From rosphotos82 at gmail.com  Sat Dec  1 00:34:18 2012
From: rosphotos82 at gmail.com (Ros P)
Date: Fri, 30 Nov 2012 23:34:18 +0200
Subject: [Freeswitch-users] WebRTC Call
Message-ID: <CANJ4vTraADdabQWs4sT-zs1KTUh59k4RXfPXQnsbqqgPftJSjw@mail.gmail.com>

Hello,

Has anyone had any luck in implementing a webrtc call? I have been able to
register using the webrtc2sip gw and sipml5 client but no audio goes
through. As this seems to be a client problem (issue
39<http://code.google.com/p/sipml5/issues/detail?id=39>),
does anyone have any other suggestions? Anything successfully tested?

Kind Regards,
ros
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From spencer at 5ninesolutions.com  Sat Dec  1 00:57:16 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 30 Nov 2012 13:57:16 -0800
Subject: [Freeswitch-users] Caller Name with FreeTDM libpri
In-Reply-To: <5F8689C5-3C82-4762-BD2B-DFE9659C19B0@freeswitch.org>
References: <F31EBEBC-5C9B-43BE-8282-91BC976D8809@5ninesolutions.com>
	<5F8689C5-3C82-4762-BD2B-DFE9659C19B0@freeswitch.org>
Message-ID: <9044DBD2-A474-42D3-9DB8-97E26C6547D0@5ninesolutions.com>

Unfortunately those only work with Sangoma spans.  I'm using ftmod_libpri.

On Nov 30, 2012, at 12:48 PM, Brian West wrote:

> facility
> facility-timeout
> 
> Are the params you want.
> 
> --
> 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
> ISN: 410*543
> 
> 
> 
> 
> 
> On Nov 30, 2012, at 1:30 PM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> 
>> Hello,
>> I have an ISDN PRI where the caller name is sent in a FACILITY message after the initial SETUP message.  Does anyone know how I can access this?  I'm using FreeTDM in DAHDI mode with libpri.
>> 
>> Thanks,
>> Spencer
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From anthony.minessale at gmail.com  Sat Dec  1 01:28:31 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Nov 2012 16:28:31 -0600
Subject: [Freeswitch-users] mod_cdr_mongodb follows json rfc4627?
In-Reply-To: <CAJOPU6XN2Wh0ROUEK45-4W-8XmEsM7qiomXFtU+odi5Vr2Qk2g@mail.gmail.com>
References: <CAJOPU6XN2Wh0ROUEK45-4W-8XmEsM7qiomXFtU+odi5Vr2Qk2g@mail.gmail.com>
Message-ID: <CAKbxfG_-BU2nOT6iK-U2hJb550_fYhq-hbt1VCvx4_+LCn2J+w@mail.gmail.com>

you are right, post the patch to JIRA please.



On Fri, Nov 30, 2012 at 1:28 PM, Jose Fco. Irles Dur? <josefu at gmail.com>wrote:

> Hello everybody, first of all, sorry for my english.
>
> I have a test server with the mod_cdr_mongodb backend configured.
>
> I'm trying to parse the cdrs with the mongodb driver for java, but i
> have a trouble with a part of the json that freeswitch sends (mongo
> saves the json document without errors). I haven't an API function to
> extract some data. Firstly I thought that the problem was in the java
> driver but I'm not sure.
>
> The problem is in the "callflow" part, with the json standard[1], a
> json document can't have two keys with the same name, but
> mod_cdr_mongodb builds the json with a 'n' callflow objects.
> Also it happends inside callflow object, with origination, originator
> and originatee objects.
>
> mod_cdr_mongodb saves the json with this format:
> http://pastebin.com/VRz6s0eb
>
> I modified the source of the backend adding json arrays around this
> objects and now I can parse without problems.
>
> The output with the modifications:
> http://pastebin.com/v05FN9Ta
>
> I'm in the correct way or I'm missing something?
>
> [1] http://www.ietf.org/rfc/rfc4627.txt Section 2.2
>
> Regards
> --
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
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From itispip-qq at hotmail.com  Sat Dec  1 10:41:44 2012
From: itispip-qq at hotmail.com (=?gb2312?B?zfXA7Q==?=)
Date: Sat, 1 Dec 2012 15:41:44 +0800
Subject: [Freeswitch-users] Missed call notification
Message-ID: <BLU169-W120766FAB52400013BA1BB28C420@phx.gbl>


I heard that Freeswitch support missed call notification (i/o VM notification). But look around Wiki no document about this feature. Anyone knows where to setup it up?

/brgds, Alex 		 	   		  
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From steveayre at gmail.com  Sat Dec  1 17:58:32 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 1 Dec 2012 14:58:32 +0000
Subject: [Freeswitch-users] Missed call notification
In-Reply-To: <BLU169-W120766FAB52400013BA1BB28C420@phx.gbl>
References: <BLU169-W120766FAB52400013BA1BB28C420@phx.gbl>
Message-ID: <F1B7119A-EA54-4612-A01E-B66949F50F36@gmail.com>

Mod_voicemail supports emailing when a vm is left

Check the mod_voicemail page on the wiki

Steve on iPhone



On 1 Dec 2012, at 07:41, ?? <itispip-qq at hotmail.com> wrote:

> I heard that Freeswitch support missed call notification (i/o VM notification). But look around Wiki no document about this feature. Anyone knows where to setup it up?
> 
> /brgds, 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From boris at tagnet.ru  Sat Dec  1 19:16:51 2012
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 01 Dec 2012 22:16:51 +0600
Subject: [Freeswitch-users] Stange problems with 1.2.5 branch
Message-ID: <50BA2D73.6070909@tagnet.ru>

Hello!

     After upgrade to 1.2.5 (and so on) I found 2 strange problems:
1) mod_fifo stop working. There are errors in log that there is no table 
fifo_callers. I looked at mod_fifo.c and created all necessary tables 
but this doesn't helps.
2) The problem looks like "unconditional VAD". I use G711A in my 
network, vad is disabled. But when I make I call - the first "hello" is 
"eaten".

Anyone may confirm same problems?

-- 
Regards,
Boris



From bdfoster at endigotech.com  Sat Dec  1 19:59:23 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 1 Dec 2012 11:59:23 -0500
Subject: [Freeswitch-users] Missed call notification
In-Reply-To: <BLU169-W120766FAB52400013BA1BB28C420@phx.gbl>
References: <BLU169-W120766FAB52400013BA1BB28C420@phx.gbl>
Message-ID: <8AE80082-5AAC-4C25-87E9-52FBD5EDB593@endigotech.com>

Message Waiting Indicator (MWI) is the voicemail notification feature for IP phones. 

Missed call notifications are provided by the phone and not the switch.

Sent from my iPhone

On Dec 1, 2012, at 2:41 AM, ?? <itispip-qq at hotmail.com> wrote:

> I heard that Freeswitch support missed call notification (i/o VM notification). But look around Wiki no document about this feature. Anyone knows where to setup it up?
> 
> /brgds, 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From yehavi.bourvine at gmail.com  Sat Dec  1 20:13:11 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 1 Dec 2012 19:13:11 +0200
Subject: [Freeswitch-users] Missed call notification
In-Reply-To: <8AE80082-5AAC-4C25-87E9-52FBD5EDB593@endigotech.com>
References: <BLU169-W120766FAB52400013BA1BB28C420@phx.gbl>
	<8AE80082-5AAC-4C25-87E9-52FBD5EDB593@endigotech.com>
Message-ID: <CAAtWKeFdgZ_MyPBiNUsDYZZsBRJMCe=d6vTSynxDtZ+ifi+KTQ@mail.gmail.com>

We have a script to find the hangup cause and send Email if the user
requested it. This way I know who is looking for me while I am on the phone
:-)

                      __Yehavi:


2012/12/1 Brian Foster <bdfoster at endigotech.com>

> Message Waiting Indicator (MWI) is the voicemail notification feature for
> IP phones.
>
> Missed call notifications are provided by the phone and not the switch.
>
> Sent from my iPhone
>
> On Dec 1, 2012, at 2:41 AM, ?? <itispip-qq at hotmail.com> wrote:
>
>  I heard that Freeswitch support missed call notification (i/o VM
> notification). But look around Wiki no document about this feature. Anyone
> knows where to setup it up?
>
> /brgds, 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 ptness at gmx.co.uk  Sat Dec  1 20:57:15 2012
From: ptness at gmx.co.uk (peter ness)
Date: Sat, 01 Dec 2012 18:57:15 +0100
Subject: [Freeswitch-users] Why does bind_digit_action invoke playback
Message-ID: <20121201175715.248640@gmx.com>

I have bind_digit_action set like this. 

<action application="digit_action_set_realm" data="live"/> 
<action application="bind_digit_action" data="live,##,exec:transfer,FOLLOWON XML cxt"/>

Btw is this correct to catch '##' to transfer it because its not been consistant in catching the digits however I would like to turn this playback off before testing more thorough

Looking at the log there is the entry "Command Execute playback(local_stream://slience)" straight after catching the '##' key press.
Where would the setting be edited to turn this off

Here is the log

2012-12-01 12:27:30.427910 [DEBUG] switch_rtp.c:3424 RTP RECV DTMF #:1600
2012-12-01 12:27:30.427910 [DEBUG] mod_dptools.c:185 sofia/external/nobody at 192.168.56.1 Digit match binding [exec:transfer][FOLLOWON XML cxt]
2012-12-01 12:27:30.427910 [DEBUG] switch_core_session.c:1014 Send signal sofia/external/nobody at 192.168.56.1 [BREAK]
2012-12-01 12:27:30.447880 [DEBUG] switch_core_session.c:731 Send signal sofia/external/nobody at 192.168.56.1 [BREAK]
2012-12-01 12:27:30.607919 [DEBUG] switch_core_session.c:1014 Send signal sofia/external/1800123456 at ip [BREAK]
2012-12-01 12:27:30.627912 [DEBUG] switch_core_session.c:731 Send signal sofia/external/1800123456 at ip[BREAK]
2012-12-01 12:27:30.747904 [DEBUG] switch_ivr.c:591 sofia/external/1800123456 at ip Command Execute playback(local_stream://slience)
EXECUTE sofia/external/1800123456 at ip playback(local_stream://slience)
2012-12-01 12:27:30.747904 [WARNING] mod_local_stream.c:393 Unknown source slience, trying 'default'
2012-12-01 12:27:30.747904 [ERR] mod_local_stream.c:402 Unknown source default
2012-12-01 12:27:30.747904 [DEBUG] switch_core_session.c:731 Send signal sofia/external/1800123456 at ip [BREAK]
2012-12-01 12:27:30.747904 [DEBUG] switch_ivr_bridge.c:329 Send signal sofia/external/nobody at 192.168.56.1 [BREAK]



From itispip-qq at hotmail.com  Sat Dec  1 21:34:44 2012
From: itispip-qq at hotmail.com (=?gb2312?B?zfXA7Q==?=)
Date: Sun, 2 Dec 2012 02:34:44 +0800
Subject: [Freeswitch-users] Missed call notification
In-Reply-To: <CAAtWKeFdgZ_MyPBiNUsDYZZsBRJMCe=d6vTSynxDtZ+ifi+KTQ@mail.gmail.com>
References: <BLU169-W120766FAB52400013BA1BB28C420@phx.gbl>,
	<8AE80082-5AAC-4C25-87E9-52FBD5EDB593@endigotech.com>,
	<CAAtWKeFdgZ_MyPBiNUsDYZZsBRJMCe=d6vTSynxDtZ+ifi+KTQ@mail.gmail.com>
Message-ID: <BLU169-W118507FC5529D5F80F26BA18C420@phx.gbl>



Hi Yehavi, seems your solution is the best fit for the situation. Is that possible to share your script? Maybe can posted it to Freeswitch Wiki as part of the documentation.
Date: Sat, 1 Dec 2012 19:13:11 +0200
From: yehavi.bourvine at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Missed call notification

We have a script to find the hangup cause and send Email if the user requested it. This way I know who is looking for me while I am on the phone :-)                       __Yehavi: 


2012/12/1 Brian Foster <bdfoster at endigotech.com>
Message Waiting Indicator (MWI) is the voicemail notification feature for IP phones. 

Missed call notifications are provided by the phone and not the switch.

Sent from my iPhone
On Dec 1, 2012, at 2:41 AM, ?? <itispip-qq at hotmail.com> wrote:






I heard that Freeswitch support missed call notification (i/o VM notification). But look around Wiki no document about this feature. Anyone knows where to setup it up?

/brgds, 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


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

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


_________________________________________________________________________

Professional FreeSWITCH Consulting 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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From jkomar at jbox.ca  Sat Dec  1 21:38:21 2012
From: jkomar at jbox.ca (Komar, Jason)
Date: Sat, 1 Dec 2012 11:38:21 -0700
Subject: [Freeswitch-users] Missed call notification
In-Reply-To: <BLU169-W118507FC5529D5F80F26BA18C420@phx.gbl>
References: <BLU169-W120766FAB52400013BA1BB28C420@phx.gbl>
	<8AE80082-5AAC-4C25-87E9-52FBD5EDB593@endigotech.com>
	<CAAtWKeFdgZ_MyPBiNUsDYZZsBRJMCe=d6vTSynxDtZ+ifi+KTQ@mail.gmail.com>
	<BLU169-W118507FC5529D5F80F26BA18C420@phx.gbl>
Message-ID: <CA+zebiYsNB0vhnDuPZo=d_uB_9tqE5W3fAzEnrdnsvVtVB4OPg@mail.gmail.com>

I would be interested to see that script too.

Thanks,
Jason


On Sat, Dec 1, 2012 at 11:34 AM, ?? <itispip-qq at hotmail.com> wrote:

>
> Hi Yehavi, seems your solution is the best fit for the situation. Is that
> possible to share your script? Maybe can posted it to Freeswitch Wiki as
> part of the documentation.
>
> ------------------------------
> Date: Sat, 1 Dec 2012 19:13:11 +0200
> From: yehavi.bourvine at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Missed call notification
>
> We have a script to find the hangup cause and send Email if the user
> requested it. This way I know who is looking for me while I am on the phone
> :-)
>
>                       __Yehavi:
>
>
> 2012/12/1 Brian Foster <bdfoster at endigotech.com>
>
> Message Waiting Indicator (MWI) is the voicemail notification feature for
> IP phones.
>
> Missed call notifications are provided by the phone and not the switch.
>
> Sent from my iPhone
>
> On Dec 1, 2012, at 2:41 AM, ?? <itispip-qq at hotmail.com> wrote:
>
>  I heard that Freeswitch support missed call notification (i/o VM
> notification). But look around Wiki no document about this feature. Anyone
> knows where to setup it up?
>
> /brgds, 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 FreeSWITCH-powered IP PBX: The CudaTel
> Communication Server  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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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>
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From steveayre at gmail.com  Sun Dec  2 00:01:57 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 1 Dec 2012 21:01:57 +0000
Subject: [Freeswitch-users] Stange problems with 1.2.5 branch
In-Reply-To: <50BA2D73.6070909@tagnet.ru>
References: <50BA2D73.6070909@tagnet.ru>
Message-ID: <CAFiqYuk+rb435CTY75uk7fmbcPBOUo4=+3BJ9g3PqfVL2N3wMA@mail.gmail.com>

>
> 1) mod_fifo stop working. There are errors in log that there is no table
> fifo_callers. I looked at mod_fifo.c and created all necessary tables
> but this doesn't helps.


Are you using ODBC? What are you using as the DSN?

"name:user:password" should still work fine, but if you're just using the
"name" form then there's a change in 1.2.4 that means that'll no longer
connect. It might mean you get no such table errors. Prefix DSNs with
odbc:// to solve it, if that's the problem.
http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn

It would be useful if you can share a debug-level log of freeswitch
starting up, with relevant configs
http://pastebin.freeswitch.org/





On 1 December 2012 16:16, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>      After upgrade to 1.2.5 (and so on) I found 2 strange problems:
> 1) mod_fifo stop working. There are errors in log that there is no table
> fifo_callers. I looked at mod_fifo.c and created all necessary tables
> but this doesn't helps.
> 2) The problem looks like "unconditional VAD". I use G711A in my
> network, vad is disabled. But when I make I call - the first "hello" is
> "eaten".
>
> Anyone may confirm same problems?
>
> --
> Regards,
> Boris
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 boris at tagnet.ru  Sun Dec  2 04:58:51 2012
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sun, 02 Dec 2012 07:58:51 +0600
Subject: [Freeswitch-users] Stange problems with 1.2.5 branch
In-Reply-To: <CAFiqYuk+rb435CTY75uk7fmbcPBOUo4=+3BJ9g3PqfVL2N3wMA@mail.gmail.com>
References: <50BA2D73.6070909@tagnet.ru>
	<CAFiqYuk+rb435CTY75uk7fmbcPBOUo4=+3BJ9g3PqfVL2N3wMA@mail.gmail.com>
Message-ID: <50BAB5DB.1010403@tagnet.ru>

Hello!

http://pastebin.freeswitch.org/20281

I use ODBC, RedHat
mysql-connector-odbc.x86_64 3.51.26r1127-2.el5           installed
unixODBC.x86_64 2.2.11-10.el5                installed
unixODBC-devel.x86_64 2.2.11-10.el5                installed
unixODBC-libs.x86_64 2.2.11-10.el5                installed

Relevant configs:
[tagnet_odbc]
Driver   = MySQL
SERVER   = X.X.X.X
PORT     = 3306
DATABASE = fspbx
OPTION  = 67108864
USER    = fspbx
PASSWORD        = ------

And from switch.conf.xml:
<param name="core-db-dsn" value="odbc://tagnet_odbc:fspbx:FsPbX" />
<param name="auto-create-schemas" value="true"/>

>     1) mod_fifo stop working. There are errors in log that there is no
>     table
>     fifo_callers. I looked at mod_fifo.c and created all necessary tables
>     but this doesn't helps.
>
>
> Are you using ODBC? What are you using as the DSN?
>
> "name:user:password" should still work fine, but if you're just using 
> the "name" form then there's a change in 1.2.4 that means that'll no 
> longer connect. It might mean you get no such table errors. Prefix 
> DSNs with odbc:// to solve it, if that's the problem.
> http://wiki.freeswitch.org/wiki/Release_Notes#odbc-dsn
>
> It would be useful if you can share a debug-level log of freeswitch 
> starting up, with relevant configs
> http://pastebin.freeswitch.org/
>
>
>
>
>
> On 1 December 2012 16:16, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>          After upgrade to 1.2.5 (and so on) I found 2 strange problems:
>     1) mod_fifo stop working. There are errors in log that there is no
>     table
>     fifo_callers. I looked at mod_fifo.c and created all necessary tables
>     but this doesn't helps.
>     2) The problem looks like "unconditional VAD". I use G711A in my
>     network, vad is disabled. But when I make I call - the first
>     "hello" is
>     "eaten".
>
>     Anyone may confirm same problems?
>
>     --
>     Regards,
>     Boris
>
>
>     _________________________________________________________________________
>     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
> 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


-- 
Regards,
Boris
  

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From yehavi.bourvine at gmail.com  Sun Dec  2 11:21:17 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 2 Dec 2012 10:21:17 +0200
Subject: [Freeswitch-users] Missed call notification
In-Reply-To: <CA+zebiYsNB0vhnDuPZo=d_uB_9tqE5W3fAzEnrdnsvVtVB4OPg@mail.gmail.com>
References: <BLU169-W120766FAB52400013BA1BB28C420@phx.gbl>
	<8AE80082-5AAC-4C25-87E9-52FBD5EDB593@endigotech.com>
	<CAAtWKeFdgZ_MyPBiNUsDYZZsBRJMCe=d6vTSynxDtZ+ifi+KTQ@mail.gmail.com>
	<BLU169-W118507FC5529D5F80F26BA18C420@phx.gbl>
	<CA+zebiYsNB0vhnDuPZo=d_uB_9tqE5W3fAzEnrdnsvVtVB4OPg@mail.gmail.com>
Message-ID: <CAAtWKeE9nFAFg1YWpb+B1BwLrTOKpOdMhJ9KEYE0E6KuGG3gNA@mail.gmail.com>

Hi,

  I've created the following wiki page with example of how we do it:
http://wiki.freeswitch.org/wiki/MailNoAsnwer

I would be greatfull if someone can add it to the examples page, as I don't
have the edit authority on it...

