bad and I highly recommend avoiding (see bottom of email for
why).

My 8 methods are: T38,ECM / ECM / T38_DISABLED,ECM / T38,SLOW /
T38_DISABLED,SLOW / BACKUP,ECM / BACKUP,SLOW /
BACKUP,SLOW,T38_DISABLED

Slashes delimit the method and commas delimit the flags for that
method.   Slow=fax_disable_v17,  T38=fax_enable_t38_request,
T38_DISABLED=fax_enable_t38=false  BACKUP=T38Faxing.com (T38faxing.com
fails if t38_request is enabled so thats why T38 is not used with
them).

If the line is busy I don't move to the next method nor if there is a
channel error.

The methods are tried in order they are not tried randomly.  Randomly
would make the below stats better but I care about fax through a fast
as possible so I try in order of what I thought had the best chance of
success.

average secs per page: 28.055
average tries until success: 1.33 (not including busy).

Method Usage Rates:
        75% T38,ECM success rate: 82% avg secs/page: 27.4
        13% ECM success rate: 46% avg secs/page: 29.5
         7% ECM,T38_DISABLED success rate: 60% avg secs/page: 28.3
         2% T38,SLOW success rate: 56% avg secs/page: 45.7
         1% SLOW,T38_DISABLED success rate: 57% avg secs/page: 40.6
         1% ECM,BACKUP success rate: 57% avg secs/page: 35.8

Failure Reasons:
        40% The call dropped prematurely
        25% No response after sending a page
        11% Invalid response after sending a page
         6% Invalid ECM response received from receiver
         5% Received no response to DCS or TCF
         2% Unexpected message received
         2% Timed out waiting for initial communication
         2% The HDLC carrier did not stop in a timely manner
         1% Received bad response to DCS or training
         1% Disconnected after permitted retries
         0% Timed out waiting for the first message

Hope you enjoy the method and stats, honestly with this rate of
success and either a few better backup providers or even a single hard
line for faxing over you could run a very respectable faxing service
using FS.

~Mitch


Providers:
As I mention on the wiki certainly one very important thing other than
trying various methods is quality providers.  Flowroute handled 99% of
the faxes above obviously and did a great job.     T38Faxing however I
cannot recommend avoiding more.  I used them as my backup provider
after surveying #freeswitch a bit and while I normally try to avoid a
brand bash it has just be a horrible experience.   They have a very
poor backend which would be livable if their support was not equally
poor.  They frequently don't answer the phone and their support email
system has been down for awhile before (emails bounce) along with
their control panel for their website.   Their entire faxing service
wasn't working for over a week when contacted they said they were
adjusting their north american routing and it took over a day before
it was back up.   So overall I would suggest finding a different
provider from t38faxing.


From sdevoy at bizfocused.com  Wed Jun 27 06:50:44 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 26 Jun 2012 22:50:44 -0400
Subject: [Freeswitch-users] Successful faxing with spandsp / T38 and
	stats
In-Reply-To: <CALn233o+3wG0WUY1Qw3rfxdnzA7UCuv2DEobFmKVyNYxS04_CA@mail.gmail.com>
References: <CALn233o+3wG0WUY1Qw3rfxdnzA7UCuv2DEobFmKVyNYxS04_CA@mail.gmail.com>
Message-ID: <115201cd540f$a63b28f0$f2b17ad0$@bizfocused.com>

Mitch,

That is very impressive.  Truly, far higher than I thought possible with
VOIP.
How about sharing your dialplan and hardware (ata brand and model) and
configuration that achieves this magic!  
Are your ATAs NAT'ed and remote from your fs server?

Details man Details!  :-)  I would love a reliable fax solution for my
customers.  I have about 99% inbound and 80% outbound for one customer and
another that we could find no way to limit its transmission speed so we have
0% for it.

Thanks,
Sean

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch
Capper
Sent: Tuesday, June 26, 2012 9:02 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Successful faxing with spandsp / T38 and stats

Hello everyone,
I have been working on faxing for awhile now and figure I would post back
with some results and method.  Much of my method I have documented on the
wiki about faxing but some more details on my method are also below.  Faxing
overall is a horrific thing and when I first looked into VOIP faxing I did
not think I would be able to achieve nearly the success rate I have been
able to with FreeSWITCH.

These stats were collected over 1000 individual faxes and over 4000
pages sent to over 500 unique fax numbers.   I would guess a lot of
these numbers are geographically clustered around the northwest of the US
but I doubt that should matter a whole lot.
Up until last week I had but 1 fax that didn't complete (it was an 80+ page
fax that failed twice mid-fax before we just emailed it so it probably would
have completed with some extra time).

In the last week I have had two I have yet to get through (not sure why or
if I have finally found fax machines I just can't send to or
what have you).   I was blown away with the current success rate.  I
did not collect as detailed stats the entire time I was logging data
(probably only 3/4's+ of the total faxes have full detailed stats).
With all that said the below numbers should be decently reflective of real
world use.

First on the method:

I try a progressive retry system,  I try one method if it fails I wait
60 seconds then try the next.  I use two providers, flowroute and
t38faxing.com.   Flowroute I highly recommend they are a great
provider and have worked very well for faxing.  T38Faxing.com / EZCALL are
quite bad and I highly recommend avoiding (see bottom of email for why).

My 8 methods are: T38,ECM / ECM / T38_DISABLED,ECM / T38,SLOW /
T38_DISABLED,SLOW / BACKUP,ECM / BACKUP,SLOW / BACKUP,SLOW,T38_DISABLED

Slashes delimit the method and commas delimit the flags for that
method.   Slow=fax_disable_v17,  T38=fax_enable_t38_request,
T38_DISABLED=fax_enable_t38=false  BACKUP=T38Faxing.com (T38faxing.com fails
if t38_request is enabled so thats why T38 is not used with them).

If the line is busy I don't move to the next method nor if there is a
channel error.

The methods are tried in order they are not tried randomly.  Randomly would
make the below stats better but I care about fax through a fast as possible
so I try in order of what I thought had the best chance of success.

average secs per page: 28.055
average tries until success: 1.33 (not including busy).

Method Usage Rates:
        75% T38,ECM success rate: 82% avg secs/page: 27.4
        13% ECM success rate: 46% avg secs/page: 29.5
         7% ECM,T38_DISABLED success rate: 60% avg secs/page: 28.3
         2% T38,SLOW success rate: 56% avg secs/page: 45.7
         1% SLOW,T38_DISABLED success rate: 57% avg secs/page: 40.6
         1% ECM,BACKUP success rate: 57% avg secs/page: 35.8

Failure Reasons:
        40% The call dropped prematurely
        25% No response after sending a page
        11% Invalid response after sending a page
         6% Invalid ECM response received from receiver
         5% Received no response to DCS or TCF
         2% Unexpected message received
         2% Timed out waiting for initial communication
         2% The HDLC carrier did not stop in a timely manner
         1% Received bad response to DCS or training
         1% Disconnected after permitted retries
         0% Timed out waiting for the first message

Hope you enjoy the method and stats, honestly with this rate of success and
either a few better backup providers or even a single hard line for faxing
over you could run a very respectable faxing service using FS.

~Mitch


Providers:
As I mention on the wiki certainly one very important thing other than
trying various methods is quality providers.  Flowroute handled 99% of
the faxes above obviously and did a great job.     T38Faxing however I
cannot recommend avoiding more.  I used them as my backup provider after
surveying #freeswitch a bit and while I normally try to avoid a
brand bash it has just be a horrible experience.   They have a very
poor backend which would be livable if their support was not equally poor.
They frequently don't answer the phone and their support email system has
been down for awhile before (emails bounce) along with
their control panel for their website.   Their entire faxing service
wasn't working for over a week when contacted they said they were adjusting
their north american routing and it took over a day before
it was back up.   So overall I would suggest finding a different
provider from t38faxing.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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 Jun 27 07:20:50 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Jun 2012 22:20:50 -0500
Subject: [Freeswitch-users] FS upgrade succeeded but failed to recover
	voicemails
In-Reply-To: <CALn233qY9sv944OZVBTTh1jFDPJay=EGaofdkDyJUtYHBr8_pg@mail.gmail.com>
References: <13825fd601c.-7467766545402872296.-2053669043336472291@zoho.com>
	<CALn233q8Nmq7ec6Vp-7f0Q7mifF+phnDceSNjNePiCDNrFWX5Q@mail.gmail.com>
	<13826761e1d.9106865966828817104.-2630298262713724873@zoho.com>
	<CALNZuw=XB4rxTDmLa1cZ=AQQBZzFwc_DdBmn0RPUPT6Q7o_WHw@mail.gmail.com>
	<CALn233qY9sv944OZVBTTh1jFDPJay=EGaofdkDyJUtYHBr8_pg@mail.gmail.com>
Message-ID: <CAKbxfG9rgv1X3zwaCb7C3bJw2FxtQNOVutVd6jrR-ic0HhGjwA@mail.gmail.com>

It does not delete anything.
There is especially nothing that overwrites voicemail.

We update like this constantly.

You made some mistake somewhere.
 On Jun 26, 2012 5:40 PM, "Mitch Capper" <mitch.capper at gmail.com> wrote:

> Don't forget about file created time and numbers embeded in your
> voicemail files so it may be possible to reconstruct most:)
>
> ~Mitch
>
> On Mon, Jun 25, 2012 at 11:19 PM, Avi Marcus <avi at avimarcus.net> wrote:
> > Inside the freeswitch folder (/opt/freeswitch, /usr/local/freeswitch,
> etc)
> > is: storage/voicemail/ etc.
> > It doesn't have the meta-data of who the messages are from or when, but
> it
> > does have the greetings and all the messages there.
> >
> > -Avi
> >
> >
> > On Tue, Jun 26, 2012 at 4:44 AM, dingdong <freeswitch at zoho.com> wrote:
> >>
> >> mitch,
> >>  id love to see where these greetings are usually saved and be able to
> re
> >> insert it,i may be paranoid but i think i can see a 100 or so people
> rather
> >> looking at me furiously and about to stab me to death as i also lost
> their
> >> saved voicemails :)
> >>
> >>
> >>
> >> ---- On Mon, 25 Jun 2012 18:38:17 -0700 Mitch
> >> Capper<mitch.capper at gmail.com> wrote ----
> >>
> >> In any upgrade always backup your database and FS before upgrading so
> >> you can rollback if needed. In addition depending on your file setup
> >> there is a good chance the greetings and voicemails are still there
> >> just not in the database and could be re-inserted and work again.
> >>
> >> ~mitch
> >>
> >> On Mon, Jun 25, 2012 at 4:32 PM, dingdong <freeswitch at zoho.com> wrote:
> >> >
> >> > Hello all,
> >> >  i have a box that has been running for like 8 months now in
> >> > production,its
> >> > setup has been using a odbc in core.  so i upgraded to a newer FS
> >> > version,and honestly didnt know that it resets the config file to the
> >> > defaults too,(without the odbc configurations). tried to reconnect the
> >> > whole
> >> > odbc thing. and it erased my old voicemail settings including ALL my
> >> > users
> >> > greetings and passwords.
> >> >
> >> > how can i prevent this from happening the next time i upgrade to newer
> >> > versions?
> >> >
> >> >
> >> > thanks
> >> >
> >> >
> >> >
> _________________________________________________________________________
> >> > Professional FreeSWITCH Consulting Services:
> >> > consulting at freeswitch.org
> >> > http://www.freeswitchsolutions.com
> >> >
> >> > 
> >> > 
> >> >
> >> > Official FreeSWITCH Sites
> >> > http://www.freeswitch.org
> >> > http://wiki.freeswitch.org
> >> > http://www.cluecon.com
> >> >
> >> > Join Us At ClueCon - Aug 7-9, 2012
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kevinfuhn at gmail.com  Wed Jun 27 07:17:14 2012
From: kevinfuhn at gmail.com (kevinfuhn)
Date: Tue, 26 Jun 2012 20:17:14 -0700 (PDT)
Subject: [Freeswitch-users] skypopen.ko insertion issue
Message-ID: <1340767034403-7580303.post@n2.nabble.com>

when running insmod ./skypopen.ko following the FS wiki instruction, got
error as below:

insmod: error inserting './skypopen.ko': -1 Device or resource busy

The other info, like this
#dmesg
[ 2232.486152] skypopen OSS driver loading (www.freeswitch.org)
[ 2232.486162] skypopen OSS driver: can't get major 14

This has been stopping me from successfully integrating skype with FS for
quite a while. Can anyone out there help me out?



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/skypopen-ko-insertion-issue-tp7580303.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From tahir at ictinnovations.com  Wed Jun 27 08:30:22 2012
From: tahir at ictinnovations.com (tahir almas)
Date: Wed, 27 Jun 2012 09:30:22 +0500
Subject: [Freeswitch-users] Successful faxing with spandsp / T38 and
	stats
In-Reply-To: <115201cd540f$a63b28f0$f2b17ad0$@bizfocused.com>
References: <CALn233o+3wG0WUY1Qw3rfxdnzA7UCuv2DEobFmKVyNYxS04_CA@mail.gmail.com>
	<115201cd540f$a63b28f0$f2b17ad0$@bizfocused.com>
Message-ID: <CAOL0t++AKDjzsEYNmpz4HYhd+mDOxVNkDYGquWw94P_m2jH8_g@mail.gmail.com>

Thanks for sharing value able information, Will give a try to flowroute

I have bad experience with T38faxing too, used  gafachi and found them ok
for T.38 faxing

We have also contributed following open source fax related projects based
on plivo / freeswitch

*ICTFAX http://www.ictfax.org *
Open Source Email to Fax , Fax to Email software

*ICTDialer http://www.ictdialer.org *
Open Source SMS, Fax and Voice broadcasting solution

Regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT






On Wed, Jun 27, 2012 at 7:50 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Mitch,
>
> That is very impressive.  Truly, far higher than I thought possible with
> VOIP.
> How about sharing your dialplan and hardware (ata brand and model) and
> configuration that achieves this magic!
> Are your ATAs NAT'ed and remote from your fs server?
>
> Details man Details!  :-)  I would love a reliable fax solution for my
> customers.  I have about 99% inbound and 80% outbound for one customer and
> another that we could find no way to limit its transmission speed so we
> have
> 0% for it.
>
> Thanks,
> Sean
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch
> Capper
> Sent: Tuesday, June 26, 2012 9:02 PM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Successful faxing with spandsp / T38 and stats
>
> Hello everyone,
> I have been working on faxing for awhile now and figure I would post back
> with some results and method.  Much of my method I have documented on the
> wiki about faxing but some more details on my method are also below.
>  Faxing
> overall is a horrific thing and when I first looked into VOIP faxing I did
> not think I would be able to achieve nearly the success rate I have been
> able to with FreeSWITCH.
>
> These stats were collected over 1000 individual faxes and over 4000
> pages sent to over 500 unique fax numbers.   I would guess a lot of
> these numbers are geographically clustered around the northwest of the US
> but I doubt that should matter a whole lot.
> Up until last week I had but 1 fax that didn't complete (it was an 80+ page
> fax that failed twice mid-fax before we just emailed it so it probably
> would
> have completed with some extra time).
>
> In the last week I have had two I have yet to get through (not sure why or
> if I have finally found fax machines I just can't send to or
> what have you).   I was blown away with the current success rate.  I
> did not collect as detailed stats the entire time I was logging data
> (probably only 3/4's+ of the total faxes have full detailed stats).
> With all that said the below numbers should be decently reflective of real
> world use.
>
> First on the method:
>
> I try a progressive retry system,  I try one method if it fails I wait
> 60 seconds then try the next.  I use two providers, flowroute and
> t38faxing.com.   Flowroute I highly recommend they are a great
> provider and have worked very well for faxing.  T38Faxing.com / EZCALL are
> quite bad and I highly recommend avoiding (see bottom of email for why).
>
> My 8 methods are: T38,ECM / ECM / T38_DISABLED,ECM / T38,SLOW /
> T38_DISABLED,SLOW / BACKUP,ECM / BACKUP,SLOW / BACKUP,SLOW,T38_DISABLED
>
> Slashes delimit the method and commas delimit the flags for that
> method.   Slow=fax_disable_v17,  T38=fax_enable_t38_request,
> T38_DISABLED=fax_enable_t38=false  BACKUP=T38Faxing.com (T38faxing.com
> fails
> if t38_request is enabled so thats why T38 is not used with them).
>
> If the line is busy I don't move to the next method nor if there is a
> channel error.
>
> The methods are tried in order they are not tried randomly.  Randomly would
> make the below stats better but I care about fax through a fast as possible
> so I try in order of what I thought had the best chance of success.
>
> average secs per page: 28.055
> average tries until success: 1.33 (not including busy).
>
> Method Usage Rates:
>        75% T38,ECM success rate: 82% avg secs/page: 27.4
>        13% ECM success rate: 46% avg secs/page: 29.5
>         7% ECM,T38_DISABLED success rate: 60% avg secs/page: 28.3
>         2% T38,SLOW success rate: 56% avg secs/page: 45.7
>         1% SLOW,T38_DISABLED success rate: 57% avg secs/page: 40.6
>         1% ECM,BACKUP success rate: 57% avg secs/page: 35.8
>
> Failure Reasons:
>        40% The call dropped prematurely
>        25% No response after sending a page
>        11% Invalid response after sending a page
>         6% Invalid ECM response received from receiver
>         5% Received no response to DCS or TCF
>         2% Unexpected message received
>         2% Timed out waiting for initial communication
>         2% The HDLC carrier did not stop in a timely manner
>         1% Received bad response to DCS or training
>         1% Disconnected after permitted retries
>         0% Timed out waiting for the first message
>
> Hope you enjoy the method and stats, honestly with this rate of success and
> either a few better backup providers or even a single hard line for faxing
> over you could run a very respectable faxing service using FS.
>
> ~Mitch
>
>
> Providers:
> As I mention on the wiki certainly one very important thing other than
> trying various methods is quality providers.  Flowroute handled 99% of
> the faxes above obviously and did a great job.     T38Faxing however I
> cannot recommend avoiding more.  I used them as my backup provider after
> surveying #freeswitch a bit and while I normally try to avoid a
> brand bash it has just be a horrible experience.   They have a very
> poor backend which would be livable if their support was not equally poor.
> They frequently don't answer the phone and their support email system has
> been down for awhile before (emails bounce) along with
> their control panel for their website.   Their entire faxing service
> wasn't working for over a week when contacted they said they were adjusting
> their north american routing and it took over a day before
> it was back up.   So overall I would suggest finding a different
> provider from t38faxing.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From wstephen80 at gmail.com  Wed Jun 27 12:44:41 2012
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 27 Jun 2012 10:44:41 +0200
Subject: [Freeswitch-users] Strange "timer_test" result
In-Reply-To: <1340748029314-7580297.post@n2.nabble.com>
References: <CAO4LeU18R_MgkKU=UL1t=X2qePyuKRQ9FZTssVTkYyvHnXDRkQ@mail.gmail.com>
	<CAKbxfG9rNPkqhDsB89bMydEayfmskDUhxnu9-n7JoT9ELnWyRA@mail.gmail.com>
	<CAO4LeU3Te+xPZo0BVUDX2k3pN2aW7x-o3XvBvvqtHg8ao+CCjg@mail.gmail.com>
	<CAD-qRkiV4qn3B4NNNdqXrR=mSSpbeNx=MeN8Ty8ChL=REwoCrg@mail.gmail.com>
	<CAO4LeU2a8-417E6aPgb5=_tc1ZZ-Xmai1Fyn=WZ+3pgrYv4cYQ@mail.gmail.com>
	<CAD-qRkhizVe+QsRHcgEseYbPhe2h7=VvtkMj1Nu0vLa00imwSw@mail.gmail.com>
	<CAO4LeU1ppjAo5dTgPywBOeGH0mn9pZC+ov+uu2LNNc6VxyDQcA@mail.gmail.com>
	<0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com>
	<CAO4LeU0F6kcH0tgpEx9039Xf45we2ZFLD_BHU_puhaEyY5mH7Q@mail.gmail.com>
	<F615FA10-493C-4AF5-ABE1-DB7E3AE8CA67@gmail.com>
	<1340748029314-7580297.post@n2.nabble.com>
Message-ID: <CAO4LeU144W1OAV3+GBiZzPUNbf3HA1SXptAR3e3QFR63qX1zJg@mail.gmail.com>

My experience in FS timers:

1. voice quality is related to timer_test result so if you have irregular
result, you'll have bad voice quality (and low ACD)

2. mod_posix_timer is my solution (I haven't try timerfd because I'm using
CentOS 5.8).

3. there are two version of mod_posix_timer, the "old" one (the one before
the fix of git 'FS-4064 mod_posix_timer workaround of CentOS 5 glibc
problem'  of 30 april) and the 'new' one (present in latest git)

4. the 'old' mod_posix_timer is much more efficient that 'new' one (with
'new' mod_posix_timer is half way between 'old' mod_posix_timer and soft
timer).

5. loading the 'old' mod_posix_timer there are random FS crashes, also if
you don't use the posix timer in your profile

6. the 'new' mod_posix_timer solves the crashes of FS

So my conclusion is that the 'old' mod_posix_timer is the best one but
sometimes FS crashes but the 'new' mod_posix_timer is not good as the 'old'
one but it solves the crash issue.

In my production system I'm using the 'old' one in servers with high load
and the 'new' one in serves with less load, according with timer_test
results.

Memory leak: I have done some test mixing different versions of FS and
mod_posix_timer and in any case, if I use the latest Freeswitch git the
used memory grows up continously, also if I use 'old' mod_posix_timer so,
my conclusion is that the memory leakage is related to Freeswitch. In my
production server, latest FS git + old mod_posix_timer consumes all server
memory (12MB) in 2 days. For this reason, in all my installation I'm using
the Freeswitch git of 26th march with new or old mod_posix_timer (depending
on timer_test result).

Stephen

On Wed, Jun 27, 2012 at 12:00 AM, marksly <mark at callassistants.com> wrote:

> I am having the same issue as Stephen Wilde regarding the strange
> "timer_test" results. Was it determined that the mod_posix timer is the
> best
> solution? Likewise with Stephen, any timer except 20ms works great.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Strange-timer-test-result-tp7399299p7580297.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From saami_mh at ymail.com  Wed Jun 27 13:38:48 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Wed, 27 Jun 2012 02:38:48 -0700 (PDT)
Subject: [Freeswitch-users] is there any solutions for registering only one
	sip phone at the same time?
Message-ID: <1340789928.78394.YahooMailNeo@web120101.mail.ne1.yahoo.com>

is there any solutions for registering only one sip client at the same time,
on the other hand if i issue following command :
freeswitch at internal> sofia status profile internal reg
there is one registered sip client,
but at the same time if i registered again with the other sip phone (like 3cxphone) on the same computer o other computer with different or same ip ,so i registred again , but i don't want do that..
is there any solutions?


Call-ID: ? ? ? ?NWYyNGU0N2RkOWJmYWU4NDlmZjZkZmUxZDdmZGQ5N2U.
User: ? ? ? ? ? 100 at 192.168.10.1
Contact: ? ? ? ?"100" <sip:100 at 192.168.10.1:32838;rinstance=738b4fe258f6941e>
Agent: ? ? ? ? ?eyeBeam release 1102q stamp 51814
Status: ? ? ? ? Registered(UDP)(unknown) EXP(2012-06-27 15:03:28) EXPSECS(3656)
Host: ? ? ? ? ? pbx
IP: ? ? ? ? ? ? 192.168.10.1
Port: ? ? ? ? ? 32838
Auth-User: ? ? ?100
Auth-Realm: ? ? 85.15.7.66
MWI-Account: ? ?100 at 192.168.10.1

Total items returned: 1
=================================================================================================
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From peter.olsson at visionutveckling.se  Wed Jun 27 13:50:57 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 27 Jun 2012 09:50:57 +0000
Subject: [Freeswitch-users] is there any solutions for registering only
 one	sip phone at the same time?
In-Reply-To: <1340789928.78394.YahooMailNeo@web120101.mail.ne1.yahoo.com>
References: <1340789928.78394.YahooMailNeo@web120101.mail.ne1.yahoo.com>
Message-ID: <1FFF97C269757C458224B7C895F35F15124BA5@cantor.std.visionutv.se>

Set this in the sofia profile, it will only allow one registration. <param name="multiple-registrations" value="false"/>

However, the last registered device will always "win" of course.

Mvh
Peter Olsson

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh
Skickat: den 27 juni 2012 11:39
Till: Free SWITCH Users Help
?mne: [Freeswitch-users] is there any solutions for registering only one sip phone at the same time?

is there any solutions for registering only one sip client at the same time,
on the other hand if i issue following command :
freeswitch at internal> sofia status profile internal reg
there is one registered sip client,
but at the same time if i registered again with the other sip phone (like 3cxphone) on the same computer o other computer with different or same ip ,so i registred again , but i don't want do that..
is there any solutions?

Call-ID:        NWYyNGU0N2RkOWJmYWU4NDlmZjZkZmUxZDdmZGQ5N2U.
User:           100 at 192.168.10.1<mailto:100 at 192.168.10.1>
Contact:        "100" <sip:100 at 192.168.10.1:32838;rinstance=738b4fe258f6941e>
Agent:          eyeBeam release 1102q stamp 51814
Status:         Registered(UDP)(unknown) EXP(2012-06-27 15:03:28) EXPSECS(3656)
Host:           pbx
IP:             192.168.10.1
Port:           32838
Auth-User:      100
Auth-Realm:     85.15.7.66
MWI-Account:    100 at 192.168.10.1<mailto:100 at 192.168.10.1>

Total items returned: 1
=================================================================================================



!DSPAM:4fead3af32761440319088!
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From saami_mh at ymail.com  Wed Jun 27 14:03:05 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Wed, 27 Jun 2012 03:03:05 -0700 (PDT)
Subject: [Freeswitch-users] is there any solutions for registering only
	one sip phone at the same time?
In-Reply-To: <1FFF97C269757C458224B7C895F35F15124BA5@cantor.std.visionutv.se>
References: <1340789928.78394.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<1FFF97C269757C458224B7C895F35F15124BA5@cantor.std.visionutv.se>
Message-ID: <1340791385.61723.YahooMailNeo@web120106.mail.ne1.yahoo.com>

hi,
thanks for your reply,
but i should set in?conf/sip_profiles/internal.xml or?conf/autoload_configs/switch.conf.xml ??
by set in?conf/sip_profiles/internal.xml it dosen't work !





________________________________
 From: Peter Olsson <peter.olsson at visionutveckling.se>
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org> 
Sent: Wednesday, June 27, 2012 2:20 PM
Subject: Re: [Freeswitch-users] is there any solutions for registering only one sip phone at the same time?
 

 
Set this in the sofia profile, it will only allow one registration. <param name="multiple-registrations" value="false"/>
?
However, the last registered device will always ?win? of course.
?
Mvh
Peter Olsson
?
Fr?n:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh
Skickat: den 27 juni 2012 11:39
Till: Free SWITCH Users Help
?mne: [Freeswitch-users] is there any solutions for registering only one sip phone at the same time?
?
is there any solutions for registering only one sip client at the same time,
on the other hand if i issue following command :
freeswitch at internal> sofia status profile internal reg
there is one registered sip client,
but at the same time if i registered again with the other sip phone (like 3cxphone) on the same computer o other computer with different or same ip ,so i registred again , but i don't want do that..
is there any solutions?
?
Call-ID: ? ? ? ?NWYyNGU0N2RkOWJmYWU4NDlmZjZkZmUxZDdmZGQ5N2U.
User: ? ? ? ? ? 100 at 192.168.10.1
Contact: ? ? ? ?"100" <sip:100 at 192.168.10.1:32838;rinstance=738b4fe258f6941e>
Agent: ? ? ? ? ?eyeBeam release 1102q stamp 51814
Status: ? ? ? ? Registered(UDP)(unknown) EXP(2012-06-27 15:03:28) EXPSECS(3656)
Host: ? ? ? ? ? pbx
IP: ? ? ? ? ? ? 192.168.10.1
Port: ? ? ? ? ? 32838
Auth-User: ? ? ?100
Auth-Realm: ? ? 85.15.7.66
MWI-Account: ? ?100 at 192.168.10.1
?
Total items returned: 1
=================================================================================================
?
?
?
!DSPAM:4fead3af32761440319088! 
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From avi at avimarcus.net  Wed Jun 27 15:26:30 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 27 Jun 2012 14:26:30 +0300
Subject: [Freeswitch-users] is there any solutions for registering only
 one sip phone at the same time?
In-Reply-To: <1340791385.61723.YahooMailNeo@web120106.mail.ne1.yahoo.com>
References: <1340789928.78394.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<1FFF97C269757C458224B7C895F35F15124BA5@cantor.std.visionutv.se>
	<1340791385.61723.YahooMailNeo@web120106.mail.ne1.yahoo.com>
Message-ID: <CALNZuw=HGRiJX1yS6aN0OXWVvw_gY3TZ0cJhG9Yh3e9KiBKgLA@mail.gmail.com>

Did you reload the internal profile? reloadxml isn't good enough. You don't
need to restart the profile, though.

-Avi


On Wed, Jun 27, 2012 at 1:03 PM, Samira Mh <saami_mh at ymail.com> wrote:

> hi,
> thanks for your reply,
> but i should set in conf/sip_profiles/internal.xml or
> conf/autoload_configs/switch.conf.xml  ?
> by set in conf/sip_profiles/internal.xml it dosen't work !
>
>
>
>
>   ------------------------------
> *From:* Peter Olsson <peter.olsson at visionutveckling.se>
> *To:* 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, June 27, 2012 2:20 PM
> *Subject:* Re: [Freeswitch-users] is there any solutions for registering
> only one sip phone at the same time?
>
>   Set this in the sofia profile, it will only allow one registration. <param name="multiple-registrations" value="false"/>
>
>
> However, the last registered device will always ?win? of course.
>
>  Mvh
> Peter Olsson
>
>  *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Samira Mh
> *Skickat:* den 27 juni 2012 11:39
> *Till:* Free SWITCH Users Help
> *?mne:* [Freeswitch-users] is there any solutions for registering only
> one sip phone at the same time?
>
>  is there any solutions for registering only one sip client at the same
> time,
>  on the other hand if i issue following command :
>  freeswitch at internal> sofia status profile internal reg
>  there is one registered sip client,
>  but at the same time if i registered again with the other sip phone
> (like 3cxphone) on the same computer o other computer with different or
> same ip ,so i registred again , but i don't want do that..
>  is there any solutions?
>
>  Call-ID:        NWYyNGU0N2RkOWJmYWU4NDlmZjZkZmUxZDdmZGQ5N2U.
>  User:           100 at 192.168.10.1
>  Contact:        "100" <
> sip:100 at 192.168.10.1:32838;rinstance=738b4fe258f6941e>
>  Agent:          eyeBeam release 1102q stamp 51814
>  Status:         Registered(UDP)(unknown) EXP(2012-06-27 15:03:28)
> EXPSECS(3656)
>  Host:           pbx
>  IP:             192.168.10.1
>  Port:           32838
>  Auth-User:      100
>  Auth-Realm:     85.15.7.66
>  MWI-Account:    100 at 192.168.10.1
>
>  Total items returned: 1
>
> =================================================================================================
>
>
>
>   !DSPAM:4fead3af32761440319088!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cmrienzo at gmail.com  Wed Jun 27 15:46:56 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 27 Jun 2012 07:46:56 -0400
Subject: [Freeswitch-users] Strange "timer_test" result
In-Reply-To: <CAO4LeU144W1OAV3+GBiZzPUNbf3HA1SXptAR3e3QFR63qX1zJg@mail.gmail.com>
References: <CAO4LeU18R_MgkKU=UL1t=X2qePyuKRQ9FZTssVTkYyvHnXDRkQ@mail.gmail.com>
	<CAKbxfG9rNPkqhDsB89bMydEayfmskDUhxnu9-n7JoT9ELnWyRA@mail.gmail.com>
	<CAO4LeU3Te+xPZo0BVUDX2k3pN2aW7x-o3XvBvvqtHg8ao+CCjg@mail.gmail.com>
	<CAD-qRkiV4qn3B4NNNdqXrR=mSSpbeNx=MeN8Ty8ChL=REwoCrg@mail.gmail.com>
	<CAO4LeU2a8-417E6aPgb5=_tc1ZZ-Xmai1Fyn=WZ+3pgrYv4cYQ@mail.gmail.com>
	<CAD-qRkhizVe+QsRHcgEseYbPhe2h7=VvtkMj1Nu0vLa00imwSw@mail.gmail.com>
	<CAO4LeU1ppjAo5dTgPywBOeGH0mn9pZC+ov+uu2LNNc6VxyDQcA@mail.gmail.com>
	<0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com>
	<CAO4LeU0F6kcH0tgpEx9039Xf45we2ZFLD_BHU_puhaEyY5mH7Q@mail.gmail.com>
	<F615FA10-493C-4AF5-ABE1-DB7E3AE8CA67@gmail.com>
	<1340748029314-7580297.post@n2.nabble.com>
	<CAO4LeU144W1OAV3+GBiZzPUNbf3HA1SXptAR3e3QFR63qX1zJg@mail.gmail.com>
Message-ID: <CAD-qRkjmQSwtAGiY8MXWH=HuqwePMZU3xR1dJdYWo0ui243z5Q@mail.gmail.com>

The crash in the old posix timer is actually caused by a glibc
timer_create() race condition when using SIGEV_THREAD.  So, you could
should be able to use the old timer if you patch glibc.  I figured that was
not a good solution for end users, so I switched to SIGEV_SIGNAL which did
degrade performance.

Chris



On Wed, Jun 27, 2012 at 4:44 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> My experience in FS timers:
>
> 1. voice quality is related to timer_test result so if you have irregular
> result, you'll have bad voice quality (and low ACD)
>
> 2. mod_posix_timer is my solution (I haven't try timerfd because I'm using
> CentOS 5.8).
>
> 3. there are two version of mod_posix_timer, the "old" one (the one before
> the fix of git 'FS-4064 mod_posix_timer workaround of CentOS 5 glibc
> problem'  of 30 april) and the 'new' one (present in latest git)
>
> 4. the 'old' mod_posix_timer is much more efficient that 'new' one (with
> 'new' mod_posix_timer is half way between 'old' mod_posix_timer and soft
> timer).
>
> 5. loading the 'old' mod_posix_timer there are random FS crashes, also if
> you don't use the posix timer in your profile
>
> 6. the 'new' mod_posix_timer solves the crashes of FS
>
> So my conclusion is that the 'old' mod_posix_timer is the best one but
> sometimes FS crashes but the 'new' mod_posix_timer is not good as the 'old'
> one but it solves the crash issue.
>
> In my production system I'm using the 'old' one in servers with high load
> and the 'new' one in serves with less load, according with timer_test
> results.
>
> Memory leak: I have done some test mixing different versions of FS and
> mod_posix_timer and in any case, if I use the latest Freeswitch git the
> used memory grows up continously, also if I use 'old' mod_posix_timer so,
> my conclusion is that the memory leakage is related to Freeswitch. In my
> production server, latest FS git + old mod_posix_timer consumes all server
> memory (12MB) in 2 days. For this reason, in all my installation I'm using
> the Freeswitch git of 26th march with new or old mod_posix_timer (depending
> on timer_test result).
>
> Stephen
>
> On Wed, Jun 27, 2012 at 12:00 AM, marksly <mark at callassistants.com> wrote:
>
>> I am having the same issue as Stephen Wilde regarding the strange
>> "timer_test" results. Was it determined that the mod_posix timer is the
>> best
>> solution? Likewise with Stephen, any timer except 20ms works great.
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Strange-timer-test-result-tp7399299p7580297.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From saami_mh at ymail.com  Wed Jun 27 15:55:47 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Wed, 27 Jun 2012 04:55:47 -0700 (PDT)
Subject: [Freeswitch-users] is there any solutions for registering only
	one sip phone at the same time?
In-Reply-To: <CALNZuw=HGRiJX1yS6aN0OXWVvw_gY3TZ0cJhG9Yh3e9KiBKgLA@mail.gmail.com>
References: <1340789928.78394.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<1FFF97C269757C458224B7C895F35F15124BA5@cantor.std.visionutv.se>
	<1340791385.61723.YahooMailNeo@web120106.mail.ne1.yahoo.com>
	<CALNZuw=HGRiJX1yS6aN0OXWVvw_gY3TZ0cJhG9Yh3e9KiBKgLA@mail.gmail.com>
Message-ID: <1340798147.53903.YahooMailNeo@web120102.mail.ne1.yahoo.com>

i restart freeswitch service as well,but didn't work !


________________________________
 From: Avi Marcus <avi at avimarcus.net>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Wednesday, June 27, 2012 3:56 PM
Subject: Re: [Freeswitch-users] is there any solutions for registering only one sip phone at the same time?
 

Did you reload the internal profile? reloadxml isn't good enough. You don't need to restart the profile, though.


-Avi


On Wed, Jun 27, 2012 at 1:03 PM, Samira Mh <saami_mh at ymail.com> wrote:

hi,
>thanks for your reply,
>but i should set in?conf/sip_profiles/internal.xml or?conf/autoload_configs/switch.conf.xml ??
>by set in?conf/sip_profiles/internal.xml it dosen't work !
>
>
>
>
>
>
>
>
>
>________________________________
> From: Peter Olsson <peter.olsson at visionutveckling.se>
>To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org> 
>Sent: Wednesday, June 27, 2012 2:20 PM
>Subject: Re: [Freeswitch-users] is there any solutions for registering only one sip phone at the same time?
> 
>
>
>Set this in the sofia profile, it will only allow one registration. <param name="multiple-registrations" value="false"/>
>?
>However, the last registered device will always ?win? of course.
>?
>Mvh
>Peter Olsson
>?
>Fr?n:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh
>Skickat: den 27 juni 2012 11:39
>Till: Free SWITCH Users Help
>?mne: [Freeswitch-users] is there any solutions for registering only one sip phone at the same time?
>?
>is there any solutions for registering only one sip client at the same time,
>on the other hand if i issue following command :
>freeswitch at internal> sofia status profile internal reg
>there is one registered sip client,
>but at the same time if i registered again with the other sip phone (like 3cxphone) on the same computer o other computer with different or same ip ,so i registred again , but i don't want do that..
>is there any solutions?
>?
>Call-ID: ? ? ? ?NWYyNGU0N2RkOWJmYWU4NDlmZjZkZmUxZDdmZGQ5N2U.
>User: ? ? ? ? ? 100 at 192.168.10.1
>Contact: ? ? ? ?"100" <sip:100 at 192.168.10.1:32838;rinstance=738b4fe258f6941e>
>Agent: ? ? ? ? ?eyeBeam release 1102q stamp 51814
>Status: ? ? ? ? Registered(UDP)(unknown) EXP(2012-06-27 15:03:28) EXPSECS(3656)
>Host: ? ? ? ? ? pbx
>IP: ? ? ? ? ? ? 192.168.10.1
>Port: ? ? ? ? ? 32838
>Auth-User: ? ? ?100
>Auth-Realm: ? ? 85.15.7.66
>MWI-Account: ? ?100 at 192.168.10.1
>?
>Total items returned: 1
>=================================================================================================
>?
>?
>?
>!DSPAM:4fead3af32761440319088! 
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com/
>
>
>/
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org/
>http://wiki.freeswitch.org/
>http://www.cluecon.com/
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From mbodbg at gmx.net  Wed Jun 27 15:00:41 2012
From: mbodbg at gmx.net (mbo)
Date: Wed, 27 Jun 2012 13:00:41 +0200
Subject: [Freeswitch-users] user variables on outbound calls
In-Reply-To: <CAKzWOxVZdHcDRgKtj1ZskU8cOA9zy-0ydV=23nNVTrs0LwpyAw@mail.gmail.com>
References: <CC0B0AB6-0745-4296-8199-CED3805F8099@gmx.net>
	<CAKzWOxVZdHcDRgKtj1ZskU8cOA9zy-0ydV=23nNVTrs0LwpyAw@mail.gmail.com>
Message-ID: <4313BC13-66E7-4625-BEDD-E181657E6EFA@gmx.net>

ok, if I'm using digest based authentication the variables are filled. Do you know why it is like that?

Thanks 

Markus


Am 26.06.2012 um 21:22 schrieb Michael Collins:

> Are you using digest authentication or ACL? Get a console debug log and drop it onto pastebin.freeswitch.org.
> -MC
> 
> On Tue, Jun 26, 2012 at 11:27 AM, mbo <mbodbg at gmx.net> wrote:
> Hello,
> 
> I've configured the following sip extension :
> 
>  <user id="1005">
>    <params>
>      <param name="password" value="$${default_password}"/>
>      <param name="vm-password" value="1005"/>
>    </params>
>    <variables>
>      <variable name="effective_caller_id_name" value="0221123456789"/>
>      <variable name="effective_caller_id_number" value="02212345679" />
>      <variable name="outbound_caller_id_name" value="02212345679"/>
>      <variable name="outbound_caller_id_number" value="02212345679"/>
>      <variable name="user_context" value="sipoutbound"/>
>    </variables>
>  </user>
> 
> When I'm dialing out from my sip client, the variables effective_caller_id_number and outbound_caller_id_number are not set.
> 
>  <context name="sipoutbound">
>         <extension name="socket">
>                <condition field="destination_number" expression="^(\d{5,20})$">
>                        <action application="set" data="calltype=OUTBOUND" />
>                        <action application="log" data="CRIT effective_caller_id_number ${effective_caller_id_number}"/>
>                        <action application="log" data="CRIT outbound_caller_id_number ${outbound_caller_id_number}"/>
>                        <action application="socket" data="192.168.1.185:8025 async full" />
>                        <action application="log" data="DEBUG Back from Socket"/>
>                </condition>
>        </extension>
>  </context>
> 
> 
> should't the variables be see with that configuration?
> 
> Thanks
> 
> Markus
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jun 27 17:37:45 2012
From: brian at freeswitch.org (Brian West)
Date: Wed, 27 Jun 2012 08:37:45 -0500
Subject: [Freeswitch-users] Successful faxing with spandsp / T38 and
	stats
In-Reply-To: <CAOL0t++AKDjzsEYNmpz4HYhd+mDOxVNkDYGquWw94P_m2jH8_g@mail.gmail.com>
References: <CALn233o+3wG0WUY1Qw3rfxdnzA7UCuv2DEobFmKVyNYxS04_CA@mail.gmail.com>
	<115201cd540f$a63b28f0$f2b17ad0$@bizfocused.com>
	<CAOL0t++AKDjzsEYNmpz4HYhd+mDOxVNkDYGquWw94P_m2jH8_g@mail.gmail.com>
Message-ID: <EFB04B2E-EE79-465A-9157-25860CB94EF0@freeswitch.org>

I've had nothing but good tests with t38faxing.com but I was mostly doing passthru with an SPA3102,  

For flowroute you may need to set sip_require_timer=false on some destinations.  

--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.918.420.9266  |  F: +1.918.420.9267  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9266
UK: +44 20 3298 4900




On Jun 26, 2012, at 11:30 PM, tahir almas wrote:

> Thanks for sharing value able information, Will give a try to flowroute 
> 
> I have bad experience with T38faxing too, used  gafachi and found them ok  for T.38 faxing 
> 
> We have also contributed following open source fax related projects based on plivo / freeswitch 

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From wstephen80 at gmail.com  Wed Jun 27 17:53:52 2012
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 27 Jun 2012 15:53:52 +0200
Subject: [Freeswitch-users] Strange "timer_test" result
In-Reply-To: <CAD-qRkjmQSwtAGiY8MXWH=HuqwePMZU3xR1dJdYWo0ui243z5Q@mail.gmail.com>
References: <CAO4LeU18R_MgkKU=UL1t=X2qePyuKRQ9FZTssVTkYyvHnXDRkQ@mail.gmail.com>
	<CAKbxfG9rNPkqhDsB89bMydEayfmskDUhxnu9-n7JoT9ELnWyRA@mail.gmail.com>
	<CAO4LeU3Te+xPZo0BVUDX2k3pN2aW7x-o3XvBvvqtHg8ao+CCjg@mail.gmail.com>
	<CAD-qRkiV4qn3B4NNNdqXrR=mSSpbeNx=MeN8Ty8ChL=REwoCrg@mail.gmail.com>
	<CAO4LeU2a8-417E6aPgb5=_tc1ZZ-Xmai1Fyn=WZ+3pgrYv4cYQ@mail.gmail.com>
	<CAD-qRkhizVe+QsRHcgEseYbPhe2h7=VvtkMj1Nu0vLa00imwSw@mail.gmail.com>
	<CAO4LeU1ppjAo5dTgPywBOeGH0mn9pZC+ov+uu2LNNc6VxyDQcA@mail.gmail.com>
	<0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com>
	<CAO4LeU0F6kcH0tgpEx9039Xf45we2ZFLD_BHU_puhaEyY5mH7Q@mail.gmail.com>
	<F615FA10-493C-4AF5-ABE1-DB7E3AE8CA67@gmail.com>
	<1340748029314-7580297.post@n2.nabble.com>
	<CAO4LeU144W1OAV3+GBiZzPUNbf3HA1SXptAR3e3QFR63qX1zJg@mail.gmail.com>
	<CAD-qRkjmQSwtAGiY8MXWH=HuqwePMZU3xR1dJdYWo0ui243z5Q@mail.gmail.com>
Message-ID: <CAO4LeU08sUKbzC1Ee4fjHa3vB_XEdRQgc=ePiqrVMJNZssYKsw@mail.gmail.com>

Thank you Chirs,
do you have any reference to how to patch the glibc for this issue?