                    Thanks, __Yehavi:


2012/12/1 Komar, Jason <jkomar at jbox.ca>

> I would be interested to see that script too.
>
> Thanks,
> Jason
>
>
> On Sat, Dec 1, 2012 at 11:34 AM, ?? <itispip-qq at hotmail.com> wrote:
>
>>
>> Hi Yehavi, seems your solution is the best fit for the situation. Is that
>> possible to share your script? Maybe can posted it to Freeswitch Wiki as
>> part of the documentation.
>>
>> ------------------------------
>> Date: Sat, 1 Dec 2012 19:13:11 +0200
>> From: yehavi.bourvine at gmail.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Missed call notification
>>
>> We have a script to find the hangup cause and send Email if the user
>> requested it. This way I know who is looking for me while I am on the phone
>> :-)
>>
>>                       __Yehavi:
>>
>>
>> 2012/12/1 Brian Foster <bdfoster at endigotech.com>
>>
>> Message Waiting Indicator (MWI) is the voicemail notification feature for
>> IP phones.
>>
>> Missed call notifications are provided by the phone and not the switch.
>>
>> Sent from my iPhone
>>
>> On Dec 1, 2012, at 2:41 AM, ?? <itispip-qq at hotmail.com> wrote:
>>
>>  I heard that Freeswitch support missed call notification (i/o VM
>> notification). But look around Wiki no document about this feature. Anyone
>> knows where to setup it up?
>>
>> /brgds, 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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 FreeSWITCH-powered IP PBX: The
>> CudaTel Communication Server  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-usersUNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Dec  2 18:46:44 2012
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Sun, 2 Dec 2012 16:46:44 +0100
Subject: [Freeswitch-users] Is FreeSWITCH stable?
Message-ID: <CAO4LeU25SNMdXfHBFQVzNBOWrfp7+e7XSedFZycevzhq9tFgwg@mail.gmail.com>

... yes!

freeswitch at internal> status
UP 0 years, 9 days, 6 hours, 50 minutes, 17 seconds, 54 milliseconds, 237
microseconds
FreeSWITCH is ready
60649825 session(s) since startup
5481 session(s) 6/200
8000 session(s) max
min idle cpu 0.00/42.00
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From abaci64 at gmail.com  Sun Dec  2 22:48:20 2012
From: abaci64 at gmail.com (Abaci)
Date: Sun, 02 Dec 2012 14:48:20 -0500
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <CAKzWOxXhmRELfDsDv3tn==H8Ji_wFzjJJEB7CuuOU3ASKADQYg@mail.gmail.com>
References: <CAKzWOxWxm3aq9LAGWkwK_xdH-n4=qPECvWOLNMJtNiPovgDFMA@mail.gmail.com>
	<1354241384.50b81568a4579@mx.newgen.co.in>
	<CAKzWOxXhmRELfDsDv3tn==H8Ji_wFzjJJEB7CuuOU3ASKADQYg@mail.gmail.com>
Message-ID: <50BBB084.9040404@gmail.com>

you can use the valet_announce_slot channel variable to disable 
annoucement of the parking slot. see 
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park#Channel_Variables

On 11/29/2012 9:59 PM, Michael Collins wrote:
> As far as I know you can't suppress the announcement of the location 
> to the caller.
>
> For the event socket you have a lot of homework to do. I recommend:
> FS Book 
> <https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book>, 
> chapter 9
> FS Cookbook <http://link.packtpub.com/nuIOlX>, chapter 4
> Wiki event socket (see link on left)
>
> For a really quick dive into what events look like:
> launch fs_cli and type:
> /log 0
> /events plain all
>
> You'll see EVERY event that the system throws. Try this to narrow it 
> down just to valet events:
>
> /filter Event-Class valet_parking::info
>
> I typed most of this off the top of my head, so standard disclaimer 
> <http://wiki.freeswitch.org/wiki/Reporting_Bugs#Standard_Disclaimer> 
> applies. Hope this helps you get started!
> -MC
>
>
> On Thu, Nov 29, 2012 at 6:09 PM, Nitin Tomer <ntomer at newgen.co.in 
> <mailto:ntomer at newgen.co.in>> wrote:
>
>     Dear Michael,
>
>     Thanks for your help.
>
>     About thsis - "Using 'auto in' the system will announce the
>     parking location. If you are sending a call in from an IVR then
>     the caller will hear their park location. The only way to know
>     where the call went would be to watch the event socket for
>     relevant valet events. "
>
>     Yes, right now the extension where call is parked, is announced to
>     the caller. I don't want that to happen. I don't want it announced
>     to caller, rather I want it retruned to me, so that I can store it
>     in database. So that my agents can pick the call after seeing the
>     extension where it is parked.
>
>     Please tell me more details about how to watch the event socket
>     for valet events.
>
>     Regards
>
>     Nitin
>
>     On Friday, 30-11-2012 on 6:38 Michael Collins wrote:
>
>         Answers inline...
>
>         On Thu, Nov 29, 2012 at 2:15 AM, Nitin Tomer
>         <ntomer at newgen.co.in <mailto:ntomer at newgen.co.in>> wrote:
>
>             Hi,
>
>             I am using valet_park. I've configure a IVR menu of an
>             extension, based on user's input call is forwarded to
>             other extensions.
>
>             Extension on which end-users will call --
>
>             <extension name="ivr_nitin">
>                   <condition field="destination_number"
>             expression="^5002$">
>                     <action application="answer"/>
>                     <action application="sleep" data="500"/>
>                     <action application="ivr" data="nitin_ivr"/>
>                   </condition>
>             </extension>
>
>             The IVR configuration XML is --
>
>             <include>
>                     <menu name="nitin_ivr"
>             greet-long="say:Welcome to Newgen General Insurance
>             Company. Press 1 for Changing Address, 2 for Changing
>             Nominee or 3 for Close Policy."
>             greet-short="say:Welcome to Newgen. Press 1 for Changing
>             Address, 2 for Changing Nominee or 3 for Close Policy."
>             invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>             exit-sound="voicemail/vm-goodbye.wav"
>             confirm-macro=""
>             confirm-key=""
>             tts-engine="flite"
>             tts-voice="slt"
>             confirm-attempts="3"
>             timeout="3000"
>             inter-digit-timeout="2000"
>             max-failures="3"
>             max-timeouts="3"
>             digit-len="4">
>
>             <entry action="menu-exec-app" digits="1" param="transfer
>             450 XML default"/> <!-- Change Address -->
>             <entry action="menu-exec-app" digits="2" param="transfer
>             451 XML default"/> <!-- Change Nominee -->
>             <entry action="menu-exec-app" digits="3" param="transfer
>             452 XML default"/> <!-- Close Policy -->
>
>                     </menu>
>             </include>
>
>             Once user presses "1", call is forwarded to 450, for this
>             extension dialplan entry is --
>
>             <extension>
>                       <condition field="destination_number"
>             expression="^(450)$">
>                       <action application="play_and_get_digits"
>             data="10 16 3 3000 # say:'Press your account number,
>             followed by hash key' say:'Wrong Input' res \d+" />
>                       <action application="phrase" data="spell,${res}"/>
>                       <action application="db"
>             data="insert/testapp/newcall1/${res}" />
>             <action application="set" data="caller_id_name=Change
>             Address Request" />
>             <action application="set" data="call_timeout=60" />
>             <action application="set" data="originate_timeout=60" />
>             <action application="valet_park" data="my_lot auto in 8501
>             8599" />
>
>             </condition>
>             </extension>
>
>             Here, the call is parked at any available extension
>             between 8501 to 8599.
>
>             Then I've set up an extension to pick up calls --
>
>             <extension name="park-out">
>                <condition field="destination_number"
>             expression="^(85\d\d)$">
>                  <action application="answer"/>
>                  <action application="valet_park" data="my_lot $1"/>
>                </condition>
>              </extension>
>
>             I have a few questions --
>
>             1.Valet_park parks the call on any available extension
>             between 8501 to 8599 (<action application="valet_park"
>             data="my_lot auto in 8501 8599" />). Is there any way to
>             let me know on which extension the call have been parked?
>
>         Using 'auto in' the system will announce the parking location.
>         If you are sending a call in from an IVR then the caller will
>         hear their park location. The only way to know where the call
>         went would be to watch the event socket for relevant valet
>         events.
>
>             2.How can I get the number from which call was made in
>             extension 450. The idea is to use the caller number as key
>             and entered value as value while making entry in database
>             (<action application="db"
>             data="insert/testapp/newcall1/${res}" />)?
>
>         Do you mean the caller id number? That's literally in channel
>         variable ${caller_id_number}
>
>             3.If two users call on extension 5002 (where IVR menu is
>             played), what will happen? Will the second user have to
>             wait for first to finish or whether both will be connected
>             parallel?
>
>         Both can be in the IVR at the same time and they won't affect
>         each other at all.
>
>         -MC
>
>             Please help me out.
>
>             Regards
>
>             Nitin
>
>
>                         Disclaimer :- This e-mail and any attachment
>                         may contain confidential, proprietary or
>                         legally privileged information. If you are not
>                         the original intended recipient and have
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>                         disclosing the content of this message. Please
>                         delete it immediately and notify the sender.
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>
>             _________________________________________________________________________
>             Professional FreeSWITCH Consulting Services:
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>             http://www.freeswitchsolutions.com
>
>             
>             
>
>             Official FreeSWITCH Sites
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>
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>
>
>
>         -- 
>         Michael S Collins
>         Twitter: @mercutioviz
>         http://www.FreeSWITCH.org
>         http://www.ClueCon.com
>         http://www.OSTAG.org
>
>
>
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>
>
>
>     _________________________________________________________________________
>     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
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>
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>     http://www.freeswitch.org
>
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>
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> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
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From brian at freeswitch.org  Mon Dec  3 00:50:29 2012
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 Dec 2012 15:50:29 -0600
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <50BBB084.9040404@gmail.com>
References: <CAKzWOxWxm3aq9LAGWkwK_xdH-n4=qPECvWOLNMJtNiPovgDFMA@mail.gmail.com>
	<1354241384.50b81568a4579@mx.newgen.co.in>
	<CAKzWOxXhmRELfDsDv3tn==H8Ji_wFzjJJEB7CuuOU3ASKADQYg@mail.gmail.com>
	<50BBB084.9040404@gmail.com>
Message-ID: <-6777481922424422356@unknownmsgid>

If the caller is hearing the location your doing the transfer wrong!

Sent from my iPhone

On Dec 2, 2012, at 1:52 PM, Abaci <abaci64 at gmail.com> wrote:

 you can use the valet_announce_slot channel variable to disable
annoucement of the parking slot. see
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park#Channel_Variables

On 11/29/2012 9:59 PM, Michael Collins wrote:

As far as I know you can't suppress the announcement of the location to the
caller.

For the event socket you have a lot of homework to do. I recommend:
FS Book<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book>,
chapter 9
FS Cookbook <http://link.packtpub.com/nuIOlX>, chapter 4
Wiki event socket (see link on left)

For a really quick dive into what events look like:
launch fs_cli and type:
/log 0
/events plain all

You'll see EVERY event that the system throws. Try this to narrow it down
just to valet events:

/filter Event-Class valet_parking::info

I typed most of this off the top of my head, so standard
disclaimer<http://wiki.freeswitch.org/wiki/Reporting_Bugs#Standard_Disclaimer>applies.
Hope this helps you get started!
-MC


On Thu, Nov 29, 2012 at 6:09 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:

>  Dear Michael,
>
> Thanks for your help.
>
> About thsis - "Using 'auto in' the system will announce the parking
> location. If you are sending a call in from an IVR then the caller will
> hear their park location. The only way to know where the call went would be
> to watch the event socket for relevant valet events. "
>
> Yes, right now the extension where call is parked, is announced to the
> caller. I don't want that to happen. I don't want it announced to caller,
> rather I want it retruned to me, so that I can store it in database. So
> that my agents can pick the call after seeing the extension where it is
> parked.
>
> Please tell me more details about how to watch the event socket for valet
> events.
>
> Regards
>
> Nitin
>
> On Friday, 30-11-2012 on 6:38 Michael Collins wrote:
>
> Answers inline...
>
> On Thu, Nov 29, 2012 at 2:15 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:
>
>>   Hi,
>>
>>
>>
>> I am using valet_park. I?ve configure a IVR menu of an extension, based
>> on user?s input call is forwarded to other extensions.
>>
>>
>>
>> Extension on which end-users will call ?
>>
>>
>>
>> <extension name="ivr_nitin">
>>       <condition field="destination_number" expression="^5002$">
>>         <action application="answer"/>
>>         <action application="sleep" data="500"/>
>>         <action application="ivr" data="nitin_ivr"/>
>>       </condition>
>> </extension>
>>
>>
>>
>> The IVR configuration XML is ?
>>
>>
>>
>> <include>
>>         <menu name="nitin_ivr"
>>                 greet-long="say:Welcome to Newgen General Insurance
>> Company. Press 1 for Changing Address, 2 for Changing Nominee or 3 for
>> Close Policy."
>>                 greet-short="say:Welcome to Newgen. Press 1 for Changing
>> Address, 2 for Changing Nominee or 3 for Close Policy."
>>                 invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>                 exit-sound="voicemail/vm-goodbye.wav"
>>                 confirm-macro=""
>>                 confirm-key=""
>>                 tts-engine="flite"
>>                 tts-voice="slt"
>>                 confirm-attempts="3"
>>                 timeout="3000"
>>                 inter-digit-timeout="2000"
>>                 max-failures="3"
>>                 max-timeouts="3"
>>                 digit-len="4">
>>
>>                 <entry action="menu-exec-app" digits="1" param="transfer
>> 450 XML default"/>    <!-- Change Address -->
>>                 <entry action="menu-exec-app" digits="2" param="transfer
>> 451 XML default"/>    <!-- Change Nominee -->
>>                 <entry action="menu-exec-app" digits="3" param="transfer
>> 452 XML default"/>    <!-- Close Policy -->
>>
>>         </menu>
>> </include>
>>
>>  Once user presses ?1?, call is forwarded to 450, for this extension
>> dialplan entry is ?
>>
>>
>>
>> <extension>
>>           <condition field="destination_number" expression="^(450)$">
>>           <action application="play_and_get_digits" data="10 16 3 3000 #
>> say:'Press your account number, followed by hash key' say:'Wrong Input' res
>> \d+" />
>>           <action application="phrase" data="spell,${res}"/>
>>           <action application="db" data="insert/testapp/newcall1/${res}"
>> />
>>               <action application="set" data="caller_id_name=Change
>> Address Request" />
>>               <action application="set" data="call_timeout=60" />
>>               <action application="set" data="originate_timeout=60" />
>>               <action application="valet_park" data="my_lot auto in 8501
>> 8599" />
>>
>>           </condition>
>> </extension>
>>
>>
>>
>> Here, the call is parked at any available extension between 8501 to 8599.
>>
>>
>>
>> Then I?ve set up an extension to pick up calls ?
>>
>>
>>
>> <extension name="park-out">
>>    <condition field="destination_number" expression="^(85\d\d)$">
>>      <action application="answer"/>
>>      <action application="valet_park" data="my_lot $1"/>
>>    </condition>
>>  </extension>
>>
>>
>>
>> I have a few questions ?
>>
>>
>>
>> 1.       Valet_park parks the call on any available extension between
>> 8501 to 8599 (<action application="valet_park" data="my_lot auto in 8501
>> 8599" />). Is there any way to let me know on which extension the call have
>> been parked?
>>
> Using 'auto in' the system will announce the parking location. If you are
> sending a call in from an IVR then the caller will hear their park
> location. The only way to know where the call went would be to watch the
> event socket for relevant valet events.
>
>>   2.       How can I get the number from which call was made in
>> extension 450. The idea is to use the caller number as key and entered
>> value as value while making entry in database (<action application="db"
>> data="insert/testapp/newcall1/${res}" />)?
>>
> Do you mean the caller id number? That's literally in channel variable
> ${caller_id_number}
>
>>   3.       If two users call on extension 5002 (where IVR menu is
>> played), what will happen? Will the second user have to wait for first to
>> finish or whether both will be connected parallel?
>>
> Both can be in the IVR at the same time and they won't affect each other
> at all.
>
> -MC
>
>>
>>
>> Please help me out.
>>
>>
>>
>> Regards
>>
>>
>>
>> Nitin
>>
>>  Disclaimer :- This e-mail and any attachment may contain confidential,
>> proprietary or legally privileged information. If you are not the original
>> intended recipient and have erroneously received this message, you are
>> prohibited from using, copying, altering or disclosing the content of this
>> message. Please delete it immediately and notify the sender. Newgen
>> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
>> damage arising from the use of the information transmitted by this email
>> including damages from virus and further acknowledges that no binding
>> nature of the message shall be implied or assumed unless the sender does so
>> expressly with due authority of NSTL.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>  Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
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>
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>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org




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From anthony.minessale at gmail.com  Mon Dec  3 01:17:44 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 2 Dec 2012 16:17:44 -0600
Subject: [Freeswitch-users] Why does bind_digit_action invoke playback
In-Reply-To: <20121201175715.248640@gmx.com>
References: <20121201175715.248640@gmx.com>
Message-ID: <CAKbxfG_38-sYeKrx_U4a7FHL6g=B3RFcsW2E24s0+PNVLY4KjQ@mail.gmail.com>

It's playing hold music for the other side of the call while its dealing
with your digit action...

Look at full bind digit syntax I think you can disable.....
On Dec 1, 2012 12:23 PM, "peter ness" <ptness at gmx.co.uk> wrote:

> I have bind_digit_action set like this.
>
> <action application="digit_action_set_realm" data="live"/>
> <action application="bind_digit_action"
> data="live,##,exec:transfer,FOLLOWON XML cxt"/>
>
> Btw is this correct to catch '##' to transfer it because its not been
> consistant in catching the digits however I would like to turn this
> playback off before testing more thorough
>
> Looking at the log there is the entry "Command Execute
> playback(local_stream://slience)" straight after catching the '##' key
> press.
> Where would the setting be edited to turn this off
>
> Here is the log
>
> 2012-12-01 12:27:30.427910 [DEBUG] switch_rtp.c:3424 RTP RECV DTMF #:1600
> 2012-12-01 12:27:30.427910 [DEBUG] mod_dptools.c:185 sofia/external/
> nobody at 192.168.56.1 Digit match binding [exec:transfer][FOLLOWON XML cxt]
> 2012-12-01 12:27:30.427910 [DEBUG] switch_core_session.c:1014 Send signal
> sofia/external/nobody at 192.168.56.1 [BREAK]
> 2012-12-01 12:27:30.447880 [DEBUG] switch_core_session.c:731 Send signal
> sofia/external/nobody at 192.168.56.1 [BREAK]
> 2012-12-01 12:27:30.607919 [DEBUG] switch_core_session.c:1014 Send signal
> sofia/external/1800123456 at ip [BREAK]
> 2012-12-01 12:27:30.627912 [DEBUG] switch_core_session.c:731 Send signal
> sofia/external/1800123456 at ip[BREAK]
> 2012-12-01 12:27:30.747904 [DEBUG] switch_ivr.c:591
> sofia/external/1800123456 at ip Command Execute
> playback(local_stream://slience)
> EXECUTE sofia/external/1800123456 at ip playback(local_stream://slience)
> 2012-12-01 12:27:30.747904 [WARNING] mod_local_stream.c:393 Unknown source
> slience, trying 'default'
> 2012-12-01 12:27:30.747904 [ERR] mod_local_stream.c:402 Unknown source
> default
> 2012-12-01 12:27:30.747904 [DEBUG] switch_core_session.c:731 Send signal
> sofia/external/1800123456 at ip [BREAK]
> 2012-12-01 12:27:30.747904 [DEBUG] switch_ivr_bridge.c:329 Send signal
> sofia/external/nobody at 192.168.56.1 [BREAK]
>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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From ntomer at newgen.co.in  Mon Dec  3 05:23:51 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Mon, 03 Dec 2012 07:53:51 +0530
Subject: [Freeswitch-users] valet_park help needed
Message-ID: <cdo25qex5xplrot14oad7fr7.1354501431119@email.android.com>

Hi Brian,

I've posted the contents of dialplan. Please tell me what I am doing wrong.

Regards

Nitin

Brian West <brian at freeswitch.org> wrote:

>If the caller is hearing the location your doing the transfer wrong!
>
>Sent from my iPhone
>
>On Dec 2, 2012, at 1:52 PM, Abaci <abaci64 at gmail.com> wrote:
>
> you can use the valet_announce_slot channel variable to disable
>annoucement of the parking slot. see
>http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park#Channel_Variables
>
>On 11/29/2012 9:59 PM, Michael Collins wrote:
>
>As far as I know you can't suppress the announcement of the location to the
>caller.
>
>For the event socket you have a lot of homework to do. I recommend:
>FS Book<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book>,
>chapter 9
>FS Cookbook <http://link.packtpub.com/nuIOlX>, chapter 4
>Wiki event socket (see link on left)
>
>For a really quick dive into what events look like:
>launch fs_cli and type:
>/log 0
>/events plain all
>
>You'll see EVERY event that the system throws. Try this to narrow it down
>just to valet events:
>
>/filter Event-Class valet_parking::info
>
>I typed most of this off the top of my head, so standard
>disclaimer<http://wiki.freeswitch.org/wiki/Reporting_Bugs#Standard_Disclaimer>applies.
>Hope this helps you get started!
>-MC
>
>
>On Thu, Nov 29, 2012 at 6:09 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:
>
>>  Dear Michael,
>>
>> Thanks for your help.
>>
>> About thsis - "Using 'auto in' the system will announce the parking
>> location. If you are sending a call in from an IVR then the caller will
>> hear their park location. The only way to know where the call went would be
>> to watch the event socket for relevant valet events. "
>>
>> Yes, right now the extension where call is parked, is announced to the
>> caller. I don't want that to happen. I don't want it announced to caller,
>> rather I want it retruned to me, so that I can store it in database. So
>> that my agents can pick the call after seeing the extension where it is
>> parked.
>>
>> Please tell me more details about how to watch the event socket for valet
>> events.
>>
>> Regards
>>
>> Nitin
>>
>> On Friday, 30-11-2012 on 6:38 Michael Collins wrote:
>>
>> Answers inline...
>>
>> On Thu, Nov 29, 2012 at 2:15 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:
>>
>>>   Hi,
>>>
>>>
>>>
>>> I am using valet_park. I?ve configure a IVR menu of an extension, based
>>> on user?s input call is forwarded to other extensions.
>>>
>>>
>>>
>>> Extension on which end-users will call ?
>>>
>>>
>>>
>>> <extension name="ivr_nitin">
>>>       <condition field="destination_number" expression="^5002$">
>>>         <action application="answer"/>
>>>         <action application="sleep" data="500"/>
>>>         <action application="ivr" data="nitin_ivr"/>
>>>       </condition>
>>> </extension>
>>>
>>>
>>>
>>> The IVR configuration XML is ?
>>>
>>>
>>>
>>> <include>
>>>         <menu name="nitin_ivr"
>>>                 greet-long="say:Welcome to Newgen General Insurance
>>> Company. Press 1 for Changing Address, 2 for Changing Nominee or 3 for
>>> Close Policy."
>>>                 greet-short="say:Welcome to Newgen. Press 1 for Changing
>>> Address, 2 for Changing Nominee or 3 for Close Policy."
>>>                 invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>>                 exit-sound="voicemail/vm-goodbye.wav"
>>>                 confirm-macro=""
>>>                 confirm-key=""
>>>                 tts-engine="flite"
>>>                 tts-voice="slt"
>>>                 confirm-attempts="3"
>>>                 timeout="3000"
>>>                 inter-digit-timeout="2000"
>>>                 max-failures="3"
>>>                 max-timeouts="3"
>>>                 digit-len="4">
>>>
>>>                 <entry action="menu-exec-app" digits="1" param="transfer
>>> 450 XML default"/>    <!-- Change Address -->
>>>                 <entry action="menu-exec-app" digits="2" param="transfer
>>> 451 XML default"/>    <!-- Change Nominee -->
>>>                 <entry action="menu-exec-app" digits="3" param="transfer
>>> 452 XML default"/>    <!-- Close Policy -->
>>>
>>>         </menu>
>>> </include>
>>>
>>>  Once user presses ?1?, call is forwarded to 450, for this extension
>>> dialplan entry is ?
>>>
>>>
>>>
>>> <extension>
>>>           <condition field="destination_number" expression="^(450)$">
>>>           <action application="play_and_get_digits" data="10 16 3 3000 #
>>> say:'Press your account number, followed by hash key' say:'Wrong Input' res
>>> \d+" />
>>>           <action application="phrase" data="spell,${res}"/>
>>>           <action application="db" data="insert/testapp/newcall1/${res}"
>>> />
>>>               <action application="set" data="caller_id_name=Change
>>> Address Request" />
>>>               <action application="set" data="call_timeout=60" />
>>>               <action application="set" data="originate_timeout=60" />
>>>               <action application="valet_park" data="my_lot auto in 8501
>>> 8599" />
>>>
>>>           </condition>
>>> </extension>
>>>
>>>
>>>
>>> Here, the call is parked at any available extension between 8501 to 8599.
>>>
>>>
>>>
>>> Then I?ve set up an extension to pick up calls ?
>>>
>>>
>>>
>>> <extension name="park-out">
>>>    <condition field="destination_number" expression="^(85\d\d)$">
>>>      <action application="answer"/>
>>>      <action application="valet_park" data="my_lot $1"/>
>>>    </condition>
>>>  </extension>
>>>
>>>
>>>
>>> I have a few questions ?
>>>
>>>
>>>
>>> 1.       Valet_park parks the call on any available extension between
>>> 8501 to 8599 (<action application="valet_park" data="my_lot auto in 8501
>>> 8599" />). Is there any way to let me know on which extension the call have
>>> been parked?
>>>
>> Using 'auto in' the system will announce the parking location. If you are
>> sending a call in from an IVR then the caller will hear their park
>> location. The only way to know where the call went would be to watch the
>> event socket for relevant valet events.
>>
>>>   2.       How can I get the number from which call was made in
>>> extension 450. The idea is to use the caller number as key and entered
>>> value as value while making entry in database (<action application="db"
>>> data="insert/testapp/newcall1/${res}" />)?
>>>
>> Do you mean the caller id number? That's literally in channel variable
>> ${caller_id_number}
>>
>>>   3.       If two users call on extension 5002 (where IVR menu is
>>> played), what will happen? Will the second user have to wait for first to
>>> finish or whether both will be connected parallel?
>>>
>> Both can be in the IVR at the same time and they won't affect each other
>> at all.
>>
>> -MC
>>
>>>
>>>
>>> Please help me out.
>>>
>>>
>>>
>>> Regards
>>>
>>>
>>>
>>> Nitin
>>>
>>>  Disclaimer :- This e-mail and any attachment may contain confidential,
>>> proprietary or legally privileged information. If you are not the original
>>> intended recipient and have erroneously received this message, you are
>>> prohibited from using, copying, altering or disclosing the content of this
>>> message. Please delete it immediately and notify the sender. Newgen
>>> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
>>> damage arising from the use of the information transmitted by this email
>>> including damages from virus and further acknowledges that no binding
>>> nature of the message shall be implied or assumed unless the sender does so
>>> expressly with due authority of NSTL.
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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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>>> FreeSWITCH-users at lists.freeswitch.org
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>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>>
>>  Disclaimer :- This e-mail and any attachment may contain confidential,
>> proprietary or legally privileged information. If you are not the original
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>>
>> 
>> 
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>
>
>-- 
>Michael S Collins
>Twitter: @mercutioviz
>http://www.FreeSWITCH.org
>http://www.ClueCon.com
>http://www.OSTAG.org
>
>
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting
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>
>FreeSWITCH-powered IP PBX: The CudaTel Communication
>Server
>
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>Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
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>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
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>http://wiki.freeswitch.org
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>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>http://www.freeswitch.org
>
>_________________________________________________________________________
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>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
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>http://wiki.freeswitch.org
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From yudha2008 at gmail.com  Mon Dec  3 10:37:10 2012
From: yudha2008 at gmail.com (baskar)
Date: Sun, 2 Dec 2012 23:37:10 -0800 (PST)
Subject: [Freeswitch-users] Freeswitch stable version 1.2.51
Message-ID: <1354520230656-7585111.post@n2.nabble.com>

Hi,

In installed new version 1.2.51 but in that Readme it is mentioned that
"This configuration, generally known as the "default configuration" for
FreeSWITCH, is *NOT* designed to be put into a production environment
without some important modifications. Please keep in mind that the default
configuration is designed to demonstrate what FreeSWITCH *can* do, not what
it *should* do in your specific scenario". To put into a production
enviroment what are changes to be done. Could you please guide me to
configure into a prodcution enviroment. I am using 1.0.6 stable version. 