Stephen

On Wed, Jun 27, 2012 at 1:46 PM, Christopher Rienzo <cmrienzo at gmail.com>wrote:

> The crash in the old posix timer is actually caused by a glibc
> timer_create() race condition when using SIGEV_THREAD.  So, you could
> should be able to use the old timer if you patch glibc.  I figured that was
> not a good solution for end users, so I switched to SIGEV_SIGNAL which did
> degrade performance.
>
> Chris
>
>
>
> On Wed, Jun 27, 2012 at 4:44 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> My experience in FS timers:
>>
>> 1. voice quality is related to timer_test result so if you have irregular
>> result, you'll have bad voice quality (and low ACD)
>>
>> 2. mod_posix_timer is my solution (I haven't try timerfd because I'm
>> using CentOS 5.8).
>>
>> 3. there are two version of mod_posix_timer, the "old" one (the one
>> before the fix of git 'FS-4064 mod_posix_timer workaround of CentOS 5 glibc
>> problem'  of 30 april) and the 'new' one (present in latest git)
>>
>> 4. the 'old' mod_posix_timer is much more efficient that 'new' one (with
>> 'new' mod_posix_timer is half way between 'old' mod_posix_timer and soft
>> timer).
>>
>>  5. loading the 'old' mod_posix_timer there are random FS crashes, also
>> if you don't use the posix timer in your profile
>>
>> 6. the 'new' mod_posix_timer solves the crashes of FS
>>
>> So my conclusion is that the 'old' mod_posix_timer is the best one but
>> sometimes FS crashes but the 'new' mod_posix_timer is not good as the 'old'
>> one but it solves the crash issue.
>>
>> In my production system I'm using the 'old' one in servers with high load
>> and the 'new' one in serves with less load, according with timer_test
>> results.
>>
>> Memory leak: I have done some test mixing different versions of FS and
>> mod_posix_timer and in any case, if I use the latest Freeswitch git the
>> used memory grows up continously, also if I use 'old' mod_posix_timer so,
>> my conclusion is that the memory leakage is related to Freeswitch. In my
>> production server, latest FS git + old mod_posix_timer consumes all server
>> memory (12MB) in 2 days. For this reason, in all my installation I'm using
>> the Freeswitch git of 26th march with new or old mod_posix_timer (depending
>> on timer_test result).
>>
>> Stephen
>>
>> On Wed, Jun 27, 2012 at 12:00 AM, marksly <mark at callassistants.com>wrote:
>>
>>> I am having the same issue as Stephen Wilde regarding the strange
>>> "timer_test" results. Was it determined that the mod_posix timer is the
>>> best
>>> solution? Likewise with Stephen, any timer except 20ms works great.
>>>
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/Strange-timer-test-result-tp7399299p7580297.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveu at coppice.org  Wed Jun 27 18:24:55 2012
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 27 Jun 2012 22:24:55 +0800
Subject: [Freeswitch-users] Successful faxing with spandsp / T38 and
	stats
In-Reply-To: <CALn233o+3wG0WUY1Qw3rfxdnzA7UCuv2DEobFmKVyNYxS04_CA@mail.gmail.com>
References: <CALn233o+3wG0WUY1Qw3rfxdnzA7UCuv2DEobFmKVyNYxS04_CA@mail.gmail.com>
Message-ID: <4FEB17B7.1010503@coppice.org>

On 06/27/2012 09:01 AM, Mitch Capper wrote:
> Hello everyone,
> I have been working on faxing for awhile now and figure I would post
> back with some results and method.  Much of my method I have
> documented on the wiki about faxing but some more details on my method
> are also below.  Faxing overall is a horrific thing and when I first
> looked into VOIP faxing I did not think I would be able to achieve
> nearly the success rate I have been able to with FreeSWITCH.
>
> These stats were collected over 1000 individual faxes and over 4000
> pages sent to over 500 unique fax numbers.   I would guess a lot of
> these numbers are geographically clustered around the northwest of the
> US but I doubt that should matter a whole lot.
> Up until last week I had but 1 fax that didn't complete (it was an 80+
> page fax that failed twice mid-fax before we just emailed it so it
> probably would have completed with some extra time).
>
> In the last week I have had two I have yet to get through (not sure
> why or if I have finally found fax machines I just can't send to or
> what have you).   I was blown away with the current success rate.  I
> did not collect as detailed stats the entire time I was logging data
> (probably only 3/4's+ of the total faxes have full detailed stats).
> With all that said the below numbers should be decently reflective of
> real world use.
>
> First on the method:
>
> I try a progressive retry system,  I try one method if it fails I wait
> 60 seconds then try the next.  I use two providers, flowroute and
> t38faxing.com.   Flowroute I highly recommend they are a great
> provider and have worked very well for faxing.  T38Faxing.com / EZCALL
> are quite bad and I highly recommend avoiding (see bottom of email for
> why).
>
> My 8 methods are: T38,ECM / ECM / T38_DISABLED,ECM / T38,SLOW /
> T38_DISABLED,SLOW / BACKUP,ECM / BACKUP,SLOW /
> BACKUP,SLOW,T38_DISABLED
>
> Slashes delimit the method and commas delimit the flags for that
> method.   Slow=fax_disable_v17,  T38=fax_enable_t38_request,
> T38_DISABLED=fax_enable_t38=false  BACKUP=T38Faxing.com (T38faxing.com
> fails if t38_request is enabled so thats why T38 is not used with
> them).
>
> If the line is busy I don't move to the next method nor if there is a
> channel error.
>
> The methods are tried in order they are not tried randomly.  Randomly
> would make the below stats better but I care about fax through a fast
> as possible so I try in order of what I thought had the best chance of
> success.
>
> average secs per page: 28.055
> average tries until success: 1.33 (not including busy).
>
> Method Usage Rates:
>          75% T38,ECM success rate: 82% avg secs/page: 27.4
>          13% ECM success rate: 46% avg secs/page: 29.5
>           7% ECM,T38_DISABLED success rate: 60% avg secs/page: 28.3
>           2% T38,SLOW success rate: 56% avg secs/page: 45.7
>           1% SLOW,T38_DISABLED success rate: 57% avg secs/page: 40.6
>           1% ECM,BACKUP success rate: 57% avg secs/page: 35.8
How should I interpret those percentages? Does it mean 75% of call 
attempts were with T.38 and ECM, and 82% of those attempts were 
successful? If so, that's somewhat encouraging, as a lot of T.38 boxes 
have broken ECM support, and quite a lot of operators suppress ECM mode.

Overall, its rather sad. On the PSTN we would get 99.5% success first 
time as a bare minimum, and then look to improve the results. :-)

>
> Failure Reasons:
>          40% The call dropped prematurely
>          25% No response after sending a page
>          11% Invalid response after sending a page
>           6% Invalid ECM response received from receiver
>           5% Received no response to DCS or TCF
>           2% Unexpected message received
>           2% Timed out waiting for initial communication
>           2% The HDLC carrier did not stop in a timely manner
>           1% Received bad response to DCS or training
>           1% Disconnected after permitted retries
>           0% Timed out waiting for the first message
I don't think you can get much from those figures. Its a real mixed bag 
of results. If there were a dominant pattern it might represent a bug or 
design weakness I might be able to address. This just suggests poor 
channels.

Steve



From mitch.capper at gmail.com  Wed Jun 27 18:28:49 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 27 Jun 2012 07:28:49 -0700
Subject: [Freeswitch-users] Successful faxing with spandsp / T38 and
	stats
In-Reply-To: <EFB04B2E-EE79-465A-9157-25860CB94EF0@freeswitch.org>
References: <CALn233o+3wG0WUY1Qw3rfxdnzA7UCuv2DEobFmKVyNYxS04_CA@mail.gmail.com>
	<115201cd540f$a63b28f0$f2b17ad0$@bizfocused.com>
	<CAOL0t++AKDjzsEYNmpz4HYhd+mDOxVNkDYGquWw94P_m2jH8_g@mail.gmail.com>
	<EFB04B2E-EE79-465A-9157-25860CB94EF0@freeswitch.org>
Message-ID: <CALn233oQ+p-9hooKb7vSPQFfgUmS2_5Xr7g5xzCFofkwOmY5Hw@mail.gmail.com>

BKW Thanks for the tip I will add it into the methods tried.

Sean between this and the mod_spandsp wiki page I have virtually all
of my details public at this point.   My three biggest tips would be:
1) Always resize the fax down to the most common resolution and size
(well technically I support two resolutions letter and legal)
2) Use quality providers and decent hardware for FS (nothing huge but
given the importance of timing you certainly may want to start without
using a VM to ensure that is not a factor)
3) Try try again with different methods, while faxing should be more
of a negotiation I can't tell you how many fax machines just can't
negotiate well.

~Mitch

On Wed, Jun 27, 2012 at 6:37 AM, Brian West <brian at freeswitch.org> wrote:
> I've had nothing but good tests with t38faxing.com but I was mostly doing
> passthru with an SPA3102,
>
> For flowroute you may need to set?sip_require_timer=false on some
> destinations.
>
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> T: +1.918.420.9266? | ?F: +1.918.420.9267 ?| ?M: +1.918.424.WEST
> iNUM: +883 5100 1420 9266
> UK: +44 20 3298 4900
>
>
>
>
> On Jun 26, 2012, at 11:30 PM, tahir almas wrote:
>
> Thanks for sharing value able information, Will give a try to flowroute
>
> I have bad experience with T38faxing too, used? gafachi and found them ok
> for T.38 faxing
>
> We have also contributed following open source fax related projects based on
> plivo / freeswitch
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From avi at avimarcus.net  Wed Jun 27 18:33:05 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 27 Jun 2012 17:33:05 +0300
Subject: [Freeswitch-users] user variables on outbound calls
In-Reply-To: <4313BC13-66E7-4625-BEDD-E181657E6EFA@gmx.net>
References: <CC0B0AB6-0745-4296-8199-CED3805F8099@gmx.net>
	<CAKzWOxVZdHcDRgKtj1ZskU8cOA9zy-0ydV=23nNVTrs0LwpyAw@mail.gmail.com>
	<4313BC13-66E7-4625-BEDD-E181657E6EFA@gmx.net>
Message-ID: <CALNZuwkL4_sEx_uf3tGiFXdK5DwAz+LOdOFUAEMAzYSurMr6Hg@mail.gmail.com>

If you specify the ACL within the user as a CIDR then it will work too.
See: http://wiki.freeswitch.org/wiki/Acl#Users
If you use the generic ACL then you haven't instructed FreeSWITCH which
account information it should use.

-Avi


On Wed, Jun 27, 2012 at 2:00 PM, mbo <mbodbg at gmx.net> wrote:

> ok, if I'm using digest based authentication the variables are filled. Do
> you know why it is like that?
>
> Thanks
>
> Markus
>
>
> Am 26.06.2012 um 21:22 schrieb Michael Collins:
>
> Are you using digest authentication or ACL? Get a console debug log and
> drop it onto pastebin.freeswitch.org.
> -MC
>
> On Tue, Jun 26, 2012 at 11:27 AM, mbo <mbodbg at gmx.net> wrote:
>
>> Hello,
>>
>> I've configured the following sip extension :
>>
>>  <user id="1005">
>>    <params>
>>      <param name="password" value="$${default_password}"/>
>>      <param name="vm-password" value="1005"/>
>>    </params>
>>    <variables>
>>      <variable name="effective_caller_id_name" value="0221123456789"/>
>>      <variable name="effective_caller_id_number" value="02212345679" />
>>      <variable name="outbound_caller_id_name" value="02212345679"/>
>>      <variable name="outbound_caller_id_number" value="02212345679"/>
>>      <variable name="user_context" value="sipoutbound"/>
>>    </variables>
>>  </user>
>>
>> When I'm dialing out from my sip client, the variables
>> effective_caller_id_number and outbound_caller_id_number are not set.
>>
>>  <context name="sipoutbound">
>>         <extension name="socket">
>>                <condition field="destination_number"
>> expression="^(\d{5,20})$">
>>                        <action application="set" data="calltype=OUTBOUND"
>> />
>>                        <action application="log" data="CRIT
>> effective_caller_id_number ${effective_caller_id_number}"/>
>>                        <action application="log" data="CRIT
>> outbound_caller_id_number ${outbound_caller_id_number}"/>
>>                        <action application="socket" data="
>> 192.168.1.185:8025 async full" />
>>                        <action application="log" data="DEBUG Back from
>> Socket"/>
>>                </condition>
>>        </extension>
>>  </context>
>>
>>
>> should't the variables be see with that configuration?
>>
>> Thanks
>>
>> Markus
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mitch.capper at gmail.com  Wed Jun 27 18:39:31 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 27 Jun 2012 07:39:31 -0700
Subject: [Freeswitch-users] Successful faxing with spandsp / T38 and
	stats
In-Reply-To: <4FEB17B7.1010503@coppice.org>
References: <CALn233o+3wG0WUY1Qw3rfxdnzA7UCuv2DEobFmKVyNYxS04_CA@mail.gmail.com>
	<4FEB17B7.1010503@coppice.org>
Message-ID: <CALn233p1wCgKLkhTwi9xRnW9-W-B2-fZVs1JULPTQJeo60+iMA@mail.gmail.com>

Sorry Steve for the confusion so:
75% T38,ECM success rate: 82%

does mean that 75% of the tries used the T38,ECM flags  and 82%
succeed.    Whats important to note is that while I turn ECM on
frequently it gets disabled as the remote faxing does not support it
or the operator suppresses it.  I actually had a hard time getting ECM
to be enabled to most places.  What I show in the flags are the
options I pass to FS not necessarily how the fax ends up going.  So a
T38,ECM flag means I tell FS to use ECM and to invite with T38 but
obviously they could turn around and negotiate no T38 and no ECM.
Also remember that T38/ECM is the first method tried and with most
faxes only taking 1.3 tries there is a good chance the first method
would always have a high percent.    For the stats to be super useful
it would be better if I chose the method at random unfortunately I
care more about getting the faxes through faster than providing the
perfect FS faxing stats.

It is sad that VOIP faxing has such a lower first try success rate,
maybe with some tweaks it could get above where I have it now for the
first try, but (atleast for my situation) getting to the 99%+ success
rate overall for outbound faxing was something I did not initially get
the impression was highly likely to achieve.  (And sorry these are all
Outbound faxing stats I did not include any inbound as those are less
interesting) .


Finally yes the Failure reasons stats I assume are not super useful
but as I had them I figured I would throw them up.   Keep in mind the
% for each failure reason is not how often is occurs just how often it
is the reason a fax fails.   The idea being if you are trying to do
faxing and seeing percentages hugely different than that you may be
doing something else wrong.      There certainly could be a better
provider for T38 faxing than flowroute, but I was already using them
and as they were quite successful in delivering faxes for me I did not
do a huge search for other providers.  If however anyone does have
other recommendations I do want to replace T38faxing as my backup
provider.


~Mitch

On Wed, Jun 27, 2012 at 7:24 AM, Steve Underwood <steveu at coppice.org> wrote:


From mitch.capper at gmail.com  Wed Jun 27 18:43:44 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 27 Jun 2012 07:43:44 -0700
Subject: [Freeswitch-users] Successful faxing with spandsp / T38 and
	stats
In-Reply-To: <EFB04B2E-EE79-465A-9157-25860CB94EF0@freeswitch.org>
References: <CALn233o+3wG0WUY1Qw3rfxdnzA7UCuv2DEobFmKVyNYxS04_CA@mail.gmail.com>
	<115201cd540f$a63b28f0$f2b17ad0$@bizfocused.com>
	<CAOL0t++AKDjzsEYNmpz4HYhd+mDOxVNkDYGquWw94P_m2jH8_g@mail.gmail.com>
	<EFB04B2E-EE79-465A-9157-25860CB94EF0@freeswitch.org>
Message-ID: <CALn233pno1QOivLGU9VidANmwWbyW-KLD-i1Bna9T4-vW-1R1w@mail.gmail.com>

Also as for t38faxing the quality of their routes may be fine, but it
was down for a decent period and their CS/support is just the worst I
have seen so as long as they are just one of the providers in your
pool it may be a good option.

~Mitch

On Wed, Jun 27, 2012 at 6:37 AM, Brian West <brian at freeswitch.org> wrote:
> I've had nothing but good tests with t38faxing.com but I was mostly doing
> passthru with an SPA3102,
>


From cmrienzo at gmail.com  Wed Jun 27 19:04:56 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 27 Jun 2012 11:04:56 -0400
Subject: [Freeswitch-users] Strange "timer_test" result
In-Reply-To: <CAO4LeU08sUKbzC1Ee4fjHa3vB_XEdRQgc=ePiqrVMJNZssYKsw@mail.gmail.com>
References: <CAO4LeU18R_MgkKU=UL1t=X2qePyuKRQ9FZTssVTkYyvHnXDRkQ@mail.gmail.com>
	<CAKbxfG9rNPkqhDsB89bMydEayfmskDUhxnu9-n7JoT9ELnWyRA@mail.gmail.com>
	<CAO4LeU3Te+xPZo0BVUDX2k3pN2aW7x-o3XvBvvqtHg8ao+CCjg@mail.gmail.com>
	<CAD-qRkiV4qn3B4NNNdqXrR=mSSpbeNx=MeN8Ty8ChL=REwoCrg@mail.gmail.com>
	<CAO4LeU2a8-417E6aPgb5=_tc1ZZ-Xmai1Fyn=WZ+3pgrYv4cYQ@mail.gmail.com>
	<CAD-qRkhizVe+QsRHcgEseYbPhe2h7=VvtkMj1Nu0vLa00imwSw@mail.gmail.com>
	<CAO4LeU1ppjAo5dTgPywBOeGH0mn9pZC+ov+uu2LNNc6VxyDQcA@mail.gmail.com>
	<0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com>
	<CAO4LeU0F6kcH0tgpEx9039Xf45we2ZFLD_BHU_puhaEyY5mH7Q@mail.gmail.com>
	<F615FA10-493C-4AF5-ABE1-DB7E3AE8CA67@gmail.com>
	<1340748029314-7580297.post@n2.nabble.com>
	<CAO4LeU144W1OAV3+GBiZzPUNbf3HA1SXptAR3e3QFR63qX1zJg@mail.gmail.com>
	<CAD-qRkjmQSwtAGiY8MXWH=HuqwePMZU3xR1dJdYWo0ui243z5Q@mail.gmail.com>
	<CAO4LeU08sUKbzC1Ee4fjHa3vB_XEdRQgc=ePiqrVMJNZssYKsw@mail.gmail.com>
Message-ID: <CAD-qRkhT0ZBca+Zz0wp3Fpc42pcV6x=eR6E_NErfeTDHz4G7nA@mail.gmail.com>

I linked to it in the Jira ticket you had opened..

Bug description here:
http://sourceware.org/bugzilla/show_bug.cgi?id=7094


Patch here:
http://sourceware.org/bugzilla/attachment.cgi?id=3681

I've never tried to patch a system library, so you're on your own if you
want to open that can of worms.

Chris



On Wed, Jun 27, 2012 at 9:53 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> Thank you Chirs,
> do you have any reference to how to patch the glibc for this issue?
>
> Stephen
>
>
> On Wed, Jun 27, 2012 at 1:46 PM, Christopher Rienzo <cmrienzo at gmail.com>wrote:
>
>> The crash in the old posix timer is actually caused by a glibc
>> timer_create() race condition when using SIGEV_THREAD.  So, you could
>> should be able to use the old timer if you patch glibc.  I figured that was
>> not a good solution for end users, so I switched to SIGEV_SIGNAL which did
>> degrade performance.
>>
>> Chris
>>
>>
>>
>> On Wed, Jun 27, 2012 at 4:44 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> My experience in FS timers:
>>>
>>> 1. voice quality is related to timer_test result so if you have
>>> irregular result, you'll have bad voice quality (and low ACD)
>>>
>>> 2. mod_posix_timer is my solution (I haven't try timerfd because I'm
>>> using CentOS 5.8).
>>>
>>> 3. there are two version of mod_posix_timer, the "old" one (the one
>>> before the fix of git 'FS-4064 mod_posix_timer workaround of CentOS 5 glibc
>>> problem'  of 30 april) and the 'new' one (present in latest git)
>>>
>>> 4. the 'old' mod_posix_timer is much more efficient that 'new' one (with
>>> 'new' mod_posix_timer is half way between 'old' mod_posix_timer and soft
>>> timer).
>>>
>>>  5. loading the 'old' mod_posix_timer there are random FS crashes, also
>>> if you don't use the posix timer in your profile
>>>
>>> 6. the 'new' mod_posix_timer solves the crashes of FS
>>>
>>> So my conclusion is that the 'old' mod_posix_timer is the best one but
>>> sometimes FS crashes but the 'new' mod_posix_timer is not good as the 'old'
>>> one but it solves the crash issue.
>>>
>>> In my production system I'm using the 'old' one in servers with high
>>> load and the 'new' one in serves with less load, according with timer_test
>>> results.
>>>
>>> Memory leak: I have done some test mixing different versions of FS and
>>> mod_posix_timer and in any case, if I use the latest Freeswitch git the
>>> used memory grows up continously, also if I use 'old' mod_posix_timer so,
>>> my conclusion is that the memory leakage is related to Freeswitch. In my
>>> production server, latest FS git + old mod_posix_timer consumes all server
>>> memory (12MB) in 2 days. For this reason, in all my installation I'm using
>>> the Freeswitch git of 26th march with new or old mod_posix_timer (depending
>>> on timer_test result).
>>>
>>> Stephen
>>>
>>> On Wed, Jun 27, 2012 at 12:00 AM, marksly <mark at callassistants.com>wrote:
>>>
>>>> I am having the same issue as Stephen Wilde regarding the strange
>>>> "timer_test" results. Was it determined that the mod_posix timer is the
>>>> best
>>>> solution? Likewise with Stephen, any timer except 20ms works great.
>>>>
>>>> --
>>>> View this message in context:
>>>> http://freeswitch-users.2379917.n2.nabble.com/Strange-timer-test-result-tp7399299p7580297.html
>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sdevoy at bizfocused.com  Wed Jun 27 19:23:25 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 27 Jun 2012 11:23:25 -0400
Subject: [Freeswitch-users] Successful faxing with spandsp / T38
	and	stats
In-Reply-To: <CALn233p1wCgKLkhTwi9xRnW9-W-B2-fZVs1JULPTQJeo60+iMA@mail.gmail.com>
References: <CALn233o+3wG0WUY1Qw3rfxdnzA7UCuv2DEobFmKVyNYxS04_CA@mail.gmail.com>	<4FEB17B7.1010503@coppice.org>
	<CALn233p1wCgKLkhTwi9xRnW9-W-B2-fZVs1JULPTQJeo60+iMA@mail.gmail.com>
Message-ID: <145501cd5478$cc41b780$64c52680$@bizfocused.com>

I seem to be the only one not following this, so forgive me if this is a
stupid question.

Is the schema here that you "send a fax" from a fax machine on an ATA but
have Freeswitch intercept it with RXFAX and then try to send  it with TXFAX
using successively different setups until it goes through?

Thanks,
Sean

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch
Capper
Sent: Wednesday, June 27, 2012 10:40 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Successful faxing with spandsp / T38 and
stats

Sorry Steve for the confusion so:
75% T38,ECM success rate: 82%

does mean that 75% of the tries used the T38,ECM flags  and 82%
succeed.    Whats important to note is that while I turn ECM on
frequently it gets disabled as the remote faxing does not support it or the
operator suppresses it.  I actually had a hard time getting ECM to be
enabled to most places.  What I show in the flags are the options I pass to
FS not necessarily how the fax ends up going.  So a T38,ECM flag means I
tell FS to use ECM and to invite with T38 but obviously they could turn
around and negotiate no T38 and no ECM.
Also remember that T38/ECM is the first method tried and with most faxes
only taking 1.3 tries there is a good chance the first method
would always have a high percent.    For the stats to be super useful
it would be better if I chose the method at random unfortunately I care more
about getting the faxes through faster than providing the perfect FS faxing
stats.

It is sad that VOIP faxing has such a lower first try success rate, maybe
with some tweaks it could get above where I have it now for the first try,
but (atleast for my situation) getting to the 99%+ success rate overall for
outbound faxing was something I did not initially get the impression was
highly likely to achieve.  (And sorry these are all Outbound faxing stats I
did not include any inbound as those are less
interesting) .


Finally yes the Failure reasons stats I assume are not super useful
but as I had them I figured I would throw them up.   Keep in mind the
% for each failure reason is not how often is occurs just how often it
is the reason a fax fails.   The idea being if you are trying to do
faxing and seeing percentages hugely different than that you may be
doing something else wrong.      There certainly could be a better
provider for T38 faxing than flowroute, but I was already using them and as
they were quite successful in delivering faxes for me I did not do a huge
search for other providers.  If however anyone does have other
recommendations I do want to replace T38faxing as my backup provider.


~Mitch

On Wed, Jun 27, 2012 at 7:24 AM, Steve Underwood <steveu at coppice.org> wrote:

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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




From mitch.capper at gmail.com  Wed Jun 27 19:45:16 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 27 Jun 2012 08:45:16 -0700
Subject: [Freeswitch-users] Successful faxing with spandsp / T38 and
	stats
In-Reply-To: <145501cd5478$cc41b780$64c52680$@bizfocused.com>
References: <CALn233o+3wG0WUY1Qw3rfxdnzA7UCuv2DEobFmKVyNYxS04_CA@mail.gmail.com>
	<4FEB17B7.1010503@coppice.org>
	<CALn233p1wCgKLkhTwi9xRnW9-W-B2-fZVs1JULPTQJeo60+iMA@mail.gmail.com>
	<145501cd5478$cc41b780$64c52680$@bizfocused.com>
Message-ID: <CALn233q3QuHrBz2MrK0H=We4qybeHUgtE76icH1Snz_UNU+f-A@mail.gmail.com>

Sorry this is Freeswitch sending the fax using TXFAX.   What you
describe is possible and possibly desirable.  I did some early testing
using an actual fax machine over VOIP but given our situation scan to
fax was a much better solution than trying to work harder on getting
the actual fax machine to work.  By using Freeswitch as the faxing
agent it allows for better control of options and retry protocol.s

~Mitch

On Wed, Jun 27, 2012 at 8:23 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> I seem to be the only one not following this, so forgive me if this is a
> stupid question.
>
> Is the schema here that you "send a fax" from a fax machine on an ATA but
> have Freeswitch intercept it with RXFAX and then try to send ?it with TXFAX
> using successively different setups until it goes through?
>
> Thanks,
> Sean
>


From msc at freeswitch.org  Wed Jun 27 20:07:32 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Jun 2012 09:07:32 -0700
Subject: [Freeswitch-users] FreeSWITCH Conf Call Today
Message-ID: <CAKzWOxUKfqx-F8dkj84p4XpMW_Fiq4ragUFjR-ushAxZLeyyPA@mail.gmail.com>

Hello gang,

This week's conference call will be starting in less than an hour. Here's
the agenda page:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_27

Talk to you soon!

-Michael
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From sparfitt at thrupoint.com  Wed Jun 27 20:24:59 2012
From: sparfitt at thrupoint.com (Parfitt, Stephen)
Date: Wed, 27 Jun 2012 17:24:59 +0100
Subject: [Freeswitch-users] FreeSwitch Sending NOTIFY to incorrect port
 based on SUBSCRIBE request over UDP
Message-ID: <CAJ9PoEtk59T=f5veFEmUfd6H_=hyMf4oOQnMd79FVNE2k68jwQ@mail.gmail.com>

Hi All,

I've been doing some testing against FreeSwitch 1.2.0 and noticed what
looks like a bug.

root at Freeswitch> version

FreeSWITCH Version
1.2.0-rc2+git~20120616T193227Z~068586f28f+unclean~20120618T093505Z


192.168.8.238 = Freeswitch
192.168.8.168 = proxyhost.test.net


A Client sends the following SUBSCRIBE to freeswitch


SUBSCRIBE sip:2001 at proxyhost.test.net SIP/2.0
Call-ID: 2039516157894306129 at 192.168.8.168
Via: SIP/2.0/UDP 192.168.8.168:5070;branch=z9hG4bKC0A808A82E8652DB03
Via: SIP/2.0/UDP 192.168.8.168:5070;branch=z9hG4bKC0A808A82E8652DB11
To: sip:2001 at proxyhost.test.net
CSeq: 1 SUBSCRIBE
Contact: <sip:proxyhost.test.net:5060>
From: sip:2001 at proxyhost.test.net
;tag=1297284766*476741369860*990463165~476741369860~fmc.sar~
Content-Length: 0
Expires: 3600
Event: message-summary
Accept: application/simple-message-summary
Route: <sip:192.168.8.238;lr>
Max-Forwards: 69


FreeSwitch responds with:


SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.8.168:5070;branch=z9hG4bKC0A808A82E8652DB03
Via: SIP/2.0/UDP 192.168.8.168:5070;branch=z9hG4bKC0A808A82E8652DB11
From: <sip:2001 at proxyhost.test.net
>;tag=1297284766*476741369860*990463165~476741369860~fmc.sar~
To: <sip:2001 at proxyhost.test.net>;tag=9t8fu6DedmnQ
Call-ID: 2039516157894306129 at 192.168.8.168
CSeq: 1 SUBSCRIBE
Contact: <sip:2001 at 192.168.8.238:5060>
Expires: 3600
User-Agent:
FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120616T193227Z~068586f28f+unclean~20120618T093505Z
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Subscription-State: active;expires=3600
Content-Length: 0

So I would expect subsequent NOTIFY messages to be send to the address &
port in the contact: sip: proxyhost.test.net:5060 (contact header from
SUBSCRIBE)

However I do not see the NOTIFY requests arrive at sip:proxyhost.test.net on
port 5060, a wireshark trace on FreeSwitch says is sending them to a
different port, port 33406 which is the port Freeswitch received the
request on.


This is how the packet looks in WireShark:

Ethernet II, Src: 42:61:a8:ef:c3:de (42:61:a8:ef:c3:de), Dst:
96:cd:bb:a4:a4:2e (96:cd:bb:a4:a4:2e)
Internet Protocol Version 4, Src: 192.168.8.238 (192.168.8.238), Dst:
192.168.8.168 (192.168.8.168)
User Datagram Protocol, Src Port: sip (5060), Dst Port: 34406 (34406)
 **Would expect this to be 5060**
Request-Line: NOTIFY sip:proxyhost.test.net:5060 SIP/2.0

I thought this defect might be in the ball park:

http://jira.freeswitch.org/browse/FS-4014?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel


Anyone seen this?

To resolve I used TCP instead of UDP but wanted to flag it up to see if
anyone else had seen it


Regards

Steve P

 
--------------------
Note: The information contained in this message may be privileged and confidential 
and protected from disclosure. If the reader of this message is not the intended 
recipient, or an employee or agent responsible for delivering this message to the 
intended recipient, you are hereby notified that any dissemination, distribution or 
copying of this communication is strictly prohibited. If you have received this 
communication in error, please notify us immediately by replying to the message and 
deleting it from your computer. Thank you. Thrupoint, Inc.
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From Gary.Chen at lightyear.net  Wed Jun 27 20:28:42 2012
From: Gary.Chen at lightyear.net (Gary Chen)
Date: Wed, 27 Jun 2012 12:28:42 -0400
Subject: [Freeswitch-users] How to get last bridged call leg phone number?
Message-ID: <A6C2711F9706AF4992FFEC48EA19CA901740FC98CB@lyepkyexc.LYEPLYEXC.local>

FreeSWITCH Version 1.2.0-rc2+git~20120625T160048Z~690cae5de1

I am trying to setup hunt group using bridge app. I want it to ring the first available phone, if that person does not answer, then it should go to that person's voicemail. Here is my part of dialplan:
    <extension name="Local_Extension1">
      <condition field="destination_number" expression="(^(5554447579$)">
        <action application="set" data="ignore_early_media=true"/>
        <action application="bridge" data="{originate_continue_on_timeout=false}[ leg_timeout=10]sofia/internal/5554447580 at 192.168.1.22|[leg_timeout=10]sofia/internal/5554447578 at 192.168.1.22"/>
      </condition>
      <condition>
      Send call to voicemail .....
     </condition>
    </extension>

If no answer on 5554447580, I want it to goto 5554447580's voicemail. If no answer on 5554447578, I want it to goto 5554447578's voicemail. But I could not find the variable which contains the last bridged number information.
How do I get the last dialed number in bridge app?
I tried to  add variable in each leg like following, but it did not work:
        <action application="bridge" data="{originate_continue_on_timeout=false}[ mailbox=7580,leg_timeout=10]sofia/internal/5554447580 at 192.168.1.22|[mailbox=7578,leg_timeout=10]sofia/internal/5554447578 at 192.168.1.22"/>


Gary


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From msc at freeswitch.org  Wed Jun 27 20:45:30 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Jun 2012 09:45:30 -0700
Subject: [Freeswitch-users] How to get last bridged call leg phone
	number?
In-Reply-To: <A6C2711F9706AF4992FFEC48EA19CA901740FC98CB@lyepkyexc.LYEPLYEXC.local>
References: <A6C2711F9706AF4992FFEC48EA19CA901740FC98CB@lyepkyexc.LYEPLYEXC.local>
Message-ID: <CAKzWOxX=2ZheuETaCk+KqnfOzB3d81KNPYUVZvbuqyM2neRh=g@mail.gmail.com>

Shouldn't you be ignoring early media on these calls? I would assume that
it would never make it to the second leg because as soon as the first leg
sends back media then the bridge is "successful" whether 7580 answers or
not.

-MC

On Wed, Jun 27, 2012 at 9:28 AM, Gary Chen <Gary.Chen at lightyear.net> wrote:

> FreeSWITCH Version 1.2.0-rc2+git~20120625T160048Z~690cae5de1****
>
> ** **
>
> I am trying to setup hunt group using bridge app. I want it to ring the
> first available phone, if that person does not answer, then it should go to
> that person?s voicemail. Here is my part of dialplan:****
>
>     <extension name="Local_Extension1">****
>
>       <condition field="destination_number" expression="(^(5554447579$)">*
> ***
>
>         <action application="set" data="ignore_early_media=true"/>****
>
>         <action application="bridge"
> data="{originate_continue_on_timeout=false}[ leg_timeout=10]sofia/internal/
> 5554447580 at 192.168.1.22|[leg_timeout=10]sofia/internal/
> 5554447578 at 192.168.1.22"/>****
>
>       </condition>****
>
>       <condition>****
>
>       Send call to voicemail ?..****
>
>      </condition>****
>
>     </extension>****
>
> ** **
>
> If no answer on 5554447580, I want it to goto 5554447580?s voicemail. If
> no answer on 5554447578, I want it to goto 5554447578?s voicemail. But I
> could not find the variable which contains the last bridged number
> information.****
>
> How do I get the last dialed number in bridge app? ****
>
> I tried to  add variable in each leg like following, but it did not work:*
> ***
>
>         <action application="bridge"
> data="{originate_continue_on_timeout=false}[
> mailbox=7580,leg_timeout=10]sofia/internal/5554447580 at 192.168.1.22
> |[mailbox=7578,leg_timeout=10]sofia/internal/5554447578 at 192.168.1.22"/>***
> *
>
> ** **
>
> ** **
>
> Gary****
>
> ** **
>
>
>
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From Gary.Chen at lightyear.net  Wed Jun 27 21:26:18 2012
From: Gary.Chen at lightyear.net (Gary Chen)
Date: Wed, 27 Jun 2012 13:26:18 -0400
Subject: [Freeswitch-users] How to get last bridged call leg
	phone	number?
In-Reply-To: <CAKzWOxX=2ZheuETaCk+KqnfOzB3d81KNPYUVZvbuqyM2neRh=g@mail.gmail.com>
References: <A6C2711F9706AF4992FFEC48EA19CA901740FC98CB@lyepkyexc.LYEPLYEXC.local>
	<CAKzWOxX=2ZheuETaCk+KqnfOzB3d81KNPYUVZvbuqyM2neRh=g@mail.gmail.com>
Message-ID: <A6C2711F9706AF4992FFEC48EA19CA901740FC98CF@lyepkyexc.LYEPLYEXC.local>

In this setting, it will ring the first available phone, it the first phone is busy, it will ring the second one. That is what I want. The only problem is that I can not find number it just rang. I need to this number to access the correct voicemail box.

If I remove ignore_early_media=true, it will ring the first number forever and never be able to goto voicemail.

Gary

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Wednesday, June 27, 2012 12:46 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to get last bridged call leg phone number?

Shouldn't you be ignoring early media on these calls? I would assume that it would never make it to the second leg because as soon as the first leg sends back media then the bridge is "successful" whether 7580 answers or not.

-MC
On Wed, Jun 27, 2012 at 9:28 AM, Gary Chen <Gary.Chen at lightyear.net<mailto:Gary.Chen at lightyear.net>> wrote:
FreeSWITCH Version 1.2.0-rc2+git~20120625T160048Z~690cae5de1

I am trying to setup hunt group using bridge app. I want it to ring the first available phone, if that person does not answer, then it should go to that person's voicemail. Here is my part of dialplan:
    <extension name="Local_Extension1">
      <condition field="destination_number" expression="(^(5554447579$)">
        <action application="set" data="ignore_early_media=true"/>
        <action application="bridge" data="{originate_continue_on_timeout=false}[ leg_timeout=10]sofia/internal/5554447580 at 192.168.1.22<mailto:5554447580 at 192.168.1.22>|[leg_timeout=10]sofia/internal/5554447578 at 192.168.1.22<mailto:5554447578 at 192.168.1.22>"/>
      </condition>
      <condition>
      Send call to voicemail .....
     </condition>
    </extension>

If no answer on 5554447580, I want it to goto 5554447580's voicemail. If no answer on 5554447578, I want it to goto 5554447578's voicemail. But I could not find the variable which contains the last bridged number information.
How do I get the last dialed number in bridge app?
I tried to  add variable in each leg like following, but it did not work:
        <action application="bridge" data="{originate_continue_on_timeout=false}[ mailbox=7580,leg_timeout=10]sofia/internal/5554447580 at 192.168.1.22<mailto:5554447580 at 192.168.1.22>|[mailbox=7578,leg_timeout=10]sofia/internal/5554447578 at 192.168.1.22<mailto:5554447578 at 192.168.1.22>"/>


Gary



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From msc at freeswitch.org  Wed Jun 27 21:42:26 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Jun 2012 10:42:26 -0700
Subject: [Freeswitch-users] How to get last bridged call leg phone
	number?
In-Reply-To: <A6C2711F9706AF4992FFEC48EA19CA901740FC98CF@lyepkyexc.LYEPLYEXC.local>
References: <A6C2711F9706AF4992FFEC48EA19CA901740FC98CB@lyepkyexc.LYEPLYEXC.local>
	<CAKzWOxX=2ZheuETaCk+KqnfOzB3d81KNPYUVZvbuqyM2neRh=g@mail.gmail.com>
	<A6C2711F9706AF4992FFEC48EA19CA901740FC98CF@lyepkyexc.LYEPLYEXC.local>
Message-ID: <CAKzWOxUiDrigv0V7NfFqGXU=fsZUeG4kfzCYuLfaU0RY23CTcw@mail.gmail.com>

Ok, sounds good. My next question is this: what do you see in the "mailbox"
chan var that you added to the call legs? Are you actually looking in the B
leg CDR for this value?

-MC

On Wed, Jun 27, 2012 at 10:26 AM, Gary Chen <Gary.Chen at lightyear.net> wrote:

> In this setting, it will ring the first available phone, it the first
> phone is busy, it will ring the second one. That is what I want. The only
> problem is that I can not find number it just rang. I need to this number
> to access the correct voicemail box.****
>
> ** **
>
> If I remove ignore_early_media=true, it will ring the first number
> forever and never be able to goto voicemail.****
>
> ** **
>
> Gary****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Wednesday, June 27, 2012 12:46 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] How to get last bridged call leg phone
> number?****
>
> ** **
>
> Shouldn't you be ignoring early media on these calls? I would assume that
> it would never make it to the second leg because as soon as the first leg
> sends back media then the bridge is "successful" whether 7580 answers or
> not.
>
> -MC****
>
> On Wed, Jun 27, 2012 at 9:28 AM, Gary Chen <Gary.Chen at lightyear.net>
> wrote:****
>
> FreeSWITCH Version 1.2.0-rc2+git~20120625T160048Z~690cae5de1****
>
>  ****
>
> I am trying to setup hunt group using bridge app. I want it to ring the
> first available phone, if that person does not answer, then it should go to
> that person?s voicemail. Here is my part of dialplan:****
>
>     <extension name="Local_Extension1">****
>
>       <condition field="destination_number" expression="(^(5554447579$)">*
> ***
>
>         <action application="set" data="ignore_early_media=true"/>****
>
>         <action application="bridge"
> data="{originate_continue_on_timeout=false}[ leg_timeout=10]sofia/internal/
> 5554447580 at 192.168.1.22|[leg_timeout=10]sofia/internal/
> 5554447578 at 192.168.1.22"/>****
>
>       </condition>****
>
>       <condition>****
>
>       Send call to voicemail ?..****
>
>      </condition>****
>
>     </extension>****
>
>  ****
>
> If no answer on 5554447580, I want it to goto 5554447580?s voicemail. If
> no answer on 5554447578, I want it to goto 5554447578?s voicemail. But I
> could not find the variable which contains the last bridged number
> information.****
>
> How do I get the last dialed number in bridge app? ****
>
> I tried to  add variable in each leg like following, but it did not work:*
> ***
>
>         <action application="bridge"
> data="{originate_continue_on_timeout=false}[
> mailbox=7580,leg_timeout=10]sofia/internal/5554447580 at 192.168.1.22
> |[mailbox=7578,leg_timeout=10]sofia/internal/5554447578 at 192.168.1.22"/>***
> *
>
>  ****
>
>  ****
>
> Gary****
>
>  ****
>
>  ****
>
>
>
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From Gary.Chen at lightyear.net  Wed Jun 27 21:51:29 2012
From: Gary.Chen at lightyear.net (Gary Chen)
Date: Wed, 27 Jun 2012 13:51:29 -0400
Subject: [Freeswitch-users] How to get last bridged call leg
	phone	number?
In-Reply-To: <CAKzWOxUiDrigv0V7NfFqGXU=fsZUeG4kfzCYuLfaU0RY23CTcw@mail.gmail.com>
References: <A6C2711F9706AF4992FFEC48EA19CA901740FC98CB@lyepkyexc.LYEPLYEXC.local>
	<CAKzWOxX=2ZheuETaCk+KqnfOzB3d81KNPYUVZvbuqyM2neRh=g@mail.gmail.com>
	<A6C2711F9706AF4992FFEC48EA19CA901740FC98CF@lyepkyexc.LYEPLYEXC.local>
	<CAKzWOxUiDrigv0V7NfFqGXU=fsZUeG4kfzCYuLfaU0RY23CTcw@mail.gmail.com>
Message-ID: <A6C2711F9706AF4992FFEC48EA19CA901740FC98D7@lyepkyexc.LYEPLYEXC.local>

I tried to use app=log or app=info to look at the voicemail variable and it does not exist or never set it up.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Wednesday, June 27, 2012 1:42 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to get last bridged call leg phone number?

Ok, sounds good. My next question is this: what do you see in the "mailbox" chan var that you added to the call legs? Are you actually looking in the B leg CDR for this value?

-MC
On Wed, Jun 27, 2012 at 10:26 AM, Gary Chen <Gary.Chen at lightyear.net<mailto:Gary.Chen at lightyear.net>> wrote:
In this setting, it will ring the first available phone, it the first phone is busy, it will ring the second one. That is what I want. The only problem is that I can not find number it just rang. I need to this number to access the correct voicemail box.

If I remove ignore_early_media=true, it will ring the first number forever and never be able to goto voicemail.

Gary

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Wednesday, June 27, 2012 12:46 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to get last bridged call leg phone number?

Shouldn't you be ignoring early media on these calls? I would assume that it would never make it to the second leg because as soon as the first leg sends back media then the bridge is "successful" whether 7580 answers or not.