Thanks in advance,
N.Baskar




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From krice at freeswitch.org  Mon Dec  3 10:45:46 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 03 Dec 2012 01:45:46 -0600
Subject: [Freeswitch-users] Freeswitch stable version 1.2.51
In-Reply-To: <1354520230656-7585111.post@n2.nabble.com>
Message-ID: <CCE1B4CA.614A6%krice@freeswitch.org>

This is documented on the wiki and that has been the policy for since before
1.0.6... Keep in mind that if you use the example configs, there are users
and passwords that are fairly well known in the FS user directory....


On 12/3/12 1:37 AM, "baskar" <yudha2008 at gmail.com> wrote:

> Hi,
> 
> In installed new version 1.2.51 but in that Readme it is mentioned that
> "This configuration, generally known as the "default configuration" for
> FreeSWITCH, is *NOT* designed to be put into a production environment
> without some important modifications. Please keep in mind that the default
> configuration is designed to demonstrate what FreeSWITCH *can* do, not what
> it *should* do in your specific scenario". To put into a production
> enviroment what are changes to be done. Could you please guide me to
> configure into a prodcution enviroment. I am using 1.0.6 stable version.
> 
> Thanks in advance,
> N.Baskar
> 
> 
> 
> 
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-stable-version-1-2-51
> -tp7585111.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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From yudha2008 at gmail.com  Mon Dec  3 11:23:16 2012
From: yudha2008 at gmail.com (baskar)
Date: Mon, 3 Dec 2012 00:23:16 -0800 (PST)
Subject: [Freeswitch-users] Freeswitch stable version 1.2.51
In-Reply-To: <CCE1B4CA.614A6%krice@freeswitch.org>
References: <1354520230656-7585111.post@n2.nabble.com>
	<CCE1B4CA.614A6%krice@freeswitch.org>
Message-ID: <1354522996442-7585113.post@n2.nabble.com>

Hi ken,

Thanks for the quick response. I have my own configuration for outbound
dialing, I am not using example configs. I need to know if i put freeswitch
1.2.5 in production it will cause any issues or downtime. Could you please
help me to get an idea about the README. 

Thanks in advance,
N.Baskar



--
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From steveayre at gmail.com  Mon Dec  3 11:38:28 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 3 Dec 2012 08:38:28 +0000
Subject: [Freeswitch-users] Freeswitch stable version 1.2.51
In-Reply-To: <1354522996442-7585113.post@n2.nabble.com>
References: <1354520230656-7585111.post@n2.nabble.com>
	<CCE1B4CA.614A6%krice@freeswitch.org>
	<1354522996442-7585113.post@n2.nabble.com>
Message-ID: <CAFiqYum96WCZedi+t11BjPycMk2r+W5g0A-eD-ug7JDp2HDK3A@mail.gmail.com>

If you're running with an existing 1.0.6 then the config you're already
using or that should work just fine.

There have been a few tweaks (you should now prefix ODBC DSNs with odbc://
if you're not specifying the username and password in the odbc-dsn) but the
developers try to keep FS as backwards-compatible as possible.


On 3 December 2012 08:23, baskar <yudha2008 at gmail.com> wrote:

> Hi ken,
>
> Thanks for the quick response. I have my own configuration for outbound
> dialing, I am not using example configs. I need to know if i put freeswitch
> 1.2.5 in production it will cause any issues or downtime. Could you please
> help me to get an idea about the README.
>
> Thanks in advance,
> N.Baskar
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-stable-version-1-2-51-tp7585111p7585113.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Mon Dec  3 11:43:53 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 03 Dec 2012 02:43:53 -0600
Subject: [Freeswitch-users] Freeswitch stable version 1.2.51
In-Reply-To: <1354522996442-7585113.post@n2.nabble.com>
Message-ID: <CCE1C269.614AA%krice@freeswitch.org>

You should test any updates to any software before just throwing it into
production...  But 1.2.X should be fine for production use... 1.2.5.1 has
known issues and you probably want to get the stable branch from git (git
checkout v1.2.stable) if you want the latest patches

K


On 12/3/12 2:23 AM, "baskar" <yudha2008 at gmail.com> wrote:

> Hi ken,
> 
> Thanks for the quick response. I have my own configuration for outbound
> dialing, I am not using example configs. I need to know if i put freeswitch
> 1.2.5 in production it will cause any issues or downtime. Could you please
> help me to get an idea about the README.
> 
> Thanks in advance,
> N.Baskar
> 
> 
> 
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-stable-version-1-2-51
> -tp7585111p7585113.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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From yudha2008 at gmail.com  Mon Dec  3 12:20:23 2012
From: yudha2008 at gmail.com (baskar)
Date: Mon, 3 Dec 2012 01:20:23 -0800 (PST)
Subject: [Freeswitch-users] Freeswitch stable version 1.2.51
In-Reply-To: <CCE1C269.614AA%krice@freeswitch.org>
References: <1354520230656-7585111.post@n2.nabble.com>
	<CCE1B4CA.614A6%krice@freeswitch.org>
	<1354522996442-7585113.post@n2.nabble.com>
	<CCE1C269.614AA%krice@freeswitch.org>
Message-ID: <1354526423398-7585116.post@n2.nabble.com>

Hi Steven and Ken,

Thanks for the Quick response to my following quires.I just worried about
the README. But before putting new version into a production environment
surely i will test all the feature for a week and put them in live.

Once again thanks for the reply. 

Thanks,
N.Baskar



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From NuwanW at unifybusiness.co.uk  Mon Dec  3 12:53:03 2012
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Mon, 3 Dec 2012 09:53:03 +0000
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
Message-ID: <78990CE7CC964442A7C2CA5F4689695EA82D29A2@BARXB0003.UnifyBusiness.local>

Hello All,

I have added my comments to JIRA case FS-4884

Thank you,
Nuwan.
This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.
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From rentmycoder at gmail.com  Mon Dec  3 14:19:39 2012
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Mon, 3 Dec 2012 12:19:39 +0100
Subject: [Freeswitch-users] Qos on Windows
Message-ID: <CAEW218JSrS9B7LK_WRs9irQr_0GHRQ5kfvNt=UAVuL+8=-yisg@mail.gmail.com>

It is possible to use iptables on Linux to add Qos mark to ip packets...
Any hint how to set QOS on windows?
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From chmp99 at gmail.com  Mon Dec  3 16:45:16 2012
From: chmp99 at gmail.com (=?ISO-8859-1?Q?Germ=E1n_Ruiz?=)
Date: Mon, 03 Dec 2012 10:45:16 -0300
Subject: [Freeswitch-users] Questions about RTMP endpoint
In-Reply-To: <1726AEE3708448019DFAD73E4C2F8ACB@gmail.com>
References: <50B631BC.8060001@gmail.com>
	<1726AEE3708448019DFAD73E4C2F8ACB@gmail.com>
Message-ID: <50BCACEC.9040507@gmail.com>

Ok,
but could you indicate which branch?  I don't found in the source.

Thanks
Germ?n


> We have text messages via rtmp in our branch. Just need time to find
> them out.
>
> -- 
> Seven Du
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Wednesday, November 28, 2012 at 11:46 PM, Germ?n Ruiz wrote:
>
>> Hi,
>> I'm adding voice calls to a web site. I'm using RTMP endpoint and Flex
>> clientprovided. I have the following questions:
>> - Can flex clients (using RTMP protocol) register their presence and are
>> visible by others connected using SIP?
>> - Is it possible to send and receive text messages with RTMP protocol?
>>
>> Thanks
>> Germ?n
>>
>> _________________________________________________________________________
>> 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
> 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 ksrigo at gmail.com  Mon Dec  3 17:17:50 2012
From: ksrigo at gmail.com (Srigo)
Date: Mon, 3 Dec 2012 06:17:50 -0800 (PST)
Subject: [Freeswitch-users] Freeswitch is crashing when httapi playback
 application is executed
In-Reply-To: <1354210814465-7585041.post@n2.nabble.com>
References: <1354210814465-7585041.post@n2.nabble.com>
Message-ID: <1354544270822-7585120.post@n2.nabble.com>

Hi,

I have now more details about this crashed.
Here is what i did (I used the FS you compiled as 'minessale"):

 - Start FS as minessale:
 
 su minessale
 /usr/local/stow/freeswitch_20121130/bin/freeswitch -nonat -rp -nc -conf
/etc/freeswitch/ -db /var/lib/freeswitch/db/ -log /var/log/freeswitch
 
 Everthing worked fine, NO CRASH!!
 
 - Then i tried samething by starting FS as "freeswitch"
  su - freeswitch -s /bin/bash
 /usr/local/stow/freeswitch_20121130/bin/freeswitch -nonat -rp -nc -conf
/etc/freeswitch/ -db /var/lib/freeswitch/db/ -log /var/log/freeswitch
 
 This time FS crashed. 
 According to "strace", we have a permission denied when accessing to
/usr/local/stow/freeswitch_20121130/storage/http_file_cache. By changing the
rights on "storage" I was able to run httapi without crash with the user
"freeswitch"
 
 I joined the trace file.
 
 Additionnaly, we got the same issue with the compiled debian package: It's
looking for its cache in /usr/storage which does nt exist.
 
 I would like to suggest to you:
- Checking return code when opening a file in "storage/http_file_cache" and
log failures.
- Fix the build system to create the storage cache with the correct rights
- Add in httapi configuration file parameters to set the storage path


 Regards,
 Srigo



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From miha at softnet.si  Mon Dec  3 17:27:20 2012
From: miha at softnet.si (Miha)
Date: Mon, 03 Dec 2012 15:27:20 +0100
Subject: [Freeswitch-users] Fax problem Reinvite Codec Error!
Message-ID: <50BCB6C8.4070208@softnet.si>

Hi,

i have problem with faxes which are not working as they should. In my 
log file I can see this error:

2012-12-03 14:56:43.477028 [DEBUG] sofia.c:6293 Remote SDP:
v=0
o=sbc2 1325546 1 IN IP4 212.13.249.90
s=sip call
c=IN IP4 212.13.249.91
t=0 0
m=audio 16158 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -

2012-12-03 14:56:43.477028 [DEBUG] sofia_glue.c:5094 Audio Codec Compare 
[PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-12-03 14:56:43.477028 [DEBUG] sofia_glue.c:5094 Audio Codec Compare 
[telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
2012-12-03 14:56:43.477028 [DEBUG] sofia_glue.c:5219 Set 2833 dtmf 
send/recv payload to 101
2012-12-03 14:56:43.477028 [ERR] sofia.c:6828 Reinvite Codec Error!

I am bridging call like this:

  <action application="bridge" 
data="{ignore_early_media=true,fax_enable_t38=true,fax_verbose=true,fax_use_ecm=false,fax_enable_t38_request=true}user/${destination_number}.fs_kabelvoip1 at xxx.xxx.xxx.xxxx" 
/>


I tired with t38 reuqest false, ecm false...



from cli:

EXECUTE sofia/internal/074961627 at xxx.xxx.xxx.xxx 
bridge({ignore_early_media=true,fax_enable_t38=true,fax_verbose=true,fax_use_ecm=true,fax_enable_t38_request=true}user/074978115.fs_kabelvoip1 at xxx.xxx.xxx.xxx)
2012-12-03 14:56:28.397064 [DEBUG] switch_channel.c:1072 
sofia/internal/074961627 at xxx.xxx.xxxx.xxxx EXPORTING[export_vars] 
[RFC2822_DATE]=[Mon, 03 Dec 2012 14:56:28 +0100] to event
2012-12-03 14:56:28.397064 [DEBUG] switch_ivr_originate.c:2005 Parsing 
global variables
2012-12-03 14:56:28.397064 [DEBUG] switch_event.c:1569 Parsing variable 
[ignore_early_media]=[true]
2012-12-03 14:56:28.397064 [DEBUG] switch_event.c:1569 Parsing variable 
[fax_enable_t38]=[true]
2012-12-03 14:56:28.397064 [DEBUG] switch_event.c:1569 Parsing variable 
[fax_verbose]=[true]
2012-12-03 14:56:28.397064 [DEBUG] switch_event.c:1569 Parsing variable 
[fax_use_ecm]=[true]
2012-12-03 14:56:28.397064 [DEBUG] switch_event.c:1569 Parsing variable 
[fax_enable_t38_request]=[true]


What else can I do are there any other ways to debug this:)?

Thanks!

Miha





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From Chad.Engler at patlive.com  Mon Dec  3 17:28:25 2012
From: Chad.Engler at patlive.com (Chad Engler)
Date: Mon, 3 Dec 2012 09:28:25 -0500
Subject: [Freeswitch-users] mod_cdr_mongodb follows json rfc4627?
In-Reply-To: <CAKbxfG_-BU2nOT6iK-U2hJb550_fYhq-hbt1VCvx4_+LCn2J+w@mail.gmail.com>
References: <CAJOPU6XN2Wh0ROUEK45-4W-8XmEsM7qiomXFtU+odi5Vr2Qk2g@mail.gmail.com>
	<CAKbxfG_-BU2nOT6iK-U2hJb550_fYhq-hbt1VCvx4_+LCn2J+w@mail.gmail.com>
Message-ID: <BD87D3B1554F614190EB6E54CE95DDB606E7D59B@mail2.patlive.local>

I submitted a patch that should fix this a while ago, but haven't heard much back:

 

http://jira.freeswitch.org/browse/FS-4830

 

-Chad

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Friday, November 30, 2012 5:29 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_cdr_mongodb follows json rfc4627?

 

you are right, post the patch to JIRA please.

 

 

On Fri, Nov 30, 2012 at 1:28 PM, Jose Fco. Irles Dur? <josefu at gmail.com> wrote:

Hello everybody, first of all, sorry for my english.

I have a test server with the mod_cdr_mongodb backend configured.

I'm trying to parse the cdrs with the mongodb driver for java, but i
have a trouble with a part of the json that freeswitch sends (mongo
saves the json document without errors). I haven't an API function to
extract some data. Firstly I thought that the problem was in the java
driver but I'm not sure.

The problem is in the "callflow" part, with the json standard[1], a
json document can't have two keys with the same name, but
mod_cdr_mongodb builds the json with a 'n' callflow objects.
Also it happends inside callflow object, with origination, originator
and originatee objects.

mod_cdr_mongodb saves the json with this format:
http://pastebin.com/VRz6s0eb

I modified the source of the backend adding json arrays around this
objects and now I can parse without problems.

The output with the modifications:
http://pastebin.com/v05FN9Ta

I'm in the correct way or I'm missing something?

[1] http://www.ietf.org/rfc/rfc4627.txt Section 2.2

Regards
--
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

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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From Chad.Engler at patlive.com  Mon Dec  3 17:39:13 2012
From: Chad.Engler at patlive.com (Chad Engler)
Date: Mon, 3 Dec 2012 09:39:13 -0500
Subject: [Freeswitch-users] mod_cdr_mongodb follows json rfc4627?
In-Reply-To: <BD87D3B1554F614190EB6E54CE95DDB606E7D59B@mail2.patlive.local>
References: <CAJOPU6XN2Wh0ROUEK45-4W-8XmEsM7qiomXFtU+odi5Vr2Qk2g@mail.gmail.com><CAKbxfG_-BU2nOT6iK-U2hJb550_fYhq-hbt1VCvx4_+LCn2J+w@mail.gmail.com>
	<BD87D3B1554F614190EB6E54CE95DDB606E7D59B@mail2.patlive.local>
Message-ID: <BD87D3B1554F614190EB6E54CE95DDB606E7D59F@mail2.patlive.local>

Well atleast it is related, I see that the MongoDB CDR generator uses some bson methods instead of json.

 

-Chad

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chad Engler
Sent: Monday, December 03, 2012 9:28 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_cdr_mongodb follows json rfc4627?

 

I submitted a patch that should fix this a while ago, but haven't heard much back:

 

http://jira.freeswitch.org/browse/FS-4830

 

-Chad

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Friday, November 30, 2012 5:29 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_cdr_mongodb follows json rfc4627?

 

you are right, post the patch to JIRA please.

 

 

On Fri, Nov 30, 2012 at 1:28 PM, Jose Fco. Irles Dur? <josefu at gmail.com> wrote:

Hello everybody, first of all, sorry for my english.

I have a test server with the mod_cdr_mongodb backend configured.

I'm trying to parse the cdrs with the mongodb driver for java, but i
have a trouble with a part of the json that freeswitch sends (mongo
saves the json document without errors). I haven't an API function to
extract some data. Firstly I thought that the problem was in the java
driver but I'm not sure.

The problem is in the "callflow" part, with the json standard[1], a
json document can't have two keys with the same name, but
mod_cdr_mongodb builds the json with a 'n' callflow objects.
Also it happends inside callflow object, with origination, originator
and originatee objects.

mod_cdr_mongodb saves the json with this format:
http://pastebin.com/VRz6s0eb

I modified the source of the backend adding json arrays around this
objects and now I can parse without problems.

The output with the modifications:
http://pastebin.com/v05FN9Ta

I'm in the correct way or I'm missing something?

[1] http://www.ietf.org/rfc/rfc4627.txt Section 2.2

Regards
--
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

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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From josefu at gmail.com  Mon Dec  3 17:41:29 2012
From: josefu at gmail.com (=?ISO-8859-1?Q?Jose_Fco=2E_Irles_Dur=E1?=)
Date: Mon, 3 Dec 2012 15:41:29 +0100
Subject: [Freeswitch-users] mod_cdr_mongodb follows json rfc4627?
In-Reply-To: <BD87D3B1554F614190EB6E54CE95DDB606E7D59B@mail2.patlive.local>
References: <CAJOPU6XN2Wh0ROUEK45-4W-8XmEsM7qiomXFtU+odi5Vr2Qk2g@mail.gmail.com>
	<CAKbxfG_-BU2nOT6iK-U2hJb550_fYhq-hbt1VCvx4_+LCn2J+w@mail.gmail.com>
	<BD87D3B1554F614190EB6E54CE95DDB606E7D59B@mail2.patlive.local>
Message-ID: <CAJOPU6VX7Aeh_9=xU_Rae7=5Tw-12G3-WDekaHXzN+jciBV0_g@mail.gmail.com>

I also sent a patch for the mod_cdr_mongodb module:

http://jira.freeswitch.org/browse/FS-4902

2012/12/3 Chad Engler <Chad.Engler at patlive.com>:
> I submitted a patch that should fix this a while ago, but haven?t heard much
> back:
>
>
>
> http://jira.freeswitch.org/browse/FS-4830
>
>
>
> -Chad
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
> Minessale
> Sent: Friday, November 30, 2012 5:29 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] mod_cdr_mongodb follows json rfc4627?
>
>
>
> you are right, post the patch to JIRA please.
>
>
>
>
>
> On Fri, Nov 30, 2012 at 1:28 PM, Jose Fco. Irles Dur? <josefu at gmail.com>
> wrote:
>
> Hello everybody, first of all, sorry for my english.
>
> I have a test server with the mod_cdr_mongodb backend configured.
>
> I'm trying to parse the cdrs with the mongodb driver for java, but i
> have a trouble with a part of the json that freeswitch sends (mongo
> saves the json document without errors). I haven't an API function to
> extract some data. Firstly I thought that the problem was in the java
> driver but I'm not sure.
>
> The problem is in the "callflow" part, with the json standard[1], a
> json document can't have two keys with the same name, but
> mod_cdr_mongodb builds the json with a 'n' callflow objects.
> Also it happends inside callflow object, with origination, originator
> and originatee objects.
>
> mod_cdr_mongodb saves the json with this format:
> http://pastebin.com/VRz6s0eb
>
> I modified the source of the backend adding json arrays around this
> objects and now I can parse without problems.
>
> The output with the modifications:
> http://pastebin.com/v05FN9Ta
>
> I'm in the correct way or I'm missing something?
>
> [1] http://www.ietf.org/rfc/rfc4627.txt Section 2.2
>
> Regards
> --
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ben122uk at gmail.com  Mon Dec  3 18:31:35 2012
From: ben122uk at gmail.com (Ben)
Date: Mon, 3 Dec 2012 15:31:35 -0000
Subject: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes
In-Reply-To: <CADaqzFOGTva6mZBgewCKc8gdHmGdDcMn-AMPPnehHTNA4n-H_w@mail.gmail.com>
References: <CADaqzFMD+Y8yc_qWNtp0drmceSU6EDogN7F6TyhtSDLbT2JeWA@mail.gmail.com>	<6036422a0e8b5982ed7af8823015bb38@www.webmail.co.za>
	<CADaqzFOGTva6mZBgewCKc8gdHmGdDcMn-AMPPnehHTNA4n-H_w@mail.gmail.com>
Message-ID: <002d01cdd16b$492b9030$db82b090$@gmail.com>

I tried setting the {passthru_ptime_mismatch=true} variable in the dialplan,
but no joy unfortunately.  I'm also fairly sure that the codecs are matched
on both call legs, so shouldn't be having that sort of issue.