-MC
On Wed, Jun 27, 2012 at 9:28 AM, Gary Chen <Gary.Chen at lightyear.net<mailto:Gary.Chen at lightyear.net>> wrote:
FreeSWITCH Version 1.2.0-rc2+git~20120625T160048Z~690cae5de1

I am trying to setup hunt group using bridge app. I want it to ring the first available phone, if that person does not answer, then it should go to that person's voicemail. Here is my part of dialplan:
    <extension name="Local_Extension1">
      <condition field="destination_number" expression="(^(5554447579$)">
        <action application="set" data="ignore_early_media=true"/>
        <action application="bridge" data="{originate_continue_on_timeout=false}[ leg_timeout=10]sofia/internal/5554447580 at 192.168.1.22<mailto:5554447580 at 192.168.1.22>|[leg_timeout=10]sofia/internal/5554447578 at 192.168.1.22<mailto:5554447578 at 192.168.1.22>"/>
      </condition>
      <condition>
      Send call to voicemail .....
     </condition>
    </extension>

If no answer on 5554447580, I want it to goto 5554447580's voicemail. If no answer on 5554447578, I want it to goto 5554447578's voicemail. But I could not find the variable which contains the last bridged number information.
How do I get the last dialed number in bridge app?
I tried to  add variable in each leg like following, but it did not work:
        <action application="bridge" data="{originate_continue_on_timeout=false}[ mailbox=7580,leg_timeout=10]sofia/internal/5554447580 at 192.168.1.22<mailto:5554447580 at 192.168.1.22>|[mailbox=7578,leg_timeout=10]sofia/internal/5554447578 at 192.168.1.22<mailto:5554447578 at 192.168.1.22>"/>


Gary




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From msc at freeswitch.org  Wed Jun 27 22:34:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Jun 2012 11:34:27 -0700
Subject: [Freeswitch-users] How to get last bridged call leg phone
	number?
In-Reply-To: <A6C2711F9706AF4992FFEC48EA19CA901740FC98D7@lyepkyexc.LYEPLYEXC.local>
References: <A6C2711F9706AF4992FFEC48EA19CA901740FC98CB@lyepkyexc.LYEPLYEXC.local>
	<CAKzWOxX=2ZheuETaCk+KqnfOzB3d81KNPYUVZvbuqyM2neRh=g@mail.gmail.com>
	<A6C2711F9706AF4992FFEC48EA19CA901740FC98CF@lyepkyexc.LYEPLYEXC.local>
	<CAKzWOxUiDrigv0V7NfFqGXU=fsZUeG4kfzCYuLfaU0RY23CTcw@mail.gmail.com>
	<A6C2711F9706AF4992FFEC48EA19CA901740FC98D7@lyepkyexc.LYEPLYEXC.local>
Message-ID: <CAKzWOxVW-fVBfEj9x=nR0uuqiag=pUgbPKrGUfDL4i2D0D9iJA@mail.gmail.com>

Yeah, because you're probably doing that on the A leg and not the B leg.
The variable is only set on the B leg. At what point in the call flow do
you need the value? Do you need it after the call for CDR processing?
-MC

On Wed, Jun 27, 2012 at 10:51 AM, Gary Chen <Gary.Chen at lightyear.net> wrote:

> I tried to use app=log or app=info to look at the voicemail variable and
> it does not exist or never set it up.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Wednesday, June 27, 2012 1:42 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] How to get last bridged call leg phone
> number?****
>
> ** **
>
> Ok, sounds good. My next question is this: what do you see in the
> "mailbox" chan var that you added to the call legs? Are you actually
> looking in the B leg CDR for this value?
>
> -MC****
>
> On Wed, Jun 27, 2012 at 10:26 AM, Gary Chen <Gary.Chen at lightyear.net>
> wrote:****
>
> In this setting, it will ring the first available phone, it the first
> phone is busy, it will ring the second one. That is what I want. The only
> problem is that I can not find number it just rang. I need to this number
> to access the correct voicemail box.****
>
>  ****
>
> If I remove ignore_early_media=true, it will ring the first number
> forever and never be able to goto voicemail.****
>
>  ****
>
> Gary****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Wednesday, June 27, 2012 12:46 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] How to get last bridged call leg phone
> number?****
>
>  ****
>
> Shouldn't you be ignoring early media on these calls? I would assume that
> it would never make it to the second leg because as soon as the first leg
> sends back media then the bridge is "successful" whether 7580 answers or
> not.
>
> -MC****
>
> On Wed, Jun 27, 2012 at 9:28 AM, Gary Chen <Gary.Chen at lightyear.net>
> wrote:****
>
> FreeSWITCH Version 1.2.0-rc2+git~20120625T160048Z~690cae5de1****
>
>  ****
>
> I am trying to setup hunt group using bridge app. I want it to ring the
> first available phone, if that person does not answer, then it should go to
> that person?s voicemail. Here is my part of dialplan:****
>
>     <extension name="Local_Extension1">****
>
>       <condition field="destination_number" expression="(^(5554447579$)">*
> ***
>
>         <action application="set" data="ignore_early_media=true"/>****
>
>         <action application="bridge"
> data="{originate_continue_on_timeout=false}[ leg_timeout=10]sofia/internal/
> 5554447580 at 192.168.1.22|[leg_timeout=10]sofia/internal/
> 5554447578 at 192.168.1.22"/>****
>
>       </condition>****
>
>       <condition>****
>
>       Send call to voicemail ?..****
>
>      </condition>****
>
>     </extension>****
>
>  ****
>
> If no answer on 5554447580, I want it to goto 5554447580?s voicemail. If
> no answer on 5554447578, I want it to goto 5554447578?s voicemail. But I
> could not find the variable which contains the last bridged number
> information.****
>
> How do I get the last dialed number in bridge app? ****
>
> I tried to  add variable in each leg like following, but it did not work:*
> ***
>
>         <action application="bridge"
> data="{originate_continue_on_timeout=false}[
> mailbox=7580,leg_timeout=10]sofia/internal/5554447580 at 192.168.1.22
> |[mailbox=7578,leg_timeout=10]sofia/internal/5554447578 at 192.168.1.22"/>***
> *
>
>  ****
>
>  ****
>
> Gary****
>
>  ****
>
>
>
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From avi at avimarcus.net  Wed Jun 27 22:40:15 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 27 Jun 2012 21:40:15 +0300
Subject: [Freeswitch-users] implementation of freeswitch in javascript
	/node
In-Reply-To: <1953A95D11D1ED42B24E60E7A59C5AB411B468@venus.Noble.local>
References: <1953A95D11D1ED42B24E60E7A59C5AB411B451@venus.Noble.local>
	<CALNZuwmHbJFe4u_5fCB4CowHynHqVHXqB=5OsZzw3SVwTktn7g@mail.gmail.com>
	<1953A95D11D1ED42B24E60E7A59C5AB411B468@venus.Noble.local>
Message-ID: <CALNZuwn_QZMQTc7ckgobxTqB7PbNi3TLNi1jcQK7cG73xwiAMw@mail.gmail.com>

All I've seen is now listed here:
http://wiki.freeswitch.org/wiki/Mod_event_socket#Javascript_.2F_Node.js_library

But much of it is in coffeescript which is a little different.

-Avi


On Thu, Jun 21, 2012 at 1:28 PM, michal <michal at noblesolutions.co.uk> wrote:

>  Do you have some links to what you have seen?****
>
> ** **
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
> *Sent:* 21 June 2012 10:43
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] implementation of freeswitch in
> javascript /node****
>
> ** **
>
> Me too. I've not seen many examples for node using ESL.. samples (and
> explanation?!) of a some code doing some useful things would be great!****
>
>
> ****
>
> -Avi****
>
> ** **
>
> On Thu, Jun 21, 2012 at 11:27 AM, michal <michal at noblesolutions.co.uk>
> wrote:****
>
> Hi there!****
>
>  ****
>
> Can someone recommend some resources on freeswitch and javascript/node?
> Ideally I would like a tutorial or guide on how it all works together (how
> to talk to FS)****
>
>  ****
>
>  ****
>
>  ****
>
> Thanks in advance! ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Gary.Chen at lightyear.net  Wed Jun 27 22:48:25 2012
From: Gary.Chen at lightyear.net (Gary Chen)
Date: Wed, 27 Jun 2012 14:48:25 -0400
Subject: [Freeswitch-users] How to get last bridged call leg
	phone	number?
In-Reply-To: <CAKzWOxVW-fVBfEj9x=nR0uuqiag=pUgbPKrGUfDL4i2D0D9iJA@mail.gmail.com>
References: <A6C2711F9706AF4992FFEC48EA19CA901740FC98CB@lyepkyexc.LYEPLYEXC.local>
	<CAKzWOxX=2ZheuETaCk+KqnfOzB3d81KNPYUVZvbuqyM2neRh=g@mail.gmail.com>
	<A6C2711F9706AF4992FFEC48EA19CA901740FC98CF@lyepkyexc.LYEPLYEXC.local>
	<CAKzWOxUiDrigv0V7NfFqGXU=fsZUeG4kfzCYuLfaU0RY23CTcw@mail.gmail.com>
	<A6C2711F9706AF4992FFEC48EA19CA901740FC98D7@lyepkyexc.LYEPLYEXC.local>
	<CAKzWOxVW-fVBfEj9x=nR0uuqiag=pUgbPKrGUfDL4i2D0D9iJA@mail.gmail.com>
Message-ID: <A6C2711F9706AF4992FFEC48EA19CA901740FC98DF@lyepkyexc.LYEPLYEXC.local>

I need it for sending the call to voicemail, if no answer. The problem is that I can not tell which phone it rang and no answer so I do not know which voicemail box should I send the call to?
 Can I export variable from B leg to A leg in bridge app?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Wednesday, June 27, 2012 2:34 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to get last bridged call leg phone number?

Yeah, because you're probably doing that on the A leg and not the B leg. The variable is only set on the B leg. At what point in the call flow do you need the value? Do you need it after the call for CDR processing?
-MC
On Wed, Jun 27, 2012 at 10:51 AM, Gary Chen <Gary.Chen at lightyear.net<mailto:Gary.Chen at lightyear.net>> wrote:
I tried to use app=log or app=info to look at the voicemail variable and it does not exist or never set it up.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Wednesday, June 27, 2012 1:42 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to get last bridged call leg phone number?

Ok, sounds good. My next question is this: what do you see in the "mailbox" chan var that you added to the call legs? Are you actually looking in the B leg CDR for this value?

-MC
On Wed, Jun 27, 2012 at 10:26 AM, Gary Chen <Gary.Chen at lightyear.net<mailto:Gary.Chen at lightyear.net>> wrote:
In this setting, it will ring the first available phone, it the first phone is busy, it will ring the second one. That is what I want. The only problem is that I can not find number it just rang. I need to this number to access the correct voicemail box.

If I remove ignore_early_media=true, it will ring the first number forever and never be able to goto voicemail.

Gary

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Wednesday, June 27, 2012 12:46 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to get last bridged call leg phone number?

Shouldn't you be ignoring early media on these calls? I would assume that it would never make it to the second leg because as soon as the first leg sends back media then the bridge is "successful" whether 7580 answers or not.

-MC
On Wed, Jun 27, 2012 at 9:28 AM, Gary Chen <Gary.Chen at lightyear.net<mailto:Gary.Chen at lightyear.net>> wrote:
FreeSWITCH Version 1.2.0-rc2+git~20120625T160048Z~690cae5de1

I am trying to setup hunt group using bridge app. I want it to ring the first available phone, if that person does not answer, then it should go to that person's voicemail. Here is my part of dialplan:
    <extension name="Local_Extension1">
      <condition field="destination_number" expression="(^(5554447579$)">
        <action application="set" data="ignore_early_media=true"/>
        <action application="bridge" data="{originate_continue_on_timeout=false}[ leg_timeout=10]sofia/internal/5554447580 at 192.168.1.22<mailto:5554447580 at 192.168.1.22>|[leg_timeout=10]sofia/internal/5554447578 at 192.168.1.22<mailto:5554447578 at 192.168.1.22>"/>
      </condition>
      <condition>
      Send call to voicemail .....
     </condition>
    </extension>

If no answer on 5554447580, I want it to goto 5554447580's voicemail. If no answer on 5554447578, I want it to goto 5554447578's voicemail. But I could not find the variable which contains the last bridged number information.
How do I get the last dialed number in bridge app?
I tried to  add variable in each leg like following, but it did not work:
        <action application="bridge" data="{originate_continue_on_timeout=false}[ mailbox=7580,leg_timeout=10]sofia/internal/5554447580 at 192.168.1.22<mailto:5554447580 at 192.168.1.22>|[mailbox=7578,leg_timeout=10]sofia/internal/5554447578 at 192.168.1.22<mailto:5554447578 at 192.168.1.22>"/>


Gary



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From saami_mh at ymail.com  Wed Jun 27 23:05:12 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Wed, 27 Jun 2012 12:05:12 -0700 (PDT)
Subject: [Freeswitch-users] multiple registerations coudln't worked
	correctly!
Message-ID: <1340823912.36352.YahooMailNeo@web120101.mail.ne1.yahoo.com>



hi,
i have set?<param name="multiple-registrations" value="false"/> on internal.xml ?and switch.conf.xml then
?restart freeswitch
but I ?could register 2 sip phones using the?same?extension number at the same time!

but i want to only one extension could be register using the same extensions?
plz help
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From peter.olsson at visionutveckling.se  Wed Jun 27 23:16:55 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 27 Jun 2012 19:16:55 +0000
Subject: [Freeswitch-users] multiple registerations coudln't
	worked	correctly!
In-Reply-To: <1340823912.36352.YahooMailNeo@web120101.mail.ne1.yahoo.com>
References: <1340823912.36352.YahooMailNeo@web120101.mail.ne1.yahoo.com>
Message-ID: <1FFF97C269757C458224B7C895F35F15125995@cantor.std.visionutv.se>

If they end up with only one registration within FS it's just normal. Since the second registration will overwrite the first.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Samira Mh [saami_mh at ymail.com]
Skickat: den 27 juni 2012 21:05
Till: Free SWITCH Users Help
?mne: [Freeswitch-users] multiple registerations coudln't worked correctly!


hi,
i have set <param name="multiple-registrations" value="false"/> on internal.xml  and switch.conf.xml then
 restart freeswitch
but I  could register 2 sip phones using the same extension number at the same time!
but i want to only one extension could be register using the same extensions
plz help
!DSPAM:4feb57b332761635117794!


From saami_mh at ymail.com  Wed Jun 27 23:26:02 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Wed, 27 Jun 2012 12:26:02 -0700 (PDT)
Subject: [Freeswitch-users] how can run fs_cli commands  in lua script?
Message-ID: <1340825162.71942.YahooMailNeo@web120102.mail.ne1.yahoo.com>

hi all,
i want to run the following command in lua script that is located on (/usr/local/freeswitch/scripts/lua1.lua)
sofia_count_reg?1001-Q0ErXNX1RuYwPdctyuGeiA at public.gmane.org

i ned to use the output of the above command
how can i do that?
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From msc at freeswitch.org  Wed Jun 27 23:29:07 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Jun 2012 12:29:07 -0700
Subject: [Freeswitch-users] how can run fs_cli commands in lua script?
In-Reply-To: <1340825162.71942.YahooMailNeo@web120102.mail.ne1.yahoo.com>
References: <1340825162.71942.YahooMailNeo@web120102.mail.ne1.yahoo.com>
Message-ID: <CAKzWOxUca_i2bndXHbh8uWPQpBwXU+TnB=hx8Hnzg1vjGWVpGA@mail.gmail.com>

What is the calling process?
-MC

On Wed, Jun 27, 2012 at 12:26 PM, Samira Mh <saami_mh at ymail.com> wrote:

> hi all,
> i want to run the following command in lua script that is located on
> (/usr/local/freeswitch/scripts/lua1.lua)
> sofia_count_reg 1001-Q0ErXNX1RuYwPdctyuGeiA at public.gmane.org
> i ned to use the output of the above command
> how can i do that?
>
>
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From avi at avimarcus.net  Wed Jun 27 23:31:41 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 27 Jun 2012 22:31:41 +0300
Subject: [Freeswitch-users] how can run fs_cli commands in lua script?
In-Reply-To: <1340825162.71942.YahooMailNeo@web120102.mail.ne1.yahoo.com>
References: <1340825162.71942.YahooMailNeo@web120102.mail.ne1.yahoo.com>
Message-ID: <CALNZuwm6tvD=gC6pT_H5GKtmqnYce0UpG48HO3gDkz0ZH39OVA@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Lua#For_making_API_calls
or
The page links to:
http://wiki.freeswitch.org/wiki/Make_API_calls_directly_from_Lua_code

Not sure of the difference between api:executeString() and api:execute() if
someone could please document the difference...

-Avi
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From peter.olsson at visionutveckling.se  Wed Jun 27 23:35:49 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 27 Jun 2012 19:35:49 +0000
Subject: [Freeswitch-users] how can run fs_cli commands  in lua script?
In-Reply-To: <1340825162.71942.YahooMailNeo@web120102.mail.ne1.yahoo.com>
References: <1340825162.71942.YahooMailNeo@web120102.mail.ne1.yahoo.com>
Message-ID: <1FFF97C269757C458224B7C895F35F15125A08@cantor.std.visionutv.se>

Please have a look here: http://wiki.freeswitch.org/wiki/Make_API_calls_directly_from_Lua_code

There is a description for how to execute API commands from within lua.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Samira Mh [saami_mh at ymail.com]
Skickat: den 27 juni 2012 21:26
Till: Free SWITCH Users Help
?mne: [Freeswitch-users] how can run fs_cli commands in lua script?

hi all,
i want to run the following command in lua script that is located on (/usr/local/freeswitch/scripts/lua1.lua)
sofia_count_reg 1001-Q0ErXNX1RuYwPdctyuGeiA at public.gmane.org<mailto:1001-Q0ErXNX1RuYwPdctyuGeiA at public.gmane.org>
i ned to use the output of the above command
how can i do that?
!DSPAM:4feb5c4d32761687116813!


From drk at drkngs.net  Wed Jun 27 23:51:48 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Wed, 27 Jun 2012 12:51:48 -0700
Subject: [Freeswitch-users]
	=?iso-8859-1?q?how_can_run_fs=5Fcli_commands_i?=
	=?iso-8859-1?q?n_lua_script=3F?=
In-Reply-To: <CALNZuwm6tvD=gC6pT_H5GKtmqnYce0UpG48HO3gDkz0ZH39OVA@mail.gmail.com>
Message-ID: <20120627195148.69758d6c@mail.tritonwest.net>

Execute, takes 2 args, the API command and the arguments seprate, where executeString takes it as a single argument.  
   
--Dave
      _____  

  From: Avi Marcus [mailto:avi at avimarcus.net]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Wed, 27 Jun 2012 12:31:41 -0700
Subject: Re: [Freeswitch-users] how can run fs_cli commands in lua script?

  
http://wiki.freeswitch.org/wiki/Lua#For_making_API_calls
or  
The page links to:  
http://wiki.freeswitch.org/wiki/Make_API_calls_directly_from_Lua_code  

  
Not sure of the difference between api:executeString() and api:execute() if someone could please document the difference...  

  
-Avi
    

      
   
 
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From jmsucasas at me.com  Thu Jun 28 00:04:55 2012
From: jmsucasas at me.com (Jose Miguel Sucasas Mejuto)
Date: Wed, 27 Jun 2012 22:04:55 +0200
Subject: [Freeswitch-users] Always PAUSE Jitterbuffer
Message-ID: <8FCBF56C-DB66-4BF6-96FE-CACBB2F6D9D1@me.com>

Hello,

	I have installed Freeswitch from GIT (currenty I have installed 1.2.0-rc2+git~20120625T203158Z~7dc9a9cacc version).  Because some users connects from 3G network using a smartphone,  I try to activate JitterBuffer and Packet Loss Concealing in both directions (in a call with U1 and U2,  I try to activate 90ms jitterbuffer in audio flow U1->U2 and another 90ms jitterbuffer in audio flow U2->U1).  To achieve this,  I have this in dialplan (default.xml file):

	<action application="cng_plc"/>
	<!-- A-leg JitterBuffer -->
	<action application="set" data="jitterbuffer_msec=90"/>
	<action application="set" data="sip_jitter_buffer_during_bridge=true"/>
	<!-- A-leg JitterBuffer debug, doesn't work -->
	<action application="jitterbuffer" data="debug:${uuid}"/>
	<!-- In bridge application, I try to activate B-leg JitterBuffer -->
	<action application="bridge" data="{jitterbuffer_msec=90,absolute_codec_string='SILK at 16000h,iLBC at 30i,G729,AMR,AMR-WB,H263,H264'}user/${dialed_extension}@${domain_name}"/>

	User 1002 is on the same network as Freeswitch box,  connected by ethernet line,  and user 1001 is on 3G,  when user 1002 calls to user 1001 (both of them uses iLBC30 codec),  on fs_cli it seems that JitterBuffer is activated (in bold green)

2012-06-27 20:55:14.929462 [NOTICE] sofia.c:5930 Ring-Ready sofia/internal/sip:1001 at 10.10.57.82:50203!
2012-06-27 20:55:14.929462 [INFO] switch_ivr_originate.c:1156 Sending early media
2012-06-27 20:55:14.929462 [INFO] switch_nat.c:590 NAT port mapping disabled
2012-06-27 20:55:14.929462 [INFO] switch_nat.c:590 NAT port mapping disabled
2012-06-27 20:55:14.929462 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [proceeding][180]
2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3269 AUDIO RTP [sofia/internal/1002 at sukmad.dyndns-home.com:58180] 79.150.184.57 port 11154 -> 192.168.0.15 port 10000 codec: 98 ms: 30
2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:1861 Starting timer [soft] 240 bytes per 30ms
2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:2207 RTCP send rate is: 5000 and packet rate is: 30000 Remote Port: 10001
2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 192.168.0.15:10001
2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to 90ms (3 frames)
2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send payload to 101
2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive payload to 101
2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3566 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Set rtp dtmf delay to 40
2012-06-27 20:55:14.929462 [DEBUG] mod_sofia.c:2606 Ring SDP:
v=0
o=FreeSWITCH 1340812160 1340812161 IN IP4 79.150.184.57
s=FreeSWITCH
c=IN IP4 79.150.184.57
t=0 0
m=audio 11154 RTP/AVP 98 101
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:30
a=sendrecv

2012-06-27 20:55:14.929462 [NOTICE] mod_sofia.c:2609 Pre-Answer sofia/internal/1002 at sukmad.dyndns-home.com:58180!
2012-06-27 20:55:14.929462 [DEBUG] switch_channel.c:3042 (sofia/internal/1002 at sukmad.dyndns-home.com:58180) Callstate Change RINGING -> EARLY
2012-06-27 20:55:14.929462 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:14.929462 [DEBUG] switch_ivr_originate.c:1205 Raw Codec Activation Success L16 at 8000hz 1 channel 30ms
2012-06-27 20:55:14.929462 [DEBUG] switch_core_codec.c:219 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Push codec L16:70
2012-06-27 20:55:14.929462 [DEBUG] switch_ivr_originate.c:1268 Play Ringback Tone [%(2000,4000,440,480)]
2012-06-27 20:55:14.929462 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:14.929462 [DEBUG] sofia.c:5831 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 skipping state [early][183]
2012-06-27 20:55:14.939460 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
2012-06-27 20:55:14.939460 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
2012-06-27 20:55:14.939460 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 entering state [proceeding][180]
2012-06-27 20:55:14.939460 [NOTICE] sofia.c:5930 Ring-Ready sofia/internal/sip:1001 at 31.4.238.246:22268!
2012-06-27 20:55:15.029459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
2012-06-27 20:55:15.029459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
2012-06-27 20:55:15.029459 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 entering state [proceeding][180]
2012-06-27 20:55:15.069457 [DEBUG] switch_rtp.c:3527 Correct ip/port confirmed.

	When user 1001 answer the call, JitterBuffer is paused (in bold red):

2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [completing][200]
2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5849 Remote SDP:
v=0
o=- 3549812097 3549812099 IN IP4 10.10.57.82
s=cpc_med
c=IN IP4 10.10.57.82
b=AS:226
t=0 0
m=audio 22010 RTP/AVP 99 101
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 103
c=IN IP4 10.10.57.82
b=AS:200000
a=rtpmap:103 H264/90000

2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3919 Deciding whether to pass zrtp-hash between legs
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [ready][200]
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3919 Deciding whether to pass zrtp-hash between legs
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [iLBC:99:8000:30:0]/[SILK:119:16000:20:30000]
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5148 Set 2833 dtmf send payload to 101
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [iLBC:99:8000:30:0]/[iLBC:97:8000:30:13330]
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3020 Set Codec sofia/internal/sip:1001 at 10.10.57.82:50203 iLBC/8000 30 ms 240 samples 13330 bits
2012-06-27 20:55:26.079458 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:1001 at 10.10.57.82:50203 Original read codec set to iLBC:97
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5148 Set 2833 dtmf send payload to 101
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3269 AUDIO RTP [sofia/internal/sip:1001 at 10.10.57.82:50203] 79.150.184.57 port 10388 -> 10.10.57.82 port 22010 codec: 99 ms: 30
2012-06-27 20:55:26.079458 [DEBUG] switch_rtp.c:1861 Starting timer [soft] 240 bytes per 30ms
2012-06-27 20:55:26.089461 [DEBUG] switch_rtp.c:2207 RTCP send rate is: 5000 and packet rate is: 30000 Remote Port: 22011
2012-06-27 20:55:26.089461 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 10.10.57.82:22011
2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to 90ms (3 frames)
2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send payload to 101
2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive payload to 101
2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3566 sofia/internal/sip:1001 at 10.10.57.82:50203 Set rtp dtmf delay to 40
2012-06-27 20:55:26.089461 [DEBUG] switch_channel.c:3301 (sofia/internal/sip:1001 at 10.10.57.82:50203) Callstate Change RINGING -> ACTIVE
2012-06-27 20:55:26.089461 [DEBUG] switch_channel.c:3313 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.089461 [NOTICE] sofia.c:6573 Channel [sofia/internal/sip:1001 at 10.10.57.82:50203] has been answered
2012-06-27 20:55:26.109459 [DEBUG] switch_core_codec.c:244 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Restore previous codec iLBC:97.
2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2899 (sofia/internal/sip:1001 at 192.168.0.11:50199) Callstate Change RINGING -> HANGUP
2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3257 Hangup sofia/internal/sip:1001 at 192.168.0.11:50199 [CS_CONSUME_MEDIA] [LOSE_RACE]
2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2922 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [KILL]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_HANGUP
2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2899 (sofia/internal/sip:1001 at 31.4.238.246:22268) Callstate Change RINGING -> HANGUP
2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3257 Hangup sofia/internal/sip:1001 at 31.4.238.246:22268 [CS_CONSUME_MEDIA] [LOSE_RACE]
2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2922 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [KILL]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 192.168.0.11:50199) State HANGUP
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001 at 192.168.0.11:50199 hanging up, cause: LOSE_RACE
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_HANGUP
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:527 Sending CANCEL to sofia/internal/sip:1001 at 192.168.0.11:50199
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard HANGUP, cause: LOSE_RACE
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 192.168.0.11:50199) State HANGUP going to sleep
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/sip:1001 at 192.168.0.11:50199) State Change CS_HANGUP -> CS_REPORTING
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_REPORTING
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 192.168.0.11:50199) State REPORTING
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard REPORTING, cause: LOSE_RACE
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 192.168.0.11:50199) State REPORTING going to sleep
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 31.4.238.246:22268) State HANGUP
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 hanging up, cause: LOSE_RACE
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at 192.168.0.11:50199) State Change CS_REPORTING -> CS_DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1429 Session 15 (sofia/internal/sip:1001 at 192.168.0.11:50199) Locked, Waiting on external entities
2012-06-27 20:55:26.109459 [INFO] switch_nat.c:590 NAT port mapping disabled
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:527 Sending CANCEL to sofia/internal/sip:1001 at 31.4.238.246:22268
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:754 Local SDP sofia/internal/1002 at sukmad.dyndns-home.com:58180:
v=0
o=FreeSWITCH 1340812160 1340812162 IN IP4 79.150.184.57
s=FreeSWITCH
c=IN IP4 79.150.184.57
t=0 0
m=audio 11154 RTP/AVP 98 101
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:30
a=sendrecv
m=video 10766 RTP/AVP 34 103
a=rtpmap:34 H263/90000
a=rtpmap:103 H264/90000

2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard HANGUP, cause: LOSE_RACE
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 31.4.238.246:22268) State HANGUP going to sleep
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/sip:1001 at 31.4.238.246:22268) State Change CS_HANGUP -> CS_REPORTING
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_REPORTING
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:3301 (sofia/internal/1002 at sukmad.dyndns-home.com:58180) Callstate Change EARLY -> ACTIVE
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 31.4.238.246:22268) State REPORTING
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard REPORTING, cause: LOSE_RACE
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 31.4.238.246:22268) State REPORTING going to sleep
2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3284 Channel [sofia/internal/1002 at sukmad.dyndns-home.com:58180] has been answered
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at 31.4.238.246:22268) State Change CS_REPORTING -> CS_DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1429 Session 17 (sofia/internal/sip:1001 at 31.4.238.246:22268) Locked, Waiting on external entities
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] sofia.c:5838 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 entering state [completed][200]
2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_originate.c:3341 Originate Resulted in Success: [sofia/internal/sip:1001 at 10.10.57.82:50203]
2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1447 Session 15 (sofia/internal/sip:1001 at 192.168.0.11:50199) Ended
2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1447 Session 17 (sofia/internal/sip:1001 at 31.4.238.246:22268) Ended
2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/sip:1001 at 192.168.0.11:50199 [CS_DESTROY]
2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/sip:1001 at 31.4.238.246:22268 [CS_DESTROY]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/sip:1001 at 31.4.238.246:22268) Callstate Change HANGUP -> DOWN
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/sip:1001 at 192.168.0.11:50199) Callstate Change HANGUP -> DOWN
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 31.4.238.246:22268) State DESTROY
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001 at 31.4.238.246:22268 SOFIA DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 31.4.238.246:22268) State DESTROY going to sleep
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 192.168.0.11:50199) State DESTROY
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001 at 192.168.0.11:50199 SOFIA DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 192.168.0.11:50199) State DESTROY going to sleep
2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_originate.c:3341 Originate Resulted in Success: [sofia/internal/sip:1001 at 10.10.57.82:50203]
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:1603 sofia/internal/sip:1001 at 10.10.57.82:50203 PAUSE Jitterbuffer
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_bridge.c:1359 (sofia/internal/sip:1001 at 10.10.57.82:50203) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 10.10.57.82:50203) Running State Change CS_EXCHANGE_MEDIA
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:443 (sofia/internal/sip:1001 at 10.10.57.82:50203) State EXCHANGE_MEDIA
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:582 SOFIA EXCHANGE_MEDIA
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] sofia.c:5838 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 entering state [ready][200]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.469460 [INFO] switch_rtp.c:3510 Auto Changing port from 10.10.57.82:22010 to 31.4.238.246:22267
2012-06-27 20:55:26.469460 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 31.4.238.246:22268
2012-06-27 20:55:38.009460 [DEBUG] sofia_reg.c:371 Registered sip.netelip.com


	I try a lot of test and all of them have the same issue,  Why JitterBuffer is paused?.  Is it possible to force JitterBuffer?


Thanks in advance,

Regards



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From peter.olsson at visionutveckling.se  Thu Jun 28 00:17:11 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 27 Jun 2012 20:17:11 +0000
Subject: [Freeswitch-users] Always PAUSE Jitterbuffer
In-Reply-To: <8FCBF56C-DB66-4BF6-96FE-CACBB2F6D9D1@me.com>
References: <8FCBF56C-DB66-4BF6-96FE-CACBB2F6D9D1@me.com>
Message-ID: <D32BDE57-EE45-4961-AD62-5651575C1572@visionutveckling.se>

Maybe this is what you are looking for? I think FS usually wants the endpoints to handle this during a bridge, setting this var should force it to stay enabled.

http://wiki.freeswitch.org/wiki/Variable_sip_jitter_buffer_during_bridge

/Peter

27 jun 2012 kl. 22:11 skrev "Jose Miguel Sucasas Mejuto" <jmsucasas at me.com<mailto:jmsucasas at me.com>>:

Hello,

I have installed Freeswitch from GIT (currenty I have installed 1.2.0-rc2+git~20120625T203158Z~7dc9a9cacc version).  Because some users connects from 3G network using a smartphone,  I try to activate JitterBuffer and Packet Loss Concealing in both directions (in a call with U1 and U2,  I try to activate 90ms jitterbuffer in audio flow U1->U2 and another 90ms jitterbuffer in audio flow U2->U1).  To achieve this,  I have this in dialplan (default.xml file):

<action application="cng_plc"/>
<!-- A-leg JitterBuffer -->
<action application="set" data="jitterbuffer_msec=90"/>
<action application="set" data="sip_jitter_buffer_during_bridge=true"/>
<!-- A-leg JitterBuffer debug, doesn't work -->
<action application="jitterbuffer" data="debug:${uuid}"/>
<!-- In bridge application, I try to activate B-leg JitterBuffer -->
<action application="bridge" data="{jitterbuffer_msec=90,absolute_codec_string='SILK at 16000h,iLBC at 30i,G729,AMR,AMR-WB,H263,H264'}user/${dialed_extension}@${domain_name}"/>

User 1002 is on the same network as Freeswitch box,  connected by ethernet line,  and user 1001 is on 3G,  when user 1002 calls to user 1001 (both of them uses iLBC30 codec),  on fs_cli it seems that JitterBuffer is activated (in bold green)

2012-06-27 20:55:14.929462 [NOTICE] sofia.c:5930 Ring-Ready sofia/internal/sip:1001 at 10.10.57.82:50203!
2012-06-27 20:55:14.929462 [INFO] switch_ivr_originate.c:1156 Sending early media
2012-06-27 20:55:14.929462 [INFO] switch_nat.c:590 NAT port mapping disabled
2012-06-27 20:55:14.929462 [INFO] switch_nat.c:590 NAT port mapping disabled
2012-06-27 20:55:14.929462 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [proceeding][180]
2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3269 AUDIO RTP [sofia/internal/1002 at sukmad.dyndns-home.com:58180] 79.150.184.57 port 11154 -> 192.168.0.15 port 10000 codec: 98 ms: 30
2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:1861 Starting timer [soft] 240 bytes per 30ms
2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:2207 RTCP send rate is: 5000 and packet rate is: 30000 Remote Port: 10001
2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 192.168.0.15:10001
2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to 90ms (3 frames)
2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send payload to 101
2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive payload to 101
2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3566 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Set rtp dtmf delay to 40
2012-06-27 20:55:14.929462 [DEBUG] mod_sofia.c:2606 Ring SDP:
v=0
o=FreeSWITCH 1340812160 1340812161 IN IP4 79.150.184.57
s=FreeSWITCH
c=IN IP4 79.150.184.57
t=0 0
m=audio 11154 RTP/AVP 98 101
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:30
a=sendrecv

2012-06-27 20:55:14.929462 [NOTICE] mod_sofia.c:2609 Pre-Answer sofia/internal/1002 at sukmad.dyndns-home.com:58180!
2012-06-27 20:55:14.929462 [DEBUG] switch_channel.c:3042 (sofia/internal/1002 at sukmad.dyndns-home.com:58180) Callstate Change RINGING -> EARLY
2012-06-27 20:55:14.929462 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:14.929462 [DEBUG] switch_ivr_originate.c:1205 Raw Codec Activation Success L16 at 8000hz 1 channel 30ms
2012-06-27 20:55:14.929462 [DEBUG] switch_core_codec.c:219 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Push codec L16:70
2012-06-27 20:55:14.929462 [DEBUG] switch_ivr_originate.c:1268 Play Ringback Tone [%(2000,4000,440,480)]
2012-06-27 20:55:14.929462 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:14.929462 [DEBUG] sofia.c:5831 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 skipping state [early][183]
2012-06-27 20:55:14.939460 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
2012-06-27 20:55:14.939460 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
2012-06-27 20:55:14.939460 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 entering state [proceeding][180]
2012-06-27 20:55:14.939460 [NOTICE] sofia.c:5930 Ring-Ready sofia/internal/sip:1001 at 31.4.238.246:22268!
2012-06-27 20:55:15.029459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
2012-06-27 20:55:15.029459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
2012-06-27 20:55:15.029459 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 entering state [proceeding][180]
2012-06-27 20:55:15.069457 [DEBUG] switch_rtp.c:3527 Correct ip/port confirmed.

When user 1001 answer the call, JitterBuffer is paused (in bold red):

2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [completing][200]
2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5849 Remote SDP:
v=0
o=- 3549812097 3549812099 IN IP4 10.10.57.82
s=cpc_med
c=IN IP4 10.10.57.82
b=AS:226
t=0 0
m=audio 22010 RTP/AVP 99 101
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 103
c=IN IP4 10.10.57.82
b=AS:200000
a=rtpmap:103 H264/90000

2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3919 Deciding whether to pass zrtp-hash between legs
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [ready][200]
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3919 Deciding whether to pass zrtp-hash between legs
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [iLBC:99:8000:30:0]/[SILK:119:16000:20:30000]
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5148 Set 2833 dtmf send payload to 101
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [iLBC:99:8000:30:0]/[iLBC:97:8000:30:13330]
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3020 Set Codec sofia/internal/sip:1001 at 10.10.57.82:50203 iLBC/8000 30 ms 240 samples 13330 bits
2012-06-27 20:55:26.079458 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:1001 at 10.10.57.82:50203 Original read codec set to iLBC:97
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5148 Set 2833 dtmf send payload to 101
2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3269 AUDIO RTP [sofia/internal/sip:1001 at 10.10.57.82:50203] 79.150.184.57 port 10388 -> 10.10.57.82 port 22010 codec: 99 ms: 30
2012-06-27 20:55:26.079458 [DEBUG] switch_rtp.c:1861 Starting timer [soft] 240 bytes per 30ms
2012-06-27 20:55:26.089461 [DEBUG] switch_rtp.c:2207 RTCP send rate is: 5000 and packet rate is: 30000 Remote Port: 22011
2012-06-27 20:55:26.089461 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 10.10.57.82:22011
2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to 90ms (3 frames)
2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send payload to 101
2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive payload to 101
2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3566 sofia/internal/sip:1001 at 10.10.57.82:50203 Set rtp dtmf delay to 40
2012-06-27 20:55:26.089461 [DEBUG] switch_channel.c:3301 (sofia/internal/sip:1001 at 10.10.57.82:50203) Callstate Change RINGING -> ACTIVE
2012-06-27 20:55:26.089461 [DEBUG] switch_channel.c:3313 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.089461 [NOTICE] sofia.c:6573 Channel [sofia/internal/sip:1001 at 10.10.57.82:50203] has been answered
2012-06-27 20:55:26.109459 [DEBUG] switch_core_codec.c:244 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Restore previous codec iLBC:97.
2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2899 (sofia/internal/sip:1001 at 192.168.0.11:50199) Callstate Change RINGING -> HANGUP
2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3257 Hangup sofia/internal/sip:1001 at 192.168.0.11:50199 [CS_CONSUME_MEDIA] [LOSE_RACE]
2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2922 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [KILL]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_HANGUP
2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2899 (sofia/internal/sip:1001 at 31.4.238.246:22268) Callstate Change RINGING -> HANGUP
2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3257 Hangup sofia/internal/sip:1001 at 31.4.238.246:22268 [CS_CONSUME_MEDIA] [LOSE_RACE]
2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2922 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [KILL]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 192.168.0.11:50199) State HANGUP
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001 at 192.168.0.11:50199 hanging up, cause: LOSE_RACE
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_HANGUP
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:527 Sending CANCEL to sofia/internal/sip:1001 at 192.168.0.11:50199
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard HANGUP, cause: LOSE_RACE
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 192.168.0.11:50199) State HANGUP going to sleep
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/sip:1001 at 192.168.0.11:50199) State Change CS_HANGUP -> CS_REPORTING
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_REPORTING
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 192.168.0.11:50199) State REPORTING
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard REPORTING, cause: LOSE_RACE
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 192.168.0.11:50199) State REPORTING going to sleep
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 31.4.238.246:22268) State HANGUP
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 hanging up, cause: LOSE_RACE
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at 192.168.0.11:50199) State Change CS_REPORTING -> CS_DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1429 Session 15 (sofia/internal/sip:1001 at 192.168.0.11:50199) Locked, Waiting on external entities
2012-06-27 20:55:26.109459 [INFO] switch_nat.c:590 NAT port mapping disabled
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:527 Sending CANCEL to sofia/internal/sip:1001 at 31.4.238.246:22268
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:754 Local SDP sofia/internal/1002 at sukmad.dyndns-home.com:58180:
v=0
o=FreeSWITCH 1340812160 1340812162 IN IP4 79.150.184.57
s=FreeSWITCH
c=IN IP4 79.150.184.57
t=0 0
m=audio 11154 RTP/AVP 98 101
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:30
a=sendrecv
m=video 10766 RTP/AVP 34 103
a=rtpmap:34 H263/90000
a=rtpmap:103 H264/90000

2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard HANGUP, cause: LOSE_RACE
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 31.4.238.246:22268) State HANGUP going to sleep
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/sip:1001 at 31.4.238.246:22268) State Change CS_HANGUP -> CS_REPORTING
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_REPORTING
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:3301 (sofia/internal/1002 at sukmad.dyndns-home.com:58180) Callstate Change EARLY -> ACTIVE
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 31.4.238.246:22268) State REPORTING
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard REPORTING, cause: LOSE_RACE
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 31.4.238.246:22268) State REPORTING going to sleep
2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3284 Channel [sofia/internal/1002 at sukmad.dyndns-home.com:58180] has been answered
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at 31.4.238.246:22268) State Change CS_REPORTING -> CS_DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1429 Session 17 (sofia/internal/sip:1001 at 31.4.238.246:22268) Locked, Waiting on external entities
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] sofia.c:5838 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 entering state [completed][200]
2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_originate.c:3341 Originate Resulted in Success: [sofia/internal/sip:1001 at 10.10.57.82:50203]
2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1447 Session 15 (sofia/internal/sip:1001 at 192.168.0.11:50199) Ended
2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1447 Session 17 (sofia/internal/sip:1001 at 31.4.238.246:22268) Ended
2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/sip:1001 at 192.168.0.11:50199 [CS_DESTROY]
2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/sip:1001 at 31.4.238.246:22268 [CS_DESTROY]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/sip:1001 at 31.4.238.246:22268) Callstate Change HANGUP -> DOWN
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/sip:1001 at 192.168.0.11:50199) Callstate Change HANGUP -> DOWN
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 31.4.238.246:22268) State DESTROY
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001 at 31.4.238.246:22268 SOFIA DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 31.4.238.246:22268) State DESTROY going to sleep
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 192.168.0.11:50199) State DESTROY
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001 at 192.168.0.11:50199 SOFIA DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard DESTROY
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 192.168.0.11:50199) State DESTROY going to sleep
2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_originate.c:3341 Originate Resulted in Success: [sofia/internal/sip:1001 at 10.10.57.82:50203]
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:1603 sofia/internal/sip:1001 at 10.10.57.82:50203 PAUSE Jitterbuffer
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_bridge.c:1359 (sofia/internal/sip:1001 at 10.10.57.82:50203) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 10.10.57.82:50203) Running State Change CS_EXCHANGE_MEDIA
2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:443 (sofia/internal/sip:1001 at 10.10.57.82:50203) State EXCHANGE_MEDIA
2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:582 SOFIA EXCHANGE_MEDIA
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] sofia.c:5838 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 entering state [ready][200]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
2012-06-27 20:55:26.469460 [INFO] switch_rtp.c:3510 Auto Changing port from 10.10.57.82:22010 to 31.4.238.246:22267
2012-06-27 20:55:26.469460 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 31.4.238.246:22268
2012-06-27 20:55:38.009460 [DEBUG] sofia_reg.c:371 Registered sip.netelip.com<http://sip.netelip.com>


I try a lot of test and all of them have the same issue,  Why JitterBuffer is paused?.  Is it possible to force JitterBuffer?


Thanks in advance,

Regards



!DSPAM:4feb659c32769194324459!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


!DSPAM:4feb659c32769194324459!


From admin at blindi.net  Thu Jun 28 05:12:32 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Thu, 28 Jun 2012 03:12:32 +0200 (CEST)
Subject: [Freeswitch-users] opensource project voicechat for freeswitch
In-Reply-To: <CALQZz0zLpRBnFu3jce07Ycq=mn_OnAGrVFZgm-ibY_3eGg93Xw@mail.gmail.com>
References: <CAPu0Fr2qSJgm54i9bWvqQZrfmqJvV8i9EaWy7aeG9Pw9BtNq9w@mail.gmail.com>
	<2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com>
	<1338558318.79806.YahooMailNeo@web120405.mail.ne1.yahoo.com>
	<Pine.LNX.4.64.1206041700001.22234@tomback.antagus.de>
	<CALQZz0zLpRBnFu3jce07Ycq=mn_OnAGrVFZgm-ibY_3eGg93Xw@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1206280309390.30414@tomback.antagus.de>

Hi Sanath,
I reply on your messages.
But i don.t become a reaction.
You mean you have a voicemailsystem in lua.
But i don.t become a needed help from your.

Is your message a joke??? I don.t understand it.

Thanks.


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From saami_mh at ymail.com  Thu Jun 28 08:11:18 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Wed, 27 Jun 2012 21:11:18 -0700 (PDT)
Subject: [Freeswitch-users] multiple registerations coudln't worked
	correctly!
In-Reply-To: <1FFF97C269757C458224B7C895F35F15125995@cantor.std.visionutv.se>
References: <1340823912.36352.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<1FFF97C269757C458224B7C895F35F15125995@cantor.std.visionutv.se>
Message-ID: <1340856678.61841.YahooMailNeo@web120102.mail.ne1.yahoo.com>

is there any solutions for?the second registration couldn't overwrite the first one registeration ?



________________________________
 From: Peter Olsson <peter.olsson at visionutveckling.se>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Wednesday, June 27, 2012 11:46 PM
Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!
 
If they end up with only one registration within FS it's just normal. Since the second registration will overwrite the first.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Samira Mh [saami_mh at ymail.com]
Skickat: den 27 juni 2012 21:05
Till: Free SWITCH Users Help
?mne: [Freeswitch-users] multiple registerations coudln't worked correctly!


hi,
i have set <param name="multiple-registrations" value="false"/> on internal.xml? and switch.conf.xml then
restart freeswitch
but I? could register 2 sip phones using the same extension number at the same time!
but i want to only one extension could be register using the same extensions
plz help
!DSPAM:4feb57b332761635117794!

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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From peter.olsson at visionutveckling.se  Thu Jun 28 09:14:39 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 28 Jun 2012 05:14:39 +0000
Subject: [Freeswitch-users] multiple registerations coudln't
	worked	correctly!
In-Reply-To: <1340856678.61841.YahooMailNeo@web120102.mail.ne1.yahoo.com>
References: <1340823912.36352.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<1FFF97C269757C458224B7C895F35F15125995@cantor.std.visionutv.se>,
	<1340856678.61841.YahooMailNeo@web120102.mail.ne1.yahoo.com>
Message-ID: <692ADD6A-960C-4DA5-A49C-ACB6D2FB7556@visionutveckling.se>

The option is to enable multiple-registrations, which will make both registrations work simultaneously.

Why is this even a problem, what problem are you trying to solve?

/Peter

28 jun 2012 kl. 06:21 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com>>:

is there any solutions for the second registration couldn't overwrite the first one registeration ?


________________________________
From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Wednesday, June 27, 2012 11:46 PM
Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!

If they end up with only one registration within FS it's just normal. Since the second registration will overwrite the first.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] f?r Samira Mh [saami_mh at ymail.com<mailto:saami_mh at ymail.com>]
Skickat: den 27 juni 2012 21:05
Till: Free SWITCH Users Help
?mne: [Freeswitch-users] multiple registerations coudln't worked correctly!


hi,
i have set <param name="multiple-registrations" value="false"/> on internal.xml  and switch.conf.xml then
restart freeswitch
but I  could register 2 sip phones using the same extension number at the same time!
but i want to only one extension could be register using the same extensions
plz help


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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/>


!DSPAM:4febd86932764854317000!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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!DSPAM:4febd86932764854317000!


From saami_mh at ymail.com  Thu Jun 28 12:04:30 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Thu, 28 Jun 2012 01:04:30 -0700 (PDT)
Subject: [Freeswitch-users] multiple registerations coudln't worked
	correctly!
In-Reply-To: <692ADD6A-960C-4DA5-A49C-ACB6D2FB7556@visionutveckling.se>
References: <1340823912.36352.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<1FFF97C269757C458224B7C895F35F15125995@cantor.std.visionutv.se>,
	<1340856678.61841.YahooMailNeo@web120102.mail.ne1.yahoo.com>
	<692ADD6A-960C-4DA5-A49C-ACB6D2FB7556@visionutveckling.se>
Message-ID: <1340870670.48416.YahooMailNeo@web120105.mail.ne1.yahoo.com>

i have implemented VODSL service using freeswitch,and create accounts for user on the server,but i don't want the userA use the account of userB,
i want to restrict each user only use her/his account not able to use each other account that don't belong to his/her...
also i don't want to restrict this using ip addresses ,want to limit using account


________________________________
 From: Peter Olsson <peter.olsson at visionutveckling.se>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Thursday, June 28, 2012 9:44 AM
Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!
 
The option is to enable multiple-registrations, which will make both registrations work simultaneously.

Why is this even a problem, what problem are you trying to solve?

/Peter

28 jun 2012 kl. 06:21 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com>>:

is there any solutions for the second registration couldn't overwrite the first one registeration ?


________________________________
From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Wednesday, June 27, 2012 11:46 PM
Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!

If they end up with only one registration within FS it's just normal. Since the second registration will overwrite the first.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] f?r Samira Mh [saami_mh at ymail.com<mailto:saami_mh at ymail.com>]
Skickat: den 27 juni 2012 21:05
Till: Free SWITCH Users Help
?mne: [Freeswitch-users] multiple registerations coudln't worked correctly!


hi,
i have set <param name="multiple-registrations" value="false"/> on internal.xml? and switch.conf.xml then
restart freeswitch
but I? could register 2 sip phones using the same extension number at the same time!
but i want to only one extension could be register using the same extensions
plz help


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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/>


!DSPAM:4febd86932764854317000!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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http://www.freeswitch.org


!DSPAM:4febd86932764854317000!

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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From andrew at cassidywebservices.co.uk  Thu Jun 28 13:09:42 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 28 Jun 2012 10:09:42 +0100
Subject: [Freeswitch-users] Looking up user data in dialplan
Message-ID: <CAMbW9oKg=1U3YUUAJMwfWP+H-2DQ=+Lf6W=ZR6Jd7pNv74nXTg@mail.gmail.com>

Hi All,

Right, I have a very basic application I'm building around voicemail. Each
user has 2 attrs that I need to search by. user_exists works fine, but
doesn't return me any data other then the fact the user exists, and
user_data only searches the id attr.

Is there any way I can do that in just the dialplan with what is already
available?

In the most simple form I need the user id for a number_alias.

-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From leon at scarlet-internet.nl  Thu Jun 28 17:36:20 2012
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Thu, 28 Jun 2012 15:36:20 +0200
Subject: [Freeswitch-users] Looking up user data in dialplan
In-Reply-To: <CAMbW9oKg=1U3YUUAJMwfWP+H-2DQ=+Lf6W=ZR6Jd7pNv74nXTg@mail.gmail.com>
References: <CAMbW9oKg=1U3YUUAJMwfWP+H-2DQ=+Lf6W=ZR6Jd7pNv74nXTg@mail.gmail.com>
Message-ID: <004ED0CB-4E46-4F01-B9F8-F82456D4F7F9@scarlet-internet.nl>

You can use the set_user dp app to set all user variables as channel variables. You can also supply a prefix string to keep these new variables separate from wha was already set.

See here: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user

regards,

Leon


On Jun 28, 2012, at 11:09, Andrew Cassidy wrote:

> Hi All,
> 
> Right, I have a very basic application I'm building around voicemail. Each user has 2 attrs that I need to search by. user_exists works fine, but doesn't return me any data other then the fact the user exists, and user_data only searches the id attr.
> 
> Is there any way I can do that in just the dialplan with what is already available?
> 
> In the most simple form I need the user id for a number_alias.
> 
> -- 
> Andrew Cassidy BSc (Hons) MBCS SSCA
> Managing Director
> 
> 
> T 03300 100 960  F 03300 100 961
> E andrew at cassidywebservices.co.uk
> W www.cassidywebservices.co.uk
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From peter.olsson at visionutveckling.se  Thu Jun 28 18:21:15 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 28 Jun 2012 14:21:15 +0000
Subject: [Freeswitch-users] multiple registerations coudln't
	worked	correctly!
Message-ID: <1FFF97C269757C458224B7C895F35F151268CD@cantor.std.visionutv.se>

I still don't understand.. :) How could UserA use UserB's account? Aren't you using different passwords for the user?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh
Skickat: den 28 juni 2012 10:05
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!

i have implemented VODSL service using freeswitch,and create accounts for user on the server,but i don't want the userA use the account of userB,
i want to restrict each user only use her/his account not able to use each other account that don't belong to his/her...
also i don't want to restrict this using ip addresses ,want to limit using account

________________________________
From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Thursday, June 28, 2012 9:44 AM
Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!

The option is to enable multiple-registrations, which will make both registrations work simultaneously.

Why is this even a problem, what problem are you trying to solve?

/Peter

28 jun 2012 kl. 06:21 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com><mailto:saami_mh at ymail.com<mailto:saami_mh at ymail.com>>>:

is there any solutions for the second registration couldn't overwrite the first one registeration ?


________________________________
From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se><mailto:peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>>
To: FreeSWITCH Users Help <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>>>
Sent: Wednesday, June 27, 2012 11:46 PM
Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!

If they end up with only one registration within FS it's just normal. Since the second registration will overwrite the first.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org><mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org><mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>>] f?r Samira Mh [saami_mh at ymail.com<mailto:saami_mh at ymail.com><mailto:saami_mh at ymail.com<mailto:saami_mh at ymail.com>>]
Skickat: den 27 juni 2012 21:05
Till: Free SWITCH Users Help
?mne: [Freeswitch-users] multiple registerations coudln't worked correctly!


hi,
i have set <param name="multiple-registrations" value="false"/> on internal.xml  and switch.conf.xml then
restart freeswitch
but I  could register 2 sip phones using the same extension number at the same time!
but i want to only one extension could be register using the same extensions
plz help


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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<http://www.freeswitch.org/><http://www.freeswitch.org/>



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>


</>

Official FreeSWITCH Sites
http://www.freeswitch.org<http://www.freeswitch.org/>
http://wiki.freeswitch.org<http://wiki.freeswitch.org/>
http://www.cluecon.com<http://www.cluecon.com/>

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
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!DSPAM:4febd86932764854317000!

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>


</>

Official FreeSWITCH Sites
http://www.freeswitch.org<http://www.freeswitch.org/>
http://wiki.freeswitch.org<http://wiki.freeswitch.org/>
http://www.cluecon.com<http://www.cluecon.com/>

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
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From shahzad.bhatti at g-r-v.com  Thu Jun 28 18:40:32 2012
From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti)
Date: Thu, 28 Jun 2012 19:40:32 +0500
Subject: [Freeswitch-users] Handling Multi Call in Lua
Message-ID: <CAN+NqUYWiq2+nodqeS583v0Cxh3dLYNJbTH86-S1y4uXe3Q0Yw@mail.gmail.com>

i want to originate 5 calls after one second sleep using lua and get the
response using playandgetdigits(), but scipt originate call one by one.

please guide me how i can call 5 numbers at the same time,

here is my code

===============================================================================================================================
*-- TestCall.lua*

function PlayMessageAndGetResponse(message)
        count    = 1
        invalid  = 'say:flite:slt:nvalid input , please enter again'
        audio_for_prompt = 'say:flite:slt:press 1 for YES, Press 2 for No,
Press 3 for Record Message and Press 4 for Replay'
        while session:ready() == true and count <= 3 do
                session:sleep(250)

                session:streamFile(message)

                session:sleep(250)

                digits = session:playAndGetDigits(1, 1, 3, 5, "#",
audio_for_prompt, invalid, "^(1|2|3|4)$")

                if digits == "" then
                        return 0
                elseif digits == "4" then
                        count = count + 1
                else
                        return digits
                end
        end
        return 0
end

number = argv[1]
message= argv[2]

greeting    = "Welcome! You have a Mesage"
bye_message = "Bye bye."
response    = 0
        freeswitch.consoleLog("info", "The Number "..number..", message
"..message.."\n")
        session =
freeswitch.Session("{ignore_early_media=true,origination_caller_id_number=321}user/"..number.."")

        session:setAutoHangup(false)
        session:set_tts_parms("flite", "slt")

        dispoA = "None"
        while(session:ready() and dispoA ~= "ANSWER") do
                dispoA = session:getVariable("endpoint_disposition")
                freeswitch.consoleLog("INFO","Leg A disposition is '" ..
dispoA .. "'\n")
                os.execute("sleep 1")
        end -- legA while

        if session:ready() == true and dispoA == "ANSWER" then
                uuid = session:getVariable("uuid")

                session:speak(greeting)

                response = PlayMessageAndGetResponse(message)

                session:speak(bye_message)
                session:hangup()
        end
        cause = session:hangupCause()
        session:destroy()

        freeswitch.consoleLog('info',"Number "..number..", respond
"..response.." hangup cause is "..cause.."\n")

===============================================================================================================================
*-- campaign.lua*

count = 1000
number = {}
for i = 1, 5 do
        number[i] = count
        count = count + 1
end

for i = 1, 5 do
        freeswitch.consoleLog("warning", "Calling " ..number[i].. "\n")
        api = freeswitch.API();
        reply = api:executeString("luarun TestCall.lua "..number[i].."
".."test_msg.wav")
        os.execute("sleep 1")
end


===============================================================================================================================

Regards

Shahzad Bhatti
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From gmaruzz at gmail.com  Thu Jun 28 18:46:12 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 28 Jun 2012 16:46:12 +0200
Subject: [Freeswitch-users] skypopen.ko insertion issue
In-Reply-To: <1340767034403-7580303.post@n2.nabble.com>
References: <1340767034403-7580303.post@n2.nabble.com>
Message-ID: <CALXCt0po-OC0ECQzUOx+g2GfM6n9=hoUO+x4W4NP7OEpYssVzQ@mail.gmail.com>

On Wed, Jun 27, 2012 at 5:17 AM, kevinfuhn <kevinfuhn at gmail.com> wrote:
> when running insmod ./skypopen.ko following the FS wiki instruction, got
> error as below:
>
> insmod: error inserting './skypopen.ko': -1 Device or resource busy
>
> The other info, like this
> #dmesg
> [ 2232.486152] skypopen OSS driver loading (www.freeswitch.org)
> [ 2232.486162] skypopen OSS driver: can't get major 14
>
> This has been stopping me from successfully integrating skype with FS for
> quite a while. Can anyone out there help me out?

You have posted this in Jira too, and you has been told to follow the
wiki page step by step. In the while, the wiki page has been updated
to reflect your exact configuration too, and the process has been
retested for your configuration too (Ubuntu 12.04 LTS).

There were no problems.

So... Please follow the wiki page :).

-giovanni


>
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/skypopen-ko-insertion-issue-tp7580303.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 julian.pawlowski at gmail.com  Thu Jun 28 16:21:04 2012
From: julian.pawlowski at gmail.com (Julian Pawlowski)
Date: Thu, 28 Jun 2012 14:21:04 +0200
Subject: [Freeswitch-users] Building Debian packages for FS and 3rd-party
	libs
Message-ID: <CADFnKK89BWnh+Zi6VV8fEy2bFrd7fFyuB_4amgNgw=2En7ppOg@mail.gmail.com>

Hi folks,

I'm currently working on the system integration part of the
GemeinschaftPBX version 5.0 (see github.com/jpawlowski/GBE). As
compilation of FreeSWITCH and all dependent 3rd party libraries takes
major time I would like to start using Debian packages which I can
prepare once and reuse during the build phase of the appliance ISO.

I understand your own Debian repository is not ready yet (or at least
it's not public) so I wanted to implement a job on our build system
(running Jenkins CI) to have a continuous build of FS packages based
on changes in the FS Git.
I started by reading documents in /freeswitch/debian/, especially
README.source but as I am new to create Debian packages I can't work
around this error when I try those command mentioned in the readme:

---------
freeswitch_1.2~rc3~n20120628T140917Z.orig.tar.xz does not exist,
creating from 'master'
Building upstream tarball with compression 'xz -1v'
  100 %        29.1 MiB / 143.4 MiB = 0.203    15 MiB/s       0:09
git-pbuilder ['-b', '-us', '-uc'] []
Building for distribution squeeze
Base directory /var/cache/pbuilder/base-squeeze.cow does not exist
git-pbuilder returned 1
Couldn't run 'git-pbuilder -b -us -uc'
---------

I wasn't able to find anything useful regarding missing
base-squeeze.cow file for pbuilder and would appreciate any assistance
here.

In addition, as I plan to have a full set of available modules for FS
in our appliance distribution: Are there already Debian packages
available for the needed 3rd-party modules like all the stuff for
mod_freetdm (especially Sangoma parts)? Just wanted to ask before I
reinvent the wheel ;-)


Cheers,
Julian


From simon.gucas.ac at gmail.com  Thu Jun 28 18:00:34 2012
From: simon.gucas.ac at gmail.com (simon jin)
Date: Thu, 28 Jun 2012 22:00:34 +0800
Subject: [Freeswitch-users] about FS symbols
Message-ID: <CACd2BXpo46ttNorcvttF_P6p3UY5nrF=Prb0v-NP8grJabs8OA@mail.gmail.com>

HI everyone

i have compiled the FS with  CFLAGS="-g -ggdb -finstrument-functions"
and MOD_CFLAGS="-g
-ggdb -finstrument-functions", and it works done.
but the problem comes when i use addr2line command to translate some
function address to its name.
This is the detail:
$ addr2line -e freeswitch -f -a 0x2a955d3f8a
0x0000002a955d3f8a
??
??:0

Note: 0x2a955d3f8a is some function's address

I think that addr2line can NOT find freeswitch's symbols, but where it is,
anyone any advice? thanks for your reply!

yours simon
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From julian.pawlowski at gmail.com  Thu Jun 28 18:10:55 2012
From: julian.pawlowski at gmail.com (Julian Pawlowski)
Date: Thu, 28 Jun 2012 16:10:55 +0200
Subject: [Freeswitch-users] Building Debian packages for FS and
	3rd-party libs
In-Reply-To: <CADFnKK89BWnh+Zi6VV8fEy2bFrd7fFyuB_4amgNgw=2En7ppOg@mail.gmail.com>
References: <CADFnKK89BWnh+Zi6VV8fEy2bFrd7fFyuB_4amgNgw=2En7ppOg@mail.gmail.com>
Message-ID: <CADFnKK_e=gGXvLOm0A+qOgw67NGn1yMvjPRKYR=1w2SiRjgqrw@mail.gmail.com>

I'm sorry. As it is always - as soon as you have asked somebody else
you will surely find an answer yourself a few minutes later...

I got the Debian package build working.
For those who are interested, this is a slightly adjusted version of
the build example script from README.source:

------
#!/bin/bash -ex
#

# settings
#
distro=squeeze

# Install basic Debian maintainer packages
apt-get -y --force-yes install -qq build-essential git-buildpackage
udev autoconf automake autotools-dev debhelper dh-make devscripts
fakeroot file gfortran git gnupg gpc lintian patch patchutils pbuilder
perl python python xutils-dev

# Install FreeSWITCH specific build packages
apt-get -y --force-yes install -qq debhelper automake autoconf libtool
dpkg-dev gcc g++ libc6-dev make wget pkg-config libssl-dev
unixodbc-dev libncurses5-dev libjpeg62-dev python-dev erlang-dev
doxygen uuid-dev libexpat1-dev libgdbm-dev libdb-dev bison zlib1g-dev
ladspa-sdk libogg-dev libasound2-dev libx11-dev libpq-dev libsnmp-dev
default-jdk gcj-jdk libperl-dev libyaml-dev

[ ! -d /var/cache/pbuilder/base-${distro}.cow ] && DIST=${distro}
git-pbuilder create
[ ! -L /dev/fd ] && ln -s /proc/self/fd /dev/fd
[ ! -d ./freeswitch ] && git clone git://git.freeswitch.org/freeswitch
freeswitch


cd freeswitch

ver="$(cat build/next-release.txt | sed -e 's/-/~/g')~n$(date
+%Y%m%dT%H%M%SZ)-1~${distro}+1"
git clean -fdx && git reset --hard HEAD
./build/set-fs-version.sh "$ver"
git add configure.in && git commit -m "bump to custom v$ver"
(cd debian && ./bootstrap.sh -c $distro)
dch -b -m -v "$ver" --force-distribution -D "$suite" "Custom build."
git-buildpackage -b -us -uc \
  --git-verbose \
  --git-pbuilder --git-dist=$distro \
  --git-compression-level=1v --git-compression=xz
git reset --hard HEAD^

cd -
----------

Regards,
Julian


On Thu, Jun 28, 2012 at 2:21 PM, Julian Pawlowski
<julian.pawlowski at gmail.com> wrote:
> Hi folks,
>
> I'm currently working on the system integration part of the
> GemeinschaftPBX version 5.0 (see github.com/jpawlowski/GBE). As
> compilation of FreeSWITCH and all dependent 3rd party libraries takes
> major time I would like to start using Debian packages which I can
> prepare once and reuse during the build phase of the appliance ISO.
>
> I understand your own Debian repository is not ready yet (or at least
> it's not public) so I wanted to implement a job on our build system
> (running Jenkins CI) to have a continuous build of FS packages based
> on changes in the FS Git.
> I started by reading documents in /freeswitch/debian/, especially
> README.source but as I am new to create Debian packages I can't work
> around this error when I try those command mentioned in the readme:
>
> ---------
> freeswitch_1.2~rc3~n20120628T140917Z.orig.tar.xz does not exist,
> creating from 'master'
> Building upstream tarball with compression 'xz -1v'
> ?100 % ? ? ? ?29.1 MiB / 143.4 MiB = 0.203 ? ?15 MiB/s ? ? ? 0:09
> git-pbuilder ['-b', '-us', '-uc'] []
> Building for distribution squeeze
> Base directory /var/cache/pbuilder/base-squeeze.cow does not exist
> git-pbuilder returned 1
> Couldn't run 'git-pbuilder -b -us -uc'
> ---------
>
> I wasn't able to find anything useful regarding missing
> base-squeeze.cow file for pbuilder and would appreciate any assistance
> here.
>
> In addition, as I plan to have a full set of available modules for FS
> in our appliance distribution: Are there already Debian packages
> available for the needed 3rd-party modules like all the stuff for
> mod_freetdm (especially Sangoma parts)? Just wanted to ask before I
> reinvent the wheel ;-)
>
>
> Cheers,
> Julian


From ramesh_mind at yahoo.com  Thu Jun 28 18:21:48 2012
From: ramesh_mind at yahoo.com (ramesh)
Date: Thu, 28 Jun 2012 07:21:48 -0700 (PDT)
Subject: [Freeswitch-users] Voice delay in normal calls
Message-ID: <1340893308741-7580349.post@n2.nabble.com>

 Hi,

During an outbound or inbound calls from freeswitch , their a voice dealy of
about 4 to 5secs  and also we are facing delay issue in rtmp calls too. we
are hardly trying to figure the reason for this.

Any Help or Suggestions would be really helpful!.

Thanks,
Ramesh 

--
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From saami_mh at ymail.com  Thu Jun 28 19:21:53 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Thu, 28 Jun 2012 08:21:53 -0700 (PDT)
Subject: [Freeswitch-users] multiple registerations coudln't worked
	correctly!
In-Reply-To: <1FFF97C269757C458224B7C895F35F151268CD@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F151268CD@cantor.std.visionutv.se>
Message-ID: <1340896913.16633.YahooMailNeo@web120101.mail.ne1.yahoo.com>

suppose we have account with the following information:

extension=1000
secret=12345
callerid=userA
--------------------------
now the user(B,C,D ..) -NOT IMPORTANT who?- stolen ?the account informations of userA and want to login using userA account

while the userA is now connect to server,but at the same time userB register on sip phone using userA iformations(extesion=1000,secret=12345,...)
and by set multi_registering=false the userA is not connected now, and userB is now connected because only one user able to register
but i don't want this situation appear,i want to set if userA is now connected the other user couldn't connect using same informations?
thanksss




________________________________
 From: Peter Olsson <peter.olsson at visionutveckling.se>
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org> 
Sent: Thursday, June 28, 2012 6:51 PM
Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!
 

 
I still don?t understand.. :) How could UserA use UserB?s account? Aren?t you using different passwords for the user?
?
/Peter
?
Fr?n:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh
Skickat: den 28 juni 2012 10:05
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!
?
i have implemented VODSL service using freeswitch,and create accounts for user on the server,but i don't want the userA use the account of userB,
i want to restrict each user only use her/his account not able to use each other account that don't belong to his/her...
also i don't want to restrict this using ip addresses ,want to limit using account
?

________________________________
 
From:Peter Olsson <peter.olsson at visionutveckling.se>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Thursday, June 28, 2012 9:44 AM
Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!

The option is to enable multiple-registrations, which will make both registrations work simultaneously.

Why is this even a problem, what problem are you trying to solve?

/Peter

28 jun 2012 kl. 06:21 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com>>:

is there any solutions for the second registration couldn't overwrite the first one registeration ?


________________________________
From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Wednesday, June 27, 2012 11:46 PM
Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!

If they end up with only one registration within FS it's just normal. Since the second registration will overwrite the first.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] f?r Samira Mh [saami_mh at ymail.com<mailto:saami_mh at ymail.com>]
Skickat: den 27 juni 2012 21:05
Till: Free SWITCH Users Help
?mne: [Freeswitch-users] multiple registerations coudln't worked correctly!


hi,
i have set <param name="multiple-registrations" value="false"/> on internal.xml? and switch.conf.xml then
restart freeswitch
but I? could register 2 sip phones using the same extension number at the same time!
but i want to only one extension could be register using the same extensions
plz help


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




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http://www.freeswitch.org
http://wiki.freeswitch.org
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consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




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http://wiki.freeswitch.org
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Join Us At ClueCon - Aug 7-9, 2012

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_________________________________________________________________________
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http://www.freeswitchsolutions.com




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http://wiki.freeswitch.org
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Join Us At ClueCon - Aug 7-9, 2012

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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_________________________________________________________________________
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http://www.freeswitchsolutions.com




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http://wiki.freeswitch.org
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Join Us At ClueCon - Aug 7-9, 2012

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From ramesh_mind at yahoo.com  Thu Jun 28 19:38:01 2012
From: ramesh_mind at yahoo.com (ramesh)
Date: Thu, 28 Jun 2012 08:38:01 -0700 (PDT)
Subject: [Freeswitch-users] Error while trying to install freeswitch latest
	source from git
Message-ID: <1340897881933-7580357.post@n2.nabble.com>

Hi All, 

We are to trying to install latest freeswitch source into our new server ,
while trying to make the source , we are getting error and unable to proceed
futhur .

Any Idea? to make it install?

Thanks,
Ramesh

--
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From peter.olsson at visionutveckling.se  Thu Jun 28 19:44:24 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 28 Jun 2012 15:44:24 +0000
Subject: [Freeswitch-users] multiple registerations coudln't
	worked	correctly!
In-Reply-To: <1340896913.16633.YahooMailNeo@web120101.mail.ne1.yahoo.com>
References: <1FFF97C269757C458224B7C895F35F151268CD@cantor.std.visionutv.se>,
	<1340896913.16633.YahooMailNeo@web120101.mail.ne1.yahoo.com>
Message-ID: <28A241AE-75DB-42E3-A28D-2AF5EA00380A@visionutveckling.se>

This is just not possible. If someone steals the user credentials, that account must be locked, revoked or given a new password. With your suggested setup it would mean that UserA wouldn't even be able to register after a progam crash, or IP address change.

The only way for the server to know if the REGISTER is ok, is to chck the credentials, if you have the correct vredentials, you will be registered.

/Peter

28 jun 2012 kl. 17:29 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com>>:

suppose we have account with the following information:
extension=1000
secret=12345
callerid=userA
--------------------------
now the user(B,C,D ..) -NOT IMPORTANT who?- stolen  the account informations of userA and want to login using userA account
while the userA is now connect to server,but at the same time userB register on sip phone using userA iformations(extesion=1000,secret=12345,...)
and by set multi_registering=false the userA is not connected now, and userB is now connected because only one user able to register
but i don't want this situation appear,i want to set if userA is now connected the other user couldn't connect using same informations
thanksss



________________________________
From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Thursday, June 28, 2012 6:51 PM
Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!

I still don?t understand.. :) How could UserA use UserB?s account? Aren?t you using different passwords for the user?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh
Skickat: den 28 juni 2012 10:05
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!

i have implemented VODSL service using freeswitch,and create accounts for user on the server,but i don't want the userA use the account of userB,
i want to restrict each user only use her/his account not able to use each other account that don't belong to his/her...
also i don't want to restrict this using ip addresses ,want to limit using account

________________________________
From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Thursday, June 28, 2012 9:44 AM
Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!

The option is to enable multiple-registrations, which will make both registrations work simultaneously.

Why is this even a problem, what problem are you trying to solve?

/Peter

28 jun 2012 kl. 06:21 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com><mailto:saami_mh at ymail.com<mailto:saami_mh at ymail.com>>>:

is there any solutions for the second registration couldn't overwrite the first one registeration ?


________________________________
From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se><mailto:peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>>
To: FreeSWITCH Users Help <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>>>
Sent: Wednesday, June 27, 2012 11:46 PM
Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!

If they end up with only one registration within FS it's just normal. Since the second registration will overwrite the first.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org><mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org><mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>>] f?r Samira Mh [saami_mh at ymail.com<mailto:saami_mh at ymail.com><mailto:saami_mh at ymail.com<mailto:saami_mh at ymail.com>>]
Skickat: den 27 juni 2012 21:05
Till: Free SWITCH Users Help
?mne: [Freeswitch-users] multiple registerations coudln't worked correctly!


hi,
i have set <param name="multiple-registrations" value="false"/> on internal.xml  and switch.conf.xml then
restart freeswitch
but I  could register 2 sip phones using the same extension number at the same time!
but i want to only one extension could be register using the same extensions
plz help


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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>>
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>


</>

Official FreeSWITCH Sites
http://www.freeswitch.org<http://www.freeswitch.org/>
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consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>


</>

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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
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</>

Official FreeSWITCH Sites
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Join Us At ClueCon - Aug 7-9, 2012

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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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!DSPAM:4fec751032765692133399!
_________________________________________________________________________
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From peter.olsson at visionutveckling.se  Thu Jun 28 19:45:58 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 28 Jun 2012 15:45:58 +0000
Subject: [Freeswitch-users] Error while trying to install freeswitch
 latest	source from git
In-Reply-To: <1340897881933-7580357.post@n2.nabble.com>
References: <1340897881933-7580357.post@n2.nabble.com>
Message-ID: <F4B93598-AEA3-4913-8BD1-86FE70891031@visionutveckling.se>

A good start would be to give us the error :) send the relevant parts of the build errors, and we'll go from there.

/Peter

28 jun 2012 kl. 17:43 skrev "ramesh" <ramesh_mind at yahoo.com>:

> Hi All, 
> 
> We are to trying to install latest freeswitch source into our new server ,
> while trying to make the source , we are getting error and unable to proceed
> futhur .
> 
> Any Idea? to make it install?
> 
> Thanks,
> Ramesh
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-while-trying-to-install-freeswitch-latest-source-from-git-tp7580357.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> !DSPAM:4fec784032761027615756!
> 


From msc at freeswitch.org  Thu Jun 28 19:46:39 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Jun 2012 08:46:39 -0700
Subject: [Freeswitch-users] multiple registerations coudln't worked
	correctly!
In-Reply-To: <1340896913.16633.YahooMailNeo@web120101.mail.ne1.yahoo.com>
References: <1FFF97C269757C458224B7C895F35F151268CD@cantor.std.visionutv.se>
	<1340896913.16633.YahooMailNeo@web120101.mail.ne1.yahoo.com>
Message-ID: <CAKzWOxX5+z5e6MF41d23=eaE1r5pBjYkB2zQkpkud9ienmvKrg@mail.gmail.com>

This is probably the wrong way to handle the situation. There are
definitely drawbacks. It sounds like you're wanting a "first come, first
served" scenario. The problem is that if the thief gets userA's information
then all he would have to do is disconnect the real userA's phone, thereby
making that registration go away, then he could register his bogus "userA"
and no one would know any better. The only symptom would be that the real
userA would not be able to register, but by the time you found out about
that the thief could have made all sorts of calls while pretending to be
userA.

I believe you need to rethink your security strategy. If the real issue is
that someone can steal user account information and use it nefariously then
you need to solve THAT problem, or at least mitigate it. For example, just
how did the thief steal the credentials?

I get the feeling that trying to do this "I'm first, so no one else can
register" thing is a poor way to make things secure. It's kind of like
leaving the front door unlocked and then super gluing all your stuff to the
floor so no one can walk in and take it.

-MC

On Thu, Jun 28, 2012 at 8:21 AM, Samira Mh <saami_mh at ymail.com> wrote:

> suppose we have account with the following information:
> extension=1000
> secret=12345
> callerid=userA
> --------------------------
> now the user(B,C,D ..) -NOT IMPORTANT who?- stolen  the account
> informations of userA and want to login using userA account
> while the userA is now connect to server,but at the same time userB
> register on sip phone using userA
> iformations(extesion=1000,secret=12345,...)
> and by set multi_registering=false the userA is not connected now, and
> userB is now connected because only one user able to register
> but i don't want this situation appear,i want to set if userA is now
> connected the other user couldn't connect using same informations
> thanksss
>
>
>
>   ------------------------------
> *From:* Peter Olsson <peter.olsson at visionutveckling.se>
> *To:* 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, June 28, 2012 6:51 PM
> *Subject:* Re: [Freeswitch-users] multiple registerations coudln't worked
> correctly!
>
>   I still don?t understand.. :) How could UserA use UserB?s account?
> Aren?t you using different passwords for the user?
>
> /Peter
>
>  *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Samira Mh
> *Skickat:* den 28 juni 2012 10:05
> *Till:* FreeSWITCH Users Help
> *?mne:* Re: [Freeswitch-users] multiple registerations coudln't worked
> correctly!
>
>  i have implemented VODSL service using freeswitch,and create accounts
> for user on the server,but i don't want the userA use the account of userB,
>  i want to restrict each user only use her/his account not able to use
> each other account that don't belong to his/her...
>  also i don't want to restrict this using ip addresses ,want to limit
> using account
>
>    ------------------------------
>  *From:* Peter Olsson <peter.olsson at visionutveckling.se>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, June 28, 2012 9:44 AM
> *Subject:* Re: [Freeswitch-users] multiple registerations coudln't worked
> correctly!
>
> The option is to enable multiple-registrations, which will make both
> registrations work simultaneously.
>
> Why is this even a problem, what problem are you trying to solve?
>
> /Peter
>
> 28 jun 2012 kl. 06:21 skrev "Samira Mh" <saami_mh at ymail.com<mailto:
> saami_mh at ymail.com>>:
>
> is there any solutions for the second registration couldn't overwrite the
> first one registeration ?
>
>
> ________________________________
> From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:
> peter.olsson at visionutveckling.se>>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:
> freeswitch-users at lists.freeswitch.org>>
> Sent: Wednesday, June 27, 2012 11:46 PM
> Subject: Re: [Freeswitch-users] multiple registerations coudln't worked
> correctly!
>
> If they end up with only one registration within FS it's just normal.
> Since the second registration will overwrite the first.
>
> /Peter
> ________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:
> freeswitch-users-bounces at lists.freeswitch.org> [
> freeswitch-users-bounces at lists.freeswitch.org<mailto:
> freeswitch-users-bounces at lists.freeswitch.org>] f?r Samira Mh [
> saami_mh at ymail.com<mailto:saami_mh at ymail.com>]
> Skickat: den 27 juni 2012 21:05
> Till: Free SWITCH Users Help
> ?mne: [Freeswitch-users] multiple registerations coudln't worked correctly!
>
>
> hi,
> i have set <param name="multiple-registrations" value="false"/> on
> internal.xml  and switch.conf.xml then
> restart freeswitch
> but I  could register 2 sip phones using the same extension number at the
> same time!
> but i want to only one extension could be register using the same
> extensions
> plz help
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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/>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>
> !DSPAM:4febd86932764854317000!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   !DSPAM:4fec0eac32768939220457!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Jun 28 20:00:11 2012
From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Thu, 28 Jun 2012 13:00:11 -0300
Subject: [Freeswitch-users] Error while trying to install freeswitch
 latest source from git
In-Reply-To: <1340897881933-7580357.post@n2.nabble.com>
References: <1340897881933-7580357.post@n2.nabble.com>
Message-ID: <79CCEB6B0EAA4750A0C0557143CDECF6@freeswitch.org>

Ramesh, a better description of the error would make more sense when posting this message.

A console output to say the very least.

Regards,  

--  
Jo?o Mesquita
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Thursday, June 28, 2012 at 12:38 PM, ramesh wrote:

> Hi All,  
>  
> We are to trying to install latest freeswitch source into our new server ,
> while trying to make the source , we are getting error and unable to proceed
> futhur .
>  
> Any Idea? to make it install?
>  
> Thanks,
> Ramesh
>  
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-while-trying-to-install-freeswitch-latest-source-from-git-tp7580357.html
> Sent from the freeswitch-users mailing list archive at Nabble.com (http://Nabble.com).
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> Join Us At ClueCon - Aug 7-9, 2012
>  
> 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
>  
>  


-------------- next part --------------
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From saami_mh at ymail.com  Thu Jun 28 20:00:22 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Thu, 28 Jun 2012 09:00:22 -0700 (PDT)
Subject: [Freeswitch-users] multiple registerations coudln't worked
	correctly!
In-Reply-To: <CAKzWOxX5+z5e6MF41d23=eaE1r5pBjYkB2zQkpkud9ienmvKrg@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F151268CD@cantor.std.visionutv.se>
	<1340896913.16633.YahooMailNeo@web120101.mail.ne1.yahoo.com>
	<CAKzWOxX5+z5e6MF41d23=eaE1r5pBjYkB2zQkpkud9ienmvKrg@mail.gmail.com>
Message-ID: <1340899222.90252.YahooMailNeo@web120103.mail.ne1.yahoo.com>

thanks alot for great reply,
using sip-to-host , ip, domain ?on realtime directory we found out who steal account informations,because we only serve this servive for users who have ADSL account of our ISP compay...so restict on the range of ip-addresses
but i think there is not any soltion for implement limit only one user login,
thanks so much


________________________________
 From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Thursday, June 28, 2012 8:16 PM
Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!
 

This is probably the wrong way to handle the situation. There are definitely drawbacks. It sounds like you're wanting a "first come, first served" scenario. The problem is that if the thief gets userA's information then all he would have to do is disconnect the real userA's phone, thereby making that registration go away, then he could register his bogus "userA" and no one would know any better. The only symptom would be that the real userA would not be able to register, but by the time you found out about that the thief could have made all sorts of calls while pretending to be userA.

I believe you need to rethink your security strategy. If the real issue is that someone can steal user account information and use it nefariously then you need to solve THAT problem, or at least mitigate it. For example, just how did the thief steal the credentials? 

I get the feeling that trying to do this "I'm first, so no one else can register" thing is a poor way to make things secure. It's kind of like leaving the front door unlocked and then super gluing all your stuff to the floor so no one can walk in and take it.

-MC


On Thu, Jun 28, 2012 at 8:21 AM, Samira Mh <saami_mh at ymail.com> wrote:

suppose we have account with the following information:
>
>extension=1000
>secret=12345
>callerid=userA
>--------------------------
>now the user(B,C,D ..) -NOT IMPORTANT who?- stolen ?the account informations of userA and want to login using userA account
>
>while the userA is now connect to server,but at the same time userB register on sip phone using userA iformations(extesion=1000,secret=12345,...)
>and by set multi_registering=false the userA is not connected now, and userB is now connected because only one user able to register
>but i don't want this situation appear,i want to set if userA is now connected the other user couldn't connect using same informations?
>thanksss
>
>
>
>
>
>
>
>________________________________
> From: Peter Olsson <peter.olsson at visionutveckling.se>
>To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org> 
>Sent: Thursday, June 28, 2012 6:51 PM
>Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!
> 
>
>I still don?t understand.. :) How could UserA use UserB?s account? Aren?t you using different passwords for the user?
>?
>/Peter
>?
>Fr?n:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh
>Skickat: den 28 juni 2012 10:05
>Till: FreeSWITCH Users Help
>?mne: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!
>?
>i have implemented VODSL service using freeswitch,and create accounts for user on the server,but i don't want the userA use the account of userB,
>i want to restrict each user only use her/his account not able to use each other account that don't belong to his/her...
>also i don't want to restrict this using ip addresses ,want to limit using account
>?
>
>________________________________
> 
>From:Peter Olsson <peter.olsson at visionutveckling.se>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Thursday, June 28, 2012 9:44 AM
>Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!
>
>The option is to enable multiple-registrations, which will make both registrations work simultaneously.
>
>Why is this even a problem, what problem are you trying to solve?
>
>/Peter
>
>28 jun 2012 kl. 06:21 skrev "Samira Mh" <saami_mh at ymail.com<mailto:saami_mh at ymail.com>>:
>
>is there any solutions for the second registration couldn't overwrite the first one registeration ?
>
>
>________________________________
>From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
>Sent: Wednesday, June 27, 2012 11:46 PM
>Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly!
>
>If they end up with only one registration within FS it's just normal. Since the second registration will overwrite the first.
>
>/Peter
>________________________________
>Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] f?r Samira Mh [saami_mh at ymail.com<mailto:saami_mh at ymail.com>]
>Skickat: den 27 juni 2012 21:05
>Till: Free SWITCH Users Help
>?mne: [Freeswitch-users] multiple registerations coudln't worked correctly!
>
>
>hi,
>i have set <param name="multiple-registrations" value="false"/> on internal.xml? and switch.conf.xml then
>restart freeswitch
>but I? could register 2 sip phones using the same extension number at the same time!
>but i want to only one extension could be register using the same extensions
>plz help
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>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/>
>
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>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
>
>
>!DSPAM:4febd86932764854317000!
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>!DSPAM:4fec0eac32768939220457! 
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
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From jmsucasas at me.com  Thu Jun 28 20:19:37 2012
From: jmsucasas at me.com (Jose Miguel Sucasas Mejuto)
Date: Thu, 28 Jun 2012 18:19:37 +0200
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 72, Issue 253
In-Reply-To: <mailman.13092.1340856710.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.13092.1340856710.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <2FB61709-BC38-4906-BFE0-1848BA50A5DA@me.com>

Hello Peter,

	As you can see in the 4th line of my dialplan,  I have this variable setted,  but have no effect, and pauses jitterbuffer:

> <action application="cng_plc"/>
> <!-- A-leg JitterBuffer -->
> <action application="set" data="jitterbuffer_msec=90"/>
> <action application="set" data="sip_jitter_buffer_during_bridge=true"/>
> <!-- A-leg JitterBuffer debug, doesn't work -->
> <action application="jitterbuffer" data="debug:${uuid}"/>
> <!-- In bridge application, I try to activate B-leg JitterBuffer -->
> <action application="bridge" data="{jitterbuffer_msec=90,absolute_codec_string='SILK at 16000h,iLBC at 30i,G729,AMR,AMR-WB,H263,H264'}user/${dialed_extension}@${domain_name}"/>

	Following FreeSwitch documentation, I dont understand why jitterbuffer is paused.


Thanks for your reply,



> De: Peter Olsson <peter.olsson at visionutveckling.se>
> Asunto: Re: [Freeswitch-users] Always PAUSE Jitterbuffer
> Fecha: 27 de junio de 2012 22:17:11 GMT+02:00
> Para: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Responder a: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Maybe this is what you are looking for? I think FS usually wants the endpoints to handle this during a bridge, setting this var should force it to stay enabled.
> 
> http://wiki.freeswitch.org/wiki/Variable_sip_jitter_buffer_during_bridge
> 
> /Peter
> 
> 27 jun 2012 kl. 22:11 skrev "Jose Miguel Sucasas Mejuto" <jmsucasas at me.com<mailto:jmsucasas at me.com>>:
> 
> Hello,
> 
> I have installed Freeswitch from GIT (currenty I have installed 1.2.0-rc2+git~20120625T203158Z~7dc9a9cacc version).  Because some users connects from 3G network using a smartphone,  I try to activate JitterBuffer and Packet Loss Concealing in both directions (in a call with U1 and U2,  I try to activate 90ms jitterbuffer in audio flow U1->U2 and another 90ms jitterbuffer in audio flow U2->U1).  To achieve this,  I have this in dialplan (default.xml file):
> 
> <action application="cng_plc"/>
> <!-- A-leg JitterBuffer -->
> <action application="set" data="jitterbuffer_msec=90"/>
> <action application="set" data="sip_jitter_buffer_during_bridge=true"/>
> <!-- A-leg JitterBuffer debug, doesn't work -->
> <action application="jitterbuffer" data="debug:${uuid}"/>
> <!-- In bridge application, I try to activate B-leg JitterBuffer -->
> <action application="bridge" data="{jitterbuffer_msec=90,absolute_codec_string='SILK at 16000h,iLBC at 30i,G729,AMR,AMR-WB,H263,H264'}user/${dialed_extension}@${domain_name}"/>
> 
> User 1002 is on the same network as Freeswitch box,  connected by ethernet line,  and user 1001 is on 3G,  when user 1002 calls to user 1001 (both of them uses iLBC30 codec),  on fs_cli it seems that JitterBuffer is activated (in bold green)
> 
> 2012-06-27 20:55:14.929462 [NOTICE] sofia.c:5930 Ring-Ready sofia/internal/sip:1001 at 10.10.57.82:50203!
> 2012-06-27 20:55:14.929462 [INFO] switch_ivr_originate.c:1156 Sending early media
> 2012-06-27 20:55:14.929462 [INFO] switch_nat.c:590 NAT port mapping disabled
> 2012-06-27 20:55:14.929462 [INFO] switch_nat.c:590 NAT port mapping disabled
> 2012-06-27 20:55:14.929462 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [proceeding][180]
> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3269 AUDIO RTP [sofia/internal/1002 at sukmad.dyndns-home.com:58180] 79.150.184.57 port 11154 -> 192.168.0.15 port 10000 codec: 98 ms: 30
> 2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:1861 Starting timer [soft] 240 bytes per 30ms
> 2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:2207 RTCP send rate is: 5000 and packet rate is: 30000 Remote Port: 10001
> 2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 192.168.0.15:10001
> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to 90ms (3 frames)
> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send payload to 101
> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive payload to 101
> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3566 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Set rtp dtmf delay to 40
> 2012-06-27 20:55:14.929462 [DEBUG] mod_sofia.c:2606 Ring SDP:
> v=0
> o=FreeSWITCH 1340812160 1340812161 IN IP4 79.150.184.57
> s=FreeSWITCH
> c=IN IP4 79.150.184.57
> t=0 0
> m=audio 11154 RTP/AVP 98 101
> a=rtpmap:98 iLBC/8000
> a=fmtp:98 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:30
> a=sendrecv
> 
> 2012-06-27 20:55:14.929462 [NOTICE] mod_sofia.c:2609 Pre-Answer sofia/internal/1002 at sukmad.dyndns-home.com:58180!
> 2012-06-27 20:55:14.929462 [DEBUG] switch_channel.c:3042 (sofia/internal/1002 at sukmad.dyndns-home.com:58180) Callstate Change RINGING -> EARLY
> 2012-06-27 20:55:14.929462 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:14.929462 [DEBUG] switch_ivr_originate.c:1205 Raw Codec Activation Success L16 at 8000hz 1 channel 30ms
> 2012-06-27 20:55:14.929462 [DEBUG] switch_core_codec.c:219 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Push codec L16:70
> 2012-06-27 20:55:14.929462 [DEBUG] switch_ivr_originate.c:1268 Play Ringback Tone [%(2000,4000,440,480)]
> 2012-06-27 20:55:14.929462 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:14.929462 [DEBUG] sofia.c:5831 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 skipping state [early][183]
> 2012-06-27 20:55:14.939460 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
> 2012-06-27 20:55:14.939460 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
> 2012-06-27 20:55:14.939460 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 entering state [proceeding][180]
> 2012-06-27 20:55:14.939460 [NOTICE] sofia.c:5930 Ring-Ready sofia/internal/sip:1001 at 31.4.238.246:22268!
> 2012-06-27 20:55:15.029459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
> 2012-06-27 20:55:15.029459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
> 2012-06-27 20:55:15.029459 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 entering state [proceeding][180]
> 2012-06-27 20:55:15.069457 [DEBUG] switch_rtp.c:3527 Correct ip/port confirmed.
> 
> When user 1001 answer the call, JitterBuffer is paused (in bold red):
> 
> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
> 2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [completing][200]
> 2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5849 Remote SDP:
> v=0
> o=- 3549812097 3549812099 IN IP4 10.10.57.82
> s=cpc_med
> c=IN IP4 10.10.57.82
> b=AS:226
> t=0 0
> m=audio 22010 RTP/AVP 99 101
> a=rtpmap:99 iLBC/8000
> a=fmtp:99 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> m=video 0 RTP/AVP 103
> c=IN IP4 10.10.57.82
> b=AS:200000
> a=rtpmap:103 H264/90000
> 
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3919 Deciding whether to pass zrtp-hash between legs
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
> 2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [ready][200]
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3919 Deciding whether to pass zrtp-hash between legs
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [iLBC:99:8000:30:0]/[SILK:119:16000:20:30000]
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5148 Set 2833 dtmf send payload to 101
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [iLBC:99:8000:30:0]/[iLBC:97:8000:30:13330]
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3020 Set Codec sofia/internal/sip:1001 at 10.10.57.82:50203 iLBC/8000 30 ms 240 samples 13330 bits
> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:1001 at 10.10.57.82:50203 Original read codec set to iLBC:97
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5148 Set 2833 dtmf send payload to 101
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3269 AUDIO RTP [sofia/internal/sip:1001 at 10.10.57.82:50203] 79.150.184.57 port 10388 -> 10.10.57.82 port 22010 codec: 99 ms: 30
> 2012-06-27 20:55:26.079458 [DEBUG] switch_rtp.c:1861 Starting timer [soft] 240 bytes per 30ms
> 2012-06-27 20:55:26.089461 [DEBUG] switch_rtp.c:2207 RTCP send rate is: 5000 and packet rate is: 30000 Remote Port: 22011
> 2012-06-27 20:55:26.089461 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 10.10.57.82:22011
> 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to 90ms (3 frames)
> 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send payload to 101
> 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive payload to 101
> 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3566 sofia/internal/sip:1001 at 10.10.57.82:50203 Set rtp dtmf delay to 40
> 2012-06-27 20:55:26.089461 [DEBUG] switch_channel.c:3301 (sofia/internal/sip:1001 at 10.10.57.82:50203) Callstate Change RINGING -> ACTIVE
> 2012-06-27 20:55:26.089461 [DEBUG] switch_channel.c:3313 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.089461 [NOTICE] sofia.c:6573 Channel [sofia/internal/sip:1001 at 10.10.57.82:50203] has been answered
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_codec.c:244 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Restore previous codec iLBC:97.
> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2899 (sofia/internal/sip:1001 at 192.168.0.11:50199) Callstate Change RINGING -> HANGUP
> 2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3257 Hangup sofia/internal/sip:1001 at 192.168.0.11:50199 [CS_CONSUME_MEDIA] [LOSE_RACE]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2922 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [KILL]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_HANGUP
> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2899 (sofia/internal/sip:1001 at 31.4.238.246:22268) Callstate Change RINGING -> HANGUP
> 2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3257 Hangup sofia/internal/sip:1001 at 31.4.238.246:22268 [CS_CONSUME_MEDIA] [LOSE_RACE]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2922 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [KILL]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 192.168.0.11:50199) State HANGUP
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001 at 192.168.0.11:50199 hanging up, cause: LOSE_RACE
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_HANGUP
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:527 Sending CANCEL to sofia/internal/sip:1001 at 192.168.0.11:50199
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard HANGUP, cause: LOSE_RACE
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 192.168.0.11:50199) State HANGUP going to sleep
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/sip:1001 at 192.168.0.11:50199) State Change CS_HANGUP -> CS_REPORTING
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_REPORTING
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 192.168.0.11:50199) State REPORTING
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard REPORTING, cause: LOSE_RACE
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 192.168.0.11:50199) State REPORTING going to sleep
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 31.4.238.246:22268) State HANGUP
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 hanging up, cause: LOSE_RACE
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at 192.168.0.11:50199) State Change CS_REPORTING -> CS_DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1429 Session 15 (sofia/internal/sip:1001 at 192.168.0.11:50199) Locked, Waiting on external entities
> 2012-06-27 20:55:26.109459 [INFO] switch_nat.c:590 NAT port mapping disabled
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:527 Sending CANCEL to sofia/internal/sip:1001 at 31.4.238.246:22268
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:754 Local SDP sofia/internal/1002 at sukmad.dyndns-home.com:58180:
> v=0
> o=FreeSWITCH 1340812160 1340812162 IN IP4 79.150.184.57
> s=FreeSWITCH
> c=IN IP4 79.150.184.57
> t=0 0
> m=audio 11154 RTP/AVP 98 101
> a=rtpmap:98 iLBC/8000
> a=fmtp:98 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:30
> a=sendrecv
> m=video 10766 RTP/AVP 34 103
> a=rtpmap:34 H263/90000
> a=rtpmap:103 H264/90000
> 
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard HANGUP, cause: LOSE_RACE
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 31.4.238.246:22268) State HANGUP going to sleep
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/sip:1001 at 31.4.238.246:22268) State Change CS_HANGUP -> CS_REPORTING
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_REPORTING
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:3301 (sofia/internal/1002 at sukmad.dyndns-home.com:58180) Callstate Change EARLY -> ACTIVE
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 31.4.238.246:22268) State REPORTING
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard REPORTING, cause: LOSE_RACE
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 31.4.238.246:22268) State REPORTING going to sleep
> 2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3284 Channel [sofia/internal/1002 at sukmad.dyndns-home.com:58180] has been answered
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at 31.4.238.246:22268) State Change CS_REPORTING -> CS_DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1429 Session 17 (sofia/internal/sip:1001 at 31.4.238.246:22268) Locked, Waiting on external entities
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] sofia.c:5838 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 entering state [completed][200]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_originate.c:3341 Originate Resulted in Success: [sofia/internal/sip:1001 at 10.10.57.82:50203]
> 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1447 Session 15 (sofia/internal/sip:1001 at 192.168.0.11:50199) Ended
> 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1447 Session 17 (sofia/internal/sip:1001 at 31.4.238.246:22268) Ended
> 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/sip:1001 at 192.168.0.11:50199 [CS_DESTROY]
> 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/sip:1001 at 31.4.238.246:22268 [CS_DESTROY]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/sip:1001 at 31.4.238.246:22268) Callstate Change HANGUP -> DOWN
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/sip:1001 at 192.168.0.11:50199) Callstate Change HANGUP -> DOWN
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 31.4.238.246:22268) State DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001 at 31.4.238.246:22268 SOFIA DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 31.4.238.246:22268) State DESTROY going to sleep
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 192.168.0.11:50199) State DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001 at 192.168.0.11:50199 SOFIA DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 192.168.0.11:50199) State DESTROY going to sleep
> 2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_originate.c:3341 Originate Resulted in Success: [sofia/internal/sip:1001 at 10.10.57.82:50203]
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:1603 sofia/internal/sip:1001 at 10.10.57.82:50203 PAUSE Jitterbuffer
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_bridge.c:1359 (sofia/internal/sip:1001 at 10.10.57.82:50203) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 10.10.57.82:50203) Running State Change CS_EXCHANGE_MEDIA
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:443 (sofia/internal/sip:1001 at 10.10.57.82:50203) State EXCHANGE_MEDIA
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:582 SOFIA EXCHANGE_MEDIA
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] sofia.c:5838 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 entering state [ready][200]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.469460 [INFO] switch_rtp.c:3510 Auto Changing port from 10.10.57.82:22010 to 31.4.238.246:22267
> 2012-06-27 20:55:26.469460 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 31.4.238.246:22268
> 2012-06-27 20:55:38.009460 [DEBUG] sofia_reg.c:371 Registered sip.netelip.com<http://sip.netelip.com>
> 
> 
> I try a lot of test and all of them have the same issue,  Why JitterBuffer is paused?.  Is it possible to force JitterBuffer?
> 
> 
> Thanks in advance,
> 
> Regards
> 
> 
> 
> !DSPAM:4feb659c32769194324459!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> 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
> 
> 
> !DSPAM:4feb659c32769194324459!
> 
> 
> 

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From ramesh_mind at yahoo.com  Thu Jun 28 20:45:59 2012
From: ramesh_mind at yahoo.com (ramesh)
Date: Thu, 28 Jun 2012 09:45:59 -0700 (PDT)
Subject: [Freeswitch-users] Error while trying to install freeswitch
	latest source from git
In-Reply-To: <79CCEB6B0EAA4750A0C0557143CDECF6@freeswitch.org>
References: <1340897881933-7580357.post@n2.nabble.com>
	<79CCEB6B0EAA4750A0C0557143CDECF6@freeswitch.org>
Message-ID: <1340901959487-7580364.post@n2.nabble.com>

Okey , after configuring freeswitch , when i give make cmd , i get the
following error.