 

I'm going to gather up some evidence and console logs/traces etc so that I
can put something together for a bug report on Jira.  Is there anything else
that someone can suggest I try before it goes on Jira?  Is anyone able to
re-produce the same problem?  Admittedly, I'm not on the latest GIT, but I'm
on a recent (last couple of months) stable release. 

 

Thanks in advance for any assistance!

 

Regards,

 

Ben

 

From: Ben N [mailto:ben122uk at gmail.com] 
Sent: 29 November 2012 09:58
To: clive engelberg
Subject: Re: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes

 

Hi Brian, yes I've set G729 at 50i on both the external and internal sofia
profiles statically, and confirmed the allowed codecs by running "sofia
status profile profilename".  

 

I can also confirm in the FS console that codec comparisons for both the
a-leg, and b-leg match instantly to G729 at 50i.

 

Clive, thanks for the pointer, although in my case I don't think there's a
mis-match as each side of the call supports sending/receiving G729 50ms
ptimes, and is negotiated correctly in the SDP.  However I'm still willing
to give it a try as I need to try everything!  I'll let you know what
happens.

 

Cheers,

 

Ben

 

On Thu, Nov 29, 2012 at 8:08 AM, clive engelberg <clive18 at webmail.co.za>
wrote:

Hi

 

I have had limited success with :

 <action application="bridge"
data="{passthru_ptime_mismatch=true}sofia/gateway/icall/$1 at 1.2.3.4"/>

 

Sometimes the sound was garbled, sometimes worked fine, not sure why it was
sometimes garbled, so I dont use it anymore, as complaints are not what I
want :)

 

Hope this helps.

regards

Clive

 

 

 

On Tue, 27 Nov 2012 11:06:18 +0000 Ben N wrote

Hi All, 

I'm wondering if anyone out there is using G729 with 50ms ptime through
Freeswitch, without proxy media?

I am unable to get this to work, even in a very basic lab with just two
clients and an up to date FS server on the same LAN. I can get G729 with
20ms working, but 50ms causes Freeswitch to hang the call.

It might be that mod_g729 and mod_com_g729 simply don't support it, so if
anyone in the know can tell me that would be great!

Further info can be supplied, I believe I have tried pretty much all
options....

Cheers,

Ben

 

  _____  

South Africa premier free email service - webmail.co.za
<http://www.webmail.co.za/>  
 <http://b.wm.co.za/click.pwm?cid=20053491&loc=N-MT&seq=50b717ec> 

 <http://b.wm.co.za/click.pwm?cid=20053491&loc=N-MT&seq=50b717ec>  

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From itispip-qq at hotmail.com  Mon Dec  3 19:44:05 2012
From: itispip-qq at hotmail.com (=?gb2312?B?zfXA7Q==?=)
Date: Tue, 4 Dec 2012 00:44:05 +0800
Subject: [Freeswitch-users] Missed call notification
In-Reply-To: <CAAtWKeE9nFAFg1YWpb+B1BwLrTOKpOdMhJ9KEYE0E6KuGG3gNA@mail.gmail.com>
References: <BLU169-W120766FAB52400013BA1BB28C420@phx.gbl>,
	<8AE80082-5AAC-4C25-87E9-52FBD5EDB593@endigotech.com>,
	<CAAtWKeFdgZ_MyPBiNUsDYZZsBRJMCe=d6vTSynxDtZ+ifi+KTQ@mail.gmail.com>,
	<BLU169-W118507FC5529D5F80F26BA18C420@phx.gbl>,
	<CA+zebiYsNB0vhnDuPZo=d_uB_9tqE5W3fAzEnrdnsvVtVB4OPg@mail.gmail.com>,
	<CAAtWKeE9nFAFg1YWpb+B1BwLrTOKpOdMhJ9KEYE0E6KuGG3gNA@mail.gmail.com>
Message-ID: <BLU169-W2283D9F938CCE5C8E60C2D8C400@phx.gbl>


Thanks Yehavi, it's really handy to implement this script to my environment :)
/Alex

Date: Sun, 2 Dec 2012 10:21:17 +0200
From: yehavi.bourvine at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Missed call notification

Hi,   I've created the following wiki page with example of how we do it:http://wiki.freeswitch.org/wiki/MailNoAsnwer
 I would be greatfull if someone can add it to the examples page, as I don't have the edit authority on it...                     Thanks, __Yehavi: 


2012/12/1 Komar, Jason <jkomar at jbox.ca>
I would be interested to see that script too.
Thanks,Jason

On Sat, Dec 1, 2012 at 11:34 AM, ?? <itispip-qq at hotmail.com> wrote:







Hi Yehavi, seems your solution is the best fit for the situation. Is that possible to share your script? Maybe can posted it to Freeswitch Wiki as part of the documentation.


Date: Sat, 1 Dec 2012 19:13:11 +0200
From: yehavi.bourvine at gmail.com


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Missed call notification

We have a script to find the hangup cause and send Email if the user requested it. This way I know who is looking for me while I am on the phone :-)

                       __Yehavi: 


2012/12/1 Brian Foster <bdfoster at endigotech.com>
Message Waiting Indicator (MWI) is the voicemail notification feature for IP phones. 


Missed call notifications are provided by the phone and not the switch.

Sent from my iPhone
On Dec 1, 2012, at 2:41 AM, ?? <itispip-qq at hotmail.com> wrote:








I heard that Freeswitch support missed call notification (i/o VM notification). But look around Wiki no document about this feature. Anyone knows where to setup it up?

/brgds, 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




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



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




_________________________________________________________________________

Professional FreeSWITCH Consulting 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

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

_________________________________________________________________________

Professional FreeSWITCH Consulting 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



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

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

http://www.freeswitch.org





_________________________________________________________________________
Professional FreeSWITCH Consulting 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
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From anthony.minessale at gmail.com  Mon Dec  3 20:20:19 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 3 Dec 2012 11:20:19 -0600
Subject: [Freeswitch-users] Freeswitch is crashing when httapi playback
 application is executed
In-Reply-To: <1354544270822-7585120.post@n2.nabble.com>
References: <1354210814465-7585041.post@n2.nabble.com>
	<1354544270822-7585120.post@n2.nabble.com>
Message-ID: <CAKbxfG-vT3rvD9cPGqcSLm0ypRMUZsB9kFYubwJuZHUhaAcUNw@mail.gmail.com>

Once we have a jira open, its ok to just put the details there.



On Mon, Dec 3, 2012 at 8:17 AM, Srigo <ksrigo at gmail.com> wrote:

> Hi,
>
> I have now more details about this crashed.
> Here is what i did (I used the FS you compiled as 'minessale"):
>
>  - Start FS as minessale:
>
>  su minessale
>  /usr/local/stow/freeswitch_20121130/bin/freeswitch -nonat -rp -nc -conf
> /etc/freeswitch/ -db /var/lib/freeswitch/db/ -log /var/log/freeswitch
>
>  Everthing worked fine, NO CRASH!!
>
>  - Then i tried samething by starting FS as "freeswitch"
>   su - freeswitch -s /bin/bash
>  /usr/local/stow/freeswitch_20121130/bin/freeswitch -nonat -rp -nc -conf
> /etc/freeswitch/ -db /var/lib/freeswitch/db/ -log /var/log/freeswitch
>
>  This time FS crashed.
>  According to "strace", we have a permission denied when accessing to
> /usr/local/stow/freeswitch_20121130/storage/http_file_cache. By changing
> the
> rights on "storage" I was able to run httapi without crash with the user
> "freeswitch"
>
>  I joined the trace file.
>
>  Additionnaly, we got the same issue with the compiled debian package: It's
> looking for its cache in /usr/storage which does nt exist.
>
>  I would like to suggest to you:
> - Checking return code when opening a file in "storage/http_file_cache" and
> log failures.
> - Fix the build system to create the storage cache with the correct rights
> - Add in httapi configuration file parameters to set the storage path
>
>
>  Regards,
>  Srigo
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-is-crashing-when-httapi-playback-application-is-executed-tp7585041p7585120.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
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From curriegrad2004 at gmail.com  Mon Dec  3 22:09:01 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 3 Dec 2012 11:09:01 -0800
Subject: [Freeswitch-users] Qos on Windows
In-Reply-To: <CAEW218JSrS9B7LK_WRs9irQr_0GHRQ5kfvNt=UAVuL+8=-yisg@mail.gmail.com>
References: <CAEW218JSrS9B7LK_WRs9irQr_0GHRQ5kfvNt=UAVuL+8=-yisg@mail.gmail.com>
Message-ID: <CAMWiMy9a+UspuViwVcXypxnycUQZTpfvyvzMxMGq3855r3fD5w@mail.gmail.com>

I would take a look under Policy-based QoS where you can set a DSCP
value and let the upstream router handle the rest

On Mon, Dec 3, 2012 at 3:19 AM, rentmycoder rentmycoder
<rentmycoder at gmail.com> wrote:
> It is possible to use iptables on Linux to add Qos mark to ip packets...
> Any hint how to set QOS on windows?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>


From gavin.henry at gmail.com  Mon Dec  3 22:41:44 2012
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 3 Dec 2012 19:41:44 +0000
Subject: [Freeswitch-users] XMPP Presence?
In-Reply-To: <CAMbW9oL7=XFkcA=cSTQiFn5RatPz+VH3OREvhpm8HT78nsPutA@mail.gmail.com>
References: <CAA8_NKB+NNq_AhMzH_pR-41mVhGV6Wi6h1qU8VkgXYi8m3JjTQ@mail.gmail.com>
	<1350630407387-7583767.post@n2.nabble.com>
	<CAHWtr1QS0s1B8jd6TtfQMt7crE3vKykLy19LM+MQbPFy0J9kwg@mail.gmail.com>
	<1350697094856-7583786.post@n2.nabble.com>
	<CAMbW9oL7=XFkcA=cSTQiFn5RatPz+VH3OREvhpm8HT78nsPutA@mail.gmail.com>
Message-ID: <CAA8_NKCOGb0uN0YzhdEbd_kEKFmf5Z1H=O77ZtW8MLBd8B4VEw@mail.gmail.com>

We ended up sponsoring the FreeSWITCH project for this integration to be
documented and configured for us. It should be on the wiki now?


On 20 October 2012 09:12, Andrew Cassidy <andrew at cassidywebservices.co.uk>wrote:

> I'm also interested in this, but not sure when I'll get a free window to
> play with it.
>
>
> On 20 October 2012 02:38, emnvn <nghiaem at gmail.com> wrote:
>
>> Hi,
>>
>> http://wiki.freeswitch.org/wiki/Mod_dingaling
>>
>> You can see this document. I have configured  successfully, two server can
>> communicate together. But it is not success all cases. In my case two
>> client
>> (xmpp and sip) can initialize the call but the media stream can't be
>> transmitted between two clients.
>>
>> If i use google and the xmpp client is google web i can voice and video
>> call
>> between two accounts. others client are not success.
>>
>> I have not jut ejabberd, if you can test on it please show me the results.
>>
>> Thanks
>>
>>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/XMPP-Presence-tp7580571p7583786.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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk
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From gavin.henry at gmail.com  Mon Dec  3 22:47:44 2012
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 3 Dec 2012 19:47:44 +0000
Subject: [Freeswitch-users] Genband S3 Intelligent SBC
Message-ID: <CAA8_NKBba2=QC9G+ipVmaoM3fi04-xbP4D1Bgv2WDCuL_3Q-pA@mail.gmail.com>

Hi all,

It looks like as we order more and more capacity for our UK Ofcom number
allocations that we host with BT Wholesale, we are going to be migrating
to Genband S3 Intelligent SBC from the ACME packets we currently
interconnect with using FreeSWITCH:

http://www.genband.com/media-center/press-releases/bt-offers-innovative-global-ip-exchange-interconnectivity-service-powere

Anyone else using FreeSWITCH with the Genband S3 SBC? Issues?
Recommendations?

Thanks,

Gavin.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk
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From amkusmirek at gmail.com  Mon Dec  3 16:10:17 2012
From: amkusmirek at gmail.com (=?ISO-8859-2?Q?Adam_Ku=B6mirek?=)
Date: Mon, 3 Dec 2012 14:10:17 +0100
Subject: [Freeswitch-users] voice announcement with Reason sip header
Message-ID: <CAMJTtGVM6KwQwSqkDP=NSttH=O08frShxM4SKKoRYiqFGD2eyA@mail.gmail.com>

Hi All

I have Freeswitch connected to SIP Trunk from local Service Provider.
Service Provider has NGN platform based on Broadworks applications.
When making failed outbound connections I never get SIP responses with
hangup cause, but they play 20 sec. voice announcement (eg. unallocated
number ...) and send 487 Request Terminated containing Reason header after
that.

---> INVITE
<--- 100 Trying
<---  183 Session Progress
<---  183 Session Progress with header Reason: Q.850;cause=?
<---  20 sec. voice announcement
<---  487 Request Terminated with header Reason: Q.850;cause=?
---> ACK

I'd like to hangup the call after 183 Session Progress
containing header Reason: Q.850;cause=?.

First i tried execute_on_media and execute_on_pre_answer, but the scripts
are triggered only for the first 183 Session Progress that does not contain
Reason header.

Then I tried  to resolve the problem by connecting to Outbound Socket, but
I can see only one CHANNEL_PROGRESS_MEDIA event  for the first 183 Session
Progress.

Can anyone help ?

Regards Adam
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From marketing at cluecon.com  Mon Dec  3 23:04:08 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 3 Dec 2012 12:04:08 -0800
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxUssdCHM-NXL9pQyAZixU7=0mA5c-a=fE9kS=iz_7hApA@mail.gmail.com>

Happy December to everyone!

Last week was painful for many of us as we were dealing with a sustained
DDoS attack on most of our infrastructure. Kudos to the guys for working
through it. It seems the worst is over and we can get back to the business
at hand: doing FreeSWITCH stuff. :)

In spite of the drama last week we did have a conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_28>and we
released 1.2.5.2! We discussed mostly the details of the DDoS we
experienced and how the community can assist in the future so that we can
mitigate the effects of such an occurrence. With the community's help we
will be more resistant to the effects of any future attacks. We appreciate
the outpouring of support we received from everyone.

This week <http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_05> we will go
back to discussing FreeSWITCH. We are still finalizing future guests so
this week we'll do another installment of tips and tricks from the
FreeSWITCH community. Among other things I will be showing how Chris Rienzo
(IRC: crienzo) and I used the source this weekend to figure out what the
XML preprocessor can do and get the wiki updated. I'll then show a simple
example of the always-present-but-previously-
undocumented command can do. As an added bonus we'll have an update on the
ClueCon <http://www.cluecon.com> 2012 videos!

Thanks and have a great week.

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From philq at qsystemsengineering.com  Mon Dec  3 23:28:23 2012
From: philq at qsystemsengineering.com (PhilQ)
Date: Mon, 3 Dec 2012 12:28:23 -0800 (PST)
Subject: [Freeswitch-users] Call disconnect upon re-invite - FS tears
	down call
In-Reply-To: <017f01cdb138$e00325c0$a0097140$@com>
References: <017f01cdb138$e00325c0$a0097140$@com>
Message-ID: <1354566503731-7585135.post@n2.nabble.com>

FYI - the solution to this problem is to ensure that "Outbound Proxy Server"
under "Basic SIP Network Settings" is populated on the Aastra phones.

- Phil



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Call-disconnect-upon-re-invite-FS-tears-down-call-tp7583888p7585135.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From paul at cupis.co.uk  Tue Dec  4 00:20:30 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Mon, 03 Dec 2012 21:20:30 +0000
Subject: [Freeswitch-users] Genband S3 Intelligent SBC
In-Reply-To: <CAA8_NKBba2=QC9G+ipVmaoM3fi04-xbP4D1Bgv2WDCuL_3Q-pA@mail.gmail.com>
References: <CAA8_NKBba2=QC9G+ipVmaoM3fi04-xbP4D1Bgv2WDCuL_3Q-pA@mail.gmail.com>
Message-ID: <50BD179E.5010805@cupis.co.uk>

On 03/12/12 19:47, Gavin Henry wrote:
> Anyone else using FreeSWITCH with the Genband S3 SBC? Issues?
> Recommendations?

I've found that FreeSWITCH works perfectly well with Genband S3 in 
general, including with BTwholesales implementation. Inter-op was 
relatively straightforward, taking into account BTs testing requirements.

Regards,


From nbhatti at gmail.com  Tue Dec  4 01:14:46 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Tue, 4 Dec 2012 01:14:46 +0300
Subject: [Freeswitch-users] Different dtmf type for different gateways in
	single profile
Message-ID: <CACak_yxsejacKz6jAkdfdhdAV5tquAF4eBVR4a4yyC81kvuOvg@mail.gmail.com>

Is there a way to set different dtmf type for different gateways in same
profile? I am originating from esl, call is bridged with audio file.
(Dialer setup). Some of the providers send dtmf  inband and others rfc2833.
I am currently checking in my lua script for a custom param, if present
then execute start_dtmf to listen for inband dtmf. While this breaks the
rest of dtmf. Is there a better way to deal with this or do I have to end
up with different profiles each with different dtmf-type.

--
Sent via a mobile device.
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From gavin.henry at gmail.com  Tue Dec  4 01:25:14 2012
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 3 Dec 2012 22:25:14 +0000
Subject: [Freeswitch-users] Genband S3 Intelligent SBC
In-Reply-To: <50BD179E.5010805@cupis.co.uk>
References: <CAA8_NKBba2=QC9G+ipVmaoM3fi04-xbP4D1Bgv2WDCuL_3Q-pA@mail.gmail.com>
	<50BD179E.5010805@cupis.co.uk>
Message-ID: <CAA8_NKCR=1bJ-YqnncOP8JdyQ-Qip-xq3-d3X2JpJ4LZTVociA@mail.gmail.com>

On 3 December 2012 21:20, Paul Cupis <paul at cupis.co.uk> wrote:
>
> On 03/12/12 19:47, Gavin Henry wrote:
> > Anyone else using FreeSWITCH with the Genband S3 SBC? Issues?
> > Recommendations?
>
> I've found that FreeSWITCH works perfectly well with Genband S3 in
> general, including with BTwholesales implementation. Inter-op was
> relatively straightforward, taking into account BTs testing requirements.

Thanks Paul. How recent was your last interop?


--
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk


From msc at freeswitch.org  Tue Dec  4 01:37:36 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Dec 2012 14:37:36 -0800
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <cdo25qex5xplrot14oad7fr7.1354501431119@email.android.com>
References: <cdo25qex5xplrot14oad7fr7.1354501431119@email.android.com>
Message-ID: <CAKzWOxVLewGaOzAPoHT7dtEV9odJFSc-4JKXD4tBLsg+nSnPoQ@mail.gmail.com>

On Sun, Dec 2, 2012 at 6:23 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:

> Hi Brian,
>
> I've posted the contents of dialplan. Please tell me what I am doing wrong.
>
>
What he means is that you are doing a blind transfer to an extension that
is designed for attended transfers. We already discussed this earlier in
the thread. In the meantime kudos to Abaci for pointing out a channel
variable that is very clearly mentioned on the valet_park wiki page:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park#Channel_Variables

Shame on us both for not looking that up. Well, mostly shame on you because
you're supposed to look at the wiki before you post here. :P

-MC
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From msc at freeswitch.org  Tue Dec  4 01:40:11 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Dec 2012 14:40:11 -0800
Subject: [Freeswitch-users] Freeswitch stable version 1.2.51
In-Reply-To: <1354526423398-7585116.post@n2.nabble.com>
References: <1354520230656-7585111.post@n2.nabble.com>
	<CCE1B4CA.614A6%krice@freeswitch.org>
	<1354522996442-7585113.post@n2.nabble.com>
	<CCE1C269.614AA%krice@freeswitch.org>
	<1354526423398-7585116.post@n2.nabble.com>
Message-ID: <CAKzWOxWxArZau5gVvZhEXvCtmO2Z_3Fzj9Qxi+eCJ=TW-XXiRA@mail.gmail.com>

FS 1.2.5.2 is definitely more stable/advanced/bug-free than 1.0.6,
regardless of the configuration you use. Just be sure that you test your
configuration when going from 1.0.6 to 1.2.5.2 before you throw it directly
into production. Sometimes people have come to rely on buggy behavior as if
it's a feature and when we fix the bug it breaks their "feature" and they
get very unhappy.

-MC

On Mon, Dec 3, 2012 at 1:20 AM, baskar <yudha2008 at gmail.com> wrote:

> Hi Steven and Ken,
>
> Thanks for the Quick response to my following quires.I just worried about
> the README. But before putting new version into a production environment
> surely i will test all the feature for a week and put them in live.
>
> Once again thanks for the reply.
>
> Thanks,
> N.Baskar
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-stable-version-1-2-51-tp7585111p7585116.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From avi at avimarcus.net  Tue Dec  4 01:50:02 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 4 Dec 2012 00:50:02 +0200
Subject: [Freeswitch-users] mod_cdr_mongodb follows json rfc4627?
In-Reply-To: <CAJOPU6VX7Aeh_9=xU_Rae7=5Tw-12G3-WDekaHXzN+jciBV0_g@mail.gmail.com>
References: <CAJOPU6XN2Wh0ROUEK45-4W-8XmEsM7qiomXFtU+odi5Vr2Qk2g@mail.gmail.com>
	<CAKbxfG_-BU2nOT6iK-U2hJb550_fYhq-hbt1VCvx4_+LCn2J+w@mail.gmail.com>
	<BD87D3B1554F614190EB6E54CE95DDB606E7D59B@mail2.patlive.local>
	<CAJOPU6VX7Aeh_9=xU_Rae7=5Tw-12G3-WDekaHXzN+jciBV0_g@mail.gmail.com>
Message-ID: <CALNZuwmk980XPR_0fDKb3sc+qwWO405SsaW_MkN9pr5pzoJ+1g@mail.gmail.com>

Hmm, this is sort of hijacking.

Does json_cdr - and possibly all these others -- skip the app_log - the
actual call*flow* -- it's in the xml_cdr.

I looked at a bunch of json_cdrs but I don't seem to see it. I just see
variable and then a callflow that just has times and origination numbers
and the like.