Making all in nua
...
./../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level:
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1198:34: error: expected ')'
before 'const'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1201:35: error: expected ')'
before 'const'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1636:1: error: unknown type
name 'sip_p_asserted_identity_t'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1636:67: error: unknown type
name 'sip_p_asserted_identity_t'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1702:1: error: unknown type
name 'sip_p_asserted_identity_t'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1702:73: error: unknown type
name 'sip_p_asserted_identity_t'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1707:1: error: unknown type
name 'sip_p_asserted_identity_t'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1707:73: error: unknown type
name 'sip_p_asserted_identity_t'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1746:1: error: unknown type
name 'sip_p_asserted_identity_t'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1746:74: error: unknown type
name 'sip_p_asserted_identity_t'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1751:1: error: unknown type
name 'sip_p_asserted_identity_t'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1751:74: error: unknown type
name 'sip_p_asserted_identity_t'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1777:1: error: unknown type
name 'sip_p_asserted_identity_t'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1781:1: error: unknown type
name 'sip_p_asserted_identity_t'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function
'sip_p_asserted_identity_make':
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1783:11: error:
'sip_p_asserted_identity_t' undeclared (first use in this function)
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1783:38: error: expected
expression before ')' token
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level:
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1809:1: error: unknown type
name 'sip_p_asserted_identity_t'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1813:1: error: unknown type
name 'sip_p_asserted_identity_t'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function
'sip_p_asserted_identity_format':
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1822:11: error:
'sip_p_asserted_identity_t' undeclared (first use in this function)
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1822:38: error: expected
expression before ')' token
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1815:17: error: variable 'h'
set but not used [-Werror=unused-but-set-variable]
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level:
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1890:56: error: expected ')'
before 'const'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1893:57: error: expected ')'
before 'const'
../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:2945:2: error: #endif
without #if
nua_subnotref.c: In function 'nua_refer_client_response':
nua_subnotref.c:915:7: error: 'sip_refer_sub_t' undeclared (first use in
this function)
nua_subnotref.c:915:23: error: expected ';' before 'const'
nua_subnotref.c:917:11: error: 'rs' undeclared (first use in this function)
cc1: all warnings being treated as errors

make[9]: *** [nua_subnotref.lo] Error 1
make[8]: *** [all] Error 2
make[8]: *** No rule to make target `nua/libnua.la', needed by
`libsofia-sip-ua.la'.  Stop.
make[7]: *** [all-recursive] Error 1
Making all in packages
make[6]: *** [all-recursive] Error 1
make[5]: *** [all] Error 2
make[4]: ***
[/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la]
Error 2
make[3]: *** [mod_sofia-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2


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From sdevoy at bizfocused.com  Thu Jun 28 20:59:53 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 28 Jun 2012 12:59:53 -0400
Subject: [Freeswitch-users] Voice delay in normal calls
In-Reply-To: <1340893308741-7580349.post@n2.nabble.com>
References: <1340893308741-7580349.post@n2.nabble.com>
Message-ID: <053701cd554f$7015b2e0$504118a0$@bizfocused.com>

HI Ramesh,

I had the same problem, especially with remote (NATed) phones.  We upgraded
our server hardware to dual core (and then quad core) faster processors and
solved the problem.  We found that the delay was really bad if two calls
arrived or were dialed at the same time.  The multiple processors solved the
concurrent call issue and faster processors solved the issue all around.  We
are now on dual hyper-threaded multicore xeon processors @ 3.0Ghz!  NO
performance issues (of course).  I picked up a used Dell PE2850 on ebay for
$250 with RAID and 4 disks.

Hope that helps.

Sean

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ramesh
Sent: Thursday, June 28, 2012 10:22 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Voice delay in normal calls

 Hi,

During an outbound or inbound calls from freeswitch , their a voice dealy of
about 4 to 5secs  and also we are facing delay issue in rtmp calls too. we
are hardly trying to figure the reason for this.

Any Help or Suggestions would be really helpful!.

Thanks,
Ramesh 

--
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Voice-delay-in-normal-calls-tp
7580349.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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




From notlikeme75 at yahoo.com  Thu Jun 28 21:06:44 2012
From: notlikeme75 at yahoo.com (Rodney)
Date: Thu, 28 Jun 2012 10:06:44 -0700 (PDT)
Subject: [Freeswitch-users] (no subject)
Message-ID: <1340903204.74428.YahooMailNeo@web65511.mail.ac4.yahoo.com>

http://feiyankj.com/test.php?meet207.png
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From ramesh_mind at yahoo.com  Thu Jun 28 21:08:39 2012
From: ramesh_mind at yahoo.com (ramesh)
Date: Thu, 28 Jun 2012 10:08:39 -0700 (PDT)
Subject: [Freeswitch-users] Error while trying to install freeswitch
	latest source from git
In-Reply-To: <1340901959487-7580364.post@n2.nabble.com>
References: <1340897881933-7580357.post@n2.nabble.com>
	<79CCEB6B0EAA4750A0C0557143CDECF6@freeswitch.org>
	<1340901959487-7580364.post@n2.nabble.com>
Message-ID: <1340903319321-7580366.post@n2.nabble.com>

Actually, we are running this on amazon EC2 cloud with better instance type,
and there was no issue before. this happened just now


High-CPU Extra Large Instance

7 GB of memory
20 EC2 Compute Units (8 virtual cores with 2.5 EC2 Compute Units each)
1690 GB of instance storage
64-bit platform
I/O Performance: High
API name: c1.xlarge

Thanks
Ramesh

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From ramesh_mind at yahoo.com  Thu Jun 28 21:09:48 2012
From: ramesh_mind at yahoo.com (ramesh)
Date: Thu, 28 Jun 2012 10:09:48 -0700 (PDT)
Subject: [Freeswitch-users] Voice delay in normal calls
In-Reply-To: <053701cd554f$7015b2e0$504118a0$@bizfocused.com>
References: <1340893308741-7580349.post@n2.nabble.com>
	<053701cd554f$7015b2e0$504118a0$@bizfocused.com>
Message-ID: <1340903388060-7580367.post@n2.nabble.com>

Actually, we are running this on amazon EC2 cloud with better instance type,
and there was no issue before. this happened just now


High-CPU Extra Large Instance

7 GB of memory
20 EC2 Compute Units (8 virtual cores with 2.5 EC2 Compute Units each)
1690 GB of instance storage
64-bit platform
I/O Performance: High
API name: c1.xlarge


Thanks
Ramesh

--
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From msc at freeswitch.org  Thu Jun 28 21:15:13 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Jun 2012 10:15:13 -0700
Subject: [Freeswitch-users] Handling Multi Call in Lua
In-Reply-To: <CAN+NqUYWiq2+nodqeS583v0Cxh3dLYNJbTH86-S1y4uXe3Q0Yw@mail.gmail.com>
References: <CAN+NqUYWiq2+nodqeS583v0Cxh3dLYNJbTH86-S1y4uXe3Q0Yw@mail.gmail.com>
Message-ID: <CAKzWOxWY-od8ShRtrW4T8Q=Po7Do8Fgbz=rYjq_-MGAJGrP+pA@mail.gmail.com>

On Thu, Jun 28, 2012 at 7:40 AM, Shahzad Bhatti <shahzad.bhatti at g-r-v.com>wrote:

>
> i want to originate 5 calls after one second sleep using lua and get the
> response using playandgetdigits(), but scipt originate call one by one.
>
> please guide me how i can call 5 numbers at the same time,
>
Short answer: you can't.

Each Lua instance corresponds to a single session/channel. Can't have more
than one when doing it like this. If you need to control multiple channels
then you really, really need to be using the event socket, and ESL in
particular. I recommend chapter 9 of the original FS book and chapter 4 of
the FS cookbook. There are also lots of examples in the source tree, in
freeswitch-contrib, and on the wiki.
-MC
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From anthony.minessale at gmail.com  Thu Jun 28 21:33:16 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Jun 2012 12:33:16 -0500
Subject: [Freeswitch-users] Always PAUSE Jitterbuffer
In-Reply-To: <D32BDE57-EE45-4961-AD62-5651575C1572@visionutveckling.se>
References: <8FCBF56C-DB66-4BF6-96FE-CACBB2F6D9D1@me.com>
	<D32BDE57-EE45-4961-AD62-5651575C1572@visionutveckling.se>
Message-ID: <CAKbxfG_r_1yX2DUNRqJLwTUbPrAK8tmTfi2teHDmeee9Vk0M3w@mail.gmail.com>

set sip_jitter_buffer_during_bridge=true globally or per channel basis


On Wed, Jun 27, 2012 at 3:17 PM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Maybe this is what you are looking for? I think FS usually wants the endpoints to handle this during a bridge, setting this var should force it to stay enabled.
>
> http://wiki.freeswitch.org/wiki/Variable_sip_jitter_buffer_during_bridge
>
> /Peter
>
> 27 jun 2012 kl. 22:11 skrev "Jose Miguel Sucasas Mejuto" <jmsucasas at me.com<mailto:jmsucasas at me.com>>:
>
> Hello,
>
> I have installed Freeswitch from GIT (currenty I have installed 1.2.0-rc2+git~20120625T203158Z~7dc9a9cacc version). ?Because some users connects from 3G network using a smartphone, ?I try to activate JitterBuffer and Packet Loss Concealing in both directions (in a call with U1 and U2, ?I try to activate 90ms jitterbuffer in audio flow U1->U2 and another 90ms jitterbuffer in audio flow U2->U1). ?To achieve this, ?I have this in dialplan (default.xml file):
>
> <action application="cng_plc"/>
> <!-- A-leg JitterBuffer -->
> <action application="set" data="jitterbuffer_msec=90"/>
> <action application="set" data="sip_jitter_buffer_during_bridge=true"/>
> <!-- A-leg JitterBuffer debug, doesn't work -->
> <action application="jitterbuffer" data="debug:${uuid}"/>
> <!-- In bridge application, I try to activate B-leg JitterBuffer -->
> <action application="bridge" data="{jitterbuffer_msec=90,absolute_codec_string='SILK at 16000h,iLBC at 30i,G729,AMR,AMR-WB,H263,H264'}user/${dialed_extension}@${domain_name}"/>
>
> User 1002 is on the same network as Freeswitch box, ?connected by ethernet line, ?and user 1001 is on 3G, ?when user 1002 calls to user 1001 (both of them uses iLBC30 codec), ?on fs_cli it seems that JitterBuffer is activated (in bold green)
>
> 2012-06-27 20:55:14.929462 [NOTICE] sofia.c:5930 Ring-Ready sofia/internal/sip:1001 at 10.10.57.82:50203!
> 2012-06-27 20:55:14.929462 [INFO] switch_ivr_originate.c:1156 Sending early media
> 2012-06-27 20:55:14.929462 [INFO] switch_nat.c:590 NAT port mapping disabled
> 2012-06-27 20:55:14.929462 [INFO] switch_nat.c:590 NAT port mapping disabled
> 2012-06-27 20:55:14.929462 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [proceeding][180]
> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3269 AUDIO RTP [sofia/internal/1002 at sukmad.dyndns-home.com:58180] 79.150.184.57 port 11154 -> 192.168.0.15 port 10000 codec: 98 ms: 30
> 2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:1861 Starting timer [soft] 240 bytes per 30ms
> 2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:2207 RTCP send rate is: 5000 and packet rate is: 30000 Remote Port: 10001
> 2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 192.168.0.15:10001
> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to 90ms (3 frames)
> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send payload to 101
> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive payload to 101
> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3566 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Set rtp dtmf delay to 40
> 2012-06-27 20:55:14.929462 [DEBUG] mod_sofia.c:2606 Ring SDP:
> v=0
> o=FreeSWITCH 1340812160 1340812161 IN IP4 79.150.184.57
> s=FreeSWITCH
> c=IN IP4 79.150.184.57
> t=0 0
> m=audio 11154 RTP/AVP 98 101
> a=rtpmap:98 iLBC/8000
> a=fmtp:98 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:30
> a=sendrecv
>
> 2012-06-27 20:55:14.929462 [NOTICE] mod_sofia.c:2609 Pre-Answer sofia/internal/1002 at sukmad.dyndns-home.com:58180!
> 2012-06-27 20:55:14.929462 [DEBUG] switch_channel.c:3042 (sofia/internal/1002 at sukmad.dyndns-home.com:58180) Callstate Change RINGING -> EARLY
> 2012-06-27 20:55:14.929462 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:14.929462 [DEBUG] switch_ivr_originate.c:1205 Raw Codec Activation Success L16 at 8000hz 1 channel 30ms
> 2012-06-27 20:55:14.929462 [DEBUG] switch_core_codec.c:219 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Push codec L16:70
> 2012-06-27 20:55:14.929462 [DEBUG] switch_ivr_originate.c:1268 Play Ringback Tone [%(2000,4000,440,480)]
> 2012-06-27 20:55:14.929462 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:14.929462 [DEBUG] sofia.c:5831 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 skipping state [early][183]
> 2012-06-27 20:55:14.939460 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
> 2012-06-27 20:55:14.939460 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
> 2012-06-27 20:55:14.939460 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 entering state [proceeding][180]
> 2012-06-27 20:55:14.939460 [NOTICE] sofia.c:5930 Ring-Ready sofia/internal/sip:1001 at 31.4.238.246:22268!
> 2012-06-27 20:55:15.029459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
> 2012-06-27 20:55:15.029459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
> 2012-06-27 20:55:15.029459 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 entering state [proceeding][180]
> 2012-06-27 20:55:15.069457 [DEBUG] switch_rtp.c:3527 Correct ip/port confirmed.
>
> When user 1001 answer the call, JitterBuffer is paused (in bold red):
>
> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
> 2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [completing][200]
> 2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5849 Remote SDP:
> v=0
> o=- 3549812097 3549812099 IN IP4 10.10.57.82
> s=cpc_med
> c=IN IP4 10.10.57.82
> b=AS:226
> t=0 0
> m=audio 22010 RTP/AVP 99 101
> a=rtpmap:99 iLBC/8000
> a=fmtp:99 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> m=video 0 RTP/AVP 103
> c=IN IP4 10.10.57.82
> b=AS:200000
> a=rtpmap:103 H264/90000
>
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3919 Deciding whether to pass zrtp-hash between legs
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
> 2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [ready][200]
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3919 Deciding whether to pass zrtp-hash between legs
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [iLBC:99:8000:30:0]/[SILK:119:16000:20:30000]
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5148 Set 2833 dtmf send payload to 101
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [iLBC:99:8000:30:0]/[iLBC:97:8000:30:13330]
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3020 Set Codec sofia/internal/sip:1001 at 10.10.57.82:50203 iLBC/8000 30 ms 240 samples 13330 bits
> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:1001 at 10.10.57.82:50203 Original read codec set to iLBC:97
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5148 Set 2833 dtmf send payload to 101
> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3269 AUDIO RTP [sofia/internal/sip:1001 at 10.10.57.82:50203] 79.150.184.57 port 10388 -> 10.10.57.82 port 22010 codec: 99 ms: 30
> 2012-06-27 20:55:26.079458 [DEBUG] switch_rtp.c:1861 Starting timer [soft] 240 bytes per 30ms
> 2012-06-27 20:55:26.089461 [DEBUG] switch_rtp.c:2207 RTCP send rate is: 5000 and packet rate is: 30000 Remote Port: 22011
> 2012-06-27 20:55:26.089461 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 10.10.57.82:22011
> 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to 90ms (3 frames)
> 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send payload to 101
> 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive payload to 101
> 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3566 sofia/internal/sip:1001 at 10.10.57.82:50203 Set rtp dtmf delay to 40
> 2012-06-27 20:55:26.089461 [DEBUG] switch_channel.c:3301 (sofia/internal/sip:1001 at 10.10.57.82:50203) Callstate Change RINGING -> ACTIVE
> 2012-06-27 20:55:26.089461 [DEBUG] switch_channel.c:3313 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.089461 [NOTICE] sofia.c:6573 Channel [sofia/internal/sip:1001 at 10.10.57.82:50203] has been answered
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_codec.c:244 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Restore previous codec iLBC:97.
> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2899 (sofia/internal/sip:1001 at 192.168.0.11:50199) Callstate Change RINGING -> HANGUP
> 2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3257 Hangup sofia/internal/sip:1001 at 192.168.0.11:50199 [CS_CONSUME_MEDIA] [LOSE_RACE]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2922 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [KILL]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_HANGUP
> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2899 (sofia/internal/sip:1001 at 31.4.238.246:22268) Callstate Change RINGING -> HANGUP
> 2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3257 Hangup sofia/internal/sip:1001 at 31.4.238.246:22268 [CS_CONSUME_MEDIA] [LOSE_RACE]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2922 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [KILL]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 192.168.0.11:50199) State HANGUP
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001 at 192.168.0.11:50199 hanging up, cause: LOSE_RACE
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_HANGUP
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:527 Sending CANCEL to sofia/internal/sip:1001 at 192.168.0.11:50199
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard HANGUP, cause: LOSE_RACE
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 192.168.0.11:50199) State HANGUP going to sleep
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/sip:1001 at 192.168.0.11:50199) State Change CS_HANGUP -> CS_REPORTING
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_REPORTING
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 192.168.0.11:50199) State REPORTING
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard REPORTING, cause: LOSE_RACE
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 192.168.0.11:50199) State REPORTING going to sleep
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 31.4.238.246:22268) State HANGUP
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 hanging up, cause: LOSE_RACE
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at 192.168.0.11:50199) State Change CS_REPORTING -> CS_DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1429 Session 15 (sofia/internal/sip:1001 at 192.168.0.11:50199) Locked, Waiting on external entities
> 2012-06-27 20:55:26.109459 [INFO] switch_nat.c:590 NAT port mapping disabled
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:527 Sending CANCEL to sofia/internal/sip:1001 at 31.4.238.246:22268
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:754 Local SDP sofia/internal/1002 at sukmad.dyndns-home.com:58180:
> v=0
> o=FreeSWITCH 1340812160 1340812162 IN IP4 79.150.184.57
> s=FreeSWITCH
> c=IN IP4 79.150.184.57
> t=0 0
> m=audio 11154 RTP/AVP 98 101
> a=rtpmap:98 iLBC/8000
> a=fmtp:98 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:30
> a=sendrecv
> m=video 10766 RTP/AVP 34 103
> a=rtpmap:34 H263/90000
> a=rtpmap:103 H264/90000
>
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard HANGUP, cause: LOSE_RACE
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 31.4.238.246:22268) State HANGUP going to sleep
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/sip:1001 at 31.4.238.246:22268) State Change CS_HANGUP -> CS_REPORTING
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_REPORTING
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:3301 (sofia/internal/1002 at sukmad.dyndns-home.com:58180) Callstate Change EARLY -> ACTIVE
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 31.4.238.246:22268) State REPORTING
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard REPORTING, cause: LOSE_RACE
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 31.4.238.246:22268) State REPORTING going to sleep
> 2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3284 Channel [sofia/internal/1002 at sukmad.dyndns-home.com:58180] has been answered
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at 31.4.238.246:22268) State Change CS_REPORTING -> CS_DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1429 Session 17 (sofia/internal/sip:1001 at 31.4.238.246:22268) Locked, Waiting on external entities
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] sofia.c:5838 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 entering state [completed][200]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_originate.c:3341 Originate Resulted in Success: [sofia/internal/sip:1001 at 10.10.57.82:50203]
> 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1447 Session 15 (sofia/internal/sip:1001 at 192.168.0.11:50199) Ended
> 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1447 Session 17 (sofia/internal/sip:1001 at 31.4.238.246:22268) Ended
> 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/sip:1001 at 192.168.0.11:50199 [CS_DESTROY]
> 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/sip:1001 at 31.4.238.246:22268 [CS_DESTROY]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/sip:1001 at 31.4.238.246:22268) Callstate Change HANGUP -> DOWN
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/sip:1001 at 192.168.0.11:50199) Callstate Change HANGUP -> DOWN
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 31.4.238.246:22268) State DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001 at 31.4.238.246:22268 SOFIA DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 31.4.238.246:22268) State DESTROY going to sleep
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 192.168.0.11:50199) State DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001 at 192.168.0.11:50199 SOFIA DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard DESTROY
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 192.168.0.11:50199) State DESTROY going to sleep
> 2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_originate.c:3341 Originate Resulted in Success: [sofia/internal/sip:1001 at 10.10.57.82:50203]
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:1603 sofia/internal/sip:1001 at 10.10.57.82:50203 PAUSE Jitterbuffer
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_bridge.c:1359 (sofia/internal/sip:1001 at 10.10.57.82:50203) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 10.10.57.82:50203) Running State Change CS_EXCHANGE_MEDIA
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:443 (sofia/internal/sip:1001 at 10.10.57.82:50203) State EXCHANGE_MEDIA
> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:582 SOFIA EXCHANGE_MEDIA
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] sofia.c:5838 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 entering state [ready][200]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
> 2012-06-27 20:55:26.469460 [INFO] switch_rtp.c:3510 Auto Changing port from 10.10.57.82:22010 to 31.4.238.246:22267
> 2012-06-27 20:55:26.469460 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 31.4.238.246:22268
> 2012-06-27 20:55:38.009460 [DEBUG] sofia_reg.c:371 Registered sip.netelip.com<http://sip.netelip.com>
>
>
> I try a lot of test and all of them have the same issue, ?Why JitterBuffer is paused?. ?Is it possible to force JitterBuffer?
>
>
> Thanks in advance,
>
> Regards
>
>
>
> !DSPAM:4feb659c32769194324459!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>
> !DSPAM:4feb659c32769194324459!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From Gary.Chen at lightyear.net  Thu Jun 28 22:10:59 2012
From: Gary.Chen at lightyear.net (Gary Chen)
Date: Thu, 28 Jun 2012 14:10:59 -0400
Subject: [Freeswitch-users] Can not pass the value from read( ) app to lua
	code
Message-ID: <A6C2711F9706AF4992FFEC48EA19CA901740FC9964@lyepkyexc.LYEPLYEXC.local>

FreeSWITCH Version 1.2.0-rc2+git~20120625T160048Z~690cae5de1

I am trying to validate the input from read app using lua. Here is my dialplan:

  <extension name="account_code_setup" continue="true">
      <condition field="${use_acct_code}" expression="^YES$|^VALIDATE$">
          <action application="answer"/>
          <action application="read" data="2 15 'tone_stream://%(10000,0,350,440)' acctcode 30000 #"/>
          <action application="log" data="username: '${user_name}' acctcode: '${acctcode}'"/>
          <action application="set" data="sip_h_X-account-code=${acctcode}"/>
          <action inline="true" application="lua" data="validate_acctcode.lua  ${use_acct_code}  ${acctcode_list}  ${acctcode}"/>
      </condition>
      <condition field="${validate}" expression="^NO$">
        <anti-action application="respond" data="406 Invalid account code"/>
        <anti-action application="hangup"/>
      </condition>
    </extension>

I can assign ${acctcode} to sip_h_X-account-code. But it always has nil value when pass to lua.
What am I doing wrong?

Gary
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From msc at freeswitch.org  Thu Jun 28 22:19:43 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Jun 2012 11:19:43 -0700
Subject: [Freeswitch-users] Can not pass the value from read( ) app to
	lua code
In-Reply-To: <A6C2711F9706AF4992FFEC48EA19CA901740FC9964@lyepkyexc.LYEPLYEXC.local>
References: <A6C2711F9706AF4992FFEC48EA19CA901740FC9964@lyepkyexc.LYEPLYEXC.local>
Message-ID: <CAKzWOxWNsmD=EqQkX0wq7D9m4ihfR-sMBK0MoCxdS+Txs6e=1A@mail.gmail.com>

It's because you're executing Lua in-line. Lua is getting executed *before*
all of the applications listed above it. Remove the inline="true" and
you'll be okay.
-MC

On Thu, Jun 28, 2012 at 11:10 AM, Gary Chen <Gary.Chen at lightyear.net> wrote:

> FreeSWITCH Version 1.2.0-rc2+git~20120625T160048Z~690cae5de1****
>
> ** **
>
> I am trying to validate the input from read app using lua. Here is my
> dialplan:****
>
>     ****
>
>   <extension name="account_code_setup" continue="true">****
>
>       <condition field="${use_acct_code}" expression="^YES$|^VALIDATE$">**
> **
>
>           <action application="answer"/>****
>
>           <action application="read" data="2 15
> 'tone_stream://%(10000,0,350,440)' acctcode 30000 #"/>****
>
>           <action application="log" data="username: '${user_name}'
> acctcode: '${acctcode}'"/>****
>
>           <action application="set"
> data="sip_h_X-account-code=${acctcode}"/>****
>
>           <action inline="true" application="lua"
> data="validate_acctcode.lua  ${use_acct_code}  ${acctcode_list}
>  ${acctcode}"/>****
>
>       </condition>****
>
>       <condition field="${validate}" expression="^NO$">****
>
>         <anti-action application="respond" data="406 Invalid account
> code"/>****
>
>         <anti-action application="hangup"/>****
>
>       </condition>****
>
>     </extension>****
>
> ** **
>
> I can assign ${acctcode} to sip_h_X-account-code. But it always has nil
> value when pass to lua.****
>
> What am I doing wrong?****
>
> ** **
>
> Gary  ****
>
>
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From s.lam at keensystems.eu  Thu Jun 28 22:44:47 2012
From: s.lam at keensystems.eu (Steven Lam, KeenSystems B.V.)
Date: Thu, 28 Jun 2012 18:44:47 +0000
Subject: [Freeswitch-users] T.38 rtp auto adjust
In-Reply-To: <2AB1E3C03738764094BDD615FEE595AD097FC9C2@DB3PRD0410MB369.eurprd04.prod.outlook.com>
References: <2AB1E3C03738764094BDD615FEE595AD097FC9C2@DB3PRD0410MB369.eurprd04.prod.outlook.com>
Message-ID: <2AB1E3C03738764094BDD615FEE595AD097FF339@DB3PRD0410MB369.eurprd04.prod.outlook.com>

Never mind.

FS does do "rtp auto adjust" on T.38 streams!

The problem here was the client behind NAT was not sending enough packets for FS to do "rtp auto adjust". After configuring the client to send T.38 no-signal packets, FS did receive enough packets to do "rtp auto adjust".

T.38 NAT problem solved :-)

Steven.

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Steven Lam, KeenSystems
> B.V.
> Sent: dinsdag 26 juni 2012 17:37
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] T.38 rtp auto adjust
> 
> Hi all,
> 
> When FS has a public IP and clients from behind nat connect to this machine
> the "rtp auto adjust" function is really very handy!
> 
> Now when using T.38 (T38 Passthru), it seems FS is no longer doing "rtp auto
> adjust".
> 
> Is there a way to make FS do "rtp auto adjust" for a T.38 fax session?
> 
> Steven.
> 
> 
> __________________________________________________________
> _______________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 cassidywebservices.co.uk  Thu Jun 28 23:04:18 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 28 Jun 2012 20:04:18 +0100
Subject: [Freeswitch-users] Looking up user data in dialplan
In-Reply-To: <004ED0CB-4E46-4F01-B9F8-F82456D4F7F9@scarlet-internet.nl>
References: <CAMbW9oKg=1U3YUUAJMwfWP+H-2DQ=+Lf6W=ZR6Jd7pNv74nXTg@mail.gmail.com>
	<004ED0CB-4E46-4F01-B9F8-F82456D4F7F9@scarlet-internet.nl>
Message-ID: <CAMbW9o+BgiO7ELwk-=ZKFPPTGTOMTEYXbU=6iZMhht8xHmNgdQ@mail.gmail.com>

Hi Leon,

Does that allow me to search by attrs other than ID? I need to get the id
of a user given another attr (such as alias), but looking for a minimal
approach.

Basically all i need is to have N mailboxes, each with a DDI to leave
messages and a DDI to check them. Obviously I can just use one of the DDI's
as the user ID, but to check i need to find the user id for the other
number.

I can do this with mod_xml_curl easily enough, was just wondering if there
was a simpler way.

On 28 June 2012 14:36, Leon de Rooij <leon at scarlet-internet.nl> wrote:

> You can use the set_user dp app to set all user variables as channel
> variables. You can also supply a prefix string to keep these new variables
> separate from wha was already set.
>
> See here: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user
>
> regards,
>
> Leon
>
>
> On Jun 28, 2012, at 11:09, Andrew Cassidy wrote:
>
> Hi All,
>
> Right, I have a very basic application I'm building around voicemail. Each
> user has 2 attrs that I need to search by. user_exists works fine, but
> doesn't return me any data other then the fact the user exists, and
> user_data only searches the id attr.
>
> Is there any way I can do that in just the dialplan with what is already
> available?
>
> In the most simple form I need the user id for a number_alias.
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From curriegrad2004 at gmail.com  Thu Jun 28 23:16:44 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 28 Jun 2012 12:16:44 -0700
Subject: [Freeswitch-users] Looking up user data in dialplan
In-Reply-To: <CAMbW9o+BgiO7ELwk-=ZKFPPTGTOMTEYXbU=6iZMhht8xHmNgdQ@mail.gmail.com>
References: <CAMbW9oKg=1U3YUUAJMwfWP+H-2DQ=+Lf6W=ZR6Jd7pNv74nXTg@mail.gmail.com>
	<004ED0CB-4E46-4F01-B9F8-F82456D4F7F9@scarlet-internet.nl>
	<CAMbW9o+BgiO7ELwk-=ZKFPPTGTOMTEYXbU=6iZMhht8xHmNgdQ@mail.gmail.com>
Message-ID: <CAMWiMy9yiTE9MaesTGgHPkCfGED-Qh6W9C_ZYm=WsywHU88f-w@mail.gmail.com>

Try the user_data command

http://wiki.freeswitch.org/wiki/Mod_commands#user_data

The wiki shoves a good example on how this is used.

On Thu, Jun 28, 2012 at 12:04 PM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> Hi Leon,
>
> Does that allow me to search by attrs other than ID? I need to get the id
> of a user given another attr (such as alias), but looking for a minimal
> approach.
>
> Basically all i need is to have N mailboxes, each with a DDI to leave
> messages and a DDI to check them. Obviously I can just use one of the DDI's
> as the user ID, but to check i need to find the user id for the other
> number.
>
> I can do this with mod_xml_curl easily enough, was just wondering if there
> was a simpler way.
>
> On 28 June 2012 14:36, Leon de Rooij <leon at scarlet-internet.nl> wrote:
>
>> You can use the set_user dp app to set all user variables as channel
>> variables. You can also supply a prefix string to keep these new variables
>> separate from wha was already set.
>>
>> See here: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user
>>
>> regards,
>>
>> Leon
>>
>>
>> On Jun 28, 2012, at 11:09, Andrew Cassidy wrote:
>>
>> Hi All,
>>
>> Right, I have a very basic application I'm building around voicemail.
>> Each user has 2 attrs that I need to search by. user_exists works fine, but
>> doesn't return me any data other then the fact the user exists, and
>> user_data only searches the id attr.
>>
>> Is there any way I can do that in just the dialplan with what is already
>> available?
>>
>> In the most simple form I need the user id for a number_alias.
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>>
>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>  *03300 100 961
>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>
>>  _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmsucasas at me.com  Thu Jun 28 23:45:35 2012
From: jmsucasas at me.com (Jose Miguel Sucasas Mejuto)
Date: Thu, 28 Jun 2012 21:45:35 +0200
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 72, Issue 260
In-Reply-To: <mailman.13166.1340904827.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.13166.1340904827.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <A4E892B1-E1A8-4CA9-A736-C4D20EFCEFAD@me.com>

> Thanks Anthony,  adding:

<X-PRE-PROCESS cmd="set" data="sip_jitter_buffer_during_bridge=true"/> 

to vars.xml file solved my problem.

Regards.

> 
> De: Anthony Minessale <anthony.minessale at gmail.com>
> Asunto: Re: [Freeswitch-users] Always PAUSE Jitterbuffer
> Fecha: 28 de junio de 2012 19:33:16 GMT+02:00
> Para: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Responder a: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> set sip_jitter_buffer_during_bridge=true globally or per channel basis
> 
> 
> On Wed, Jun 27, 2012 at 3:17 PM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
>> Maybe this is what you are looking for? I think FS usually wants the endpoints to handle this during a bridge, setting this var should force it to stay enabled.
>> 
>> http://wiki.freeswitch.org/wiki/Variable_sip_jitter_buffer_during_bridge
>> 
>> /Peter
>> 
>> 27 jun 2012 kl. 22:11 skrev "Jose Miguel Sucasas Mejuto" <jmsucasas at me.com<mailto:jmsucasas at me.com>>:
>> 
>> Hello,
>> 
>> I have installed Freeswitch from GIT (currenty I have installed 1.2.0-rc2+git~20120625T203158Z~7dc9a9cacc version).  Because some users connects from 3G network using a smartphone,  I try to activate JitterBuffer and Packet Loss Concealing in both directions (in a call with U1 and U2,  I try to activate 90ms jitterbuffer in audio flow U1->U2 and another 90ms jitterbuffer in audio flow U2->U1).  To achieve this,  I have this in dialplan (default.xml file):
>> 
>> <action application="cng_plc"/>
>> <!-- A-leg JitterBuffer -->
>> <action application="set" data="jitterbuffer_msec=90"/>
>> <action application="set" data="sip_jitter_buffer_during_bridge=true"/>
>> <!-- A-leg JitterBuffer debug, doesn't work -->
>> <action application="jitterbuffer" data="debug:${uuid}"/>
>> <!-- In bridge application, I try to activate B-leg JitterBuffer -->
>> <action application="bridge" data="{jitterbuffer_msec=90,absolute_codec_string='SILK at 16000h,iLBC at 30i,G729,AMR,AMR-WB,H263,H264'}user/${dialed_extension}@${domain_name}"/>
>> 
>> User 1002 is on the same network as Freeswitch box,  connected by ethernet line,  and user 1001 is on 3G,  when user 1002 calls to user 1001 (both of them uses iLBC30 codec),  on fs_cli it seems that JitterBuffer is activated (in bold green)
>> 
>> 2012-06-27 20:55:14.929462 [NOTICE] sofia.c:5930 Ring-Ready sofia/internal/sip:1001 at 10.10.57.82:50203!
>> 2012-06-27 20:55:14.929462 [INFO] switch_ivr_originate.c:1156 Sending early media
>> 2012-06-27 20:55:14.929462 [INFO] switch_nat.c:590 NAT port mapping disabled
>> 2012-06-27 20:55:14.929462 [INFO] switch_nat.c:590 NAT port mapping disabled
>> 2012-06-27 20:55:14.929462 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [proceeding][180]
>> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3269 AUDIO RTP [sofia/internal/1002 at sukmad.dyndns-home.com:58180] 79.150.184.57 port 11154 -> 192.168.0.15 port 10000 codec: 98 ms: 30
>> 2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:1861 Starting timer [soft] 240 bytes per 30ms
>> 2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:2207 RTCP send rate is: 5000 and packet rate is: 30000 Remote Port: 10001
>> 2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 192.168.0.15:10001
>> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to 90ms (3 frames)
>> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send payload to 101
>> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive payload to 101
>> 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3566 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Set rtp dtmf delay to 40
>> 2012-06-27 20:55:14.929462 [DEBUG] mod_sofia.c:2606 Ring SDP:
>> v=0
>> o=FreeSWITCH 1340812160 1340812161 IN IP4 79.150.184.57
>> s=FreeSWITCH
>> c=IN IP4 79.150.184.57
>> t=0 0
>> m=audio 11154 RTP/AVP 98 101
>> a=rtpmap:98 iLBC/8000
>> a=fmtp:98 mode=30
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:30
>> a=sendrecv
>> 
>> 2012-06-27 20:55:14.929462 [NOTICE] mod_sofia.c:2609 Pre-Answer sofia/internal/1002 at sukmad.dyndns-home.com:58180!
>> 2012-06-27 20:55:14.929462 [DEBUG] switch_channel.c:3042 (sofia/internal/1002 at sukmad.dyndns-home.com:58180) Callstate Change RINGING -> EARLY
>> 2012-06-27 20:55:14.929462 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
>> 2012-06-27 20:55:14.929462 [DEBUG] switch_ivr_originate.c:1205 Raw Codec Activation Success L16 at 8000hz 1 channel 30ms
>> 2012-06-27 20:55:14.929462 [DEBUG] switch_core_codec.c:219 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Push codec L16:70
>> 2012-06-27 20:55:14.929462 [DEBUG] switch_ivr_originate.c:1268 Play Ringback Tone [%(2000,4000,440,480)]
>> 2012-06-27 20:55:14.929462 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
>> 2012-06-27 20:55:14.929462 [DEBUG] sofia.c:5831 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 skipping state [early][183]
>> 2012-06-27 20:55:14.939460 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
>> 2012-06-27 20:55:14.939460 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
>> 2012-06-27 20:55:14.939460 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 entering state [proceeding][180]
>> 2012-06-27 20:55:14.939460 [NOTICE] sofia.c:5930 Ring-Ready sofia/internal/sip:1001 at 31.4.238.246:22268!
>> 2012-06-27 20:55:15.029459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
>> 2012-06-27 20:55:15.029459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
>> 2012-06-27 20:55:15.029459 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 entering state [proceeding][180]
>> 2012-06-27 20:55:15.069457 [DEBUG] switch_rtp.c:3527 Correct ip/port confirmed.
>> 
>> When user 1001 answer the call, JitterBuffer is paused (in bold red):
>> 
>> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
>> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [completing][200]
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5849 Remote SDP:
>> v=0
>> o=- 3549812097 3549812099 IN IP4 10.10.57.82
>> s=cpc_med
>> c=IN IP4 10.10.57.82
>> b=AS:226
>> t=0 0
>> m=audio 22010 RTP/AVP 99 101
>> a=rtpmap:99 iLBC/8000
>> a=fmtp:99 mode=30
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> m=video 0 RTP/AVP 103
>> c=IN IP4 10.10.57.82
>> b=AS:200000
>> a=rtpmap:103 H264/90000
>> 
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3919 Deciding whether to pass zrtp-hash between legs
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
>> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
>> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001 at 10.10.57.82:50203 entering state [ready][200]
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3919 Deciding whether to pass zrtp-hash between legs
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [iLBC:99:8000:30:0]/[SILK:119:16000:20:30000]
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5148 Set 2833 dtmf send payload to 101
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [iLBC:99:8000:30:0]/[iLBC:97:8000:30:13330]
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3020 Set Codec sofia/internal/sip:1001 at 10.10.57.82:50203 iLBC/8000 30 ms 240 samples 13330 bits
>> 2012-06-27 20:55:26.079458 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:1001 at 10.10.57.82:50203 Original read codec set to iLBC:97
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5148 Set 2833 dtmf send payload to 101
>> 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3269 AUDIO RTP [sofia/internal/sip:1001 at 10.10.57.82:50203] 79.150.184.57 port 10388 -> 10.10.57.82 port 22010 codec: 99 ms: 30
>> 2012-06-27 20:55:26.079458 [DEBUG] switch_rtp.c:1861 Starting timer [soft] 240 bytes per 30ms
>> 2012-06-27 20:55:26.089461 [DEBUG] switch_rtp.c:2207 RTCP send rate is: 5000 and packet rate is: 30000 Remote Port: 22011
>> 2012-06-27 20:55:26.089461 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 10.10.57.82:22011
>> 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to 90ms (3 frames)
>> 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send payload to 101
>> 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive payload to 101
>> 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3566 sofia/internal/sip:1001 at 10.10.57.82:50203 Set rtp dtmf delay to 40
>> 2012-06-27 20:55:26.089461 [DEBUG] switch_channel.c:3301 (sofia/internal/sip:1001 at 10.10.57.82:50203) Callstate Change RINGING -> ACTIVE
>> 2012-06-27 20:55:26.089461 [DEBUG] switch_channel.c:3313 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
>> 2012-06-27 20:55:26.089461 [NOTICE] sofia.c:6573 Channel [sofia/internal/sip:1001 at 10.10.57.82:50203] has been answered
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_codec.c:244 sofia/internal/1002 at sukmad.dyndns-home.com:58180 Restore previous codec iLBC:97.
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2899 (sofia/internal/sip:1001 at 192.168.0.11:50199) Callstate Change RINGING -> HANGUP
>> 2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3257 Hangup sofia/internal/sip:1001 at 192.168.0.11:50199 [CS_CONSUME_MEDIA] [LOSE_RACE]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2922 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [KILL]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_HANGUP
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2899 (sofia/internal/sip:1001 at 31.4.238.246:22268) Callstate Change RINGING -> HANGUP
>> 2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3257 Hangup sofia/internal/sip:1001 at 31.4.238.246:22268 [CS_CONSUME_MEDIA] [LOSE_RACE]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2922 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [KILL]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 192.168.0.11:50199) State HANGUP
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001 at 192.168.0.11:50199 hanging up, cause: LOSE_RACE
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_HANGUP
>> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:527 Sending CANCEL to sofia/internal/sip:1001 at 192.168.0.11:50199
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard HANGUP, cause: LOSE_RACE
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 192.168.0.11:50199) State HANGUP going to sleep
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/sip:1001 at 192.168.0.11:50199) State Change CS_HANGUP -> CS_REPORTING
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_REPORTING
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 192.168.0.11:50199) State REPORTING
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard REPORTING, cause: LOSE_RACE
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 192.168.0.11:50199) State REPORTING going to sleep
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 31.4.238.246:22268) State HANGUP
>> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001 at 31.4.238.246:22268 hanging up, cause: LOSE_RACE
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at 192.168.0.11:50199) State Change CS_REPORTING -> CS_DESTROY
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 192.168.0.11:50199 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1429 Session 15 (sofia/internal/sip:1001 at 192.168.0.11:50199) Locked, Waiting on external entities
>> 2012-06-27 20:55:26.109459 [INFO] switch_nat.c:590 NAT port mapping disabled
>> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:527 Sending CANCEL to sofia/internal/sip:1001 at 31.4.238.246:22268
>> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:754 Local SDP sofia/internal/1002 at sukmad.dyndns-home.com:58180:
>> v=0
>> o=FreeSWITCH 1340812160 1340812162 IN IP4 79.150.184.57
>> s=FreeSWITCH
>> c=IN IP4 79.150.184.57
>> t=0 0
>> m=audio 11154 RTP/AVP 98 101
>> a=rtpmap:98 iLBC/8000
>> a=fmtp:98 mode=30
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:30
>> a=sendrecv
>> m=video 10766 RTP/AVP 34 103
>> a=rtpmap:34 H263/90000
>> a=rtpmap:103 H264/90000
>> 
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard HANGUP, cause: LOSE_RACE
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001 at 31.4.238.246:22268) State HANGUP going to sleep
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/sip:1001 at 31.4.238.246:22268) State Change CS_HANGUP -> CS_REPORTING
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_REPORTING
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:3301 (sofia/internal/1002 at sukmad.dyndns-home.com:58180) Callstate Change EARLY -> ACTIVE
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 31.4.238.246:22268) State REPORTING
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard REPORTING, cause: LOSE_RACE
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001 at 31.4.238.246:22268) State REPORTING going to sleep
>> 2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3284 Channel [sofia/internal/1002 at sukmad.dyndns-home.com:58180] has been answered
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at 31.4.238.246:22268) State Change CS_REPORTING -> CS_DESTROY
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 31.4.238.246:22268 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1429 Session 17 (sofia/internal/sip:1001 at 31.4.238.246:22268) Locked, Waiting on external entities
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] sofia.c:5838 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 entering state [completed][200]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_originate.c:3341 Originate Resulted in Success: [sofia/internal/sip:1001 at 10.10.57.82:50203]
>> 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1447 Session 15 (sofia/internal/sip:1001 at 192.168.0.11:50199) Ended
>> 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1447 Session 17 (sofia/internal/sip:1001 at 31.4.238.246:22268) Ended
>> 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/sip:1001 at 192.168.0.11:50199 [CS_DESTROY]
>> 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/sip:1001 at 31.4.238.246:22268 [CS_DESTROY]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/sip:1001 at 31.4.238.246:22268) Callstate Change HANGUP -> DOWN
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/sip:1001 at 192.168.0.11:50199) Callstate Change HANGUP -> DOWN
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/sip:1001 at 31.4.238.246:22268) Running State Change CS_DESTROY
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 31.4.238.246:22268) State DESTROY
>> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001 at 31.4.238.246:22268 SOFIA DESTROY
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at 31.4.238.246:22268 Standard DESTROY
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/sip:1001 at 192.168.0.11:50199) Running State Change CS_DESTROY
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 31.4.238.246:22268) State DESTROY going to sleep
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 192.168.0.11:50199) State DESTROY
>> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001 at 192.168.0.11:50199 SOFIA DESTROY
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at 192.168.0.11:50199 Standard DESTROY
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001 at 192.168.0.11:50199) State DESTROY going to sleep
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_originate.c:3341 Originate Resulted in Success: [sofia/internal/sip:1001 at 10.10.57.82:50203]
>> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:1603 sofia/internal/sip:1001 at 10.10.57.82:50203 PAUSE Jitterbuffer
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_bridge.c:1359 (sofia/internal/sip:1001 at 10.10.57.82:50203) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001 at 10.10.57.82:50203) Running State Change CS_EXCHANGE_MEDIA
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:443 (sofia/internal/sip:1001 at 10.10.57.82:50203) State EXCHANGE_MEDIA
>> 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:582 SOFIA EXCHANGE_MEDIA
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] sofia.c:5838 Channel sofia/internal/1002 at sukmad.dyndns-home.com:58180 entering state [ready][200]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/sip:1001 at 10.10.57.82:50203 [BREAK]
>> 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/1002 at sukmad.dyndns-home.com:58180 [BREAK]
>> 2012-06-27 20:55:26.469460 [INFO] switch_rtp.c:3510 Auto Changing port from 10.10.57.82:22010 to 31.4.238.246:22267
>> 2012-06-27 20:55:26.469460 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 31.4.238.246:22268
>> 2012-06-27 20:55:38.009460 [DEBUG] sofia_reg.c:371 Registered sip.netelip.com<http://sip.netelip.com>
>> 
>> 
>> I try a lot of test and all of them have the same issue,  Why JitterBuffer is paused?.  Is it possible to force JitterBuffer?
>> 
>> 
>> Thanks in advance,
>> 
>> Regards
>> 
>> 
>> 
>> !DSPAM:4feb659c32769194324459!
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
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From edwin.hoexum at office.ziggo.nl  Thu Jun 28 22:41:59 2012
From: edwin.hoexum at office.ziggo.nl (Hoexum, Edwin)
Date: Thu, 28 Jun 2012 20:41:59 +0200
Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with
 freeswitch as trancoding device SILK to G711a
Message-ID: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern>

Gents,

I am trying to do some load testing on a freeswitch for transcoding from SILK to G711a.  The question is:  Is  the limit  for RTP sessions limited to CPU power or are there also limits on the freeswitch application and can I tune freeswitch to do the job for 1500 sessions or more.

At the moment freeswitch is the best transcoding device for SILK to G711a that I know of. Also compared to the big companies who deliver transcoding devices.

SETUP:

3 X G7  HP DL 380 with 16 G mem 6 core 12 CPU's

1 for SIPP UAC  (1Gb interface for media and Signaling)
1 for Freeswitch (trancoding function SILK to G711a)  1Gb interface for media and Signaling
1 for SIPP UAS (1Gb interface for media and Signaling)

First tuning OS (with help from the wiki and some LINUX sysctl tuning)


SIPP UAC <-> Freeswitch


root at mnd-rc0001-srvot02:~# iperf -c 172.23.125.46 -u -d -b 400m
------------------------------------------------------------
Server listening on UDP port 5001
Receiving 1470 byte datagrams
UDP buffer size: 64.0 KByte (default)
------------------------------------------------------------
------------------------------------------------------------
Client connecting to 172.23.125.46, UDP port 5001
Sending 1470 byte datagrams
UDP buffer size: 64.0 KByte (default)
------------------------------------------------------------
[  4] local 172.23.125.42 port 46443 connected with 172.23.125.46 port 5001
[  3] local 172.23.125.42 port 5001 connected with 172.23.125.46 port 39259
[ ID] Interval       Transfer     Bandwidth
[  4]  0.0-10.0 sec   483 MBytes   405 Mbits/sec
[  4] Sent 344796 datagrams
[  3]  0.0-10.0 sec   483 MBytes   406 Mbits/sec   0.001 ms    0/344828 (0%)
[  3]  0.0-10.0 sec  1 datagrams received out-of-order
[  4] Server Report:
[  4]  0.0-10.0 sec   483 MBytes   405 Mbits/sec   0.004 ms  261/344795 (0.076%)
[  4]  0.0-10.0 sec  1 datagrams received out-of-order
root at mnd-rc0001-srvot02:~#



Linux mnd-rc0001-srvot06 2.6.32-28-vserver #55~ppa1-Ubuntu SMP Fri Feb 4 21:25:09 UTC 2011 x86_64 GNU/Linux
Ubuntu 10.04 LTS

Welcome to Ubuntu!
* Documentation:  https://help.ubuntu.com/
Last login: Thu Jun 21 15:01:35 2012 from 172.21.88.65
root at mnd-rc0001-srvot06:~# iperf -s -u -i1
------------------------------------------------------------
Server listening on UDP port 5001
Receiving 1470 byte datagrams
UDP buffer size: 64.0 KByte (default)
------------------------------------------------------------
------------------------------------------------------------
Client connecting to 172.23.125.42, UDP port 5001
Sending 1470 byte datagrams
UDP buffer size: 64.0 KByte (default)
------------------------------------------------------------
[  5] local 172.23.125.46 port 39259 connected with 172.23.125.42 port 5001
[  3] local 172.23.125.46 port 5001 connected with 172.23.125.42 port 46443
[ ID] Interval       Transfer     Bandwidth
[  5]  0.0- 1.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  0.0- 1.0 sec  48.4 MBytes    406 Mbits/sec  0.004 ms  262/34754 (0.75%)
[  5]  1.0- 2.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  1.0- 2.0 sec  48.3 MBytes    405 Mbits/sec  0.005 ms    0/34476 (0%)
[  5]  2.0- 3.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  2.0- 3.0 sec  48.3 MBytes    405 Mbits/sec  0.005 ms    0/34480 (0%)
[  5]  3.0- 4.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  3.0- 4.0 sec  48.3 MBytes    406 Mbits/sec  0.004 ms    0/34483 (0%)
[  5]  4.0- 5.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  4.0- 5.0 sec  48.3 MBytes    406 Mbits/sec  0.005 ms    0/34482 (0%)
[  5]  5.0- 6.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  5.0- 6.0 sec  48.3 MBytes    405 Mbits/sec  0.004 ms    0/34474 (0%)
[  5]  6.0- 7.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  6.0- 7.0 sec  48.3 MBytes    405 Mbits/sec  0.005 ms    0/34477 (0%)
[  5]  7.0- 8.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  7.0- 8.0 sec  48.3 MBytes    405 Mbits/sec  0.004 ms    0/34480 (0%)
[  5]  8.0- 9.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  8.0- 9.0 sec  48.3 MBytes    406 Mbits/sec  0.005 ms    0/34482 (0%)
[  5]  9.0-10.0 sec  48.3 MBytes    406 Mbits/sec
[  5]  0.0-10.0 sec    483 MBytes    406 Mbits/sec
[  5] Sent 344829 datagrams
[  3]  0.0-10.0 sec    483 MBytes    405 Mbits/sec  0.004 ms  261/344795 (0.076%)
[  3]  0.0-10.0 sec  1 datagrams received out-of-order
[  5] Server Report:
[  5]  0.0-10.0 sec    483 MBytes    406 Mbits/sec  0.000 ms    0/344828 (0%)
[  5]  0.0-10.0 sec  1 datagrams received out-of-order


UDP traffic looks fine.

SIPP (UAC) inject SILK RTP (PCAP ) and send this to Freeswitch. Freeswitch send this G711a to the SIPP( UAS).

I doing some testing and until  700 sessions it look fine (Packed loss <0.1%  and delay <40ms)

If I do more sessions 10% of packet have no packet loss but delay of more the 100ms.


SIPP uac side:

------------------------------ Scenario Screen -------- [1-9]: Change Screen --
  Call-rate(length)   Port   Total-time  Total-calls  Remote-host
  50.0(0 ms)/1.000s   5060     230.53 s         2000  172.23.125.12:5060(UDP)

  Call limit reached (-m 2000), 0.000 s period  0 ms scheduler resolution
  0 calls (limit 4500)                   Peak was 2000 calls, after 40 s
  0 Running, 1654 Paused, 0 Woken up
  0 dead call msg (discarded)            6459 out-of-call msg (discarded)
  1 open sockets

                                 Messages  Retrans   Timeout   Unexpected-Msg
       Pause [   2000ms]         2000                          0
      INVITE ---------->         2000      0         0
         100 <----------         2000      0         0         0
         183 <----------         0         0         0         0
         400 <----------         0         0         0         0
         401 <----------         0         0         0         0
         403 <----------         0         0         0         0
         404 <----------         0         0         0         0
         405 <----------         0         0         0         0
         406 <----------         0         0         0         0
         408 <----------         0         0         0         0
         415 <----------         0         0         0         0
         433 <----------         0         0         0         0
         480 <----------         0         0         0         0
         481 <----------         0         0         0         0
         484 <----------         0         0         0         0
         485 <----------         0         0         0         0
         486 <----------         0         0         0         0
         487 <----------         0         0         0         0
         488 <----------         0         0         0         0
         489 <----------         0         0         0         0
         500 <----------         0         0         0         0
         502 <----------         0         0         0         0
         503 <----------         0         0         0         0
         180 <----------         2000      0         0         0
         200 <----------  E-RTD1 2000      0         0         0
         ACK ---------->         2000      0
       Pause [    500ms]         2000                          0
              [ NOP ]
         BYE <----------         2000      0         0         0
         200 ---------->         2000      0
------------------------------ Test Terminated --------------------------------


----------------------------- Statistics Screen ------- [1-9]: Change Screen --
  Start Time             | 2012-06-28   20:17:17:312    1340907437.312927
  Last Reset Time        | 2012-06-28   20:21:07:857    1340907667.857562
  Current Time           | 2012-06-28   20:21:07:857    1340907667.857769
-------------------------+---------------------------+--------------------------
  Counter Name           | Periodic value            | Cumulative value
-------------------------+---------------------------+--------------------------
  Elapsed Time           | 00:00:00:000              | 00:03:50:544
  Call Rate              |    0.000 cps              |    8.675 cps
-------------------------+---------------------------+--------------------------
  Incoming call created  |        0                  |        0
  OutGoing call created  |        0                  |     2000
  Total Call created     |                           |     2000
  Current Call           |        0                  |



SIPP UAS side:

------------------------------ Scenario Screen -------- [1-9]: Change Screen --
  Port   Total-time  Total-calls  Transport
  5060      65.10 s         2000  UDP

  0 new calls during 1.001 s period      1 ms scheduler resolution
  2000 calls                             Peak was 2000 calls, after 44 s
  0 Running, 2002 Paused, 4 Woken up
  0 dead call msg (discarded)
  3 open sockets
  2021 Total echo RTP pckts 1st stream   5.327 last period RTP rate (kB/s)
  0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate (kB/s)

                                 Messages  Retrans   Timeout   Unexpected-Msg
 ----------> INVITE             2000      0         0         0
  <---------- 100                2000      0
  [    500ms] Pause              2000                          0
  <---------- 180                2000      0
  <---------- 200                2000      0         0
  ----------> ACK                2000      0         0         0
  [     3:08] Pause              2000                          0
  <---------- BYE                0         0         0
  ----------> 200                0         0         0         0
------------------------------ Sipp Server Mode -------------------------------


##############################################################################

MPSTAT -A 5   on freeswitch:


8:29:50     CPU    %usr   %nice    %sys %iowait    %irq   %soft  %steal  %guest   %idle
18:29:55     all   24.84    0.00   25.63    0.00    0.02    7.46    0.00    0.00   42.05
18:29:55       0   43.52    0.00   26.72    0.00    0.00    4.05    0.00    0.00   25.71
18:29:55       1   27.88    0.00   39.18    0.00    0.00    3.12    0.00    0.00   29.82
18:29:55       2   27.70    0.00   33.40    0.00    0.00    4.17    0.00    0.00   34.72
18:29:55       3   25.57    0.00   35.98    0.00    0.00    3.03    0.00    0.00   35.42
18:29:55       4   27.36    0.00   27.56    0.00    0.00    2.56    0.00    0.00   42.52
18:29:55       5   25.45    0.00   24.44    0.00    0.00    2.22    0.00    0.00   47.88
18:29:55       6   24.77    0.00   31.28    0.00    0.00    2.98    0.00    0.00   40.97
18:29:55       7   23.05    0.00   23.81    0.00    0.00    2.67    0.00    0.00   50.48
18:29:55       8   22.22    0.00   23.03    0.00    0.20    2.02    0.00    0.00   52.53
18:29:55       9    4.84    0.00    5.71    0.00    0.00   62.86    0.00    0.00   26.59
18:29:55      10   23.36    0.00   17.01    0.00    0.00    2.46    0.00    0.00   57.17
18:29:55      11   20.21    0.00   15.11    0.00    0.00    2.98    0.00    0.00   61.70

18:29:50     CPU    intr/s
18:29:55     all  30433.80
18:29:55       0      0.00
18:29:55       1      0.00
18:29:55       2      0.00
18:29:55       3      0.00
18:29:55       4      0.00
18:29:55       5      0.00
18:29:55       6      0.00
18:29:55       7      0.00
18:29:55       8      0.00
18:29:55       9      0.00
18:29:55      10      0.00
18:29:55      11      0.00

18:29:50     CPU     0/s     1/s     8/s
18:29:55       0    0.00    0.00    0.00
18:29:55       1    0.00    0.00    0.00
18:29:55       2    0.00    0.00    0.00
18:29:55       3    0.00    0.00    0.00
18:29:55       4    0.00    0.00    0.00
18:29:55       5    0.00    0.00    0.00
18:29:55       6    0.00    0.00    0.00
18:29:55       7    0.00    0.00    0.00
18:29:55       8    0.00    0.00    0.00
18:29:55       9    0.00    0.00    0.00
18:29:55      10    0.00    0.00    0.00
18:29:55      11    0.00    0.00    0.00

I tested it with G711a to G711a and got the same result.

I tested also some RTP parameters like

<param name="rtp-autoflush-during-bridge" value="true"/>  or false

<param name="rtp-autoflush" value="true"/>   or false

The lot  of the module are not activated like CDR stuff.

mvg,
Edwin


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From edwin.hoexum at office.ziggo.nl  Thu Jun 28 22:48:07 2012
From: edwin.hoexum at office.ziggo.nl (Hoexum, Edwin)
Date: Thu, 28 Jun 2012 20:48:07 +0200
Subject: [Freeswitch-users] RTP delay > 100ms by performance testing
 with freeswitch as trancoding device SILK to G711a
Message-ID: <2C18241C6768CF469C7E432A25327AD513529739DF@MAIL-WPV01.office.intern>

Some results from  tshark on the Freeswitch SBC

tshark -i any -f " portrange 6000-6030 or portrange  5430-5460 or port 5060" -w rtp_test_2000_final6.pcap

tshark -i any  -r rtp_test_2000_final6.pcap  -q -z rtp,streams

========================= RTP Streams ========================
    Src IP addr  Port    Dest IP addr  Port       SSRC          Payload  Pkts         Lost   Max Delta(ms)  Max Jitter(ms) Mean Jitter(ms) Problems?
  172.23.125.42  6000   172.23.125.46 21656 0x5D628E99             SILK  9384     8 (0.1%)            0.00            0.00            0.00 X
  172.23.125.42  6004   172.23.125.46 31046 0x5D628E99             SILK  9387     9 (0.1%)            0.00            0.00            0.00 X
  172.23.125.46 21656   172.23.125.42  6000 0xB837AEF9             SILK  5938     4 (0.1%)            0.00            0.00            0.00 X
  172.23.125.42  6008   172.23.125.46 30098 0x5D628E99             SILK  9385     9 (0.1%)            0.00            0.00            0.00 X
  172.23.125.42  6012   172.23.125.46 27704 0x5D628E99             SILK  9386     7 (0.1%)            0.00            0.00            0.00 X
  172.23.125.42  6016   172.23.125.46 31888 0x5D628E99             SILK  9387     8 (0.1%)            0.00            0.00            0.00 X
  172.23.125.42  6020   172.23.125.46 23834 0x5D628E99             SILK  9383    11 (0.1%)            0.00            0.00            0.00 X
  172.23.125.42  6024   172.23.125.46 28654 0x5D628E99             SILK  9384    10 (0.1%)            0.00            0.00            0.00 X
  172.23.125.42  6028   172.23.125.46 20508 0x5D628E99             SILK  9384     9 (0.1%)            0.00            0.00            0.00 X
172.23.125.111 17886  172.23.125.108  5430 0x8C7D01EB ITU-T G.711 PCMA  6321     7 (0.1%)          164.75          255.72           24.33 X
172.23.125.111 20028  172.23.125.108  5434 0x902B9E9B ITU-T G.711 PCMA  6250     5 (0.1%)          164.71          261.51           24.59 X
172.23.125.111 24484  172.23.125.108  5442 0x902CEEAB ITU-T G.711 PCMA  6360     3 (0.0%)          185.83          211.48           24.34 X
172.23.125.111 23012  172.23.125.108  5438 0x9904D0BB ITU-T G.711 PCMA  6359     2 (0.0%)          155.48          254.51           24.20 X
172.23.125.111 26100  172.23.125.108  5446 0x9906A10B ITU-T G.711 PCMA  6354    10 (0.2%)          170.63          230.68           24.24 X
172.23.125.111 25534  172.23.125.108  5450 0x8C7F224B ITU-T G.711 PCMA  6210     7 (0.1%)          176.21          276.95           24.67 X
172.23.125.111 24274  172.23.125.108  5454 0x8C80926B ITU-T G.711 PCMA  6294     2 (0.0%)          199.27          253.83           24.49 X
172.23.125.111 17884  172.23.125.108  5458 0x902E3EBB ITU-T G.711 PCMA  6378    10 (0.2%)          166.43          252.46           24.23 X
172.23.125.111 24240  172.23.125.108  6006 0x92E4386E ITU-T G.711 PCMA  6150     3 (0.0%)          207.51          246.84           24.95 X
172.23.125.111 27234  172.23.125.108  6002 0x8C62004E ITU-T G.711 PCMA  6280     3 (0.0%)          240.75          222.93           24.65 X
172.23.125.111 23198  172.23.125.108  6010 0x8C63706E ITU-T G.711 PCMA  6312     6 (0.1%)          152.98          210.53           24.47 X
172.23.125.111 22242  172.23.125.108  6014 0x98EF8A3E ITU-T G.711 PCMA  6358     2 (0.0%)          162.71          221.53           24.33 X
172.23.125.111 31472  172.23.125.108  6018 0x85949B9E ITU-T G.711 PCMA  6350     8 (0.1%)          162.41          205.65           24.39 X
172.23.125.111 18000  172.23.125.108  6022 0x98F0DA4E ITU-T G.711 PCMA  6382     9 (0.1%)          161.46          201.20           24.25 X
172.23.125.111 21464  172.23.125.108  6026 0x8C64FC2E ITU-T G.711 PCMA  6282     5 (0.1%)          159.87          218.82           24.69 X
172.23.125.111 31828  172.23.125.108  6030 0x98F22A5E ITU-T G.711 PCMA  6283     3 (0.0%)          151.56          223.87           24.52 X

mvg,
Edwin

Van: Hoexum, Edwin
Verzonden: donderdag 28 juni 2012 20:42
Aan: 'FreeSWITCH-users at lists.freeswitch.org'
Onderwerp: RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a

Gents,

I am trying to do some load testing on a freeswitch for transcoding from SILK to G711a.  The question is:  Is  the limit  for RTP sessions limited to CPU power or are there also limits on the freeswitch application and can I tune freeswitch to do the job for 1500 sessions or more.

At the moment freeswitch is the best transcoding device for SILK to G711a that I know of. Also compared to the big companies who deliver transcoding devices.

SETUP:

3 X G7  HP DL 380 with 16 G mem 6 core 12 CPU's

1 for SIPP UAC  (1Gb interface for media and Signaling)
1 for Freeswitch (trancoding function SILK to G711a)  1Gb interface for media and Signaling
1 for SIPP UAS (1Gb interface for media and Signaling)

First tuning OS (with help from the wiki and some LINUX sysctl tuning)


SIPP UAC <-> Freeswitch


root at mnd-rc0001-srvot02:~# iperf -c 172.23.125.46 -u -d -b 400m
------------------------------------------------------------
Server listening on UDP port 5001
Receiving 1470 byte datagrams
UDP buffer size: 64.0 KByte (default)
------------------------------------------------------------
------------------------------------------------------------
Client connecting to 172.23.125.46, UDP port 5001
Sending 1470 byte datagrams
UDP buffer size: 64.0 KByte (default)
------------------------------------------------------------
[  4] local 172.23.125.42 port 46443 connected with 172.23.125.46 port 5001
[  3] local 172.23.125.42 port 5001 connected with 172.23.125.46 port 39259
[ ID] Interval       Transfer     Bandwidth
[  4]  0.0-10.0 sec   483 MBytes   405 Mbits/sec
[  4] Sent 344796 datagrams
[  3]  0.0-10.0 sec   483 MBytes   406 Mbits/sec   0.001 ms    0/344828 (0%)
[  3]  0.0-10.0 sec  1 datagrams received out-of-order
[  4] Server Report:
[  4]  0.0-10.0 sec   483 MBytes   405 Mbits/sec   0.004 ms  261/344795 (0.076%)
[  4]  0.0-10.0 sec  1 datagrams received out-of-order
root at mnd-rc0001-srvot02:~#



Linux mnd-rc0001-srvot06 2.6.32-28-vserver #55~ppa1-Ubuntu SMP Fri Feb 4 21:25:09 UTC 2011 x86_64 GNU/Linux
Ubuntu 10.04 LTS

Welcome to Ubuntu!
* Documentation:  https://help.ubuntu.com/
Last login: Thu Jun 21 15:01:35 2012 from 172.21.88.65
root at mnd-rc0001-srvot06:~# iperf -s -u -i1
------------------------------------------------------------
Server listening on UDP port 5001
Receiving 1470 byte datagrams
UDP buffer size: 64.0 KByte (default)
------------------------------------------------------------
------------------------------------------------------------
Client connecting to 172.23.125.42, UDP port 5001
Sending 1470 byte datagrams
UDP buffer size: 64.0 KByte (default)
------------------------------------------------------------
[  5] local 172.23.125.46 port 39259 connected with 172.23.125.42 port 5001
[  3] local 172.23.125.46 port 5001 connected with 172.23.125.42 port 46443
[ ID] Interval       Transfer     Bandwidth
[  5]  0.0- 1.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  0.0- 1.0 sec  48.4 MBytes    406 Mbits/sec  0.004 ms  262/34754 (0.75%)
[  5]  1.0- 2.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  1.0- 2.0 sec  48.3 MBytes    405 Mbits/sec  0.005 ms    0/34476 (0%)
[  5]  2.0- 3.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  2.0- 3.0 sec  48.3 MBytes    405 Mbits/sec  0.005 ms    0/34480 (0%)
[  5]  3.0- 4.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  3.0- 4.0 sec  48.3 MBytes    406 Mbits/sec  0.004 ms    0/34483 (0%)
[  5]  4.0- 5.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  4.0- 5.0 sec  48.3 MBytes    406 Mbits/sec  0.005 ms    0/34482 (0%)
[  5]  5.0- 6.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  5.0- 6.0 sec  48.3 MBytes    405 Mbits/sec  0.004 ms    0/34474 (0%)
[  5]  6.0- 7.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  6.0- 7.0 sec  48.3 MBytes    405 Mbits/sec  0.005 ms    0/34477 (0%)
[  5]  7.0- 8.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  7.0- 8.0 sec  48.3 MBytes    405 Mbits/sec  0.004 ms    0/34480 (0%)
[  5]  8.0- 9.0 sec  48.3 MBytes    406 Mbits/sec
[  3]  8.0- 9.0 sec  48.3 MBytes    406 Mbits/sec  0.005 ms    0/34482 (0%)
[  5]  9.0-10.0 sec  48.3 MBytes    406 Mbits/sec
[  5]  0.0-10.0 sec    483 MBytes    406 Mbits/sec
[  5] Sent 344829 datagrams
[  3]  0.0-10.0 sec    483 MBytes    405 Mbits/sec  0.004 ms  261/344795 (0.076%)
[  3]  0.0-10.0 sec  1 datagrams received out-of-order
[  5] Server Report:
[  5]  0.0-10.0 sec    483 MBytes    406 Mbits/sec  0.000 ms    0/344828 (0%)
[  5]  0.0-10.0 sec  1 datagrams received out-of-order


UDP traffic looks fine.

SIPP (UAC) inject SILK RTP (PCAP ) and send this to Freeswitch. Freeswitch send this G711a to the SIPP( UAS).

I doing some testing and until  700 sessions it look fine (Packed loss <0.1%  and delay <40ms)

If I do more sessions 10% of packet have no packet loss but delay of more the 100ms.


SIPP uac side:

------------------------------ Scenario Screen -------- [1-9]: Change Screen --
  Call-rate(length)   Port   Total-time  Total-calls  Remote-host
  50.0(0 ms)/1.000s   5060     230.53 s         2000  172.23.125.12:5060(UDP)

  Call limit reached (-m 2000), 0.000 s period  0 ms scheduler resolution
  0 calls (limit 4500)                   Peak was 2000 calls, after 40 s
  0 Running, 1654 Paused, 0 Woken up
  0 dead call msg (discarded)            6459 out-of-call msg (discarded)
  1 open sockets

                                 Messages  Retrans   Timeout   Unexpected-Msg
       Pause [   2000ms]         2000                          0
      INVITE ---------->         2000      0         0
         100 <----------         2000      0         0         0
         183 <----------         0         0         0         0
         400 <----------         0         0         0         0
         401 <----------         0         0         0         0
         403 <----------         0         0         0         0
         404 <----------         0         0         0         0
         405 <----------         0         0         0         0
         406 <----------         0         0         0         0
         408 <----------         0         0         0         0
         415 <----------         0         0         0         0
         433 <----------         0         0         0         0
         480 <----------         0         0         0         0
         481 <----------         0         0         0         0
         484 <----------         0         0         0         0
         485 <----------         0         0         0         0
         486 <----------         0         0         0         0
         487 <----------         0         0         0         0
         488 <----------         0         0         0         0
         489 <----------         0         0         0         0
         500 <----------         0         0         0         0
         502 <----------         0         0         0         0
         503 <----------         0         0         0         0
         180 <----------         2000      0         0         0
         200 <----------  E-RTD1 2000      0         0         0
         ACK ---------->         2000      0
       Pause [    500ms]         2000                          0
              [ NOP ]
         BYE <----------         2000      0         0         0
         200 ---------->         2000      0
------------------------------ Test Terminated --------------------------------


----------------------------- Statistics Screen ------- [1-9]: Change Screen --
  Start Time             | 2012-06-28   20:17:17:312    1340907437.312927
  Last Reset Time        | 2012-06-28   20:21:07:857    1340907667.857562
  Current Time           | 2012-06-28   20:21:07:857    1340907667.857769
-------------------------+---------------------------+--------------------------
  Counter Name           | Periodic value            | Cumulative value
-------------------------+---------------------------+--------------------------
  Elapsed Time           | 00:00:00:000              | 00:03:50:544
  Call Rate              |    0.000 cps              |    8.675 cps
-------------------------+---------------------------+--------------------------
  Incoming call created  |        0                  |        0
  OutGoing call created  |        0                  |     2000
  Total Call created     |                           |     2000
  Current Call           |        0                  |



SIPP UAS side:

------------------------------ Scenario Screen -------- [1-9]: Change Screen --
  Port   Total-time  Total-calls  Transport
  5060      65.10 s         2000  UDP

  0 new calls during 1.001 s period      1 ms scheduler resolution
  2000 calls                             Peak was 2000 calls, after 44 s
  0 Running, 2002 Paused, 4 Woken up
  0 dead call msg (discarded)
  3 open sockets
  2021 Total echo RTP pckts 1st stream   5.327 last period RTP rate (kB/s)
  0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate (kB/s)

                                 Messages  Retrans   Timeout   Unexpected-Msg
 ----------> INVITE             2000      0         0         0
  <---------- 100                2000      0
  [    500ms] Pause              2000                          0
  <---------- 180                2000      0
  <---------- 200                2000      0         0
  ----------> ACK                2000      0         0         0
  [     3:08] Pause              2000                          0
  <---------- BYE                0         0         0
  ----------> 200                0         0         0         0
------------------------------ Sipp Server Mode -------------------------------


##############################################################################

MPSTAT -A 5   on freeswitch:


8:29:50     CPU    %usr   %nice    %sys %iowait    %irq   %soft  %steal  %guest   %idle
18:29:55     all   24.84    0.00   25.63    0.00    0.02    7.46    0.00    0.00   42.05
18:29:55       0   43.52    0.00   26.72    0.00    0.00    4.05    0.00    0.00   25.71
18:29:55       1   27.88    0.00   39.18    0.00    0.00    3.12    0.00    0.00   29.82
18:29:55       2   27.70    0.00   33.40    0.00    0.00    4.17    0.00    0.00   34.72
18:29:55       3   25.57    0.00   35.98    0.00    0.00    3.03    0.00    0.00   35.42
18:29:55       4   27.36    0.00   27.56    0.00    0.00    2.56    0.00    0.00   42.52
18:29:55       5   25.45    0.00   24.44    0.00    0.00    2.22    0.00    0.00   47.88
18:29:55       6   24.77    0.00   31.28    0.00    0.00    2.98    0.00    0.00   40.97
18:29:55       7   23.05    0.00   23.81    0.00    0.00    2.67    0.00    0.00   50.48
18:29:55       8   22.22    0.00   23.03    0.00    0.20    2.02    0.00    0.00   52.53
18:29:55       9    4.84    0.00    5.71    0.00    0.00   62.86    0.00    0.00   26.59
18:29:55      10   23.36    0.00   17.01    0.00    0.00    2.46    0.00    0.00   57.17
18:29:55      11   20.21    0.00   15.11    0.00    0.00    2.98    0.00    0.00   61.70

18:29:50     CPU    intr/s
18:29:55     all  30433.80
18:29:55       0      0.00
18:29:55       1      0.00
18:29:55       2      0.00
18:29:55       3      0.00
18:29:55       4      0.00
18:29:55       5      0.00
18:29:55       6      0.00
18:29:55       7      0.00
18:29:55       8      0.00
18:29:55       9      0.00
18:29:55      10      0.00
18:29:55      11      0.00

18:29:50     CPU     0/s     1/s     8/s
18:29:55       0    0.00    0.00    0.00
18:29:55       1    0.00    0.00    0.00
18:29:55       2    0.00    0.00    0.00
18:29:55       3    0.00    0.00    0.00
18:29:55       4    0.00    0.00    0.00
18:29:55       5    0.00    0.00    0.00
18:29:55       6    0.00    0.00    0.00
18:29:55       7    0.00    0.00    0.00
18:29:55       8    0.00    0.00    0.00
18:29:55       9    0.00    0.00    0.00
18:29:55      10    0.00    0.00    0.00
18:29:55      11    0.00    0.00    0.00

I tested it with G711a to G711a and got the same result.

I tested also some RTP parameters like

<param name="rtp-autoflush-during-bridge" value="true"/>  or false

<param name="rtp-autoflush" value="true"/>   or false

The lot  of the module are not activated like CDR stuff.

mvg,
Edwin


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From andrew at cassidywebservices.co.uk  Thu Jun 28 23:47:04 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 28 Jun 2012 20:47:04 +0100
Subject: [Freeswitch-users] Looking up user data in dialplan
In-Reply-To: <CAMWiMy9yiTE9MaesTGgHPkCfGED-Qh6W9C_ZYm=WsywHU88f-w@mail.gmail.com>
References: <CAMbW9oKg=1U3YUUAJMwfWP+H-2DQ=+Lf6W=ZR6Jd7pNv74nXTg@mail.gmail.com>
	<004ED0CB-4E46-4F01-B9F8-F82456D4F7F9@scarlet-internet.nl>
	<CAMbW9o+BgiO7ELwk-=ZKFPPTGTOMTEYXbU=6iZMhht8xHmNgdQ@mail.gmail.com>
	<CAMWiMy9yiTE9MaesTGgHPkCfGED-Qh6W9C_ZYm=WsywHU88f-w@mail.gmail.com>
Message-ID: <CAMbW9oKQAtWAmBrD41WC0GCXj8vrdP8y+XnH3yAQHfL4n5jJog@mail.gmail.com>

Again, that only lets me search by ID, ie I need to know the user ID in
advance. I'm trying to find the user ID by searching other fields.

On 28 June 2012 20:16, curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> Try the user_data command
>
> http://wiki.freeswitch.org/wiki/Mod_commands#user_data
>
> The wiki shoves a good example on how this is used.
>
>
> On Thu, Jun 28, 2012 at 12:04 PM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
>> Hi Leon,
>>
>> Does that allow me to search by attrs other than ID? I need to get the id
>> of a user given another attr (such as alias), but looking for a minimal
>> approach.
>>
>> Basically all i need is to have N mailboxes, each with a DDI to leave
>> messages and a DDI to check them. Obviously I can just use one of the DDI's
>> as the user ID, but to check i need to find the user id for the other
>> number.
>>
>> I can do this with mod_xml_curl easily enough, was just wondering if
>> there was a simpler way.
>>
>> On 28 June 2012 14:36, Leon de Rooij <leon at scarlet-internet.nl> wrote:
>>
>>> You can use the set_user dp app to set all user variables as channel
>>> variables. You can also supply a prefix string to keep these new variables
>>> separate from wha was already set.
>>>
>>> See here: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user
>>>
>>> regards,
>>>
>>> Leon
>>>
>>>
>>> On Jun 28, 2012, at 11:09, Andrew Cassidy wrote:
>>>
>>> Hi All,
>>>
>>> Right, I have a very basic application I'm building around voicemail.
>>> Each user has 2 attrs that I need to search by. user_exists works fine, but
>>> doesn't return me any data other then the fact the user exists, and
>>> user_data only searches the id attr.
>>>
>>> Is there any way I can do that in just the dialplan with what is already
>>> available?
>>>
>>> In the most simple form I need the user id for a number_alias.
>>>
>>> --
>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>> Managing Director
>>>
>>>
>>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>>  *03300 100 961
>>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>>
>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>  *03300 100 961
>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From andrew at cassidywebservices.co.uk  Thu Jun 28 23:48:42 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 28 Jun 2012 20:48:42 +0100
Subject: [Freeswitch-users] Looking up user data in dialplan
In-Reply-To: <CAMbW9oKQAtWAmBrD41WC0GCXj8vrdP8y+XnH3yAQHfL4n5jJog@mail.gmail.com>
References: <CAMbW9oKg=1U3YUUAJMwfWP+H-2DQ=+Lf6W=ZR6Jd7pNv74nXTg@mail.gmail.com>
	<004ED0CB-4E46-4F01-B9F8-F82456D4F7F9@scarlet-internet.nl>
	<CAMbW9o+BgiO7ELwk-=ZKFPPTGTOMTEYXbU=6iZMhht8xHmNgdQ@mail.gmail.com>
	<CAMWiMy9yiTE9MaesTGgHPkCfGED-Qh6W9C_ZYm=WsywHU88f-w@mail.gmail.com>
	<CAMbW9oKQAtWAmBrD41WC0GCXj8vrdP8y+XnH3yAQHfL4n5jJog@mail.gmail.com>
Message-ID: <CAMbW9oJnpBZgwjw8r4q-hj5oEo5beJeewFwMXyv98QmKstvq_g@mail.gmail.com>

Right, It appears that I need to use number-alias not number_alias. I'll
try that and let you know.

On 28 June 2012 20:47, Andrew Cassidy <andrew at cassidywebservices.co.uk>wrote:

> Again, that only lets me search by ID, ie I need to know the user ID in
> advance. I'm trying to find the user ID by searching other fields.
>
>
> On 28 June 2012 20:16, curriegrad2004 <curriegrad2004 at gmail.com> wrote:
>
>> Try the user_data command
>>
>> http://wiki.freeswitch.org/wiki/Mod_commands#user_data
>>
>> The wiki shoves a good example on how this is used.
>>
>>
>> On Thu, Jun 28, 2012 at 12:04 PM, Andrew Cassidy <
>> andrew at cassidywebservices.co.uk> wrote:
>>
>>> Hi Leon,
>>>
>>> Does that allow me to search by attrs other than ID? I need to get the
>>> id of a user given another attr (such as alias), but looking for a minimal
>>> approach.
>>>
>>> Basically all i need is to have N mailboxes, each with a DDI to leave
>>> messages and a DDI to check them. Obviously I can just use one of the DDI's
>>> as the user ID, but to check i need to find the user id for the other
>>> number.
>>>
>>> I can do this with mod_xml_curl easily enough, was just wondering if
>>> there was a simpler way.
>>>
>>> On 28 June 2012 14:36, Leon de Rooij <leon at scarlet-internet.nl> wrote:
>>>
>>>> You can use the set_user dp app to set all user variables as channel
>>>> variables. You can also supply a prefix string to keep these new variables
>>>> separate from wha was already set.
>>>>
>>>> See here: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user
>>>>
>>>> regards,
>>>>
>>>> Leon
>>>>
>>>>
>>>> On Jun 28, 2012, at 11:09, Andrew Cassidy wrote:
>>>>
>>>> Hi All,
>>>>
>>>> Right, I have a very basic application I'm building around voicemail.
>>>> Each user has 2 attrs that I need to search by. user_exists works fine, but
>>>> doesn't return me any data other then the fact the user exists, and
>>>> user_data only searches the id attr.
>>>>
>>>> Is there any way I can do that in just the dialplan with what is
>>>> already available?
>>>>
>>>> In the most simple form I need the user id for a number_alias.
>>>>
>>>> --
>>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>>> Managing Director
>>>>
>>>>
>>>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>>>  *03300 100 961
>>>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>>>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>> Managing Director
>>>
>>>
>>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>>  *03300 100 961
>>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From msc at freeswitch.org  Thu Jun 28 23:49:46 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Jun 2012 12:49:46 -0700
Subject: [Freeswitch-users] Looking up user data in dialplan
In-Reply-To: <CAMbW9oKQAtWAmBrD41WC0GCXj8vrdP8y+XnH3yAQHfL4n5jJog@mail.gmail.com>
References: <CAMbW9oKg=1U3YUUAJMwfWP+H-2DQ=+Lf6W=ZR6Jd7pNv74nXTg@mail.gmail.com>
	<004ED0CB-4E46-4F01-B9F8-F82456D4F7F9@scarlet-internet.nl>
	<CAMbW9o+BgiO7ELwk-=ZKFPPTGTOMTEYXbU=6iZMhht8xHmNgdQ@mail.gmail.com>
	<CAMWiMy9yiTE9MaesTGgHPkCfGED-Qh6W9C_ZYm=WsywHU88f-w@mail.gmail.com>
	<CAMbW9oKQAtWAmBrD41WC0GCXj8vrdP8y+XnH3yAQHfL4n5jJog@mail.gmail.com>
Message-ID: <CAKzWOxX_+1pKFfZ6Y6KyTwzzSr_2mtEZgnd90ObMLh9jhjWJaQ@mail.gmail.com>

On Thu, Jun 28, 2012 at 12:47 PM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> Again, that only lets me search by ID, ie I need to know the user ID in
> advance. I'm trying to find the user ID by searching other fields.
>

Can we back up a step? Can you reset this question? What information do you
already have at the point that you are attempting to look for the user_id?
-MC
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From andrew at cassidywebservices.co.uk  Thu Jun 28 23:59:54 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 28 Jun 2012 20:59:54 +0100
Subject: [Freeswitch-users] Looking up user data in dialplan
In-Reply-To: <CAKzWOxX_+1pKFfZ6Y6KyTwzzSr_2mtEZgnd90ObMLh9jhjWJaQ@mail.gmail.com>
References: <CAMbW9oKg=1U3YUUAJMwfWP+H-2DQ=+Lf6W=ZR6Jd7pNv74nXTg@mail.gmail.com>
	<004ED0CB-4E46-4F01-B9F8-F82456D4F7F9@scarlet-internet.nl>
	<CAMbW9o+BgiO7ELwk-=ZKFPPTGTOMTEYXbU=6iZMhht8xHmNgdQ@mail.gmail.com>
	<CAMWiMy9yiTE9MaesTGgHPkCfGED-Qh6W9C_ZYm=WsywHU88f-w@mail.gmail.com>
	<CAMbW9oKQAtWAmBrD41WC0GCXj8vrdP8y+XnH3yAQHfL4n5jJog@mail.gmail.com>
	<CAKzWOxX_+1pKFfZ6Y6KyTwzzSr_2mtEZgnd90ObMLh9jhjWJaQ@mail.gmail.com>
Message-ID: <CAMbW9oJo2rDL_BToYY=CB9uUrva8uBCs6M+1pNaVX1msOuPDrA@mail.gmail.com>

Right, basically I have 2 DID's that each map to a voicemail box, one to
leave messages and one to check messages. So when a call comes in all I
have is the DID the call came in on. I am trying to do this as simply as
possible and I don't know how many he end user of this system is going to
create.

So, i was going to put one number in the user id attr, and the other in
number-alias (which I now see is not an underscore as I originally thought
it was) and just pass the DID to the voicemail application, or look up the
ID if that approach didn't work.