-Avi


On Mon, Dec 3, 2012 at 4:41 PM, Jose Fco. Irles Dur? <josefu at gmail.com>wrote:

> I also sent a patch for the mod_cdr_mongodb module:
>
> http://jira.freeswitch.org/browse/FS-4902
>
> 2012/12/3 Chad Engler <Chad.Engler at patlive.com>:
> > I submitted a patch that should fix this a while ago, but haven?t heard
> much
> > back:
> >
> >
> >
> > http://jira.freeswitch.org/browse/FS-4830
> >
> >
> >
> > -Chad
> >
> >
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Anthony
> > Minessale
> > Sent: Friday, November 30, 2012 5:29 PM
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] mod_cdr_mongodb follows json rfc4627?
> >
> >
> >
> > you are right, post the patch to JIRA please.
> >
> >
> >
> >
> >
> > On Fri, Nov 30, 2012 at 1:28 PM, Jose Fco. Irles Dur? <josefu at gmail.com>
> > wrote:
> >
> > Hello everybody, first of all, sorry for my english.
> >
> > I have a test server with the mod_cdr_mongodb backend configured.
> >
> > I'm trying to parse the cdrs with the mongodb driver for java, but i
> > have a trouble with a part of the json that freeswitch sends (mongo
> > saves the json document without errors). I haven't an API function to
> > extract some data. Firstly I thought that the problem was in the java
> > driver but I'm not sure.
> >
> > The problem is in the "callflow" part, with the json standard[1], a
> > json document can't have two keys with the same name, but
> > mod_cdr_mongodb builds the json with a 'n' callflow objects.
> > Also it happends inside callflow object, with origination, originator
> > and originatee objects.
> >
> > mod_cdr_mongodb saves the json with this format:
> > http://pastebin.com/VRz6s0eb
> >
> > I modified the source of the backend adding json arrays around this
> > objects and now I can parse without problems.
> >
> > The output with the modifications:
> > http://pastebin.com/v05FN9Ta
> >
> > I'm in the correct way or I'm missing something?
> >
> > [1] http://www.ietf.org/rfc/rfc4627.txt Section 2.2
> >
> > Regards
> > --
> > 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue Dec  4 02:09:46 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Mon, 03 Dec 2012 23:09:46 +0000
Subject: [Freeswitch-users] Genband S3 Intelligent SBC
In-Reply-To: <CAA8_NKCR=1bJ-YqnncOP8JdyQ-Qip-xq3-d3X2JpJ4LZTVociA@mail.gmail.com>
References: <CAA8_NKBba2=QC9G+ipVmaoM3fi04-xbP4D1Bgv2WDCuL_3Q-pA@mail.gmail.com>
	<50BD179E.5010805@cupis.co.uk>
	<CAA8_NKCR=1bJ-YqnncOP8JdyQ-Qip-xq3-d3X2JpJ4LZTVociA@mail.gmail.com>
Message-ID: <50BD313A.5070802@cupis.co.uk>

On 03/12/12 22:25, Gavin Henry wrote:
> On 3 December 2012 21:20, Paul Cupis<paul at cupis.co.uk>  wrote:
>> I've found that FreeSWITCH works perfectly well with Genband S3 in
>> general, including with BTwholesales implementation. Inter-op was
>> relatively straightforward, taking into account BTs testing requirements.
>
> Thanks Paul. How recent was your last interop?

A little over a year ago.

Regards,


From msc at freeswitch.org  Tue Dec  4 02:48:42 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Dec 2012 15:48:42 -0800
Subject: [Freeswitch-users] Different dtmf type for different gateways
 in single profile
In-Reply-To: <CACak_yxsejacKz6jAkdfdhdAV5tquAF4eBVR4a4yyC81kvuOvg@mail.gmail.com>
References: <CACak_yxsejacKz6jAkdfdhdAV5tquAF4eBVR4a4yyC81kvuOvg@mail.gmail.com>
Message-ID: <CAKzWOxXTDzOJ7Q0X4R+m06pG4R1w6kkkO7EA_7WzSbqO=crLZw@mail.gmail.com>

How does it "break the rest of DTMF"?
-MC

On Mon, Dec 3, 2012 at 2:14 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com>wrote:

> Is there a way to set different dtmf type for different gateways in same
> profile? I am originating from esl, call is bridged with audio file.
> (Dialer setup). Some of the providers send dtmf  inband and others rfc2833.
> I am currently checking in my lua script for a custom param, if present
> then execute start_dtmf to listen for inband dtmf. While this breaks the
> rest of dtmf. Is there a better way to deal with this or do I have to end
> up with different profiles each with different dtmf-type.
>
> --
> Sent via a mobile 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Tue Dec  4 03:48:03 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Dec 2012 16:48:03 -0800
Subject: [Freeswitch-users] voice announcement with Reason sip header
In-Reply-To: <CAMJTtGVM6KwQwSqkDP=NSttH=O08frShxM4SKKoRYiqFGD2eyA@mail.gmail.com>
References: <CAMJTtGVM6KwQwSqkDP=NSttH=O08frShxM4SKKoRYiqFGD2eyA@mail.gmail.com>
Message-ID: <CAKzWOxV8k5waLQ+tV5QHozBVu8B=9Hxp_f1q+KmiM7cRHTiX8Q@mail.gmail.com>

Can you post a SIP trace of this happening? That looks pretty interesting.
Use pastebin.freeswitch.org and put the URL of the pb post in this email
thread.
-MC

On Mon, Dec 3, 2012 at 5:10 AM, Adam Ku?mirek <amkusmirek at gmail.com> wrote:

> Hi All
>
> I have Freeswitch connected to SIP Trunk from local Service Provider.
> Service Provider has NGN platform based on Broadworks applications.
> When making failed outbound connections I never get SIP responses with
> hangup cause, but they play 20 sec. voice announcement (eg. unallocated
> number ...) and send 487 Request Terminated containing Reason header after
> that.
>
> ---> INVITE
> <--- 100 Trying
> <---  183 Session Progress
> <---  183 Session Progress with header Reason: Q.850;cause=?
> <---  20 sec. voice announcement
> <---  487 Request Terminated with header Reason: Q.850;cause=?
> ---> ACK
>
> I'd like to hangup the call after 183 Session Progress
> containing header Reason: Q.850;cause=?.
>
> First i tried execute_on_media and execute_on_pre_answer, but the scripts
> are triggered only for the first 183 Session Progress that does not contain
> Reason header.
>
> Then I tried  to resolve the problem by connecting to Outbound Socket, but
> I can see only one CHANNEL_PROGRESS_MEDIA event  for the first 183 Session
> Progress.
>
> Can anyone help ?
>
> Regards 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From dujinfang at gmail.com  Tue Dec  4 04:08:48 2012
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 4 Dec 2012 09:08:48 +0800
Subject: [Freeswitch-users] Questions about RTMP endpoint
In-Reply-To: <50BCACEC.9040507@gmail.com>
References: <50B631BC.8060001@gmail.com>
	<1726AEE3708448019DFAD73E4C2F8ACB@gmail.com>
	<50BCACEC.9040507@gmail.com>
Message-ID: <E35626F11E1D4998BCDF22B1A9A44B12@gmail.com>

It's a private branch. I made that work and then did no further test and real use. It is in a large change set (with video or sth. and client side changes) and I don't have time to split that out and test at this time. I would like to share it someday, do you consider some bounty?  

--  
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Monday, December 3, 2012 at 9:45 PM, Germ?n Ruiz wrote:

> Ok,
> but could you indicate which branch?  I don't found in the source.
>  
> Thanks
> Germ?n
>  
>  
> > We have text messages via rtmp in our branch. Just need time to find them out.  
> >  
> > --   
> > Seven Du
> > Sent with Sparrow (http://www.sparrowmailapp.com/?sig)
> >  
> >  
> > On Wednesday, November 28, 2012 at 11:46 PM, Germ?n Ruiz wrote:
> >  
> > > Hi,
> > > I'm adding voice calls to a web site. I'm using RTMP endpoint and Flex
> > > clientprovided. I have the following questions:
> > > - Can flex clients (using RTMP protocol) register their presence and are
> > > visible by others connected using SIP?
> > > - Is it possible to send and receive text messages with RTMP protocol?
> > >  
> > > Thanks  
> > > Germ?n
> > >  
> > > _________________________________________________________________________  
> > > 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 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
>  
> 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 ntomer at newgen.co.in  Tue Dec  4 07:25:41 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Tue, 4 Dec 2012 09:55:41 +0530
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <CAKzWOxVLewGaOzAPoHT7dtEV9odJFSc-4JKXD4tBLsg+nSnPoQ@mail.gmail.com>
References: <cdo25qex5xplrot14oad7fr7.1354501431119@email.android.com>
	<CAKzWOxVLewGaOzAPoHT7dtEV9odJFSc-4JKXD4tBLsg+nSnPoQ@mail.gmail.com>
Message-ID: <00aa01cdd1d7$6bcea110$436be330$@co.in>

Accepted J I should have looked that up.

 

Need another help, is there is way to check from Lua whether an extension is
available?

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, December 04, 2012 4:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] valet_park help needed

 

 

On Sun, Dec 2, 2012 at 6:23 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:

Hi Brian,

I've posted the contents of dialplan. Please tell me what I am doing wrong.

 

What he means is that you are doing a blind transfer to an extension that is
designed for attended transfers. We already discussed this earlier in the
thread. In the meantime kudos to Abaci for pointing out a channel variable
that is very clearly mentioned on the valet_park wiki page:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park#Channel_Vari
ables

Shame on us both for not looking that up. Well, mostly shame on you because
you're supposed to look at the wiki before you post here. :P

-MC


Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From qasimakhan at gmail.com  Tue Dec  4 10:05:18 2012
From: qasimakhan at gmail.com (qasimakhan at gmail.com)
Date: Tue, 4 Dec 2012 12:05:18 +0500
Subject: [Freeswitch-users] RMPT module gets stuck!
Message-ID: <CAPBKugtKTzsvukjZWMu_zMkz-1BOv-6r_hetM0PNUqC7tuym3A@mail.gmail.com>

Hi,

I am using RTMP module in freeswitch and after a few days one of the
processors gets stuck on 100% processing. Here are a few logs that might
help.

freeswitch at internal> rtmp status profile default sessions
> Profile: default
> I/O Backend: tcp
> Bind address: 0.0.0.0:1935
> Active calls: 0
>
> Sessions:
> uuid,address,user,domain,flashVer
> 6094eee2-f9a2-427e-ada4-d4a9052525b0,X.X.X.X:54316,NNNNNNNNN,X.X.X.X,WIN
> 11,5,31,2
> a1bb87a4-98a7-4b1e-b4d5-080e54690c31,X.X.X.X:50667,NNNNNNNNNN,X.X.X.X,WIN
> 11,5,502,110
> f2df9e88-17b3-4685-9336-01fcdd2a5a0d,X.X.X.X:51806,NNNNNNNNNN,X.X.X.X,WIN
> 11,5,502,110
> f9d46493-a707-48a5-9cbb-032aa80919bb,X.X.X.X:14331,,,WIN 11,5,502,110
> 7a561e8b-4d34-49ba-b4e8-b6c9ec58abd5,X.X.X.X:51998,,,WIN 11,4,402,287
>

All the IP's are replaced by X.X.X.X and dialed numbers are replaced by
NNNNNNN.

My rtmp.conf.xml looks something like:

> <configuration name="rtmp.conf" description="RTMP Endpoint">
>   <profiles>
>         <profile name="default">
>                 <settings>
>                         <param name="bind-address" value="0.0.0.0:1935" />
>                         <param name="context" value="public" />
>                         <param name="dialplan" value="XML" />
>                         <!-- If this is set to true, no unauthenticated
> inbound calls will be allowed -->
>                         <param name="auth-calls" value="true" />
>                         <!-- How much time should the clients buffer the
> media stream (miliseconds) -->
>                         <param name="buffer-len" value="128" />
>                         <!-- Sets the maximum size of outbound RTMP chunks
> -->
>                         <param name="chunksize" value="128" />
>                 </settings>
>         </profile>
>   </profiles>
> </configuration>
>

I have tried several freeswitch versions but the problem persists. Please
let me know if anything else is needed i will provide it.

Regards,
Qasim
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From =?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?=  Tue Dec  4 09:08:44 2012
From: =?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?= (=?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?=)
Date: Tue, 04 Dec 2012 12:08:44 +0600
Subject: [Freeswitch-users] CPS
Message-ID: <20121204120844.c7d7a893@mail.btcom.kz>

Hi,

Could anyone explain me what impact does CPS config param has on FS?

Say it's set to 30, does that mean that FS box won't allow creation of more than 30 call per sec? Or FS just log that in a log?

Regards,
Y.


From peter.olsson at visionutveckling.se  Tue Dec  4 10:16:40 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 4 Dec 2012 07:16:40 +0000
Subject: [Freeswitch-users] RMPT module gets stuck!
In-Reply-To: <CAPBKugtKTzsvukjZWMu_zMkz-1BOv-6r_hetM0PNUqC7tuym3A@mail.gmail.com>
References: <CAPBKugtKTzsvukjZWMu_zMkz-1BOv-6r_hetM0PNUqC7tuym3A@mail.gmail.com>
Message-ID: <0378FF42-43CF-47F7-B02E-1C2D874F585F@visionutveckling.se>

Please read how to report a bug on the wiki. Then add all the needed information to a Jira.

/Peter

4 dec 2012 kl. 08:14 skrev "qasimakhan at gmail.com<mailto:qasimakhan at gmail.com>" <qasimakhan at gmail.com<mailto:qasimakhan at gmail.com>>:

Hi,

I am using RTMP module in freeswitch and after a few days one of the processors gets stuck on 100% processing. Here are a few logs that might help.

freeswitch at internal> rtmp status profile default sessions
Profile: default
I/O Backend: tcp
Bind address: 0.0.0.0:1935<http://0.0.0.0:1935>
Active calls: 0

Sessions:
uuid,address,user,domain,flashVer
6094eee2-f9a2-427e-ada4-d4a9052525b0,X.X.X.X:54316,NNNNNNNNN,X.X.X.X,WIN 11,5,31,2
a1bb87a4-98a7-4b1e-b4d5-080e54690c31,X.X.X.X:50667,NNNNNNNNNN,X.X.X.X,WIN 11,5,502,110
f2df9e88-17b3-4685-9336-01fcdd2a5a0d,X.X.X.X:51806,NNNNNNNNNN,X.X.X.X,WIN 11,5,502,110
f9d46493-a707-48a5-9cbb-032aa80919bb,X.X.X.X:14331,,,WIN 11,5,502,110
7a561e8b-4d34-49ba-b4e8-b6c9ec58abd5,X.X.X.X:51998,,,WIN 11,4,402,287

All the IP's are replaced by X.X.X.X and dialed numbers are replaced by NNNNNNN.

My rtmp.conf.xml looks something like:
<configuration name="rtmp.conf" description="RTMP Endpoint">
  <profiles>
        <profile name="default">
                <settings>
                        <param name="bind-address" value="0.0.0.0:1935<http://0.0.0.0:1935>" />
                        <param name="context" value="public" />
                        <param name="dialplan" value="XML" />
                        <!-- If this is set to true, no unauthenticated inbound calls will be allowed -->
                        <param name="auth-calls" value="true" />
                        <!-- How much time should the clients buffer the media stream (miliseconds) -->
                        <param name="buffer-len" value="128" />
                        <!-- Sets the maximum size of outbound RTMP chunks -->
                        <param name="chunksize" value="128" />
                </settings>
        </profile>
  </profiles>
</configuration>

I have tried several freeswitch versions but the problem persists. Please let me know if anything else is needed i will provide it.

Regards,
Qasim

!DSPAM:50bd9e9c32762677411666!
_________________________________________________________________________
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://lists.freeswitch.org/mailman/options/freeswitch-users>
http://www.freeswitch.org


!DSPAM:50bd9e9c32762677411666!


From steveayre at gmail.com  Tue Dec  4 11:04:25 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 4 Dec 2012 08:04:25 +0000
Subject: [Freeswitch-users] RMPT module gets stuck!
In-Reply-To: <0378FF42-43CF-47F7-B02E-1C2D874F585F@visionutveckling.se>
References: <CAPBKugtKTzsvukjZWMu_zMkz-1BOv-6r_hetM0PNUqC7tuym3A@mail.gmail.com>
	<0378FF42-43CF-47F7-B02E-1C2D874F585F@visionutveckling.se>
Message-ID: <CAFiqYunGhHj5d6548WkwVD7vDvzCeytHFCOc5Dyz+dE9rL8g6Q@mail.gmail.com>

100% sounds like something may be stuck in a loop.

Use a gcore to collect a coredump of the running process (it will keep
running after the coredump). Usage is 'gcore PID'

Attach the backtrace to the Jira ticket.
http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Getting_a_Backtrace

You should try on latest Git head - it may be something that has already
been fixed in a newer version, and ongoing development means the latest
code may have changed significantly which makes the backtrace less useful.



On 4 December 2012 07:16, Peter Olsson <peter.olsson at visionutveckling.se>wrote:

> Please read how to report a bug on the wiki. Then add all the needed
> information to a Jira.
>
> /Peter
>
> 4 dec 2012 kl. 08:14 skrev "qasimakhan at gmail.com<mailto:
> qasimakhan at gmail.com>" <qasimakhan at gmail.com<mailto:qasimakhan at gmail.com
> >>:
>
> Hi,
>
> I am using RTMP module in freeswitch and after a few days one of the
> processors gets stuck on 100% processing. Here are a few logs that might
> help.
>
> freeswitch at internal> rtmp status profile default sessions
> Profile: default
> I/O Backend: tcp
> Bind address: 0.0.0.0:1935<http://0.0.0.0:1935>
> Active calls: 0
>
> Sessions:
> uuid,address,user,domain,flashVer
> 6094eee2-f9a2-427e-ada4-d4a9052525b0,X.X.X.X:54316,NNNNNNNNN,X.X.X.X,WIN
> 11,5,31,2
> a1bb87a4-98a7-4b1e-b4d5-080e54690c31,X.X.X.X:50667,NNNNNNNNNN,X.X.X.X,WIN
> 11,5,502,110
> f2df9e88-17b3-4685-9336-01fcdd2a5a0d,X.X.X.X:51806,NNNNNNNNNN,X.X.X.X,WIN
> 11,5,502,110
> f9d46493-a707-48a5-9cbb-032aa80919bb,X.X.X.X:14331,,,WIN 11,5,502,110
> 7a561e8b-4d34-49ba-b4e8-b6c9ec58abd5,X.X.X.X:51998,,,WIN 11,4,402,287
>
> All the IP's are replaced by X.X.X.X and dialed numbers are replaced by
> NNNNNNN.
>
> My rtmp.conf.xml looks something like:
> <configuration name="rtmp.conf" description="RTMP Endpoint">
>   <profiles>
>         <profile name="default">
>                 <settings>
>                         <param name="bind-address" value="0.0.0.0:1935<
> http://0.0.0.0:1935>" />
>                         <param name="context" value="public" />
>                         <param name="dialplan" value="XML" />
>                         <!-- If this is set to true, no unauthenticated
> inbound calls will be allowed -->
>                         <param name="auth-calls" value="true" />
>                         <!-- How much time should the clients buffer the
> media stream (miliseconds) -->
>                         <param name="buffer-len" value="128" />
>                         <!-- Sets the maximum size of outbound RTMP chunks
> -->
>                         <param name="chunksize" value="128" />
>                 </settings>
>         </profile>
>   </profiles>
> </configuration>
>
> I have tried several freeswitch versions but the problem persists. Please
> let me know if anything else is needed i will provide it.
>
> Regards,
> Qasim
>
> !DSPAM:50bd9e9c32762677411666!
> _________________________________________________________________________
> 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://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org
>
>
> !DSPAM:50bd9e9c32762677411666!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 qasimakhan at gmail.com  Tue Dec  4 13:43:54 2012
From: qasimakhan at gmail.com (qasimakhan at gmail.com)
Date: Tue, 4 Dec 2012 15:43:54 +0500
Subject: [Freeswitch-users] WebRTC Call
In-Reply-To: <CANJ4vTraADdabQWs4sT-zs1KTUh59k4RXfPXQnsbqqgPftJSjw@mail.gmail.com>
References: <CANJ4vTraADdabQWs4sT-zs1KTUh59k4RXfPXQnsbqqgPftJSjw@mail.gmail.com>
Message-ID: <CAPBKuguCN4aOcqM3B2pwkX=zL0Fq1tbTpm6zkPudSNur-axtow@mail.gmail.com>

I have successfuly made SIP-SIP (Both Audio & Video) calls using OverSIPs
as WebRTC-SIP gateway and using Opensips 1.7 with VIA patch for WebRTC.

Regards,
Qasim

On Sat, Dec 1, 2012 at 2:34 AM, Ros P <rosphotos82 at gmail.com> wrote:

> Hello,
>
> Has anyone had any luck in implementing a webrtc call? I have been able to
> register using the webrtc2sip gw and sipml5 client but no audio goes
> through. As this seems to be a client problem (issue 39<http://code.google.com/p/sipml5/issues/detail?id=39>),
> does anyone have any other suggestions? Anything successfully tested?
>
> Kind Regards,
> ros
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 nbhatti at gmail.com  Tue Dec  4 14:41:51 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Tue, 4 Dec 2012 14:41:51 +0300
Subject: [Freeswitch-users] Different dtmf type for different gateways
	in single profile
In-Reply-To: <CAKzWOxXTDzOJ7Q0X4R+m06pG4R1w6kkkO7EA_7WzSbqO=crLZw@mail.gmail.com>
References: <CACak_yxsejacKz6jAkdfdhdAV5tquAF4eBVR4a4yyC81kvuOvg@mail.gmail.com>
	<CAKzWOxXTDzOJ7Q0X4R+m06pG4R1w6kkkO7EA_7WzSbqO=crLZw@mail.gmail.com>
Message-ID: <473CAD87-8625-4580-BEDC-08DE4475CE22@gmail.com>


If I set to dtmf-type=none in profile, only inband works because of start_dtmf, If set dtmf-type=rfc2833, in band starts false positive catches. I may be doing something practically wrong here, so just wanted to know what options do we have to set it up correctly? 

Thanks,
--
Muhammad Naseer Bhatti



On Dec 4, 2012, at 2:48 AM, Michael Collins <msc at freeswitch.org> wrote:

> How does it "break the rest of DTMF"?
> -MC
> 
> On Mon, Dec 3, 2012 at 2:14 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:
> Is there a way to set different dtmf type for different gateways in same profile? I am originating from esl, call is bridged with audio file. (Dialer setup). Some of the providers send dtmf  inband and others rfc2833. I am currently checking in my lua script for a custom param, if present then execute start_dtmf to listen for inband dtmf. While this breaks the rest of dtmf. Is there a better way to deal with this or do I have to end up with different profiles each with different dtmf-type.
> --
> Sent via a mobile 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 shaheryarkh at gmail.com  Tue Dec  4 15:24:29 2012
From: shaheryarkh at gmail.com (Muhammad Shahzad)
Date: Tue, 4 Dec 2012 13:24:29 +0100
Subject: [Freeswitch-users] WebRTC Call
In-Reply-To: <CAPBKuguCN4aOcqM3B2pwkX=zL0Fq1tbTpm6zkPudSNur-axtow@mail.gmail.com>
References: <CANJ4vTraADdabQWs4sT-zs1KTUh59k4RXfPXQnsbqqgPftJSjw@mail.gmail.com>
	<CAPBKuguCN4aOcqM3B2pwkX=zL0Fq1tbTpm6zkPudSNur-axtow@mail.gmail.com>
Message-ID: <CAFZQphzWJ_FL0oQ6rPWMWMOcy8fSErSrFwHUFXB=RCJoqnx4nw@mail.gmail.com>

Also you can use is WebRTC2SIP gateway which can connect to any traditional
SIP server, like freeswitch or asterisk.

http://code.google.com/p/webrtc2sip/

Thank you.