But yes, as I've now determined that number_alias should have been
number-alias, I'm going to try that and see where that gets me.

Regards,

On 28 June 2012 20:49, Michael Collins <msc at freeswitch.org> wrote:

> On Thu, Jun 28, 2012 at 12:47 PM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
>> Again, that only lets me search by ID, ie I need to know the user ID in
>> advance. I'm trying to find the user ID by searching other fields.
>>
>
> Can we back up a step? Can you reset this question? What information do
> you already have at the point that you are attempting to look for the
> user_id?
> -MC
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From andrew at cassidywebservices.co.uk  Fri Jun 29 00:05:16 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 28 Jun 2012 21:05:16 +0100
Subject: [Freeswitch-users] Looking up user data in dialplan
In-Reply-To: <CAMbW9oJo2rDL_BToYY=CB9uUrva8uBCs6M+1pNaVX1msOuPDrA@mail.gmail.com>
References: <CAMbW9oKg=1U3YUUAJMwfWP+H-2DQ=+Lf6W=ZR6Jd7pNv74nXTg@mail.gmail.com>
	<004ED0CB-4E46-4F01-B9F8-F82456D4F7F9@scarlet-internet.nl>
	<CAMbW9o+BgiO7ELwk-=ZKFPPTGTOMTEYXbU=6iZMhht8xHmNgdQ@mail.gmail.com>
	<CAMWiMy9yiTE9MaesTGgHPkCfGED-Qh6W9C_ZYm=WsywHU88f-w@mail.gmail.com>
	<CAMbW9oKQAtWAmBrD41WC0GCXj8vrdP8y+XnH3yAQHfL4n5jJog@mail.gmail.com>
	<CAKzWOxX_+1pKFfZ6Y6KyTwzzSr_2mtEZgnd90ObMLh9jhjWJaQ@mail.gmail.com>
	<CAMbW9oJo2rDL_BToYY=CB9uUrva8uBCs6M+1pNaVX1msOuPDrA@mail.gmail.com>
Message-ID: <CAMbW9oKxoA_yO9UTvZ-8a6mt=1uZf3GsT7VzYdHuR8hwoKbSWg@mail.gmail.com>

That's got it. It was just a simple typo,  number_alias should have been
number-alias.

On 28 June 2012 20:59, Andrew Cassidy <andrew at cassidywebservices.co.uk>wrote:

> Right, basically I have 2 DID's that each map to a voicemail box, one to
> leave messages and one to check messages. So when a call comes in all I
> have is the DID the call came in on. I am trying to do this as simply as
> possible and I don't know how many he end user of this system is going to
> create.
>
> So, i was going to put one number in the user id attr, and the other in
> number-alias (which I now see is not an underscore as I originally thought
> it was) and just pass the DID to the voicemail application, or look up the
> ID if that approach didn't work.
>
> But yes, as I've now determined that number_alias should have been
> number-alias, I'm going to try that and see where that gets me.
>
> Regards,
>
> On 28 June 2012 20:49, Michael Collins <msc at freeswitch.org> wrote:
>
>> On Thu, Jun 28, 2012 at 12:47 PM, Andrew Cassidy <
>> andrew at cassidywebservices.co.uk> wrote:
>>
>>> Again, that only lets me search by ID, ie I need to know the user ID in
>>> advance. I'm trying to find the user ID by searching other fields.
>>>
>>
>> Can we back up a step? Can you reset this question? What information do
>> you already have at the point that you are attempting to look for the
>> user_id?
>> -MC
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From julian.pawlowski at gmail.com  Fri Jun 29 00:31:51 2012
From: julian.pawlowski at gmail.com (Julian Pawlowski)
Date: Thu, 28 Jun 2012 22:31:51 +0200
Subject: [Freeswitch-users] Unofficial Debian repository
Message-ID: <CADFnKK_X8ughZqpkEDGPNH=0Y5yQ6L5o+ToFcvt65gyGrBiUmA@mail.gmail.com>

Hi folks,

just wanted to let you know that I have just setup an unofficial
nightly build repository for FreeSWITCH.
It currently builds for squeeze, wheezy and sid all in i386 as well as
amd64 architectures.

Current build status can be seen here:
https://build.profhost.eu/view/FreeSWITCH/

The actual repository is available here:
http://repo.profhost.eu/debian/dists/

Feel free to use it for your personal purposes. Just put the following
line into your /etc/apt/sources.list:

  dep http://repo.profhost.eu/debian DISTRIBUTION main

Change DISTRIBUTION for your actual base installation.
I will use this repo for system integration of Gemeinschaft 5 so from
my side I will keep this repo running until the official one from
FreeSWITCH guys is up and running.


Cheers,
Julian


From jnankin at gmail.com  Fri Jun 29 00:41:31 2012
From: jnankin at gmail.com (Joshua Nankin)
Date: Thu, 28 Jun 2012 15:41:31 -0500
Subject: [Freeswitch-users] Multiple network interfaces,
	can't force freeswitch to use one.
Message-ID: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>

I have two interfaces on my machine, eth0=10.0.2.15 and eth1=192.168.1.7.
 I'm trying to get freeswitch to only use eth1.  I've put the following at
the top of my vars.xml:

<X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.7"/>

However, this does not seem to be working.  When I originate a call, I see
that the initial invite is originating from 10.0.2.15, but incoming packets
from my sip provider are going to 192.168.1.7 and are not being responded
to by freeswitch.  Shouldn't freeswitch be listening on 192.168.1.7 and
sending packets from there as well?

What am I doing wrong?
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From curriegrad2004 at gmail.com  Fri Jun 29 00:52:51 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 28 Jun 2012 13:52:51 -0700
Subject: [Freeswitch-users] Multiple network interfaces,
 can't force freeswitch to use one.
In-Reply-To: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>
References: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>
Message-ID: <CAMWiMy9SuKwMvKgz2c5_rSnPCvwpY6AvZcd3jfbdNwt6RpPq1Q@mail.gmail.com>

You would want to set the IP address bindings in sofia's profile conf
page, not using vars.xml.

rtp-ip and sip-ip would probably be your best bet in this case.

On Thu, Jun 28, 2012 at 1:41 PM, Joshua Nankin <jnankin at gmail.com> wrote:
> I have two interfaces on my machine, eth0=10.0.2.15 and eth1=192.168.1.7.
> ?I'm trying to get freeswitch to only use eth1. ?I've put the following at
> the top of my vars.xml:
>
> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.7"/>
>
> However, this does not seem to be working. ?When I originate a call, I see
> that the initial invite is originating from 10.0.2.15, but incoming packets
> from my sip provider are going to 192.168.1.7 and are not being responded to
> by freeswitch. ?Shouldn't freeswitch be listening on 192.168.1.7 and sending
> packets from there as well?
>
> What am I doing wrong?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From jnankin at gmail.com  Fri Jun 29 01:08:32 2012
From: jnankin at gmail.com (Joshua Nankin)
Date: Thu, 28 Jun 2012 16:08:32 -0500
Subject: [Freeswitch-users] Multiple network interfaces,
	can't force freeswitch to use one.
In-Reply-To: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>
References: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>
Message-ID: <CAAYWZ8=bC4OGYYzqdNutsznvjjm1N6j55zkw9+-5XboMhr5pww@mail.gmail.com>

sip-ip and rtp-ip in both external.xml and internal.xml were set to
$${local_ip_v4}.  I figured that setting the ip in vars.xml would be enough.

I manually set sip-ip and rtp-ip and both of those files, but that still
did not solve the problem.

Any other ideas?

On Thu, Jun 28, 2012 at 3:41 PM, Joshua Nankin <jnankin at gmail.com> wrote:

> I have two interfaces on my machine, eth0=10.0.2.15 and eth1=192.168.1.7.
>  I'm trying to get freeswitch to only use eth1.  I've put the following at
> the top of my vars.xml:
>
> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.7"/>
>
> However, this does not seem to be working.  When I originate a call, I see
> that the initial invite is originating from 10.0.2.15, but incoming packets
> from my sip provider are going to 192.168.1.7 and are not being responded
> to by freeswitch.  Shouldn't freeswitch be listening on 192.168.1.7 and
> sending packets from there as well?
>
> What am I doing wrong?
>
>
>
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From mark at callassistants.com  Fri Jun 29 01:02:28 2012
From: mark at callassistants.com (marksly)
Date: Thu, 28 Jun 2012 14:02:28 -0700 (PDT)
Subject: [Freeswitch-users] Strange "timer_test" result
In-Reply-To: <CAD-qRkhT0ZBca+Zz0wp3Fpc42pcV6x=eR6E_NErfeTDHz4G7nA@mail.gmail.com>
References: <CAD-qRkhizVe+QsRHcgEseYbPhe2h7=VvtkMj1Nu0vLa00imwSw@mail.gmail.com>
	<CAO4LeU1ppjAo5dTgPywBOeGH0mn9pZC+ov+uu2LNNc6VxyDQcA@mail.gmail.com>
	<0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com>
	<CAO4LeU0F6kcH0tgpEx9039Xf45we2ZFLD_BHU_puhaEyY5mH7Q@mail.gmail.com>
	<F615FA10-493C-4AF5-ABE1-DB7E3AE8CA67@gmail.com>
	<1340748029314-7580297.post@n2.nabble.com>
	<CAO4LeU144W1OAV3+GBiZzPUNbf3HA1SXptAR3e3QFR63qX1zJg@mail.gmail.com>
	<CAD-qRkjmQSwtAGiY8MXWH=HuqwePMZU3xR1dJdYWo0ui243z5Q@mail.gmail.com>
	<CAO4LeU08sUKbzC1Ee4fjHa3vB_XEdRQgc=ePiqrVMJNZssYKsw@mail.gmail.com>
	<CAD-qRkhT0ZBca+Zz0wp3Fpc42pcV6x=eR6E_NErfeTDHz4G7nA@mail.gmail.com>
Message-ID: <1340917348693-7580388.post@n2.nabble.com>

To give everyone an update: I enabled the mod_posix_timer last night and
things appear to be running better today (fewer calls that have a sporadic
choppy audio). The server load is lower, but it is not where I would like to
see it. The load is around 10 on average. I think that this might be due to
the fact that I have "enable-clock-nanosleep" set to true. I am going to try
disabling this and see if the load decreases.

The timer_test results are more consistent and regular; a lot closer to the
20ms that you would expect to see.

Server Information:
2x AMD Opteron 6128 (8 cores)
8GB RAM
CentOS release 5.7
Kernel: 2.6.18-274.7.1.el5
FreeSWITCH version:
1.2.0-rc2+git~20120625T203158Z~7dc9a9cacc+unclean~20120626T041419Z

Feel free to provide other suggestions if you see fit.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Strange-timer-test-result-tp7399299p7580388.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Fri Jun 29 01:19:43 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Jun 2012 14:19:43 -0700
Subject: [Freeswitch-users] Multiple network interfaces,
 can't force freeswitch to use one.
In-Reply-To: <CAAYWZ8=bC4OGYYzqdNutsznvjjm1N6j55zkw9+-5XboMhr5pww@mail.gmail.com>
References: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>
	<CAAYWZ8=bC4OGYYzqdNutsznvjjm1N6j55zkw9+-5XboMhr5pww@mail.gmail.com>
Message-ID: <CAKzWOxUCRSwN6iXYftW9JbGxe8m3y0Nt0V1xjfjMmF8xV7NguQ@mail.gmail.com>

Did you restart the profiles after doing reloadxml?
-MC

On Thu, Jun 28, 2012 at 2:08 PM, Joshua Nankin <jnankin at gmail.com> wrote:

> sip-ip and rtp-ip in both external.xml and internal.xml were set to
> $${local_ip_v4}.  I figured that setting the ip in vars.xml would be enough.
>
> I manually set sip-ip and rtp-ip and both of those files, but that still
> did not solve the problem.
>
> Any other ideas?
>
> On Thu, Jun 28, 2012 at 3:41 PM, Joshua Nankin <jnankin at gmail.com> wrote:
>
>> I have two interfaces on my machine, eth0=10.0.2.15 and eth1=192.168.1.7.
>>  I'm trying to get freeswitch to only use eth1.  I've put the following at
>> the top of my vars.xml:
>>
>> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.7"/>
>>
>> However, this does not seem to be working.  When I originate a call, I
>> see that the initial invite is originating from 10.0.2.15, but incoming
>> packets from my sip provider are going to 192.168.1.7 and are not being
>> responded to by freeswitch.  Shouldn't freeswitch be listening on
>> 192.168.1.7 and sending packets from there as well?
>>
>> What am I doing wrong?
>>
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jnankin at gmail.com  Fri Jun 29 01:36:51 2012
From: jnankin at gmail.com (Joshua Nankin)
Date: Thu, 28 Jun 2012 16:36:51 -0500
Subject: [Freeswitch-users] Multiple network interfaces,
	can't force freeswitch to use one.
In-Reply-To: <CAAYWZ8=bC4OGYYzqdNutsznvjjm1N6j55zkw9+-5XboMhr5pww@mail.gmail.com>
References: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>
	<CAAYWZ8=bC4OGYYzqdNutsznvjjm1N6j55zkw9+-5XboMhr5pww@mail.gmail.com>
Message-ID: <CAAYWZ8nt7ecAu29f=iNXwB9jvxjJVYTwrisir0xudwTuzzYSCA@mail.gmail.com>

I did a sudo /etc/init.d/freeswitch restart.  That should be enough correct?

On Thu, Jun 28, 2012 at 4:08 PM, Joshua Nankin <jnankin at gmail.com> wrote:

> sip-ip and rtp-ip in both external.xml and internal.xml were set to
> $${local_ip_v4}.  I figured that setting the ip in vars.xml would be enough.
>
> I manually set sip-ip and rtp-ip and both of those files, but that still
> did not solve the problem.
>
> Any other ideas?
>
>
> On Thu, Jun 28, 2012 at 3:41 PM, Joshua Nankin <jnankin at gmail.com> wrote:
>
>> I have two interfaces on my machine, eth0=10.0.2.15 and eth1=192.168.1.7.
>>  I'm trying to get freeswitch to only use eth1.  I've put the following at
>> the top of my vars.xml:
>>
>> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.7"/>
>>
>> However, this does not seem to be working.  When I originate a call, I
>> see that the initial invite is originating from 10.0.2.15, but incoming
>> packets from my sip provider are going to 192.168.1.7 and are not being
>> responded to by freeswitch.  Shouldn't freeswitch be listening on
>> 192.168.1.7 and sending packets from there as well?
>>
>> What am I doing wrong?
>>
>>
>>
>
>
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From curriegrad2004 at gmail.com  Fri Jun 29 01:39:58 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 28 Jun 2012 14:39:58 -0700
Subject: [Freeswitch-users] Multiple network interfaces,
 can't force freeswitch to use one.
In-Reply-To: <CAAYWZ8nt7ecAu29f=iNXwB9jvxjJVYTwrisir0xudwTuzzYSCA@mail.gmail.com>
References: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>
	<CAAYWZ8=bC4OGYYzqdNutsznvjjm1N6j55zkw9+-5XboMhr5pww@mail.gmail.com>
	<CAAYWZ8nt7ecAu29f=iNXwB9jvxjJVYTwrisir0xudwTuzzYSCA@mail.gmail.com>
Message-ID: <CAMWiMy9ybkw2SauZNzwaiVFxGe06ghFwmTMWTm6AE9vP=Y7PeA@mail.gmail.com>

More than enough. Type in sofia status to verify your changes did take effect.

On Thu, Jun 28, 2012 at 2:36 PM, Joshua Nankin <jnankin at gmail.com> wrote:
> I did a sudo /etc/init.d/freeswitch restart. ?That should be enough correct?
>
> On Thu, Jun 28, 2012 at 4:08 PM, Joshua Nankin <jnankin at gmail.com> wrote:
>>
>> sip-ip and rtp-ip in both external.xml and internal.xml were set to
>> $${local_ip_v4}. ?I figured that setting the ip in vars.xml would be enough.
>>
>> I manually set sip-ip and rtp-ip and both of those files, but that still
>> did not solve the problem.
>>
>> Any other ideas?
>>
>>
>> On Thu, Jun 28, 2012 at 3:41 PM, Joshua Nankin <jnankin at gmail.com> wrote:
>>>
>>> I have two interfaces on my machine, eth0=10.0.2.15 and eth1=192.168.1.7.
>>> ?I'm trying to get freeswitch to only use eth1. ?I've put the following at
>>> the top of my vars.xml:
>>>
>>> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.7"/>
>>>
>>> However, this does not seem to be working. ?When I originate a call, I
>>> see that the initial invite is originating from 10.0.2.15, but incoming
>>> packets from my sip provider are going to 192.168.1.7 and are not being
>>> responded to by freeswitch. ?Shouldn't freeswitch be listening on
>>> 192.168.1.7 and sending packets from there as well?
>>>
>>> What am I doing wrong?
>>>
>>>
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From curriegrad2004 at gmail.com  Fri Jun 29 01:41:24 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 28 Jun 2012 14:41:24 -0700
Subject: [Freeswitch-users] Multiple network interfaces,
 can't force freeswitch to use one.
In-Reply-To: <CAMWiMy9ybkw2SauZNzwaiVFxGe06ghFwmTMWTm6AE9vP=Y7PeA@mail.gmail.com>
References: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>
	<CAAYWZ8=bC4OGYYzqdNutsznvjjm1N6j55zkw9+-5XboMhr5pww@mail.gmail.com>
	<CAAYWZ8nt7ecAu29f=iNXwB9jvxjJVYTwrisir0xudwTuzzYSCA@mail.gmail.com>
	<CAMWiMy9ybkw2SauZNzwaiVFxGe06ghFwmTMWTm6AE9vP=Y7PeA@mail.gmail.com>
Message-ID: <CAMWiMy_z5t36+4+=6H=G7ssQuoR4KQHrmA-Jvdf5qLKwaNYE4w@mail.gmail.com>

Forgot to mention that you enter that in the freeswitch command line
prompt (aka via fs_cli)

On Thu, Jun 28, 2012 at 2:39 PM, curriegrad2004
<curriegrad2004 at gmail.com> wrote:
> More than enough. Type in sofia status to verify your changes did take effect.
>
> On Thu, Jun 28, 2012 at 2:36 PM, Joshua Nankin <jnankin at gmail.com> wrote:
>> I did a sudo /etc/init.d/freeswitch restart. ?That should be enough correct?
>>
>> On Thu, Jun 28, 2012 at 4:08 PM, Joshua Nankin <jnankin at gmail.com> wrote:
>>>
>>> sip-ip and rtp-ip in both external.xml and internal.xml were set to
>>> $${local_ip_v4}. ?I figured that setting the ip in vars.xml would be enough.
>>>
>>> I manually set sip-ip and rtp-ip and both of those files, but that still
>>> did not solve the problem.
>>>
>>> Any other ideas?
>>>
>>>
>>> On Thu, Jun 28, 2012 at 3:41 PM, Joshua Nankin <jnankin at gmail.com> wrote:
>>>>
>>>> I have two interfaces on my machine, eth0=10.0.2.15 and eth1=192.168.1.7.
>>>> ?I'm trying to get freeswitch to only use eth1. ?I've put the following at
>>>> the top of my vars.xml:
>>>>
>>>> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.7"/>
>>>>
>>>> However, this does not seem to be working. ?When I originate a call, I
>>>> see that the initial invite is originating from 10.0.2.15, but incoming
>>>> packets from my sip provider are going to 192.168.1.7 and are not being
>>>> responded to by freeswitch. ?Shouldn't freeswitch be listening on
>>>> 192.168.1.7 and sending packets from there as well?
>>>>
>>>> What am I doing wrong?
>>>>
>>>>
>>>
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>


From jnankin at gmail.com  Fri Jun 29 01:48:52 2012
From: jnankin at gmail.com (Joshua Nankin)
Date: Thu, 28 Jun 2012 16:48:52 -0500
Subject: [Freeswitch-users] Multiple network interfaces,
	can't force freeswitch to use one.
In-Reply-To: <CAAYWZ8nt7ecAu29f=iNXwB9jvxjJVYTwrisir0xudwTuzzYSCA@mail.gmail.com>
References: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>
	<CAAYWZ8=bC4OGYYzqdNutsznvjjm1N6j55zkw9+-5XboMhr5pww@mail.gmail.com>
	<CAAYWZ8nt7ecAu29f=iNXwB9jvxjJVYTwrisir0xudwTuzzYSCA@mail.gmail.com>
Message-ID: <CAAYWZ8nV+TCduKpg3JG33F-xCi+0apn784KCr-Z-a9K6y6sOPQ@mail.gmail.com>

Right, but freeswitch is still getting packets to 192.168.1.7, but sending
from 10.0.2.15.

Not sure if this makes a difference, but this is a Vagrant virtual machine
with a bridged interface.  The 10.x address is an internal Vagrant ip, but
the 192 is from my router DHCP.  I've set the external sip and rtp ips to
my public IP address, and that IP is what appears for the profile in sofia
status.



On Thu, Jun 28, 2012 at 4:36 PM, Joshua Nankin <jnankin at gmail.com> wrote:

> I did a sudo /etc/init.d/freeswitch restart.  That should be enough
> correct?
>
> On Thu, Jun 28, 2012 at 4:08 PM, Joshua Nankin <jnankin at gmail.com> wrote:
>
>> sip-ip and rtp-ip in both external.xml and internal.xml were set to
>> $${local_ip_v4}.  I figured that setting the ip in vars.xml would be enough.
>>
>> I manually set sip-ip and rtp-ip and both of those files, but that still
>> did not solve the problem.
>>
>> Any other ideas?
>>
>>
>> On Thu, Jun 28, 2012 at 3:41 PM, Joshua Nankin <jnankin at gmail.com> wrote:
>>
>>> I have two interfaces on my machine, eth0=10.0.2.15 and
>>> eth1=192.168.1.7.  I'm trying to get freeswitch to only use eth1.  I've put
>>> the following at the top of my vars.xml:
>>>
>>> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.7"/>
>>>
>>> However, this does not seem to be working.  When I originate a call, I
>>> see that the initial invite is originating from 10.0.2.15, but incoming
>>> packets from my sip provider are going to 192.168.1.7 and are not being
>>> responded to by freeswitch.  Shouldn't freeswitch be listening on
>>> 192.168.1.7 and sending packets from there as well?
>>>
>>> What am I doing wrong?
>>>
>>>
>>>
>>
>>
>
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From anthony.minessale at gmail.com  Fri Jun 29 02:12:43 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Jun 2012 17:12:43 -0500
Subject: [Freeswitch-users] Strange "timer_test" result
In-Reply-To: <CAO4LeU144W1OAV3+GBiZzPUNbf3HA1SXptAR3e3QFR63qX1zJg@mail.gmail.com>
References: <CAO4LeU18R_MgkKU=UL1t=X2qePyuKRQ9FZTssVTkYyvHnXDRkQ@mail.gmail.com>
	<CAKbxfG9rNPkqhDsB89bMydEayfmskDUhxnu9-n7JoT9ELnWyRA@mail.gmail.com>
	<CAO4LeU3Te+xPZo0BVUDX2k3pN2aW7x-o3XvBvvqtHg8ao+CCjg@mail.gmail.com>
	<CAD-qRkiV4qn3B4NNNdqXrR=mSSpbeNx=MeN8Ty8ChL=REwoCrg@mail.gmail.com>
	<CAO4LeU2a8-417E6aPgb5=_tc1ZZ-Xmai1Fyn=WZ+3pgrYv4cYQ@mail.gmail.com>
	<CAD-qRkhizVe+QsRHcgEseYbPhe2h7=VvtkMj1Nu0vLa00imwSw@mail.gmail.com>
	<CAO4LeU1ppjAo5dTgPywBOeGH0mn9pZC+ov+uu2LNNc6VxyDQcA@mail.gmail.com>
	<0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com>
	<CAO4LeU0F6kcH0tgpEx9039Xf45we2ZFLD_BHU_puhaEyY5mH7Q@mail.gmail.com>
	<F615FA10-493C-4AF5-ABE1-DB7E3AE8CA67@gmail.com>
	<1340748029314-7580297.post@n2.nabble.com>
	<CAO4LeU144W1OAV3+GBiZzPUNbf3HA1SXptAR3e3QFR63qX1zJg@mail.gmail.com>
Message-ID: <CAKbxfG9GqrW-xUbp_g35TyyyOrq3eHJVduf9zyYFfU+rZfATOQ@mail.gmail.com>

You need to help us find that leak.
If you go back and look at the jira, I have asked for you to do some things.
Thanks.



On Wed, Jun 27, 2012 at 3:44 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> My experience in FS timers:
>
> 1. voice quality is related to timer_test result so if you have irregular
> result, you'll have bad voice quality (and low ACD)
>
> 2. mod_posix_timer is my solution (I haven't try timerfd because I'm using
> CentOS 5.8).
>
> 3. there are two version of mod_posix_timer, the "old" one (the one before
> the fix of git 'FS-4064 mod_posix_timer workaround of CentOS 5 glibc
> problem' ?of 30 april) and the 'new' one (present in latest git)
>
> 4. the 'old' mod_posix_timer is much more efficient that 'new' one (with
> 'new' mod_posix_timer is half way between 'old' mod_posix_timer and soft
> timer).
>
> 5. loading the 'old' mod_posix_timer there are random FS crashes, also if
> you don't use the posix timer in your profile
>
> 6. the 'new' mod_posix_timer solves the crashes of FS
>
> So my conclusion is that the 'old' mod_posix_timer is the best one but
> sometimes FS crashes but the 'new' mod_posix_timer is not good as the 'old'
> one but it solves the crash issue.
>
> In my production system I'm using the 'old' one in servers with high load
> and the 'new' one in serves with less load, according with timer_test
> results.
>
> Memory leak: I have done some test mixing different versions of FS and
> mod_posix_timer and in any case, if I use the latest Freeswitch git the used
> memory grows up continously, also if I use 'old' mod_posix_timer so, my
> conclusion is that the memory leakage is related to Freeswitch. In my
> production server, latest FS git + old mod_posix_timer consumes all server
> memory (12MB) in 2 days. For this reason, in all my installation I'm using
> the Freeswitch git of 26th march with new or old mod_posix_timer (depending
> on timer_test result).
>
> Stephen
>
> On Wed, Jun 27, 2012 at 12:00 AM, marksly <mark at callassistants.com> wrote:
>>
>> I am having the same issue as Stephen Wilde regarding the strange
>> "timer_test" results. Was it determined that the mod_posix timer is the
>> best
>> solution? Likewise with Stephen, any timer except 20ms works great.
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Strange-timer-test-result-tp7399299p7580297.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From wstephen80 at gmail.com  Fri Jun 29 02:21:48 2012
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Fri, 29 Jun 2012 00:21:48 +0200
Subject: [Freeswitch-users] Strange "timer_test" result
In-Reply-To: <1340917348693-7580388.post@n2.nabble.com>
References: <CAD-qRkhizVe+QsRHcgEseYbPhe2h7=VvtkMj1Nu0vLa00imwSw@mail.gmail.com>
	<CAO4LeU1ppjAo5dTgPywBOeGH0mn9pZC+ov+uu2LNNc6VxyDQcA@mail.gmail.com>
	<0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com>
	<CAO4LeU0F6kcH0tgpEx9039Xf45we2ZFLD_BHU_puhaEyY5mH7Q@mail.gmail.com>
	<F615FA10-493C-4AF5-ABE1-DB7E3AE8CA67@gmail.com>
	<1340748029314-7580297.post@n2.nabble.com>
	<CAO4LeU144W1OAV3+GBiZzPUNbf3HA1SXptAR3e3QFR63qX1zJg@mail.gmail.com>
	<CAD-qRkjmQSwtAGiY8MXWH=HuqwePMZU3xR1dJdYWo0ui243z5Q@mail.gmail.com>
	<CAO4LeU08sUKbzC1Ee4fjHa3vB_XEdRQgc=ePiqrVMJNZssYKsw@mail.gmail.com>
	<CAD-qRkhT0ZBca+Zz0wp3Fpc42pcV6x=eR6E_NErfeTDHz4G7nA@mail.gmail.com>
	<1340917348693-7580388.post@n2.nabble.com>
Message-ID: <CAO4LeU29SxnYB+tdiGV0Z3ZsNefNLG4WAAbct0dXegC3C2SsYA@mail.gmail.com>

I have bad experience with combination Opteron + RTP (choppy audio). I
think this is due to the size of the cache, for example for your 6128 is
128KB per core and, for example, for a Xeon E5649 is 12Mb per core. I own a
4 x Opteron 6176 (4 x 12 core) and after many tries to use it as Freeswitch
server, now I'm using it as database server...

Stephen

On Thu, Jun 28, 2012 at 11:02 PM, marksly <mark at callassistants.com> wrote:

> To give everyone an update: I enabled the mod_posix_timer last night and
> things appear to be running better today (fewer calls that have a sporadic
> choppy audio). The server load is lower, but it is not where I would like
> to
> see it. The load is around 10 on average. I think that this might be due to
> the fact that I have "enable-clock-nanosleep" set to true. I am going to
> try
> disabling this and see if the load decreases.
>
> The timer_test results are more consistent and regular; a lot closer to the
> 20ms that you would expect to see.
>
> Server Information:
> 2x AMD Opteron 6128 (8 cores)
> 8GB RAM
> CentOS release 5.7
> Kernel: 2.6.18-274.7.1.el5
> FreeSWITCH version:
> 1.2.0-rc2+git~20120625T203158Z~7dc9a9cacc+unclean~20120626T041419Z
>
> Feel free to provide other suggestions if you see fit.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Strange-timer-test-result-tp7399299p7580388.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rnbrady at gmail.com  Fri Jun 29 04:17:38 2012
From: rnbrady at gmail.com (Richard Brady)
Date: Fri, 29 Jun 2012 01:17:38 +0100
Subject: [Freeswitch-users] Bind extension to an IP address
In-Reply-To: <CAKzWOxWJuxd+PjBxDmRPa3UNJ6L68xEZotfCKc7HXccg7Jxtvw@mail.gmail.com>
References: <4FE7175F.90209@the800group.com> <4FE787D4.5080603@gmail.com>
	<CALNZuwm-RPQRBLMRq65fb9eVdUS2W9HURZC5y+=gQ5a2GWhtqA@mail.gmail.com>
	<CAKbxfG_iUqoU0VFAsKu5a=cwV8fh02WzrEx1R70cMgXXcFy6iQ@mail.gmail.com>
	<CAN5esQ+ZQ4v6U-EB9+6rdQ1uBr84c6sZL4S6oi2R=1wdah3k-Q@mail.gmail.com>
	<CAKzWOxWJuxd+PjBxDmRPa3UNJ6L68xEZotfCKc7HXccg7Jxtvw@mail.gmail.com>
Message-ID: <CAN5esQ+Wma3O_5TSb0Wj8=6t0TM=mZqx8b6bBrah4m+TvLz9Kw@mail.gmail.com>

>> Is there an equivalent for gateways?
>>
>> I can't find a way to bind incoming calls from a CIDR to a particular gateway.
>>
> Just curious - what functionality do you need this for?

In order to configure DTMF per provider, i.e. per gateway:

  <gateway name="provider1">
    ...
    <variables>
      <variable direction="both" name="dtmf_type" value="info">
    </variables>
  </gateway>

  <gateway name="provider2">
    ...
    <variables>
      <variable direction="both" name="dtmf_type" value="rfc2833">
    </variables>
  </gateway>

This works for outbound calls but not for inbound calls, even though I
have specified direction="both".

I believe that direction="both" and direction="inbound" will only work
on a SIP profile, not on a gateway. But I'm hoping to be corrected.

Richard


From robin.gilks at taitradio.com  Fri Jun 29 08:22:39 2012
From: robin.gilks at taitradio.com (Robin Gilks)
Date: Fri, 29 Jun 2012 16:22:39 +1200
Subject: [Freeswitch-users] Simple SPA8800 FXO connection to FreeSwitch
Message-ID: <CACMPV2CYALx0J_P7HG3T2RXtk2jo7mhER6fiKkvNHWexEt+gYg@mail.gmail.com>

Greetings

I'm a noob with FS (and VoIP generally!!) and am really struggling
with getting 2 way comms between a PABX and a soft-phone using FS and
an SPA8800.

Calls from the PABX are OK, the FXO ports have registered with FS and
when I dial the FXO extension I get the dial tone, dial the number for
the soft-phone and it all works fine.

The problem is connections *TO* the PABX. The SPA8800 AFAIK has a
separate SIP port for each of its FXO ports. My understanding of how
it should work is as follows:

I have to have 4 gateways defined in a file in the
sip-profiles/externals directory, each with a different "sip-port"
line. I'm testing with 2 to start with:

# cat cicsco_fxo.xml
<include>
  <gateway name="fxoline3">
    <param name="username" value="none"/>
    <param name="password" value="none"/>
    <param name="realm" value="172.27.1.20"/>
    <param name="sip-port" value="5260"/>
    <param name="register" value="false"/>
  </gateway>
  <gateway name="fxoline4">
    <param name="username" value="none"/>
    <param name="password" value="none"/>
    <param name="realm" value="172.27.1.20"/>
    <param name="sip-port" value="5360"/>
    <param name="register" value="false"/>
  </gateway>

I have to use mod_distributor to select one of the gateways that
define the SIP ports. Thats is configured in the autoload_configs
directory.

# cat distributor.conf.xml
<configuration name="distributor.conf" description="Distributor Configuration">
  <lists>
    <list name="cisco_fxo" total-weight="2">
      <node name="fxoline3" weight="1"/>
      <node name="fxoline4" weight="1"/>
    </list>
  </lists>
</configuration>

Finally, I use a dialplan to access this distribution of gateways when
a 4 digit number starting with 8 is dialed from the soft-phone.

# cat 33_ciscofox.xml
<include>
  <extension name="pabx">
    <condition field="destination_number" expression="^(8\d{3})$">
      <action application="bridge"
data="sofia/gateway/${distributor(cisco_fxo)}/$1"/>
    </condition>
 </extension>
</include>

Result expected - it all works.
Result I get is:

freeswitch at PBX.localdomain> 2012-06-29 04:13:44.287982 [NOTICE]
switch_channel.c:926 New Channel sofia/internal/1002 at 172.16.164.33
[d0fc47e6-c1a0-11e1-881d-f70f560259d2]
2012-06-29 04:13:44.287982 [INFO] mod_dialplan_xml.c:485 Processing
1002 <1002>->8569 in context default
2012-06-29 04:13:44.298584 [ERR] mod_sofia.c:4492 Invalid Gateway ''
2012-06-29 04:13:44.298584 [NOTICE] mod_sofia.c:4889 Close Channel N/A
[CS_NEW]
2012-06-29 04:13:44.298584 [NOTICE] switch_ivr_originate.c:2535 Cannot
create outgoing channel of type [sofia] cause: [INVALID_GATEWAY]
2012-06-29 04:13:44.298584 [INFO] mod_dptools.c:2956 Originate Failed.
 Cause: INVALID_GATEWAY
2012-06-29 04:13:44.298584 [NOTICE] mod_dptools.c:3076 Hangup
sofia/internal/1002 at 172.16.164.33 [CS_EXECUTE] [INVALID_GATEWAY]
2012-06-29 04:13:44.298584 [NOTICE] switch_core_session.c:1447 Session
11 (sofia/internal/1002 at 172.16.164.33) Ended
2012-06-29 04:13:44.298584 [NOTICE] switch_core_session.c:1449 Close
Channel sofia/internal/1002 at 172.16.164.33 [CS_DESTROY]

So what is invalid about the gateway?

Cheers

-- 
Robin Gilks

-- 

------------------------------
This email, including any attachments, is only for the intended recipient. 
It is subject to copyright, is confidential and may be the subject of legal 
or other privilege, none of which is waived or lost by reason of this 
transmission.
If you are not an intended recipient, you may not use, disseminate, 
distribute or reproduce such email, any attachments, or any part thereof. 
If you have received a message in error, please notify the sender 
immediately and erase all copies of the message and any attachments.
Unfortunately, we cannot warrant that the email has not been altered or 
corrupted during transmission nor can we guarantee that any email or any 
attachments are free from computer viruses or other conditions which may 
damage or interfere with recipient data, hardware or software. The 
recipient relies upon its own procedures and assumes all risk of use and of 
opening any attachments.
------------------------------


From shcherbatyuk at belrosbank.by  Fri Jun 29 10:32:03 2012
From: shcherbatyuk at belrosbank.by (Eugene Shcherbatyuk)
Date: Fri, 29 Jun 2012 09:32:03 +0300
Subject: [Freeswitch-users] Multiple network interfaces,
 can't force freeswitch to use one.
In-Reply-To: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>
References: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>
Message-ID: <4FED4BE3.1030101@belrosbank.by>

This problem seems to be at least two-fold:

a) FreeSwitch is to be configured to use proper address
b) Networking infrastructure must route packets from/to proper interface

By the way, after configuring FreeSwitch you can check which addresses
it actually uses with 'lsof -i' cli command.

Given your words 'packets from my sip provider are going to 192.168.1.7'
one can suspect that your network equipment forwards traffic from your
provider to that address. FreeSwitch must listen appropriate address but
it it does not control where provider's packets are sent to. That is
network must be reconfigured as well. This well might be the cause of
the problem if your FreeSwitch is behind NAT and network just routes
packets to the wrong address.

On the other hand, if you _public_ ip address is 192.168.1.7 and it is
configured on FreeSwitch itself then no network reconfiguration needed.
But consider the fact that provider is not aware of your internal
addresses and can not reach any address except for public one.

And finally: a variable 'at the top of my vars.xml' can be overwritten
later with another assignment in configuration files.

On 28/06/12 23:41, Joshua Nankin wrote:
> I have two interfaces on my machine, eth0=10.0.2.15 and
> eth1=192.168.1.7.  I'm trying to get freeswitch to only use eth1.  I've
> put the following at the top of my vars.xml:
>
> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.7"/>
>
> However, this does not seem to be working.  When I originate a call, I
> see that the initial invite is originating from 10.0.2.15, but incoming
> packets from my sip provider are going to 192.168.1.7 and are not being
> responded to by freeswitch.  Shouldn't freeswitch be listening on
> 192.168.1.7 and sending packets from there as well?
>
> What am I doing wrong?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



=========================================================

?????? ????????? ? ????? ???????? (???????????) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????.

=========================================================

This message and any attachments (the ?message?) are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC ?BELROSBANK? nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified.

=========================================================


From shcherbatyuk at belrosbank.by  Fri Jun 29 10:39:29 2012
From: shcherbatyuk at belrosbank.by (Eugene Shcherbatyuk)
Date: Fri, 29 Jun 2012 09:39:29 +0300
Subject: [Freeswitch-users] RTP delay > 100ms by performance testing
 with freeswitch as trancoding device SILK to G711a
In-Reply-To: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern>
References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern>
Message-ID: <4FED4DA1.10304@belrosbank.by>

Session limit is configurable. Have a look at switch.conf.xml file.


On 28/06/12 21:41, Hoexum, Edwin wrote:
>
> I am trying to do some load testing on a freeswitch for transcoding from
> SILK to G711a.  The question is:  Is  the limit  for RTP sessions
> limited to CPU power or are there also limits on the freeswitch
> application and can I tune freeswitch to do the job for 1500 sessions or
> more.


=========================================================

?????? ????????? ? ????? ???????? (???????????) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????.

=========================================================

This message and any attachments (the ?message?) are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC ?BELROSBANK? nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified.

=========================================================


From anton.jugatsu at gmail.com  Fri Jun 29 11:19:51 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Fri, 29 Jun 2012 11:19:51 +0400
Subject: [Freeswitch-users] Simple SPA8800 FXO connection to FreeSwitch
In-Reply-To: <CACMPV2CYALx0J_P7HG3T2RXtk2jo7mhER6fiKkvNHWexEt+gYg@mail.gmail.com>
References: <CACMPV2CYALx0J_P7HG3T2RXtk2jo7mhER6fiKkvNHWexEt+gYg@mail.gmail.com>
Message-ID: <CANBhyPB-mfAhrr09uGO=4VdDN3sEvC14nQXcG7NETdqB-Rnm_w@mail.gmail.com>

2012-06-29 04:13:44.298584 [ERR] mod_sofia.c:4492 Invalid Gateway ''

Are you sure that you compile and install mod_distributor or at least load
it. From fs_cli
module_exist mod_distributor

2012/6/29 Robin Gilks <robin.gilks at taitradio.com>

> Greetings
>
> I'm a noob with FS (and VoIP generally!!) and am really struggling
> with getting 2 way comms between a PABX and a soft-phone using FS and
> an SPA8800.
>
> Calls from the PABX are OK, the FXO ports have registered with FS and
> when I dial the FXO extension I get the dial tone, dial the number for
> the soft-phone and it all works fine.
>
> The problem is connections *TO* the PABX. The SPA8800 AFAIK has a
> separate SIP port for each of its FXO ports. My understanding of how
> it should work is as follows:
>
> I have to have 4 gateways defined in a file in the
> sip-profiles/externals directory, each with a different "sip-port"
> line. I'm testing with 2 to start with:
>
> # cat cicsco_fxo.xml
> <include>
>  <gateway name="fxoline3">
>    <param name="username" value="none"/>
>    <param name="password" value="none"/>
>    <param name="realm" value="172.27.1.20"/>
>    <param name="sip-port" value="5260"/>
>    <param name="register" value="false"/>
>  </gateway>
>  <gateway name="fxoline4">
>    <param name="username" value="none"/>
>    <param name="password" value="none"/>
>    <param name="realm" value="172.27.1.20"/>
>    <param name="sip-port" value="5360"/>
>    <param name="register" value="false"/>
>  </gateway>
>
> I have to use mod_distributor to select one of the gateways that
> define the SIP ports. Thats is configured in the autoload_configs
> directory.
>
> # cat distributor.conf.xml
> <configuration name="distributor.conf" description="Distributor
> Configuration">
>  <lists>
>    <list name="cisco_fxo" total-weight="2">
>      <node name="fxoline3" weight="1"/>
>      <node name="fxoline4" weight="1"/>
>    </list>
>  </lists>
> </configuration>
>
> Finally, I use a dialplan to access this distribution of gateways when
> a 4 digit number starting with 8 is dialed from the soft-phone.
>
> # cat 33_ciscofox.xml
> <include>
>  <extension name="pabx">
>    <condition field="destination_number" expression="^(8\d{3})$">
>      <action application="bridge"
> data="sofia/gateway/${distributor(cisco_fxo)}/$1"/>
>    </condition>
>  </extension>
> </include>
>
> Result expected - it all works.
> Result I get is:
>
> freeswitch at PBX.localdomain> 2012-06-29 04:13:44.287982 [NOTICE]
> switch_channel.c:926 New Channel sofia/internal/1002 at 172.16.164.33
> [d0fc47e6-c1a0-11e1-881d-f70f560259d2]
> 2012-06-29 04:13:44.287982 [INFO] mod_dialplan_xml.c:485 Processing
> 1002 <1002>->8569 in context default
> 2012-06-29 04:13:44.298584 [ERR] mod_sofia.c:4492 Invalid Gateway ''
> 2012-06-29 04:13:44.298584 [NOTICE] mod_sofia.c:4889 Close Channel N/A
> [CS_NEW]
> 2012-06-29 04:13:44.298584 [NOTICE] switch_ivr_originate.c:2535 Cannot
> create outgoing channel of type [sofia] cause: [INVALID_GATEWAY]
> 2012-06-29 04:13:44.298584 [INFO] mod_dptools.c:2956 Originate Failed.
>  Cause: INVALID_GATEWAY
> 2012-06-29 04:13:44.298584 [NOTICE] mod_dptools.c:3076 Hangup
> sofia/internal/1002 at 172.16.164.33 [CS_EXECUTE] [INVALID_GATEWAY]
> 2012-06-29 04:13:44.298584 [NOTICE] switch_core_session.c:1447 Session
> 11 (sofia/internal/1002 at 172.16.164.33) Ended
> 2012-06-29 04:13:44.298584 [NOTICE] switch_core_session.c:1449 Close
> Channel sofia/internal/1002 at 172.16.164.33 [CS_DESTROY]
>
> So what is invalid about the gateway?
>
> Cheers
>
> --
> Robin Gilks
>
> --
>
> ------------------------------
> This email, including any attachments, is only for the intended recipient.
> It is subject to copyright, is confidential and may be the subject of legal
> or other privilege, none of which is waived or lost by reason of this
> transmission.
> If you are not an intended recipient, you may not use, disseminate,
> distribute or reproduce such email, any attachments, or any part thereof.
> If you have received a message in error, please notify the sender
> immediately and erase all copies of the message and any attachments.
> Unfortunately, we cannot warrant that the email has not been altered or
> corrupted during transmission nor can we guarantee that any email or any
> attachments are free from computer viruses or other conditions which may
> damage or interfere with recipient data, hardware or software. The
> recipient relies upon its own procedures and assumes all risk of use and of
> opening any attachments.
> ------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From miha at softnet.si  Fri Jun 29 17:10:10 2012
From: miha at softnet.si (Miha)
Date: Fri, 29 Jun 2012 15:10:10 +0200
Subject: [Freeswitch-users] accessing variable after 302
Message-ID: <4FEDA932.3050702@softnet.si>

Hi,

I have one problem with variables when it goes for different PBX on same FS.

Scenario:

A calls B (B is registered on FS and has set 302 redirect to user who is 
on different PBX).

Problems is that FS uses varibles from user A (not on FS) which is 
calling user B. I can not use varibles from user B, which I must get for 
radius authentication.

p.s.: I have set munual redirect in sip_profiles

This is redirect.xml

<context name="redirected">

<extension name="check_auth" continue="true">
   <condition field="${sip_authorized}" expression="^true$">
     <anti-action application="respond" data="407"/>
   </condition>
</extension>

   <extension name="redirected_enterprise">
    <condition field="${domena}" expression="enterprise.fs2.xxxxx.si">
        <action application="set" data="sip_contact_user=${sip_to_user}" />
    <action application="info" />

        <action application="transfer" data="${sip_req_user} XML 
enterprise.fs2.xxxxx.si"/>

<!-- <action application="bridge" data="${sip_redirect_dialstring}" /> -->
  </condition>
   </extension>

 From info application:

Channel-State-Number: [4]
Channel-Name: [sofia/internal/051357952 at xxx.xxx.xxx.xxx]
Unique-ID: [cc021cec-c1e2-11e1-88b5-eb0ece2a11d3]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-HIT-Dialplan: [true]
Channel-Presence-ID: [051357952 at xxx.xxx.xxx.xxx]
Channel-Call-UUID: [cc021cec-c1e2-11e1-88b5-eb0ece2a11d3]
Answer-State: [ringing]
Caller-Direction: [inbound]
Caller-Username: [051357952]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [051357952]
Caller-Caller-ID-Number: [051357952]

051357952 is number, which is calling user B.