On Tue, Dec 4, 2012 at 11:43 AM, qasimakhan at gmail.com
<qasimakhan at gmail.com>wrote:

> I have successfuly made SIP-SIP (Both Audio & Video) calls using OverSIPs
> as WebRTC-SIP gateway and using Opensips 1.7 with VIA patch for WebRTC.
>
> Regards,
> Qasim
>
> On Sat, Dec 1, 2012 at 2:34 AM, Ros P <rosphotos82 at gmail.com> wrote:
>
>> Hello,
>>
>> Has anyone had any luck in implementing a webrtc call? I have been able
>> to register using the webrtc2sip gw and sipml5 client but no audio goes
>> through. As this seems to be a client problem (issue 39<http://code.google.com/p/sipml5/issues/detail?id=39>),
>> does anyone have any other suggestions? Anything successfully tested?
>>
>> Kind Regards,
>> ros
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From ksrigo at gmail.com  Tue Dec  4 17:29:08 2012
From: ksrigo at gmail.com (Srigo)
Date: Tue, 4 Dec 2012 06:29:08 -0800 (PST)
Subject: [Freeswitch-users] Freeswitch is crashing when httapi playback
 application is executed
In-Reply-To: <1354210814465-7585041.post@n2.nabble.com>
References: <1354210814465-7585041.post@n2.nabble.com>
Message-ID: <1354631348345-7585154.post@n2.nabble.com>

Ok Anthony...Good to knw for next times.

Srigo



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-is-crashing-when-httapi-playback-application-is-executed-tp7585041p7585154.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From krice at freeswitch.org  Tue Dec  4 17:57:33 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 04 Dec 2012 08:57:33 -0600
Subject: [Freeswitch-users] CPS
In-Reply-To: <20121204120844.c7d7a893@mail.btcom.kz>
Message-ID: <CCE36B7D.61537%krice@freeswitch.org>

If you mean the SPS setting this is sessions per second, it defaults to 30,
that means 30 sessions (or call legs) per second can be setup, after that,
new sessions are blocked from starting... If you need to do more simply
increase the number the setting allows


Keep in mind, this setting is there to keep you from melting down your
machine, set it for something reasonable... Same thing with Max Sessions its
there to keep you from overloading the box...


On 12/4/12 12:08 AM, "????? ???????" <e.tylepov at btcom.kz> wrote:

> Hi,
> 
> Could anyone explain me what impact does CPS config param has on FS?
> 
> Say it's set to 30, does that mean that FS box won't allow creation of more
> than 30 call per sec? Or FS just log that in a log?
> 
> Regards,
> Y.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From brian at freeswitch.org  Tue Dec  4 18:11:23 2012
From: brian at freeswitch.org (Brian West)
Date: Tue, 4 Dec 2012 09:11:23 -0600
Subject: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes
In-Reply-To: <002d01cdd16b$492b9030$db82b090$@gmail.com>
References: <CADaqzFMD+Y8yc_qWNtp0drmceSU6EDogN7F6TyhtSDLbT2JeWA@mail.gmail.com>	<6036422a0e8b5982ed7af8823015bb38@www.webmail.co.za>
	<CADaqzFOGTva6mZBgewCKc8gdHmGdDcMn-AMPPnehHTNA4n-H_w@mail.gmail.com>
	<002d01cdd16b$492b9030$db82b090$@gmail.com>
Message-ID: <0C442564-683E-4271-A76B-6D5FF3EB2797@freeswitch.org>

PCAP's will show you.... I suspect something is telling a lie about the timestamps.  Very common.
--
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
ISN: 410*543





On Dec 3, 2012, at 9:31 AM, Ben <ben122uk at gmail.com> wrote:

> I tried setting the {passthru_ptime_mismatch=true} variable in the dialplan, but no joy unfortunately.  I?m also fairly sure that the codecs are matched on both call legs, so shouldn?t be having that sort of issue.
>  
> I?m going to gather up some evidence and console logs/traces etc so that I can put something together for a bug report on Jira.  Is there anything else that someone can suggest I try before it goes on Jira?  Is anyone able to re-produce the same problem?  Admittedly, I?m not on the latest GIT, but I?m on a recent (last couple of months) stable release.
>  
> Thanks in advance for any assistance!
>  
> Regards,
>  
> Ben
>  



From brian at freeswitch.org  Tue Dec  4 18:15:23 2012
From: brian at freeswitch.org (Brian West)
Date: Tue, 4 Dec 2012 09:15:23 -0600
Subject: [Freeswitch-users] Call disconnect upon re-invite - FS tears
	down call
In-Reply-To: <CAFVUaV8TwiNqean_gs1EbNPRXjE=Cv+8Us7-c3yWG_n00NduGg@mail.gmail.com>
References: <017f01cdb138$e00325c0$a0097140$@com>
	<CAFVUaV8TwiNqean_gs1EbNPRXjE=Cv+8Us7-c3yWG_n00NduGg@mail.gmail.com>
Message-ID: <15AD3C94-1640-486D-B1C5-B961EB30F35B@freeswitch.org>

Y U NO FILE JIRA?
--
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
ISN: 410*543





On Oct 23, 2012, at 11:32 AM, Yiftach Golan <yiftah at choochee.com> wrote:

> 
> You did not send the initial invite in the Jira so I do not know if the initial request constructed correctly
> but judging by the code it looks like your request is still pending when the second invite arrives 
> 
> Thanks,
> Yiftach.



From andrew at cassidywebservices.co.uk  Tue Dec  4 18:50:39 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Tue, 4 Dec 2012 15:50:39 +0000
Subject: [Freeswitch-users] Call disconnect upon re-invite - FS tears
	down call
In-Reply-To: <15AD3C94-1640-486D-B1C5-B961EB30F35B@freeswitch.org>
References: <017f01cdb138$e00325c0$a0097140$@com>
	<CAFVUaV8TwiNqean_gs1EbNPRXjE=Cv+8Us7-c3yWG_n00NduGg@mail.gmail.com>
	<15AD3C94-1640-486D-B1C5-B961EB30F35B@freeswitch.org>
Message-ID: <CAMbW9o+9SpGpTzj5JVQajx-UO3WER4gxqCd+CG0aydOLr2UPUQ@mail.gmail.com>

I was half expecting the meme image that goes with that...

On 4 December 2012 15:15, Brian West <brian at freeswitch.org> wrote:

> Y U NO FILE JIRA?
> --
> 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
> ISN: 410*543
>
>
>
>
>
> On Oct 23, 2012, at 11:32 AM, Yiftach Golan <yiftah at choochee.com> wrote:
>
> >
> > You did not send the initial invite in the Jira so I do not know if the
> initial request constructed correctly
> > but judging by the code it looks like your request is still pending when
> the second invite arrives
> >
> > Thanks,
> > Yiftach.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
*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 freeswitch at orresta.no-ip.com  Tue Dec  4 13:42:03 2012
From: freeswitch at orresta.no-ip.com (Jakob)
Date: Tue, 04 Dec 2012 11:42:03 +0100
Subject: [Freeswitch-users] mod_sndfile can't find files
Message-ID: <50BDD37B.80308@orresta.no-ip.com>

Hi,

I did a clean install from git master and keep getting file not found in 
the logs. This was noticed in late nov on a fedora 17 git install but is 
still present in a recent (as of today dec 4:th) debian squeeze git 
master system.

This was tested using sipdroid on the same lan.

Relevant part of the log follows:

2012-11-25 21:10:35.053024 [DEBUG] switch_core_session.c:830 Send signal 
sofia/internal/1050 at 172.16.10.9 [BREAK]
2012-11-25 21:10:35.053024 [DEBUG] switch_channel.c:3380 
(sofia/internal/1050 at 172.16.10.9) Callstate Change RINGING -> ACTIVE
2012-11-25 21:10:35.053024 [DEBUG] switch_core_session.c:976 Send signal 
sofia/internal/1050 at 172.16.10.9 [BREAK]
2012-11-25 21:10:35.053024 [NOTICE] mod_dptools.c:1176 Channel 
[sofia/internal/1050 at 172.16.10.9] has been answered
2012-11-25 21:10:35.053024 [DEBUG] sofia.c:5607 Channel 
sofia/internal/1050 at 172.16.10.9 entering state [completed][200]
EXECUTE sofia/internal/1050 at 172.16.10.9 sleep(1000)
2012-11-25 21:10:35.113025 [DEBUG] switch_core_session.c:976 Send signal 
sofia/internal/1050 at 172.16.10.9 [BREAK]
2012-11-25 21:10:35.113025 [DEBUG] switch_core_session.c:976 Send signal 
sofia/internal/1050 at 172.16.10.9 [BREAK]
2012-11-25 21:10:35.113025 [DEBUG] switch_core_session.c:976 Send signal 
sofia/internal/1050 at 172.16.10.9 [BREAK]
2012-11-25 21:10:35.133027 [DEBUG] sofia.c:5607 Channel 
sofia/internal/1050 at 172.16.10.9 entering state [ready][200]
EXECUTE sofia/internal/1050 at 172.16.10.9 voicemail(check default)
2012-11-25 21:10:36.053026 [DEBUG] switch_ivr_play_say.c:67 No language 
specified - Using [en]
2012-11-25 21:10:36.253026 [DEBUG] switch_ivr_play_say.c:67 No language 
specified - Using [en]
2012-11-25 21:10:36.253026 [DEBUG] switch_ivr_play_say.c:244 Handle 
play-file:[voicemail/vm-enter_id.wav] (en:en)
2012-11-25 21:10:36.253026 [ERR] mod_sndfile.c:198 Error Opening File 
[/usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-enter_$
2012-11-25 21:10:36.353027 [DEBUG] switch_ivr_play_say.c:67 No language 
specified - Using [en]
2012-11-25 21:10:36.353027 [DEBUG] switch_ivr_play_say.c:244 Handle 
play-file:[voicemail/vm-goodbye.wav] (en:en)
2012-11-25 21:10:36.353027 [ERR] mod_sndfile.c:198 Error Opening File 
[/usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-goodby$
2012-11-25 21:10:36.453048 [NOTICE] switch_core_state_machine.c:262 
sofia/internal/1050 at 172.16.10.9 has executed the last dialplan i$
2012-11-25 21:10:36.453048 [DEBUG] switch_channel.c:2979 
(sofia/internal/1050 at 172.16.10.9) Callstate Change ACTIVE -> HANGUP

Best regards
Jakob Sundberg


From amkusmirek at gmail.com  Tue Dec  4 15:11:00 2012
From: amkusmirek at gmail.com (=?ISO-8859-2?Q?Adam_Ku=B6mirek?=)
Date: Tue, 4 Dec 2012 13:11:00 +0100
Subject: [Freeswitch-users] voice announcement with Reason sip header
Message-ID: <CAMJTtGX5+D4ihh4hfEtqMnrPNko9FszmowfyKyBByNC9+ORxDg@mail.gmail.com>

Hi,

Url of the pb post: http://pastebin.freeswitch.org/20284

Thanks for any help

Regards Adam
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From msc at freeswitch.org  Tue Dec  4 19:21:06 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Dec 2012 08:21:06 -0800
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <00aa01cdd1d7$6bcea110$436be330$@co.in>
References: <cdo25qex5xplrot14oad7fr7.1354501431119@email.android.com>
	<CAKzWOxVLewGaOzAPoHT7dtEV9odJFSc-4JKXD4tBLsg+nSnPoQ@mail.gmail.com>
	<00aa01cdd1d7$6bcea110$436be330$@co.in>
Message-ID: <CAKzWOxUBVqEnXtRa=f7TJjBup47v0_2e3eTnUn-rWx=LBe6K+Q@mail.gmail.com>

On Mon, Dec 3, 2012 at 8:25 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:

>  Accepted J I should have looked that up?****
>
> ** **
>
> Need another help, is there is way to check from Lua whether an extension
> is available?
>
What do you mean by "extension is available"?
-MC


> ****
>
> ** **
>
> Regards****
>
> ** **
>
> Nitin****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Tuesday, December 04, 2012 4:08 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] valet_park help needed****
>
> ** **
>
> ** **
>
> On Sun, Dec 2, 2012 at 6:23 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:**
> **
>
> Hi Brian,
>
> I've posted the contents of dialplan. Please tell me what I am doing wrong.
> ****
>
> ** **
>
> What he means is that you are doing a blind transfer to an extension that
> is designed for attended transfers. We already discussed this earlier in
> the thread. In the meantime kudos to Abaci for pointing out a channel
> variable that is very clearly mentioned on the valet_park wiki page:
>
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park#Channel_Variables
>
> Shame on us both for not looking that up. Well, mostly shame on you
> because you're supposed to look at the wiki before you post here. :P
>
> -MC****
>
> Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
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> http://www.freeswitch.org
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>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Tue Dec  4 19:27:30 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Dec 2012 08:27:30 -0800
Subject: [Freeswitch-users] Different dtmf type for different gateways
 in single profile
In-Reply-To: <473CAD87-8625-4580-BEDC-08DE4475CE22@gmail.com>
References: <CACak_yxsejacKz6jAkdfdhdAV5tquAF4eBVR4a4yyC81kvuOvg@mail.gmail.com>
	<CAKzWOxXTDzOJ7Q0X4R+m06pG4R1w6kkkO7EA_7WzSbqO=crLZw@mail.gmail.com>
	<473CAD87-8625-4580-BEDC-08DE4475CE22@gmail.com>
Message-ID: <CAKzWOxULqB7nubtt7zwZ2uFv-6hBW9UtO2FWms43W7CykC=waw@mail.gmail.com>

I would set inband dtmf on a per-call basis or I would use a completely
different profile.
-MC

On Tue, Dec 4, 2012 at 3:41 AM, Muhammad Naseer Bhatti <nbhatti at gmail.com>wrote:

>
> If I set to dtmf-type=none in profile, only inband works because of
> start_dtmf, If set dtmf-type=rfc2833, in band starts false positive
> catches. I may be doing something practically wrong here, so just wanted to
> know what options do we have to set it up correctly?
>
> Thanks,
> --
> Muhammad Naseer Bhatti
>
>
>
> On Dec 4, 2012, at 2:48 AM, Michael Collins <msc at freeswitch.org> wrote:
>
> How does it "break the rest of DTMF"?
> -MC
>
> On Mon, Dec 3, 2012 at 2:14 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com>wrote:
>
>> Is there a way to set different dtmf type for different gateways in same
>> profile? I am originating from esl, call is bridged with audio file.
>> (Dialer setup). Some of the providers send dtmf  inband and others rfc2833.
>> I am currently checking in my lua script for a custom param, if present
>> then execute start_dtmf to listen for inband dtmf. While this breaks the
>> rest of dtmf. Is there a better way to deal with this or do I have to end
>> up with different profiles each with different dtmf-type.
>>
>> --
>> Sent via a mobile 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
> http://www.ClueCon.com <http://www.cluecon.com/>
> http://www.OSTAG.org <http://www.ostag.org/>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Tue Dec  4 19:34:23 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Dec 2012 08:34:23 -0800
Subject: [Freeswitch-users] mod_sndfile can't find files
In-Reply-To: <50BDD37B.80308@orresta.no-ip.com>
References: <50BDD37B.80308@orresta.no-ip.com>
Message-ID: <CAKzWOxX1vC6JUO2D7rcxQcV9by+OHpatCggc7iLfkuVquVRiNg@mail.gmail.com>

I would rebuild from a clean git checkout. It looks like it's trying to
find a wonky filename. Notice the dollar sign at the end of the file name.
If it re-occurs on a clean git checkout then go ahead and open a Jira.

-MC

On Tue, Dec 4, 2012 at 2:42 AM, Jakob <freeswitch at orresta.no-ip.com> wrote:

> Hi,
>
> I did a clean install from git master and keep getting file not found in
> the logs. This was noticed in late nov on a fedora 17 git install but is
> still present in a recent (as of today dec 4:th) debian squeeze git
> master system.
>
> This was tested using sipdroid on the same lan.
>
> Relevant part of the log follows:
>
> 2012-11-25 21:10:35.053024 [DEBUG] switch_core_session.c:830 Send signal
> sofia/internal/1050 at 172.16.10.9 [BREAK]
> 2012-11-25 21:10:35.053024 [DEBUG] switch_channel.c:3380
> (sofia/internal/1050 at 172.16.10.9) Callstate Change RINGING -> ACTIVE
> 2012-11-25 21:10:35.053024 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/1050 at 172.16.10.9 [BREAK]
> 2012-11-25 21:10:35.053024 [NOTICE] mod_dptools.c:1176 Channel
> [sofia/internal/1050 at 172.16.10.9] has been answered
> 2012-11-25 21:10:35.053024 [DEBUG] sofia.c:5607 Channel
> sofia/internal/1050 at 172.16.10.9 entering state [completed][200]
> EXECUTE sofia/internal/1050 at 172.16.10.9 sleep(1000)
> 2012-11-25 21:10:35.113025 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/1050 at 172.16.10.9 [BREAK]
> 2012-11-25 21:10:35.113025 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/1050 at 172.16.10.9 [BREAK]
> 2012-11-25 21:10:35.113025 [DEBUG] switch_core_session.c:976 Send signal
> sofia/internal/1050 at 172.16.10.9 [BREAK]
> 2012-11-25 21:10:35.133027 [DEBUG] sofia.c:5607 Channel
> sofia/internal/1050 at 172.16.10.9 entering state [ready][200]
> EXECUTE sofia/internal/1050 at 172.16.10.9 voicemail(check default)
> 2012-11-25 21:10:36.053026 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]
> 2012-11-25 21:10:36.253026 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]
> 2012-11-25 21:10:36.253026 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-enter_id.wav] (en:en)
> 2012-11-25 21:10:36.253026 [ERR] mod_sndfile.c:198 Error Opening File
> [/usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-enter_$
> 2012-11-25 21:10:36.353027 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]
> 2012-11-25 21:10:36.353027 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-goodbye.wav] (en:en)
> 2012-11-25 21:10:36.353027 [ERR] mod_sndfile.c:198 Error Opening File
> [/usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-goodby$
> 2012-11-25 21:10:36.453048 [NOTICE] switch_core_state_machine.c:262
> sofia/internal/1050 at 172.16.10.9 has executed the last dialplan i$
> 2012-11-25 21:10:36.453048 [DEBUG] switch_channel.c:2979
> (sofia/internal/1050 at 172.16.10.9) Callstate Change ACTIVE -> HANGUP
>
> Best regards
> Jakob Sundberg
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From steveayre at gmail.com  Tue Dec  4 19:46:59 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 4 Dec 2012 16:46:59 +0000
Subject: [Freeswitch-users] Different dtmf type for different gateways
 in single profile
In-Reply-To: <473CAD87-8625-4580-BEDC-08DE4475CE22@gmail.com>
References: <CACak_yxsejacKz6jAkdfdhdAV5tquAF4eBVR4a4yyC81kvuOvg@mail.gmail.com>
	<CAKzWOxXTDzOJ7Q0X4R+m06pG4R1w6kkkO7EA_7WzSbqO=crLZw@mail.gmail.com>
	<473CAD87-8625-4580-BEDC-08DE4475CE22@gmail.com>
Message-ID: <CAFiqYunGtJ4qbbCyO-L-ppdpRfQDPgMK_3uAR2na61_GkjnyJA@mail.gmail.com>

Use RFC2833, and only call start_dtmf on gateways that don't support
RFC2833.

>From dialplan you could test the SDP variables to see whether the endpoint
has negotiated RFC2833 and only call start_dtmf if it hasn't, or test a
variable that you set in the user directory / sofia gateway config.



On 4 December 2012 11:41, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:

>
> If I set to dtmf-type=none in profile, only inband works because of
> start_dtmf, If set dtmf-type=rfc2833, in band starts false positive
> catches. I may be doing something practically wrong here, so just wanted to
> know what options do we have to set it up correctly?
>
> Thanks,
> --
> Muhammad Naseer Bhatti
>
>
>
> On Dec 4, 2012, at 2:48 AM, Michael Collins <msc at freeswitch.org> wrote:
>
> How does it "break the rest of DTMF"?
> -MC
>
> On Mon, Dec 3, 2012 at 2:14 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com>wrote:
>
>> Is there a way to set different dtmf type for different gateways in same
>> profile? I am originating from esl, call is bridged with audio file.
>> (Dialer setup). Some of the providers send dtmf  inband and others rfc2833.
>> I am currently checking in my lua script for a custom param, if present
>> then execute start_dtmf to listen for inband dtmf. While this breaks the
>> rest of dtmf. Is there a better way to deal with this or do I have to end
>> up with different profiles each with different dtmf-type.
>>
>> --
>> Sent via a mobile 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
> http://www.ClueCon.com <http://www.cluecon.com/>
> http://www.OSTAG.org <http://www.ostag.org/>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue Dec  4 20:52:19 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Tue, 4 Dec 2012 20:52:19 +0300
Subject: [Freeswitch-users] Different dtmf type for different gateways
 in single profile
In-Reply-To: <CAFiqYunGtJ4qbbCyO-L-ppdpRfQDPgMK_3uAR2na61_GkjnyJA@mail.gmail.com>
References: <CACak_yxsejacKz6jAkdfdhdAV5tquAF4eBVR4a4yyC81kvuOvg@mail.gmail.com>
	<CAKzWOxXTDzOJ7Q0X4R+m06pG4R1w6kkkO7EA_7WzSbqO=crLZw@mail.gmail.com>
	<473CAD87-8625-4580-BEDC-08DE4475CE22@gmail.com>
	<CAFiqYunGtJ4qbbCyO-L-ppdpRfQDPgMK_3uAR2na61_GkjnyJA@mail.gmail.com>
Message-ID: <CACak_yy3RuP+WJ3Six70PqB+h-DZz3vw+Bfd5wOhKqQzF5+H0Q@mail.gmail.com>

What about info, it should be able to detect with dtmf-type already set to
rfc2833?