Thanks!

Miha

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From guojing1982423 at sina.com  Fri Jun 29 11:15:12 2012
From: guojing1982423 at sina.com (guojing1982423 at sina.com)
Date: Fri, 29 Jun 2012 15:15:12 +0800 
Subject: [Freeswitch-users] record_session cause memory increase
Message-ID: <20120629071512.56EF3978001@webmail.sinamail.sina.com.cn>

Hi,I'm a newbie. I add 32 sip users in FreeSwitch, and use Ameritec AM-2 to test it. There are about 40 calls every minutes. When i add record_session, the memory increase slowly, about dozens of megabytes every day. Can give me some adivces?The freeswitch version is 1.2.0r2CPU: Dual Core E6600 3.06GMemory: 2G
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From msc at freeswitch.org  Fri Jun 29 19:20:57 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jun 2012 08:20:57 -0700
Subject: [Freeswitch-users] Simple SPA8800 FXO connection to FreeSwitch
In-Reply-To: <CACMPV2CYALx0J_P7HG3T2RXtk2jo7mhER6fiKkvNHWexEt+gYg@mail.gmail.com>
References: <CACMPV2CYALx0J_P7HG3T2RXtk2jo7mhER6fiKkvNHWexEt+gYg@mail.gmail.com>
Message-ID: <CAKzWOxUaakFUCWUfTyG1DamW0AHvai6zdQ9Ls5w7mCKC4XGw1g@mail.gmail.com>

Go to fs_cli and type:

distributor cisco_fxo

If you get an error then you know your distributor is not set up correctly.
If you get "fxolineX" then your dialplan call to distributor is not
correct. Most likely it's the former, and most likely mod_distributor is
not compiled and/or installed.

In FS source:
Edit modules.conf and uncomment the line with "mod_distributor"
Do "make mod_distributor-install"

In fs_cli do:
load mod_distributor

Then go to conf/autoload_configs/modules.conf.xml
Uncomment the line w/ mod_distributor
That will make mod_distributor load each time FS starts.

-MC

On Thu, Jun 28, 2012 at 9:22 PM, Robin Gilks <robin.gilks at taitradio.com>wrote:

> Greetings
>
> I'm a noob with FS (and VoIP generally!!) and am really struggling
> with getting 2 way comms between a PABX and a soft-phone using FS and
> an SPA8800.
>
> Calls from the PABX are OK, the FXO ports have registered with FS and
> when I dial the FXO extension I get the dial tone, dial the number for
> the soft-phone and it all works fine.
>
> The problem is connections *TO* the PABX. The SPA8800 AFAIK has a
> separate SIP port for each of its FXO ports. My understanding of how
> it should work is as follows:
>
> I have to have 4 gateways defined in a file in the
> sip-profiles/externals directory, each with a different "sip-port"
> line. I'm testing with 2 to start with:
>
> # cat cicsco_fxo.xml
> <include>
>  <gateway name="fxoline3">
>    <param name="username" value="none"/>
>    <param name="password" value="none"/>
>    <param name="realm" value="172.27.1.20"/>
>    <param name="sip-port" value="5260"/>
>    <param name="register" value="false"/>
>  </gateway>
>  <gateway name="fxoline4">
>    <param name="username" value="none"/>
>    <param name="password" value="none"/>
>    <param name="realm" value="172.27.1.20"/>
>    <param name="sip-port" value="5360"/>
>    <param name="register" value="false"/>
>  </gateway>
>
> I have to use mod_distributor to select one of the gateways that
> define the SIP ports. Thats is configured in the autoload_configs
> directory.
>
> # cat distributor.conf.xml
> <configuration name="distributor.conf" description="Distributor
> Configuration">
>  <lists>
>    <list name="cisco_fxo" total-weight="2">
>      <node name="fxoline3" weight="1"/>
>      <node name="fxoline4" weight="1"/>
>    </list>
>  </lists>
> </configuration>
>
> Finally, I use a dialplan to access this distribution of gateways when
> a 4 digit number starting with 8 is dialed from the soft-phone.
>
> # cat 33_ciscofox.xml
> <include>
>  <extension name="pabx">
>    <condition field="destination_number" expression="^(8\d{3})$">
>      <action application="bridge"
> data="sofia/gateway/${distributor(cisco_fxo)}/$1"/>
>    </condition>
>  </extension>
> </include>
>
> Result expected - it all works.
> Result I get is:
>
> freeswitch at PBX.localdomain> 2012-06-29 04:13:44.287982 [NOTICE]
> switch_channel.c:926 New Channel sofia/internal/1002 at 172.16.164.33
> [d0fc47e6-c1a0-11e1-881d-f70f560259d2]
> 2012-06-29 04:13:44.287982 [INFO] mod_dialplan_xml.c:485 Processing
> 1002 <1002>->8569 in context default
> 2012-06-29 04:13:44.298584 [ERR] mod_sofia.c:4492 Invalid Gateway ''
> 2012-06-29 04:13:44.298584 [NOTICE] mod_sofia.c:4889 Close Channel N/A
> [CS_NEW]
> 2012-06-29 04:13:44.298584 [NOTICE] switch_ivr_originate.c:2535 Cannot
> create outgoing channel of type [sofia] cause: [INVALID_GATEWAY]
> 2012-06-29 04:13:44.298584 [INFO] mod_dptools.c:2956 Originate Failed.
>  Cause: INVALID_GATEWAY
> 2012-06-29 04:13:44.298584 [NOTICE] mod_dptools.c:3076 Hangup
> sofia/internal/1002 at 172.16.164.33 [CS_EXECUTE] [INVALID_GATEWAY]
> 2012-06-29 04:13:44.298584 [NOTICE] switch_core_session.c:1447 Session
> 11 (sofia/internal/1002 at 172.16.164.33) Ended
> 2012-06-29 04:13:44.298584 [NOTICE] switch_core_session.c:1449 Close
> Channel sofia/internal/1002 at 172.16.164.33 [CS_DESTROY]
>
> So what is invalid about the gateway?
>
> Cheers
>
> --
> Robin Gilks
>
> --
>
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From cmrienzo at gmail.com  Fri Jun 29 20:10:31 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 29 Jun 2012 12:10:31 -0400
Subject: [Freeswitch-users] record_session cause memory increase
In-Reply-To: <20120629071512.56EF3978001@webmail.sinamail.sina.com.cn>
References: <20120629071512.56EF3978001@webmail.sinamail.sina.com.cn>
Message-ID: <CAD-qRkj5xpct34Qf7dSejp2bk5T7BPiQLfMR5KXD1MaAP0WSfg@mail.gmail.com>

Yes, report bugs at http://jira.freeswitch.org



On Fri, Jun 29, 2012 at 3:15 AM, <guojing1982423 at sina.com> wrote:

> Hi,I'm a newbie. I add 32 sip users in FreeSwitch, and use Ameritec AM-2 to
> test it. There are about 40 calls every minutes. When i add
> record_session, the memory increase slowly, about dozens of megabytes
> every day. Can give me some adivces?
> The freeswitch version is 1.2.0r2
> CPU: Dual Core E6600 3.06G
> Memory: 2G
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Jun 29 20:24:04 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 29 Jun 2012 09:24:04 -0700 (PDT)
Subject: [Freeswitch-users] record_session cause memory increase
In-Reply-To: <CAD-qRkj5xpct34Qf7dSejp2bk5T7BPiQLfMR5KXD1MaAP0WSfg@mail.gmail.com>
References: <20120629071512.56EF3978001@webmail.sinamail.sina.com.cn>
	<CAD-qRkj5xpct34Qf7dSejp2bk5T7BPiQLfMR5KXD1MaAP0WSfg@mail.gmail.com>
Message-ID: <1340987044597-7580406.post@n2.nabble.com>

I think he already did. FS-4365

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/record-session-cause-memory-increase-tp7580403p7580406.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Fri Jun 29 21:07:09 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 29 Jun 2012 12:07:09 -0500
Subject: [Freeswitch-users] RTP delay > 100ms by performance testing
 with freeswitch as trancoding device SILK to G711a
In-Reply-To: <4FED4DA1.10304@belrosbank.by>
References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern>
	<4FED4DA1.10304@belrosbank.by>
Message-ID: <CAKbxfG_a8r_kTxiwAq9fQHM183F_ZSSD4_TTQ77Mcn4BdAuQdQ@mail.gmail.com>

you could do with more than 2 cpu, the more cores the better you will do.
For similar prices you could find a 8 or 12 core box.

The 3 most important things in a FS server are,  as many CPU cores as
possible, as much ram as possible, the best possible CPU with shared
bus to RAM and CPU.



2012/6/29 Eugene Shcherbatyuk <shcherbatyuk at belrosbank.by>:
> Session limit is configurable. Have a look at switch.conf.xml file.
>
>
> On 28/06/12 21:41, Hoexum, Edwin wrote:
>>
>> I am trying to do some load testing on a freeswitch for transcoding from
>> SILK to G711a.  The question is:  Is  the limit  for RTP sessions
>> limited to CPU power or are there also limits on the freeswitch
>> application and can I tune freeswitch to do the job for 1500 sessions or
>> more.
>
>
> =========================================================
>
> ?????? ????????? ? ????? ???????? (<<?????????>>) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????.
>
> =========================================================
>
> This message and any attachments (the "message") are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC "BELROSBANK" nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified.
>
> =========================================================
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

From anthony.minessale at gmail.com  Fri Jun 29 21:09:21 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 29 Jun 2012 12:09:21 -0500
Subject: [Freeswitch-users] Strange "timer_test" result
In-Reply-To: <CAO4LeU29SxnYB+tdiGV0Z3ZsNefNLG4WAAbct0dXegC3C2SsYA@mail.gmail.com>
References: <CAD-qRkhizVe+QsRHcgEseYbPhe2h7=VvtkMj1Nu0vLa00imwSw@mail.gmail.com>
	<CAO4LeU1ppjAo5dTgPywBOeGH0mn9pZC+ov+uu2LNNc6VxyDQcA@mail.gmail.com>
	<0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com>
	<CAO4LeU0F6kcH0tgpEx9039Xf45we2ZFLD_BHU_puhaEyY5mH7Q@mail.gmail.com>
	<F615FA10-493C-4AF5-ABE1-DB7E3AE8CA67@gmail.com>
	<1340748029314-7580297.post@n2.nabble.com>
	<CAO4LeU144W1OAV3+GBiZzPUNbf3HA1SXptAR3e3QFR63qX1zJg@mail.gmail.com>
	<CAD-qRkjmQSwtAGiY8MXWH=HuqwePMZU3xR1dJdYWo0ui243z5Q@mail.gmail.com>
	<CAO4LeU08sUKbzC1Ee4fjHa3vB_XEdRQgc=ePiqrVMJNZssYKsw@mail.gmail.com>
	<CAD-qRkhT0ZBca+Zz0wp3Fpc42pcV6x=eR6E_NErfeTDHz4G7nA@mail.gmail.com>
	<1340917348693-7580388.post@n2.nabble.com>
	<CAO4LeU29SxnYB+tdiGV0Z3ZsNefNLG4WAAbct0dXegC3C2SsYA@mail.gmail.com>
Message-ID: <CAKbxfG8oO8KtesHHDqj6CvNdvCaAzoiNzJ5_-5QMMdqpcrCA2g@mail.gmail.com>

Get a nice new many core intel box with a nice motherboard and give
freeswitch a nice home.


On Thu, Jun 28, 2012 at 5:21 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> I have bad experience with combination Opteron + RTP (choppy audio).?I think
> this is due to the size of the cache, for example for your 6128 is 128KB per
> core and, for example, for a Xeon E5649 is 12Mb per core. I own a 4 x
> Opteron 6176 (4 x 12 core) and after many tries to use it as Freeswitch
> server, now I'm using it as database server...
>
> Stephen
>
>
> On Thu, Jun 28, 2012 at 11:02 PM, marksly <mark at callassistants.com> wrote:
>>
>> To give everyone an update: I enabled the mod_posix_timer last night and
>> things appear to be running better today (fewer calls that have a sporadic
>> choppy audio). The server load is lower, but it is not where I would like
>> to
>> see it. The load is around 10 on average. I think that this might be due
>> to
>> the fact that I have "enable-clock-nanosleep" set to true. I am going to
>> try
>> disabling this and see if the load decreases.
>>
>> The timer_test results are more consistent and regular; a lot closer to
>> the
>> 20ms that you would expect to see.
>>
>> Server Information:
>> 2x AMD Opteron 6128 (8 cores)
>> 8GB RAM
>> CentOS release 5.7
>> Kernel: 2.6.18-274.7.1.el5
>> FreeSWITCH version:
>> 1.2.0-rc2+git~20120625T203158Z~7dc9a9cacc+unclean~20120626T041419Z
>>
>> Feel free to provide other suggestions if you see fit.
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Strange-timer-test-result-tp7399299p7580388.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From mark at callassistants.com  Fri Jun 29 21:20:27 2012
From: mark at callassistants.com (marksly)
Date: Fri, 29 Jun 2012 10:20:27 -0700 (PDT)
Subject: [Freeswitch-users] Strange "timer_test" result
In-Reply-To: <CAKbxfG8oO8KtesHHDqj6CvNdvCaAzoiNzJ5_-5QMMdqpcrCA2g@mail.gmail.com>
References: <CAO4LeU0F6kcH0tgpEx9039Xf45we2ZFLD_BHU_puhaEyY5mH7Q@mail.gmail.com>
	<F615FA10-493C-4AF5-ABE1-DB7E3AE8CA67@gmail.com>
	<1340748029314-7580297.post@n2.nabble.com>
	<CAO4LeU144W1OAV3+GBiZzPUNbf3HA1SXptAR3e3QFR63qX1zJg@mail.gmail.com>
	<CAD-qRkjmQSwtAGiY8MXWH=HuqwePMZU3xR1dJdYWo0ui243z5Q@mail.gmail.com>
	<CAO4LeU08sUKbzC1Ee4fjHa3vB_XEdRQgc=ePiqrVMJNZssYKsw@mail.gmail.com>
	<CAD-qRkhT0ZBca+Zz0wp3Fpc42pcV6x=eR6E_NErfeTDHz4G7nA@mail.gmail.com>
	<1340917348693-7580388.post@n2.nabble.com>
	<CAO4LeU29SxnYB+tdiGV0Z3ZsNefNLG4WAAbct0dXegC3C2SsYA@mail.gmail.com>
	<CAKbxfG8oO8KtesHHDqj6CvNdvCaAzoiNzJ5_-5QMMdqpcrCA2g@mail.gmail.com>
Message-ID: <1340990427081-7580409.post@n2.nabble.com>

Anthony & Stephen - thanks for the information.

Any suggestions on motherboards/processor models? Right now the Opteron
system is handling about 300 simultaneous conferences; my plan was to scale
this closer to around 1,000 per deployment.

I have not made the change to the nanosleep setting, but will be doing that
this weekend.

-Mark

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Strange-timer-test-result-tp7399299p7580409.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Fri Jun 29 22:05:36 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 29 Jun 2012 13:05:36 -0500
Subject: [Freeswitch-users] Bind extension to an IP address
In-Reply-To: <CAN5esQ+Wma3O_5TSb0Wj8=6t0TM=mZqx8b6bBrah4m+TvLz9Kw@mail.gmail.com>
References: <4FE7175F.90209@the800group.com> <4FE787D4.5080603@gmail.com>
	<CALNZuwm-RPQRBLMRq65fb9eVdUS2W9HURZC5y+=gQ5a2GWhtqA@mail.gmail.com>
	<CAKbxfG_iUqoU0VFAsKu5a=cwV8fh02WzrEx1R70cMgXXcFy6iQ@mail.gmail.com>
	<CAN5esQ+ZQ4v6U-EB9+6rdQ1uBr84c6sZL4S6oi2R=1wdah3k-Q@mail.gmail.com>
	<CAKzWOxWJuxd+PjBxDmRPa3UNJ6L68xEZotfCKc7HXccg7Jxtvw@mail.gmail.com>
	<CAN5esQ+Wma3O_5TSb0Wj8=6t0TM=mZqx8b6bBrah4m+TvLz9Kw@mail.gmail.com>
Message-ID: <CAKbxfG8qiC92z6FZdc40XGg5nFweQgmFE96BDQ5FuJwiNFM7Ow@mail.gmail.com>

they do work on gateways

<variables> tag with <variable> tags within it

the caveat is the calls can't always be tied back to the gateway.
If you use something that handles sip correctly there should be some
hints in the contact addr we register with to find your way back to
the gateway and parse the variables.



On Thu, Jun 28, 2012 at 7:17 PM, Richard Brady <rnbrady at gmail.com> wrote:
>>> Is there an equivalent for gateways?
>>>
>>> I can't find a way to bind incoming calls from a CIDR to a particular gateway.
>>>
>> Just curious - what functionality do you need this for?
>
> In order to configure DTMF per provider, i.e. per gateway:
>
> ?<gateway name="provider1">
> ? ?...
> ? ?<variables>
> ? ? ?<variable direction="both" name="dtmf_type" value="info">
> ? ?</variables>
> ?</gateway>
>
> ?<gateway name="provider2">
> ? ?...
> ? ?<variables>
> ? ? ?<variable direction="both" name="dtmf_type" value="rfc2833">
> ? ?</variables>
> ?</gateway>
>
> This works for outbound calls but not for inbound calls, even though I
> have specified direction="both".
>
> I believe that direction="both" and direction="inbound" will only work
> on a SIP profile, not on a gateway. But I'm hoping to be corrected.
>
> Richard
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From bdfoster at endigotech.com  Fri Jun 29 23:16:17 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 29 Jun 2012 15:16:17 -0400
Subject: [Freeswitch-users] record_session cause memory increase
In-Reply-To: <1340987044597-7580406.post@n2.nabble.com>
References: <20120629071512.56EF3978001@webmail.sinamail.sina.com.cn>
	<CAD-qRkj5xpct34Qf7dSejp2bk5T7BPiQLfMR5KXD1MaAP0WSfg@mail.gmail.com>
	<1340987044597-7580406.post@n2.nabble.com>
Message-ID: <CAKV-jFySpEtm_CQzZ_oPOZbdD7HvFG65_4nDK=xJUKnJwcfDuw@mail.gmail.com>

Thread closed.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Jun 29, 2012 12:24 PM, "Jeff Lenk" <jeff at jefflenk.com> wrote:

> I think he already did. FS-4365
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/record-session-cause-memory-increase-tp7580403p7580406.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Jun 30 01:11:28 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jun 2012 14:11:28 -0700
Subject: [Freeswitch-users] Connecting FS to MS UM
Message-ID: <CAKzWOxVfZdZQ1puvLARxHQfZCEC=jz=X46Ye=1criCCT5FwBcw@mail.gmail.com>

Hey, if anyone knows about how this can be done with FreeSWITCH, please
contact me off list:

http://help.outlook.com/en-us/140/gg702670%28d=loband%29.aspx

I know someone looking to provide a solution for his customer. I'd like to
get this solution documented on the wiki...

Thanks,
MC
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From jnankin at gmail.com  Sat Jun 30 02:45:28 2012
From: jnankin at gmail.com (Joshua Nankin)
Date: Fri, 29 Jun 2012 17:45:28 -0500
Subject: [Freeswitch-users] Multiple network interfaces,
	can't force freeswitch to use one.
In-Reply-To: <CAAYWZ8mPRQkg8gkb=3OMnYECC-oNMAR_CSmVjvvmW4dAO2C2-Q@mail.gmail.com>
References: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>
	<CAAYWZ8=bC4OGYYzqdNutsznvjjm1N6j55zkw9+-5XboMhr5pww@mail.gmail.com>
	<CAAYWZ8nt7ecAu29f=iNXwB9jvxjJVYTwrisir0xudwTuzzYSCA@mail.gmail.com>
	<CAAYWZ8nV+TCduKpg3JG33F-xCi+0apn784KCr-Z-a9K6y6sOPQ@mail.gmail.com>
	<CAMWiMy9czfecmfhLsXGcs7AL1PR095Oz8hR539r5WfZUHhbO6g@mail.gmail.com>
	<CAAYWZ8mPRQkg8gkb=3OMnYECC-oNMAR_CSmVjvvmW4dAO2C2-Q@mail.gmail.com>
Message-ID: <CAAYWZ8=7pT4s2Hf4zuVW-a-rtBF+EzwNB2Ub0sT+dJCP-G2gPQ@mail.gmail.com>

Hi Eugene.

So, you are correct, I have configured my network to forward SIP and RTP
traffic to my 192.* IP, and this is working as intended.  I have also now
manually put in the 192.* IP into the sofia external.xml and internal.xml
for sip-ip and rtp-ip.

Again, on this virtual machine there are two interfaces (eth0=10.* ip, and
eth1=192.* ip).  After restarting freeswitch and doing the lsof -i that you
suggested, I get the following for freeswitch:

freeswitc 17227 freeswitch   23u  IPv4  61949      0t0  UDP
192.168.1.10:5080
freeswitc 17227 freeswitch   26u  IPv4  61950      0t0  TCP
192.168.1.10:5080 (LISTEN)
freeswitc 17227 freeswitch   30u  IPv6  61957      0t0  UDP
ip6-localhost:5080
freeswitc 17227 freeswitch   31u  IPv6  61958      0t0  TCP
ip6-localhost:5080 (LISTEN)
freeswitc 17227 freeswitch   35u  IPv4  62707      0t0  UDP 192.168.1.10:
sip
freeswitc 17227 freeswitch   36u  IPv4  62708      0t0  TCP 192.168.1.10:sip
(LISTEN)
freeswitc 17227 freeswitch   37u  IPv4  61960      0t0  UDP 10.0.2.15:33908
->chic-cns.area4.il.chicago.comcast.net:domain
freeswitc 17227 freeswitch   40u  IPv4  61964      0t0  TCP
localhost:zope-ftp (LISTEN)

I'm not sure where that 10.0.2.15 is configured in freeswitch, or even what
that port is for.

So, below I've linked to two packet captures for both eth0 and eth1.
 You'll notice that the intial INVITE is occuring on eth0, but that we're
receiving traffic from the sip provider on eth1 that is not being responded
to by freeswitch.  I still can't figure out why that is happening (both why
eth0 is even being used, and why freeswitch is not responding to traffic
directed at the other IP).

http://joshnankin.com/tcpdump0.pcap
http://joshnankin.com/tcpdump1.pcap

Thanks again for all of your help.


On Thu, Jun 28, 2012 at 7:31 PM, Joshua Nankin <jnankin at gmail.com> wrote:

> I'm stumped.  Do you think it would be possible to use a sofia alias to
> get the profile to listen to the other ip (192.x) as well?
>
>
> On Thu, Jun 28, 2012 at 5:10 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:
>
>> That could be causing the problems right there if Vagrant is doing
>> some NATing or mangling the IP packets
>>
>> On Thu, Jun 28, 2012 at 2:48 PM, Joshua Nankin <jnankin at gmail.com> wrote:
>> > Right, but freeswitch is still getting packets to 192.168.1.7, but
>> sending
>> > from 10.0.2.15.
>> >
>> > Not sure if this makes a difference, but this is a Vagrant virtual
>> machine
>> > with a bridged interface.  The 10.x address is an internal Vagrant ip,
>> but
>> > the 192 is from my router DHCP.  I've set the external sip and rtp ips
>> to my
>> > public IP address, and that IP is what appears for the profile in sofia
>> > status.
>> >
>> >
>> >
>> > On Thu, Jun 28, 2012 at 4:36 PM, Joshua Nankin <jnankin at gmail.com>
>> wrote:
>> >>
>> >> I did a sudo /etc/init.d/freeswitch restart.  That should be enough
>> >> correct?
>> >>
>> >> On Thu, Jun 28, 2012 at 4:08 PM, Joshua Nankin <jnankin at gmail.com>
>> wrote:
>> >>>
>> >>> sip-ip and rtp-ip in both external.xml and internal.xml were set to
>> >>> $${local_ip_v4}.  I figured that setting the ip in vars.xml would be
>> enough.
>> >>>
>> >>> I manually set sip-ip and rtp-ip and both of those files, but that
>> still
>> >>> did not solve the problem.
>> >>>
>> >>> Any other ideas?
>> >>>
>> >>>
>> >>> On Thu, Jun 28, 2012 at 3:41 PM, Joshua Nankin <jnankin at gmail.com>
>> wrote:
>> >>>>
>> >>>> I have two interfaces on my machine, eth0=10.0.2.15 and
>> >>>> eth1=192.168.1.7.  I'm trying to get freeswitch to only use eth1.
>>  I've put
>> >>>> the following at the top of my vars.xml:
>> >>>>
>> >>>> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.7"/>
>> >>>>
>> >>>> However, this does not seem to be working.  When I originate a call,
>> I
>> >>>> see that the initial invite is originating from 10.0.2.15, but
>> incoming
>> >>>> packets from my sip provider are going to 192.168.1.7 and are not
>> being
>> >>>> responded to by freeswitch.  Shouldn't freeswitch be listening on
>> >>>> 192.168.1.7 and sending packets from there as well?
>> >>>>
>> >>>> What am I doing wrong?
>> >>>>
>> >>>>
>> >>>
>> >>>
>> >>
>> >
>>
>
>
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From msc at freeswitch.org  Sat Jun 30 02:56:26 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jun 2012 15:56:26 -0700
Subject: [Freeswitch-users] Multiple network interfaces,
 can't force freeswitch to use one.
In-Reply-To: <CAAYWZ8=7pT4s2Hf4zuVW-a-rtBF+EzwNB2Ub0sT+dJCP-G2gPQ@mail.gmail.com>
References: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>
	<CAAYWZ8=bC4OGYYzqdNutsznvjjm1N6j55zkw9+-5XboMhr5pww@mail.gmail.com>
	<CAAYWZ8nt7ecAu29f=iNXwB9jvxjJVYTwrisir0xudwTuzzYSCA@mail.gmail.com>
	<CAAYWZ8nV+TCduKpg3JG33F-xCi+0apn784KCr-Z-a9K6y6sOPQ@mail.gmail.com>
	<CAMWiMy9czfecmfhLsXGcs7AL1PR095Oz8hR539r5WfZUHhbO6g@mail.gmail.com>
	<CAAYWZ8mPRQkg8gkb=3OMnYECC-oNMAR_CSmVjvvmW4dAO2C2-Q@mail.gmail.com>
	<CAAYWZ8=7pT4s2Hf4zuVW-a-rtBF+EzwNB2Ub0sT+dJCP-G2gPQ@mail.gmail.com>
Message-ID: <CAKzWOxXRHeK+YZQhOoBgfQEh+03cTRNhMaJtpk4V1o2Xk941ug@mail.gmail.com>

Did I see "Chicago" in that Comast domain name? I know it's still a month
off but if you're near Chicago it might behoove you to come hang out with
us at ClueCon. It's the single largest gathering of FreeSWITCH gurus in the
world...

Question: can rotate logs and do a fresh startup of FreeSWITCH? I think if
we see the log from the point FreeSWITCH starts up we might be able to tell
where that 10.x.x.x address is coming from.

-MC

On Fri, Jun 29, 2012 at 3:45 PM, Joshua Nankin <jnankin at gmail.com> wrote:

> Hi Eugene.
>
> So, you are correct, I have configured my network to forward SIP and RTP
> traffic to my 192.* IP, and this is working as intended.  I have also now
> manually put in the 192.* IP into the sofia external.xml and internal.xml
> for sip-ip and rtp-ip.
>
> Again, on this virtual machine there are two interfaces (eth0=10.* ip, and
> eth1=192.* ip).  After restarting freeswitch and doing the lsof -i that you
> suggested, I get the following for freeswitch:
>
> freeswitc 17227 freeswitch   23u  IPv4  61949      0t0  UDP
> 192.168.1.10:5080
> freeswitc 17227 freeswitch   26u  IPv4  61950      0t0  TCP
> 192.168.1.10:5080 (LISTEN)
> freeswitc 17227 freeswitch   30u  IPv6  61957      0t0  UDP
> ip6-localhost:5080
> freeswitc 17227 freeswitch   31u  IPv6  61958      0t0  TCP
> ip6-localhost:5080 (LISTEN)
> freeswitc 17227 freeswitch   35u  IPv4  62707      0t0  UDP 192.168.1.10:
> sip
> freeswitc 17227 freeswitch   36u  IPv4  62708      0t0  TCP 192.168.1.10:sip
> (LISTEN)
> freeswitc 17227 freeswitch   37u  IPv4  61960      0t0  UDP 10.0.2.15:33908
> ->chic-cns.area4.il.chicago.comcast.net:domain
> freeswitc 17227 freeswitch   40u  IPv4  61964      0t0  TCP
> localhost:zope-ftp (LISTEN)
>
> I'm not sure where that 10.0.2.15 is configured in freeswitch, or even
> what that port is for.
>
> So, below I've linked to two packet captures for both eth0 and eth1.
>  You'll notice that the intial INVITE is occuring on eth0, but that we're
> receiving traffic from the sip provider on eth1 that is not being responded
> to by freeswitch.  I still can't figure out why that is happening (both why
> eth0 is even being used, and why freeswitch is not responding to traffic
> directed at the other IP).
>
> http://joshnankin.com/tcpdump0.pcap
> http://joshnankin.com/tcpdump1.pcap
>
> Thanks again for all of your help.
>
>
> On Thu, Jun 28, 2012 at 7:31 PM, Joshua Nankin <jnankin at gmail.com> wrote:
>
>> I'm stumped.  Do you think it would be possible to use a sofia alias to
>> get the profile to listen to the other ip (192.x) as well?
>>
>>
>> On Thu, Jun 28, 2012 at 5:10 PM, curriegrad2004 <curriegrad2004 at gmail.com
>> > wrote:
>>
>>> That could be causing the problems right there if Vagrant is doing
>>> some NATing or mangling the IP packets
>>>
>>> On Thu, Jun 28, 2012 at 2:48 PM, Joshua Nankin <jnankin at gmail.com>
>>> wrote:
>>> > Right, but freeswitch is still getting packets to 192.168.1.7, but
>>> sending
>>> > from 10.0.2.15.
>>> >
>>> > Not sure if this makes a difference, but this is a Vagrant virtual
>>> machine
>>> > with a bridged interface.  The 10.x address is an internal Vagrant ip,
>>> but
>>> > the 192 is from my router DHCP.  I've set the external sip and rtp ips
>>> to my
>>> > public IP address, and that IP is what appears for the profile in sofia
>>> > status.
>>> >
>>> >
>>> >
>>> > On Thu, Jun 28, 2012 at 4:36 PM, Joshua Nankin <jnankin at gmail.com>
>>> wrote:
>>> >>
>>> >> I did a sudo /etc/init.d/freeswitch restart.  That should be enough
>>> >> correct?
>>> >>
>>> >> On Thu, Jun 28, 2012 at 4:08 PM, Joshua Nankin <jnankin at gmail.com>
>>> wrote:
>>> >>>
>>> >>> sip-ip and rtp-ip in both external.xml and internal.xml were set to
>>> >>> $${local_ip_v4}.  I figured that setting the ip in vars.xml would be
>>> enough.
>>> >>>
>>> >>> I manually set sip-ip and rtp-ip and both of those files, but that
>>> still
>>> >>> did not solve the problem.
>>> >>>
>>> >>> Any other ideas?
>>> >>>
>>> >>>
>>> >>> On Thu, Jun 28, 2012 at 3:41 PM, Joshua Nankin <jnankin at gmail.com>
>>> wrote:
>>> >>>>
>>> >>>> I have two interfaces on my machine, eth0=10.0.2.15 and
>>> >>>> eth1=192.168.1.7.  I'm trying to get freeswitch to only use eth1.
>>>  I've put
>>> >>>> the following at the top of my vars.xml:
>>> >>>>
>>> >>>> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.7"/>
>>> >>>>
>>> >>>> However, this does not seem to be working.  When I originate a
>>> call, I
>>> >>>> see that the initial invite is originating from 10.0.2.15, but
>>> incoming
>>> >>>> packets from my sip provider are going to 192.168.1.7 and are not
>>> being
>>> >>>> responded to by freeswitch.  Shouldn't freeswitch be listening on
>>> >>>> 192.168.1.7 and sending packets from there as well?
>>> >>>>
>>> >>>> What am I doing wrong?
>>> >>>>
>>> >>>>
>>> >>>
>>> >>>
>>> >>
>>> >
>>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From saami_mh at ymail.com  Sat Jun 30 11:40:33 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Sat, 30 Jun 2012 00:40:33 -0700 (PDT)
Subject: [Freeswitch-users] How to Filter Event on lua script?
Message-ID: <1341042033.12901.YahooMailNeo@web120101.mail.ne1.yahoo.com>

hi all,
i want to filter eventname of general,how can i do that suing lua ?

filter EventName GENERAL

thanks ...
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From shcherbatyuk at belrosbank.by  Sat Jun 30 12:23:08 2012
From: shcherbatyuk at belrosbank.by (Eugene Shcherbatyuk)
Date: Sat, 30 Jun 2012 11:23:08 +0300
Subject: [Freeswitch-users] Multiple network interfaces,
	can't force freeswitch to use one.
In-Reply-To: <CAAYWZ8=7pT4s2Hf4zuVW-a-rtBF+EzwNB2Ub0sT+dJCP-G2gPQ@mail.gmail.com>
References: <CAAYWZ8kSmcy7OMtUqRYWJX4D6AEz-1ZtS2bC1pEr+RF4RoF-fg@mail.gmail.com>
	<CAAYWZ8=bC4OGYYzqdNutsznvjjm1N6j55zkw9+-5XboMhr5pww@mail.gmail.com>
	<CAAYWZ8nt7ecAu29f=iNXwB9jvxjJVYTwrisir0xudwTuzzYSCA@mail.gmail.com>
	<CAAYWZ8nV+TCduKpg3JG33F-xCi+0apn784KCr-Z-a9K6y6sOPQ@mail.gmail.com>
	<CAMWiMy9czfecmfhLsXGcs7AL1PR095Oz8hR539r5WfZUHhbO6g@mail.gmail.com>
	<CAAYWZ8mPRQkg8gkb=3OMnYECC-oNMAR_CSmVjvvmW4dAO2C2-Q@mail.gmail.com>
	<CAAYWZ8=7pT4s2Hf4zuVW-a-rtBF+EzwNB2Ub0sT+dJCP-G2gPQ@mail.gmail.com>
Message-ID: <4FEEB76C.8070703@belrosbank.by>

Hi Joshua,

I am afraid your dumps are a bit strange. That is:

-- SIP session at your public address is fine;
-- there are only outgoing RTP packets from your public address and no
packets to it in the first dump;
-- the other dump shows only incoming T38 packets to your public interface;
-- none of dumps shows sip/rtp traffic on your internal address (eth0);
-- dumps show Internet (dns, ntp) traffic on your internal interface;
-- your 'public' address actually belongs to 'disconnected' network,
which is not routed in the Internet, but dump show connection with real
public addresses, that is there is NAT on the way;

That makes me suspect that problem is within network configuration and,
probably, FreeSwitch NAT traversal configuration. It seems the problem
is not that FreeSwitch uses wrong address for I can not notice a sign
for the case in your dumps.



On 30/06/12 01:45, Joshua Nankin wrote:
> Hi Eugene.
>
> So, you are correct, I have configured my network to forward SIP and RTP
> traffic to my 192.* IP, and this is working as intended.  I have also
> now manually put in the 192.* IP into the sofia external.xml and
> internal.xml for sip-ip and rtp-ip.
>
> Again, on this virtual machine there are two interfaces (eth0=10.* ip,
> and eth1=192.* ip).  After restarting freeswitch and doing the lsof -i
> that you suggested, I get the following for freeswitch:
>
> freeswitc 17227 freeswitch   23u  IPv4  61949      0t0  UDP
> 192.168.1.10:5080 <http://192.168.1.10:5080>
> freeswitc 17227 freeswitch   26u  IPv4  61950      0t0  TCP
> 192.168.1.10:5080 <http://192.168.1.10:5080> (LISTEN)
> freeswitc 17227 freeswitch   30u  IPv6  61957      0t0  UDP
> ip6-localhost:5080
> freeswitc 17227 freeswitch   31u  IPv6  61958      0t0  TCP
> ip6-localhost:5080 (LISTEN)
> freeswitc 17227 freeswitch   35u  IPv4  62707      0t0  UDP
> 192.168.1.10:sip
> freeswitc 17227 freeswitch   36u  IPv4  62708      0t0  TCP
> 192.168.1.10:sip (LISTEN)
> freeswitc 17227 freeswitch   37u  IPv4  61960      0t0  UDP
> 10.0.2.15:33908->chic-cns.area4.il.chicago.comcast.net:domain
> freeswitc 17227 freeswitch   40u  IPv4  61964      0t0  TCP
> localhost:zope-ftp (LISTEN)
>
> I'm not sure where that 10.0.2.15 is configured in freeswitch, or even
> what that port is for.
>
> So, below I've linked to two packet captures for both eth0 and eth1.
>   You'll notice that the intial INVITE is occuring on eth0, but that
> we're receiving traffic from the sip provider on eth1 that is not being
> responded to by freeswitch.  I still can't figure out why that is
> happening (both why eth0 is even being used, and why freeswitch is not
> responding to traffic directed at the other IP).
>
> http://joshnankin.com/tcpdump0.pcap
> http://joshnankin.com/tcpdump1.pcap
>
> Thanks again for all of your help.
>


=========================================================

?????? ????????? ? ????? ???????? (???????????) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????.

=========================================================

This message and any attachments (the ?message?) are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC ?BELROSBANK? nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified.

=========================================================


From saami_mh at ymail.com  Sat Jun 30 13:00:23 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Sat, 30 Jun 2012 02:00:23 -0700 (PDT)
Subject: [Freeswitch-users] is there any channel or event to findout the
	status of sip user is register or not ?
Message-ID: <1341046823.34000.YahooMailNeo@web120103.mail.ne1.yahoo.com>

hi,
is there any channel variable / events etc .. for check the status of sip user ?

i am going to check the status of sip user when register to sip phone and when exit from sip phone (not registered)
using fs_cli we can check user is register or not "?sofia status profile internal reg" but i want to check using event or channel on lua script

thanks...
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From saami_mh at ymail.com  Sat Jun 30 16:57:32 2012
From: saami_mh at ymail.com (Samira Mh)
Date: Sat, 30 Jun 2012 05:57:32 -0700 (PDT)
Subject: [Freeswitch-users] how to get the content of sip header ?
Message-ID: <1341061052.23871.YahooMailNeo@web120106.mail.ne1.yahoo.com>

hi,
how to get the content of sip header ?in lua?
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From andrew at cassidywebservices.co.uk  Sat Jun 30 17:24:33 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sat, 30 Jun 2012 14:24:33 +0100
Subject: [Freeswitch-users] is there any channel or event to findout the
 status of sip user is register or not ?
In-Reply-To: <1341046823.34000.YahooMailNeo@web120103.mail.ne1.yahoo.com>
References: <1341046823.34000.YahooMailNeo@web120103.mail.ne1.yahoo.com>
Message-ID: <CAMbW9o+yjzPf5r=0mwcsnBCcUVCvM1F0E76RTGq8AHU1OJ+AyA@mail.gmail.com>

Hi,

You can use sofia_contact, if the user isn't registered it'll return
error/not_registered and will return a valid dial string if they are.

Regards,

On 30 June 2012 10:00, Samira Mh <saami_mh at ymail.com> wrote:

> hi,
> is there any channel variable / events etc .. for check the status of sip
> user ?
> i am going to check the status of sip user when register to sip phone and
> when exit from sip phone (not registered)
> using fs_cli we can check user is register or not " sofia status profile
> internal reg" but i want to check using event or channel on lua script
>
> thanks...
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From kevinfuhn at gmail.com  Sat Jun 30 12:57:31 2012
From: kevinfuhn at gmail.com (Kevin Fu)
Date: Sat, 30 Jun 2012 01:57:31 -0700 (PDT)
Subject: [Freeswitch-users] skypopen.ko insertion issue
In-Reply-To: <CALXCt0po-OC0ECQzUOx+g2GfM6n9=hoUO+x4W4NP7OEpYssVzQ@mail.gmail.com>
References: <1340767034403-7580303.post@n2.nabble.com>
	<CALXCt0po-OC0ECQzUOx+g2GfM6n9=hoUO+x4W4NP7OEpYssVzQ@mail.gmail.com>
Message-ID: <1341046651098-7580417.post@n2.nabble.com>

Thanks Giovanni. i am reinstallig ubuntu and FS. One more thing, when
installing python2.6-dev, the system said it had been replaced by
python2.7-dev and it was not available in the repository any more. Do i need
to go ahead to install python2.7-dev or have to discard the install and then
end up with installing python2.6-dev? Thanks again.

Thanks,
Kevin

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/skypopen-ko-insertion-issue-tp7580303p7580417.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rnbrady at gmail.com  Sat Jun 30 15:21:25 2012
From: rnbrady at gmail.com (Richard Brady)
Date: Sat, 30 Jun 2012 12:21:25 +0100
Subject: [Freeswitch-users] Bind extension to an IP address
In-Reply-To: <CAKbxfG8qiC92z6FZdc40XGg5nFweQgmFE96BDQ5FuJwiNFM7Ow@mail.gmail.com>
References: <4FE7175F.90209@the800group.com> <4FE787D4.5080603@gmail.com>
	<CALNZuwm-RPQRBLMRq65fb9eVdUS2W9HURZC5y+=gQ5a2GWhtqA@mail.gmail.com>
	<CAKbxfG_iUqoU0VFAsKu5a=cwV8fh02WzrEx1R70cMgXXcFy6iQ@mail.gmail.com>
	<CAN5esQ+ZQ4v6U-EB9+6rdQ1uBr84c6sZL4S6oi2R=1wdah3k-Q@mail.gmail.com>
	<CAKzWOxWJuxd+PjBxDmRPa3UNJ6L68xEZotfCKc7HXccg7Jxtvw@mail.gmail.com>
	<CAN5esQ+Wma3O_5TSb0Wj8=6t0TM=mZqx8b6bBrah4m+TvLz9Kw@mail.gmail.com>
	<CAKbxfG8qiC92z6FZdc40XGg5nFweQgmFE96BDQ5FuJwiNFM7Ow@mail.gmail.com>
Message-ID: <CAN5esQKhqQbBt0KEDLBDbh+P0b3DEHFFYdrqg9-5hE15uefkNQ@mail.gmail.com>

On 29 June 2012 19:05, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> the caveat is the calls can't always be tied back to the gateway.
> If you use something that handles sip correctly there should be some
> hints in the contact addr we register with to find your way back to
> the gateway and parse the variables.

That's exactly what I'm saying.

Inbound calls simply cannot be tied to a gateway, or alternatively
stated: gateways are an outbound-only concept.

So the next question is: how can one set a variable on an inbound call
(before CS_INIT)?

Richard


From xyangni at gmail.com  Sat Jun 30 23:24:09 2012
From: xyangni at gmail.com (Yihui Li)
Date: Sat, 30 Jun 2012 20:24:09 +0100
Subject: [Freeswitch-users] NAT type unknown, ExtIP empty, and ACK timeout
Message-ID: <CAO1DH18VQ_b-ee=rdj7ouAr7CXecp+QQ32-UO8XaVdU5wZw+fw@mail.gmail.com>

Hi,

I am running FS behind a nat, however the nat does not seem to work and
nat_map status shows:
NAT type unknown, ExtIP empty.

My call to the server on external context always get disconnected in about
30 seconds.
Siptrace indicate a ACK timeout as the reason.

I am using a TP-Link router and already set the server as DMZ host. Can
anyone help? Thanks
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From xyangni at gmail.com  Sat Jun 30 23:48:53 2012
From: xyangni at gmail.com (Yihui Li)
Date: Sat, 30 Jun 2012 20:48:53 +0100
Subject: [Freeswitch-users] NAT type unknown, ExtIP empty,
	and ACK timeout
In-Reply-To: <CAO1DH18VQ_b-ee=rdj7ouAr7CXecp+QQ32-UO8XaVdU5wZw+fw@mail.gmail.com>
References: <CAO1DH18VQ_b-ee=rdj7ouAr7CXecp+QQ32-UO8XaVdU5wZw+fw@mail.gmail.com>
Message-ID: <CAO1DH1-iA90i6Ki3raqbWGR7Tygh+JKfjoLsMyyx7is3+8yp7Q@mail.gmail.com>

SIP trace on the external FS server give the following

send 386 bytes to udp/[192.168.0.8]:5060 at 19:44:02.932215:
   ------------------------------------------------------------------------
   ACK sip:tttt at 192.168.0.8:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 178.79.156.136:5080;rport;branch=z9hG4bKXKvp0eHNU6t8S
   Max-Forwards: 70
   From: "" <sip:0000000000 at 178.79.156.136>;tag=pUtN7g4DcUtmj
   To: <sip:tttt at 95.172.231.192>;tag=FBttS33jBre1j
   Call-ID: b66f4c06-3d8e-1230-2488-fefdb24f9c88
   CSeq: 30206641 ACK
   Contact: <sip:mod_sofia at 178.79.156.136:5080>
   Content-Length: 0

do this mean the ACK was sent to the private IP address instead of the
correct public ip address?

On Sat, Jun 30, 2012 at 8:24 PM, Yihui Li <xyangni at gmail.com> wrote:

> Hi,
>
> I am running FS behind a nat, however the nat does not seem to work and
> nat_map status shows:
> NAT type unknown, ExtIP empty.
>
> My call to the server on external context always get disconnected in about
> 30 seconds.
> Siptrace indicate a ACK timeout as the reason.
>
> I am using a TP-Link router and already set the server as DMZ host. Can
> anyone help? Thanks
>
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