--
Sent via a mobile device.
On Dec 4, 2012 7:51 PM, "Steven Ayre" <steveayre at gmail.com> wrote:

> Use RFC2833, and only call start_dtmf on gateways that don't support
> RFC2833.
>
> From dialplan you could test the SDP variables to see whether the endpoint
> has negotiated RFC2833 and only call start_dtmf if it hasn't, or test a
> variable that you set in the user directory / sofia gateway config.
>
>
>
> On 4 December 2012 11:41, Muhammad Naseer Bhatti <nbhatti at gmail.com>wrote:
>
>>
>> If I set to dtmf-type=none in profile, only inband works because of
>> start_dtmf, If set dtmf-type=rfc2833, in band starts false positive
>> catches. I may be doing something practically wrong here, so just wanted to
>> know what options do we have to set it up correctly?
>>
>> Thanks,
>> --
>> Muhammad Naseer Bhatti
>>
>>
>>
>> On Dec 4, 2012, at 2:48 AM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> How does it "break the rest of DTMF"?
>> -MC
>>
>> On Mon, Dec 3, 2012 at 2:14 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com
>> > wrote:
>>
>>> Is there a way to set different dtmf type for different gateways in same
>>> profile? I am originating from esl, call is bridged with audio file.
>>> (Dialer setup). Some of the providers send dtmf  inband and others rfc2833.
>>> I am currently checking in my lua script for a custom param, if present
>>> then execute start_dtmf to listen for inband dtmf. While this breaks the
>>> rest of dtmf. Is there a better way to deal with this or do I have to end
>>> up with different profiles each with different dtmf-type.
>>>
>>> --
>>> Sent via a mobile 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
>> http://www.ClueCon.com <http://www.cluecon.com/>
>> http://www.OSTAG.org <http://www.ostag.org/>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 a.venugopan at mundio.com  Tue Dec  4 21:00:16 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Tue, 4 Dec 2012 18:00:16 +0000
Subject: [Freeswitch-users] sip registration
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2339A49@Mail-Kilo.squay.com>

Hi,

Currently we register authentication name as say '100' in sip registration, this comes to freeswitch and it will check in our DB for 100 and if its present then registrations would be successful.

freeswitch at internal> show registrations
reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
100,fsfailover.uk01.com,e4969067f9a8c098,sofia/internal/sip:100 at 192.168.2.234:5060;transport=udp,1354638871,192.168.2.234,5060,udp,squay-laptop-1.squay.com

I want to change this 100 to some e-mail address, so instead of 100 it will be something like 'ana at gmail.com'. Can we do this? While coming to freeswitch whether there would be any issues?

Regards,
Archana

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From steveayre at gmail.com  Tue Dec  4 21:20:16 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 4 Dec 2012 18:20:16 +0000
Subject: [Freeswitch-users] Different dtmf type for different gateways
 in single profile
In-Reply-To: <CACak_yy3RuP+WJ3Six70PqB+h-DZz3vw+Bfd5wOhKqQzF5+H0Q@mail.gmail.com>
References: <CACak_yxsejacKz6jAkdfdhdAV5tquAF4eBVR4a4yyC81kvuOvg@mail.gmail.com>
	<CAKzWOxXTDzOJ7Q0X4R+m06pG4R1w6kkkO7EA_7WzSbqO=crLZw@mail.gmail.com>
	<473CAD87-8625-4580-BEDC-08DE4475CE22@gmail.com>
	<CAFiqYunGtJ4qbbCyO-L-ppdpRfQDPgMK_3uAR2na61_GkjnyJA@mail.gmail.com>
	<CACak_yy3RuP+WJ3Six70PqB+h-DZz3vw+Bfd5wOhKqQzF5+H0Q@mail.gmail.com>
Message-ID: <CAFiqYum0sFpYuffUwURooEqrVSwvTsc0e253NyfNpK2cbDN=eA@mail.gmail.com>

SIP Info doesn't show in SDP so might be harder to detect from dialplan,
I'm not sure if there's anything set that'd indicate it was in use...

You can see whether there are any useful channel variables by running the
info app or checking the xml cdr for a test call though.


On 4 December 2012 17:52, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:

> What about info, it should be able to detect with dtmf-type already set to
> rfc2833?
>
> --
> Sent via a mobile device.
> On Dec 4, 2012 7:51 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>
>> Use RFC2833, and only call start_dtmf on gateways that don't support
>> RFC2833.
>>
>> From dialplan you could test the SDP variables to see whether the
>> endpoint has negotiated RFC2833 and only call start_dtmf if it hasn't, or
>> test a variable that you set in the user directory / sofia gateway config.
>>
>>
>>
>> On 4 December 2012 11:41, Muhammad Naseer Bhatti <nbhatti at gmail.com>wrote:
>>
>>>
>>> If I set to dtmf-type=none in profile, only inband works because of
>>> start_dtmf, If set dtmf-type=rfc2833, in band starts false positive
>>> catches. I may be doing something practically wrong here, so just wanted to
>>> know what options do we have to set it up correctly?
>>>
>>> Thanks,
>>> --
>>> Muhammad Naseer Bhatti
>>>
>>>
>>>
>>> On Dec 4, 2012, at 2:48 AM, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>> How does it "break the rest of DTMF"?
>>> -MC
>>>
>>> On Mon, Dec 3, 2012 at 2:14 PM, Muhammad Naseer Bhatti <
>>> nbhatti at gmail.com> wrote:
>>>
>>>> Is there a way to set different dtmf type for different gateways in
>>>> same profile? I am originating from esl, call is bridged with audio file.
>>>> (Dialer setup). Some of the providers send dtmf  inband and others rfc2833.
>>>> I am currently checking in my lua script for a custom param, if present
>>>> then execute start_dtmf to listen for inband dtmf. While this breaks the
>>>> rest of dtmf. Is there a better way to deal with this or do I have to end
>>>> up with different profiles each with different dtmf-type.
>>>>
>>>> --
>>>> Sent via a mobile 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
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Michael S Collins
>>> Twitter: @mercutioviz
>>> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
>>> http://www.ClueCon.com <http://www.cluecon.com/>
>>> http://www.OSTAG.org <http://www.ostag.org/>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec  4 22:50:48 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 4 Dec 2012 19:50:48 +0000
Subject: [Freeswitch-users] sip registration
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2339A49@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2339A49@Mail-Kilo.squay.com>
Message-ID: <CAFiqYu=cpMkTxdE99smQBM6NgitNQ5ZpePwfSYr_R23dpj7Hng@mail.gmail.com>

You can have a user 'ana' in the domain 'gmail.com'. Though using someone
else's domain as local in your FS setup may not be a good idea.

You can't have a @ in the username itself (per the SIP standard, not
limited to FreeSWITCH).


On 4 December 2012 18:00, Archana Venugopan <a.venugopan at mundio.com> wrote:

>  Hi,****
>
> ** **
>
> Currently we register authentication name as say ?100? in sip
> registration, this comes to freeswitch and it will check in our DB for 100
> and if its present then registrations would be successful. ****
>
> ** **
>
> freeswitch at internal> show registrations****
>
>
> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
> ****
>
> 100,fsfailover.uk01.com
> ,e4969067f9a8c098,sofia/internal/sip:100 at 192.168.2.234:5060
> ;transport=udp,1354638871,192.168.2.234,5060,udp,squay-laptop-1.squay.com*
> ***
>
> ** **
>
> I want to change this 100 to some e-mail address, so instead of 100 it
> will be something like ?ana at gmail.com?. Can we do this? While coming to
> freeswitch whether there would be any issues?****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 ben122uk at gmail.com  Tue Dec  4 23:40:49 2012
From: ben122uk at gmail.com (Ben)
Date: Tue, 4 Dec 2012 20:40:49 -0000
Subject: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes
In-Reply-To: <0C442564-683E-4271-A76B-6D5FF3EB2797@freeswitch.org>
References: <CADaqzFMD+Y8yc_qWNtp0drmceSU6EDogN7F6TyhtSDLbT2JeWA@mail.gmail.com>	<6036422a0e8b5982ed7af8823015bb38@www.webmail.co.za>	<CADaqzFOGTva6mZBgewCKc8gdHmGdDcMn-AMPPnehHTNA4n-H_w@mail.gmail.com>	<002d01cdd16b$492b9030$db82b090$@gmail.com>
	<0C442564-683E-4271-A76B-6D5FF3EB2797@freeswitch.org>
Message-ID: <004801cdd25f$a5e33770$f1a9a650$@gmail.com>

Hi Brian,

I looked at the timestamps for both RTP streams that are being sent by each
client, it appears that the timestamp increments by a value of 400 for each
packet, for both streams.  I can also see this in FS for each stream - 

2012-12-03 15:51:03.374955 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
[G729:18:8000:50:8000]/[G729:18:8000:50:8000]
2012-12-03 15:51:03.374955 [DEBUG] sofia_glue.c:3077 Set Codec
sofia/internal/1002 at 192.168.17.154 G729/8000 50 ms 400 samples 8000 bits

Looking at the Freeswitch log, I'm guessing that the timestamps set by the
client correlate to the sample rate set by Freeswitch?  Or am I not on the
right track with what you were suggesting?

I've also noticed that when using 50ms ptimes, one of the RTP streams from a
client is not very 'steady'.  One stream seems acceptable, with only one or
two packets straying to 60ms, and jitter hovers around 2-3ms.  However the
other stream really struggles, the delta(ms) values in wireshark are all
over the place.  This is the summary of a 20 second RTP stream in wireshark
-

Max delta = 120.95 ms at packet no. 1250 
Max jitter = 22.00 ms. Mean jitter = 18.12 ms.
Max skew = 65.24 ms.

This seems odd since the test I did was all internal on the same LAN.  I'll
try and track down the cause tomorrow by switching around some kit so that
it cuts out any potential problems.  

Are there tolerance levels in Freeswitch for RTP streams that are affected
during transit?  If so, how does Freeswitch react when one of these
tolerance levels is exceeded?  My FS server seems to just hang during call
setup, and no RTP stream is sent out the server to the clients.

Thanks for the help, I'll let you know how I get on tomorrow with some
further tests.

Regards,

Ben	


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 04 December 2012 15:11
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes

PCAP's will show you.... I suspect something is telling a lie about the
timestamps.  Very common.
--
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
ISN: 410*543





On Dec 3, 2012, at 9:31 AM, Ben <ben122uk at gmail.com> wrote:

> I tried setting the {passthru_ptime_mismatch=true} variable in the
dialplan, but no joy unfortunately.  I'm also fairly sure that the codecs
are matched on both call legs, so shouldn't be having that sort of issue.
>  
> I'm going to gather up some evidence and console logs/traces etc so that I
can put something together for a bug report on Jira.  Is there anything else
that someone can suggest I try before it goes on Jira?  Is anyone able to
re-produce the same problem?  Admittedly, I'm not on the latest GIT, but I'm
on a recent (last couple of months) stable release.
>  
> Thanks in advance for any assistance!
>  
> Regards,
>  
> Ben
>  


_________________________________________________________________________
Professional FreeSWITCH Consulting 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



From bpriddy at bryantschools.org  Tue Dec  4 23:16:36 2012
From: bpriddy at bryantschools.org (Blake Priddy)
Date: Tue, 4 Dec 2012 14:16:36 -0600
Subject: [Freeswitch-users] 10 second delay
Message-ID: <CADK-BjuB7ugJMPLSYByJPa0jQmnRywvWLgPdwDUCkKF2e7eFbA@mail.gmail.com>

I have some secretaries here at our school district that when
they receive a call they have to give their spill 2-3 times before the
party on the other end will hear them.. I have the Epygi Gateway and
FreeSwitch box in the same network. Any thoughts would be greatly
appreciated! :)


-- 

*Blakelund Priddy*
Network Systems Engineer
Bryant Public School District
Bryant, Arkansas 72022
http://www.bryantschools.org
p 501-653-5038
f 501-847-5656
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From kris at kriskinc.com  Wed Dec  5 00:00:17 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 4 Dec 2012 16:00:17 -0500
Subject: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes
In-Reply-To: <004801cdd25f$a5e33770$f1a9a650$@gmail.com>
References: <CADaqzFMD+Y8yc_qWNtp0drmceSU6EDogN7F6TyhtSDLbT2JeWA@mail.gmail.com>
	<6036422a0e8b5982ed7af8823015bb38@www.webmail.co.za>
	<CADaqzFOGTva6mZBgewCKc8gdHmGdDcMn-AMPPnehHTNA4n-H_w@mail.gmail.com>
	<002d01cdd16b$492b9030$db82b090$@gmail.com>
	<0C442564-683E-4271-A76B-6D5FF3EB2797@freeswitch.org>
	<004801cdd25f$a5e33770$f1a9a650$@gmail.com>
Message-ID: <CAKDfjgdBT-0TbH3OAwYKkdaN5WSFPwHr8jM+-pn9To98FamWDg@mail.gmail.com>

RTP timestamps = ptime x 8, so:

PTIME:timestamp increment
10:80
20:160
30:240
40:320
50:400

  Your endpoints seem to be writing their RTP timestamps correctly.
For the endpoint that's all over the place, check the hardware.  I've
seen bad DSPs, etc cause strange RTP skew before.

On Tue, Dec 4, 2012 at 3:40 PM, Ben <ben122uk at gmail.com> wrote:
> Hi Brian,
>
> I looked at the timestamps for both RTP streams that are being sent by each
> client, it appears that the timestamp increments by a value of 400 for each
> packet, for both streams.  I can also see this in FS for each stream -
>
> 2012-12-03 15:51:03.374955 [DEBUG] sofia_glue.c:5094 Audio Codec Compare
> [G729:18:8000:50:8000]/[G729:18:8000:50:8000]
> 2012-12-03 15:51:03.374955 [DEBUG] sofia_glue.c:3077 Set Codec
> sofia/internal/1002 at 192.168.17.154 G729/8000 50 ms 400 samples 8000 bits
>
> Looking at the Freeswitch log, I'm guessing that the timestamps set by the
> client correlate to the sample rate set by Freeswitch?  Or am I not on the
> right track with what you were suggesting?
>
> I've also noticed that when using 50ms ptimes, one of the RTP streams from a
> client is not very 'steady'.  One stream seems acceptable, with only one or
> two packets straying to 60ms, and jitter hovers around 2-3ms.  However the
> other stream really struggles, the delta(ms) values in wireshark are all
> over the place.  This is the summary of a 20 second RTP stream in wireshark
> -
>
> Max delta = 120.95 ms at packet no. 1250
> Max jitter = 22.00 ms. Mean jitter = 18.12 ms.
> Max skew = 65.24 ms.
>
> This seems odd since the test I did was all internal on the same LAN.  I'll
> try and track down the cause tomorrow by switching around some kit so that
> it cuts out any potential problems.
>
> Are there tolerance levels in Freeswitch for RTP streams that are affected
> during transit?  If so, how does Freeswitch react when one of these
> tolerance levels is exceeded?  My FS server seems to just hang during call
> setup, and no RTP stream is sent out the server to the clients.
>
> Thanks for the help, I'll let you know how I get on tomorrow with some
> further tests.
>
> Regards,
>
> Ben
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: 04 December 2012 15:11
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes
>
> PCAP's will show you.... I suspect something is telling a lie about the
> timestamps.  Very common.
> --
> 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
> ISN: 410*543
>
>
>
>
>
> On Dec 3, 2012, at 9:31 AM, Ben <ben122uk at gmail.com> wrote:
>
>> I tried setting the {passthru_ptime_mismatch=true} variable in the
> dialplan, but no joy unfortunately.  I'm also fairly sure that the codecs
> are matched on both call legs, so shouldn't be having that sort of issue.
>>
>> I'm going to gather up some evidence and console logs/traces etc so that I
> can put something together for a bug report on Jira.  Is there anything else
> that someone can suggest I try before it goes on Jira?  Is anyone able to
> re-produce the same problem?  Admittedly, I'm not on the latest GIT, but I'm
> on a recent (last couple of months) stable release.
>>
>> Thanks in advance for any assistance!
>>
>> Regards,
>>
>> Ben
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 freeswitch at orresta.no-ip.com  Wed Dec  5 00:07:18 2012
From: freeswitch at orresta.no-ip.com (Jakob)
Date: Tue, 04 Dec 2012 22:07:18 +0100
Subject: [Freeswitch-users] mod_sndfile can't find files
In-Reply-To: <CAKzWOxX1vC6JUO2D7rcxQcV9by+OHpatCggc7iLfkuVquVRiNg@mail.gmail.com>
References: <50BDD37B.80308@orresta.no-ip.com>
	<CAKzWOxX1vC6JUO2D7rcxQcV9by+OHpatCggc7iLfkuVquVRiNg@mail.gmail.com>
Message-ID: <50BE6606.2030707@orresta.no-ip.com>

I found the problem in vars.xml. A previous installation were in a 
nonstandard place and the path is now correct and problem solved.

-thanks
12/04/12 17:34, Michael Collins skrev:
> I would rebuild from a clean git checkout. It looks like it's trying 
> to find a wonky filename. Notice the dollar sign at the end of the 
> file name. If it re-occurs on a clean git checkout then go ahead and 
> open a Jira.
>
> -MC
>
> On Tue, Dec 4, 2012 at 2:42 AM, Jakob <freeswitch at orresta.no-ip.com 
> <mailto:freeswitch at orresta.no-ip.com>> wrote:
>
>     Hi,
>
>     I did a clean install from git master and keep getting file not
>     found in
>     the logs. This was noticed in late nov on a fedora 17 git install
>     but is
>     still present in a recent (as of today dec 4:th) debian squeeze git
>     master system.
>
>     This was tested using sipdroid on the same lan.
>
>     Relevant part of the log follows:
>
>     2012-11-25 21:10:35.053024 [DEBUG] switch_core_session.c:830 Send
>     signal
>     sofia/internal/1050 at 172.16.10.9 <mailto:1050 at 172.16.10.9> [BREAK]
>     2012-11-25 21:10:35.053024 [DEBUG] switch_channel.c:3380
>     (sofia/internal/1050 at 172.16.10.9 <mailto:1050 at 172.16.10.9>)
>     Callstate Change RINGING -> ACTIVE
>     2012-11-25 21:10:35.053024 [DEBUG] switch_core_session.c:976 Send
>     signal
>     sofia/internal/1050 at 172.16.10.9 <mailto:1050 at 172.16.10.9> [BREAK]
>     2012-11-25 21:10:35.053024 [NOTICE] mod_dptools.c:1176 Channel
>     [sofia/internal/1050 at 172.16.10.9 <mailto:1050 at 172.16.10.9>] has
>     been answered
>     2012-11-25 21:10:35.053024 [DEBUG] sofia.c:5607 Channel
>     sofia/internal/1050 at 172.16.10.9 <mailto:1050 at 172.16.10.9> entering
>     state [completed][200]
>     EXECUTE sofia/internal/1050 at 172.16.10.9 <mailto:1050 at 172.16.10.9>
>     sleep(1000)
>     2012-11-25 21:10:35.113025 [DEBUG] switch_core_session.c:976 Send
>     signal
>     sofia/internal/1050 at 172.16.10.9 <mailto:1050 at 172.16.10.9> [BREAK]
>     2012-11-25 21:10:35.113025 [DEBUG] switch_core_session.c:976 Send
>     signal
>     sofia/internal/1050 at 172.16.10.9 <mailto:1050 at 172.16.10.9> [BREAK]
>     2012-11-25 21:10:35.113025 [DEBUG] switch_core_session.c:976 Send
>     signal
>     sofia/internal/1050 at 172.16.10.9 <mailto:1050 at 172.16.10.9> [BREAK]
>     2012-11-25 21:10:35.133027 [DEBUG] sofia.c:5607 Channel
>     sofia/internal/1050 at 172.16.10.9 <mailto:1050 at 172.16.10.9> entering
>     state [ready][200]
>     EXECUTE sofia/internal/1050 at 172.16.10.9 <mailto:1050 at 172.16.10.9>
>     voicemail(check default)
>     2012-11-25 21:10:36.053026 [DEBUG] switch_ivr_play_say.c:67 No
>     language
>     specified - Using [en]
>     2012-11-25 21:10:36.253026 [DEBUG] switch_ivr_play_say.c:67 No
>     language
>     specified - Using [en]
>     2012-11-25 21:10:36.253026 [DEBUG] switch_ivr_play_say.c:244 Handle
>     play-file:[voicemail/vm-enter_id.wav] (en:en)
>     2012-11-25 21:10:36.253026 [ERR] mod_sndfile.c:198 Error Opening File
>     [/usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-enter_$
>     2012-11-25 21:10:36.353027 [DEBUG] switch_ivr_play_say.c:67 No
>     language
>     specified - Using [en]
>     2012-11-25 21:10:36.353027 [DEBUG] switch_ivr_play_say.c:244 Handle
>     play-file:[voicemail/vm-goodbye.wav] (en:en)
>     2012-11-25 21:10:36.353027 [ERR] mod_sndfile.c:198 Error Opening File
>     [/usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-goodby$
>     2012-11-25 21:10:36.453048 [NOTICE] switch_core_state_machine.c:262
>     sofia/internal/1050 at 172.16.10.9 <mailto:1050 at 172.16.10.9> has
>     executed the last dialplan i$
>     2012-11-25 21:10:36.453048 [DEBUG] switch_channel.c:2979
>     (sofia/internal/1050 at 172.16.10.9 <mailto:1050 at 172.16.10.9>)
>     Callstate Change ACTIVE -> HANGUP
>
>     Best regards
>     Jakob Sundberg
>
>     _________________________________________________________________________
>     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
>
>
>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 ben122uk at gmail.com  Wed Dec  5 00:32:37 2012
From: ben122uk at gmail.com (Ben)
Date: Tue, 4 Dec 2012 21:32:37 -0000
Subject: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes
In-Reply-To: <CAKDfjgdBT-0TbH3OAwYKkdaN5WSFPwHr8jM+-pn9To98FamWDg@mail.gmail.com>
References: <CADaqzFMD+Y8yc_qWNtp0drmceSU6EDogN7F6TyhtSDLbT2JeWA@mail.gmail.com>	<6036422a0e8b5982ed7af8823015bb38@www.webmail.co.za>	<CADaqzFOGTva6mZBgewCKc8gdHmGdDcMn-AMPPnehHTNA4n-H_w@mail.gmail.com>	<002d01cdd16b$492b9030$db82b090$@gmail.com>	<0C442564-683E-4271-A76B-6D5FF3EB2797@freeswitch.org>	<004801cdd25f$a5e33770$f1a9a650$@gmail.com>
	<CAKDfjgdBT-0TbH3OAwYKkdaN5WSFPwHr8jM+-pn9To98FamWDg@mail.gmail.com>
Message-ID: <005e01cdd266$e24062e0$a6c128a0$@gmail.com>

Thanks for the info, I suspected that it should always add up to 8000
samples per second :-)

Yes hardware is my next area of investigation.  Both endpoints are iPhones,
I'll be checking tomorrow to see if these are the cause of the dodgy RTP
stream.

Cheers,

Ben
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: 04 December 2012 21:00
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes

RTP timestamps = ptime x 8, so:

PTIME:timestamp increment
10:80
20:160
30:240
40:320
50:400

  Your endpoints seem to be writing their RTP timestamps correctly.
For the endpoint that's all over the place, check the hardware.  I've seen
bad DSPs, etc cause strange RTP skew before.

On Tue, Dec 4, 2012 at 3:40 PM, Ben <ben122uk at gmail.com> wrote:
> Hi Brian,
>
> I looked at the timestamps for both RTP streams that are being sent by 
> each client, it appears that the timestamp increments by a value of 
> 400 for each packet, for both streams.  I can also see this in FS for 
> each stream -
>
> 2012-12-03 15:51:03.374955 [DEBUG] sofia_glue.c:5094 Audio Codec 
> Compare [G729:18:8000:50:8000]/[G729:18:8000:50:8000]
> 2012-12-03 15:51:03.374955 [DEBUG] sofia_glue.c:3077 Set Codec
> sofia/internal/1002 at 192.168.17.154 G729/8000 50 ms 400 samples 8000 
> bits
>
> Looking at the Freeswitch log, I'm guessing that the timestamps set by 
> the client correlate to the sample rate set by Freeswitch?  Or am I 
> not on the right track with what you were suggesting?
>
> I've also noticed that when using 50ms ptimes, one of the RTP streams 
> from a client is not very 'steady'.  One stream seems acceptable, with 
> only one or two packets straying to 60ms, and jitter hovers around 
> 2-3ms.  However the other stream really struggles, the delta(ms) 
> values in wireshark are all over the place.  This is the summary of a 
> 20 second RTP stream in wireshark
> -
>
> Max delta = 120.95 ms at packet no. 1250 Max jitter = 22.00 ms. Mean 
> jitter = 18.12 ms.
> Max skew = 65.24 ms.
>
> This seems odd since the test I did was all internal on the same LAN.  
> I'll try and track down the cause tomorrow by switching around some 
> kit so that it cuts out any potential problems.
>
> Are there tolerance levels in Freeswitch for RTP streams that are 
> affected during transit?  If so, how does Freeswitch react when one of 
> these tolerance levels is exceeded?  My FS server seems to just hang 
> during call setup, and no RTP stream is sent out the server to the
clients.
>
> Thanks for the help, I'll let you know how I get on tomorrow with some 
> further tests.
>
> Regards,
>
> Ben
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> Brian West
> Sent: 04 December 2012 15:11
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Doubts over G729 passthru for 50ms 
> ptimes
>
> PCAP's will show you.... I suspect something is telling a lie about 
> the timestamps.  Very common.
> --
> 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
> ISN: 410*543
>
>
>
>
>
> On Dec 3, 2012, at 9:31 AM, Ben <ben122uk at gmail.com> wrote:
>
>> I tried setting the {passthru_ptime_mismatch=true} variable in the
> dialplan, but no joy unfortunately.  I'm also fairly sure that the 
> codecs are matched on both call legs, so shouldn't be having that sort of
issue.
>>
>> I'm going to gather up some evidence and console logs/traces etc so 
>> that I
> can put something together for a bug report on Jira.  Is there 
> anything else that someone can suggest I try before it goes on Jira?  
> Is anyone able to re-produce the same problem?  Admittedly, I'm not on 
> the latest GIT, but I'm on a recent (last couple of months) stable
release.
>>
>> Thanks in advance for any assistance!
>>
>> Regards,
>>
>> Ben
>>
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting 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-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
>
> FreeSWITCH-users 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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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  Wed Dec  5 00:34:51 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 4 Dec 2012 16:34:51 -0500
Subject: [Freeswitch-users] Call disconnect upon re-invite - FS tears
	down call
In-Reply-To: <CAMbW9o+9SpGpTzj5JVQajx-UO3WER4gxqCd+CG0aydOLr2UPUQ@mail.gmail.com>
References: <017f01cdb138$e00325c0$a0097140$@com>
	<CAFVUaV8TwiNqean_gs1EbNPRXjE=Cv+8Us7-c3yWG_n00NduGg@mail.gmail.com>
	<15AD3C94-1640-486D-B1C5-B961EB30F35B@freeswitch.org>
	<CAMbW9o+9SpGpTzj5JVQajx-UO3WER4gxqCd+CG0aydOLr2UPUQ@mail.gmail.com>
Message-ID: <EB6C0153-B6EC-4B42-A3E8-893324C07BAB@endigotech.com>



Sent from my iPhone

On Dec 4, 2012, at 10:50 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk> wrote:

> I was half expecting the meme image that goes with that...
> 
> On 4 December 2012 15:15, Brian West <brian at freeswitch.org> wrote:
>> Y U NO FILE JIRA?
>> --
>> 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
>> ISN: 410*543
>> 
>> 
>> 
>> 
>> 
>> On Oct 23, 2012, at 11:32 AM, Yiftach Golan <yiftah at choochee.com> wrote:
>> 
>> >
>> > You did not send the initial invite in the Jira so I do not know if the initial request constructed correctly
>> > but judging by the code it looks like your request is still pending when the second invite arrives
>> >
>> > Thanks,
>> > Yiftach.
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
> 
> 
> 
> -- 
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec  5 00:59:09 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Dec 2012 13:59:09 -0800
Subject: [Freeswitch-users] Call disconnect upon re-invite - FS tears
	down call
In-Reply-To: <EB6C0153-B6EC-4B42-A3E8-893324C07BAB@endigotech.com>
References: <017f01cdb138$e00325c0$a0097140$@com>
	<CAFVUaV8TwiNqean_gs1EbNPRXjE=Cv+8Us7-c3yWG_n00NduGg@mail.gmail.com>
	<15AD3C94-1640-486D-B1C5-B961EB30F35B@freeswitch.org>
	<CAMbW9o+9SpGpTzj5JVQajx-UO3WER4gxqCd+CG0aydOLr2UPUQ@mail.gmail.com>
	<EB6C0153-B6EC-4B42-A3E8-893324C07BAB@endigotech.com>
Message-ID: <CAKzWOxUg_dvt4V84q+DVTiqJV0F2mCv=8U-z5qheC7CX-oMiKQ@mail.gmail.com>

Okay, this is now on the wiki:
http://wiki.freeswitch.org/wiki/File:Yunfj.jpeg
We just need a good place for it. :P
-MC

On Tue, Dec 4, 2012 at 1:34 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> [image: image.jpeg]
>
> Sent from my iPhone
>
> On Dec 4, 2012, at 10:50 AM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
> I was half expecting the meme image that goes with that...
>
> On 4 December 2012 15:15, Brian West <brian at freeswitch.org> wrote:
>
>> Y U NO FILE JIRA?
>> --
>> 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
>> ISN: 410*543
>>
>>
>>
>>
>>
>> On Oct 23, 2012, at 11:32 AM, Yiftach Golan <yiftah at choochee.com> wrote:
>>
>> >
>> > You did not send the initial invite in the Jira so I do not know if the
>> initial request constructed correctly
>> > but judging by the code it looks like your request is still pending
>> when the second invite arrives
>> >
>> > Thanks,
>> > Yiftach.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>
>
>
> --
> *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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From freeswitch at orresta.no-ip.com  Wed Dec  5 02:38:40 2012
From: freeswitch at orresta.no-ip.com (Jakob)
Date: Wed, 05 Dec 2012 00:38:40 +0100
Subject: [Freeswitch-users] Single sndcard multiple extensions?
Message-ID: <50BE8980.3050900@orresta.no-ip.com>

Hi,

I'm trying to get right and left channels from a soundcard into separate 
alsa or portaudio endpoints.

right channel into ext 3001 and left channel into ext 3002

Any hints?

Regards Jakob


From msc at freeswitch.org  Wed Dec  5 02:50:26 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Dec 2012 15:50:26 -0800
Subject: [Freeswitch-users] Single sndcard multiple extensions?
In-Reply-To: <50BE8980.3050900@orresta.no-ip.com>
References: <50BE8980.3050900@orresta.no-ip.com>
Message-ID: <CAKzWOxXiEn+Ox=eJYfj-6ooEse48x8Q7YPsyi=oU_4YoN4KvNA@mail.gmail.com>

As far as I know mod_portaudio does not support multiple channels...
-MC

On Tue, Dec 4, 2012 at 3:38 PM, Jakob <freeswitch at orresta.no-ip.com> wrote:

> Hi,
>
> I'm trying to get right and left channels from a soundcard into separate
> alsa or portaudio endpoints.
>
> right channel into ext 3001 and left channel into ext 3002
>
> Any hints?
>
> Regards Jakob
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec  5 03:29:51 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 Dec 2012 16:29:51 -0800
Subject: [Freeswitch-users] Call for assistance: splicing/editing MP4 files
Message-ID: <CAKzWOxWonrtAvcDXTwtQuLvqHeQAMTFcTcH1AOmC1wnHrpqGEA@mail.gmail.com>

Hi folks!

If you have the time, equipment, and bandwidth to assist with some
video/mp4 editing please contact me off list.

Thanks!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From richardrcruzc at gmail.com  Wed Dec  5 03:44:58 2012
From: richardrcruzc at gmail.com (Richard Cruz)
Date: Tue, 4 Dec 2012 19:44:58 -0500
Subject: [Freeswitch-users] Call for assistance: splicing/editing MP4
	files
In-Reply-To: <CAKzWOxWonrtAvcDXTwtQuLvqHeQAMTFcTcH1AOmC1wnHrpqGEA@mail.gmail.com>
References: <CAKzWOxWonrtAvcDXTwtQuLvqHeQAMTFcTcH1AOmC1wnHrpqGEA@mail.gmail.com>
Message-ID: <CAMJrzFBfPWustUpNdCpW4ud6sUBce+AbTfqH=WDnaFYvDDYo1A@mail.gmail.com>

hi !


On Tue, Dec 4, 2012 at 7:29 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hi folks!
>
> If you have the time, equipment, and bandwidth to assist with some
> video/mp4 editing please contact me off list.
>
> Thanks!
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Richard Cruz
678.394-6400
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From luis.daniel.lucio at gmail.com  Wed Dec  5 07:33:06 2012
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Tue, 4 Dec 2012 23:33:06 -0500
Subject: [Freeswitch-users] OT: what carrier do you recommend me?
Message-ID: <CAFLo2QzkBvmCxjUnviFsvVbm2WcPPL6PZROr7f_D-ho-LZUuBA@mail.gmail.com>

Helo every one,

In your experience what carrier do you recommend me to route calls,
I looking carriers for next countries:

- China
- USA/CA
- Mexico

I am using now Flowroute, but it fails sometimes and it is a little expensive.

Regards,

LD


From jkomar at jbox.ca  Wed Dec  5 07:43:36 2012
From: jkomar at jbox.ca (Komar, Jason)
Date: Tue, 4 Dec 2012 21:43:36 -0700
Subject: [Freeswitch-users] OT: what carrier do you recommend me?
In-Reply-To: <CAFLo2QzkBvmCxjUnviFsvVbm2WcPPL6PZROr7f_D-ho-LZUuBA@mail.gmail.com>
References: <CAFLo2QzkBvmCxjUnviFsvVbm2WcPPL6PZROr7f_D-ho-LZUuBA@mail.gmail.com>
Message-ID: <CA+zebiYa0uv=x4rd-Tdsm7Amp+4J-xwQB67RbqSen+bCmJWYtg@mail.gmail.com>

I use voip.ms in Canada. They have been good.


On Tue, Dec 4, 2012 at 9:33 PM, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> Helo every one,
>
> In your experience what carrier do you recommend me to route calls,
> I looking carriers for next countries:
>
> - China
> - USA/CA
> - Mexico
>
> I am using now Flowroute, but it fails sometimes and it is a little
> expensive.
>
> Regards,
>
> LD
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 krice at freeswitch.org  Wed Dec  5 07:53:24 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 04 Dec 2012 22:53:24 -0600
Subject: [Freeswitch-users] OT: what carrier do you recommend me?
In-Reply-To: <CA+zebiYa0uv=x4rd-Tdsm7Amp+4J-xwQB67RbqSen+bCmJWYtg@mail.gmail.com>
Message-ID: <CCE42F64.615A9%krice@freeswitch.org>

Whats wrong with FlowRoute? They are pretty good... You can also checkout
voicenetwork.ca they are FreeSWITCH supporters...

If you want something cheaper then FlowRoute tho, you really need to have
the Volume to support such things...

K



On 12/4/12 10:43 PM, "Komar, Jason" <jkomar at jbox.ca> wrote:

> I use voip.ms <http://voip.ms>  in Canada. They have been good.
> 
> 
> On Tue, Dec 4, 2012 at 9:33 PM, Luis Daniel Lucio Quiroz
> <luis.daniel.lucio at gmail.com> wrote:
>> Helo every one,
>> 
>> In your experience what carrier do you recommend me to route calls,
>> I looking carriers for next countries:
>> 
>> - China
>> - USA/CA
>> - Mexico
>> 
>> I am using now Flowroute, but it fails sometimes and it is a little
>> expensive.
>> 
>> Regards,
>> 
>> LD
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From fs-list at communicatefreely.net  Wed Dec  5 08:01:11 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 05 Dec 2012 00:01:11 -0500
Subject: [Freeswitch-users] 10 second delay
In-Reply-To: <CADK-BjuB7ugJMPLSYByJPa0jQmnRywvWLgPdwDUCkKF2e7eFbA@mail.gmail.com>
References: <CADK-BjuB7ugJMPLSYByJPa0jQmnRywvWLgPdwDUCkKF2e7eFbA@mail.gmail.com>
Message-ID: <50BED517.8060900@communicatefreely.net>

My best advice would be to to a packet capture, or turn on sip traces between the gateway, 
  FS, and the phones, then try a test.

When you look through the call, take a look at the difference in the time stamps.

You should see a progress or a ringing back from the phone, then a 200 OK when they pick up.

After the call is picked up, see if there is additional traffic required to setup media, 
or any other entries in the log that might suggest that some additional actions are 
happening after the answer.

Also, if you can capture with wireshark, you can play back the media stream.  Check to see 
if the first hello is there coming out of the phone, and then going to the gateway. 
Wireshark will put it all in a time line perspective for you, so you can determine if this 
is a problem with the phone, your configuration, or the gateway.

Are you using late negotiation, bypass media, or any other options like that?

-Tim

Blake Priddy wrote:
> I have some secretaries here at our school district that when 
> they receive a call they have to give their spill 2-3 times before the 
> party on the other end will hear them.. I have the Epygi Gateway and 
> FreeSwitch box in the same network. Any thoughts would be greatly 
> appreciated! :)
> 
> 
> -- 
> 
> *Blakelund Priddy*
> Network Systems Engineer
> Bryant Public School District
> Bryant, Arkansas 72022
> http://www.bryantschools.org <http://www.bryantschools.org/>
> p 501-653-5038
> f 501-847-5656
> 
> 
> ------------------------------------------------------------------------
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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



From boris at tagnet.ru  Wed Dec  5 09:37:23 2012
From: boris at tagnet.ru (Boris Kovalenko)
Date: Wed, 05 Dec 2012 12:37:23 +0600
Subject: [Freeswitch-users] 10 second delay
In-Reply-To: <50BED517.8060900@communicatefreely.net>
References: <CADK-BjuB7ugJMPLSYByJPa0jQmnRywvWLgPdwDUCkKF2e7eFbA@mail.gmail.com>
	<50BED517.8060900@communicatefreely.net>
Message-ID: <50BEEBA3.9040006@tagnet.ru>

Hello!

What version of FS are You useing? I have the same troubles with 1.2.5 
branch. 1.2.3 works fine.
> My best advice would be to to a packet capture, or turn on sip traces between the gateway,
>    FS, and the phones, then try a test.
>
> When you look through the call, take a look at the difference in the time stamps.
>
> You should see a progress or a ringing back from the phone, then a 200 OK when they pick up.
>
> After the call is picked up, see if there is additional traffic required to setup media,
> or any other entries in the log that might suggest that some additional actions are
> happening after the answer.
>
> Also, if you can capture with wireshark, you can play back the media stream.  Check to see
> if the first hello is there coming out of the phone, and then going to the gateway.
> Wireshark will put it all in a time line perspective for you, so you can determine if this
> is a problem with the phone, your configuration, or the gateway.
>
> Are you using late negotiation, bypass media, or any other options like that?
>
> -Tim
>
> Blake Priddy wrote:
>> I have some secretaries here at our school district that when
>> they receive a call they have to give their spill 2-3 times before the
>> party on the other end will hear them.. I have the Epygi Gateway and
>> FreeSwitch box in the same network. Any thoughts would be greatly
>> appreciated! :)
>>
>>
>> -- 
>>
>> *Blakelund Priddy*
>> Network Systems Engineer
>> Bryant Public School District
>> Bryant, Arkansas 72022
>> http://www.bryantschools.org <http://www.bryantschools.org/>
>> p 501-653-5038
>> f 501-847-5656
>>
>>
>> ------------------------------------------------------------------------
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005
   http://www.tagnet.ru



From scobei001 at gmail.com  Wed Dec  5 04:37:41 2012
From: scobei001 at gmail.com (Scott Beil)
Date: Tue, 04 Dec 2012 19:37:41 -0600
Subject: [Freeswitch-users] Displacing ringback audio during bridge - only
 first displacement heard
Message-ID: <50BEA565.1070508@gmail.com>

I am attempting to interrupt the ringback audio at regular intervals 
with announcements while a bridge is in progress. Everything is going 
fine - the far end is ringing, the ringback audio is playing, the first 
displacement is heard, but subsequent displacements are not.

I am using an outbound ESL connection, FreeSWITCH version 1.2.3.

First, the ringback audio is set:

         esl_execute(handle, "set", 
"ringback=file_string://'../sounds/music/8000/suite-espanola-op-47-leyenda.wav'",NULL);

Next, the bridge is initiated:

         esl_execute(handle, "bridge", "user/1001",NULL);

Now, a displacement is scheduled for 5  seconds in the future:

         esl_send_recv(handle, "api sched_api +5 none uuid_displace 
<uuid> start digits/1.wav");

After each MEDIA_BUG_STOP event is received, another displacement is 
scheduled.

The log file shows successful attempts are being made to play the 
displacement audio:

66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:39.161496 [DEBUG] 
switch_core_media_bug.c:506 Attaching BUG to 
sofia/internal/1000 at 192.168.1.136
2012-12-04 18:25:39.161496 [DEBUG] mod_commands.c:3894 Command 
uuid_displace(66017ce6-23c8-4df1-accc-6275eed8fc45 start digits/1.wav):
+OK Success

2012-12-04 18:25:39.161496 [DEBUG] switch_scheduler.c:138 Deleting task 
17 sched_api_function (none)
66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:39.641496 [DEBUG] 
switch_core_media_bug.c:724 Removing BUG from 
sofia/internal/1000 at 192.168.1.136
2012-12-04 18:25:39.761496 [DEBUG] switch_scheduler.c:214 Added task 18 
sched_api_function (none) to run at 1354667144
66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:44.161496 [DEBUG] 
switch_core_media_bug.c:506 Attaching BUG to 
sofia/internal/1000 at 192.168.1.136
2012-12-04 18:25:44.161496 [DEBUG] mod_commands.c:3894 Command 
uuid_displace(66017ce6-23c8-4df1-accc-6275eed8fc45 start digits/1.wav):
+OK Success

2012-12-04 18:25:44.161496 [DEBUG] switch_scheduler.c:138 Deleting task 
18 sched_api_function (none)
66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:44.181496 [DEBUG] 
switch_core_media_bug.c:724 Removing BUG from 
sofia/internal/1000 at 192.168.1.136
2012-12-04 18:25:44.341496 [DEBUG] switch_scheduler.c:214 Added task 19 
sched_api_function (none) to run at 1354667149
66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:49.161496 [DEBUG] 
switch_core_media_bug.c:506 Attaching BUG to 
sofia/internal/1000 at 192.168.1.136
2012-12-04 18:25:49.161496 [DEBUG] mod_commands.c:3894 Command 
uuid_displace(66017ce6-23c8-4df1-accc-6275eed8fc45 start digits/1.wav):
+OK Success

2012-12-04 18:25:49.161496 [DEBUG] switch_scheduler.c:138 Deleting task 
19 sched_api_function (none)
66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:49.181496 [DEBUG] 
switch_core_media_bug.c:724 Removing BUG from 
sofia/internal/1000 at 192.168.1.136
2012-12-04 18:25:49.341496 [DEBUG] switch_scheduler.c:214 Added task 20 
sched_api_function (none) to run at 1354667154
66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:54.161496 [DEBUG] 
switch_core_media_bug.c:506 Attaching BUG to 
sofia/internal/1000 at 192.168.1.136
2012-12-04 18:25:54.161496 [DEBUG] mod_commands.c:3894 Command 
uuid_displace(66017ce6-23c8-4df1-accc-6275eed8fc45 start digits/1.wav):
+OK Success

Any guidance would be appreciated.

Thanks,

Scott



From ash at url.net.au  Wed Dec  5 09:57:29 2012
From: ash at url.net.au (Ashley Breeden)
Date: Wed, 5 Dec 2012 17:57:29 +1100
Subject: [Freeswitch-users] 10 second delay
In-Reply-To: <50BEEBA3.9040006@tagnet.ru>
References: <CADK-BjuB7ugJMPLSYByJPa0jQmnRywvWLgPdwDUCkKF2e7eFbA@mail.gmail.com>
	<50BED517.8060900@communicatefreely.net>
	<50BEEBA3.9040006@tagnet.ru>
Message-ID: <83CE1052-D007-40D9-A5DB-1540C1A7D982@url.net.au>

Hi There,

I noticed a similar problem using 1.2.5.2.  In my dial plans I had:

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

By removing the ignore_early_media line completely from my dialplans the problem was fixed.  Do you have ignore_early_media set anywhere in the Dialplans it passes through?



- Ash.



On 05/12/2012, at 5:37 PM, Boris Kovalenko wrote:

> Hello!
> 
> What version of FS are You useing? I have the same troubles with 1.2.5 
> branch. 1.2.3 works fine.
>> My best advice would be to to a packet capture, or turn on sip traces between the gateway,
>>   FS, and the phones, then try a test.
>> 
>> When you look through the call, take a look at the difference in the time stamps.
>> 
>> You should see a progress or a ringing back from the phone, then a 200 OK when they pick up.
>> 
>> After the call is picked up, see if there is additional traffic required to setup media,
>> or any other entries in the log that might suggest that some additional actions are
>> happening after the answer.
>> 
>> Also, if you can capture with wireshark, you can play back the media stream.  Check to see
>> if the first hello is there coming out of the phone, and then going to the gateway.
>> Wireshark will put it all in a time line perspective for you, so you can determine if this
>> is a problem with the phone, your configuration, or the gateway.
>> 
>> Are you using late negotiation, bypass media, or any other options like that?
>> 
>> -Tim
>> 
>> Blake Priddy wrote:
>>> I have some secretaries here at our school district that when
>>> they receive a call they have to give their spill 2-3 times before the
>>> party on the other end will hear them.. I have the Epygi Gateway and
>>> FreeSwitch box in the same network. Any thoughts would be greatly
>>> appreciated! :)
>>> 
>>> 
>>> -- 
>>> 
>>> *Blakelund Priddy*
>>> Network Systems Engineer
>>> Bryant Public School District
>>> Bryant, Arkansas 72022
>>> http://www.bryantschools.org <http://www.bryantschools.org/>
>>> p 501-653-5038
>>> f 501-847-5656
>>> 
>>> 
>>> ------------------------------------------------------------------------
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> -- 
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435) 230001
>   ???? +7 (3435) 230005
>   http://www.tagnet.ru
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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



From boris at tagnet.ru  Wed Dec  5 10:06:38 2012
From: boris at tagnet.ru (Boris Kovalenko)
Date: Wed, 05 Dec 2012 13:06:38 +0600
Subject: [Freeswitch-users] 10 second delay
In-Reply-To: <83CE1052-D007-40D9-A5DB-1540C1A7D982@url.net.au>
References: <CADK-BjuB7ugJMPLSYByJPa0jQmnRywvWLgPdwDUCkKF2e7eFbA@mail.gmail.com>
	<50BED517.8060900@communicatefreely.net>
	<50BEEBA3.9040006@tagnet.ru>
	<83CE1052-D007-40D9-A5DB-1540C1A7D982@url.net.au>
Message-ID: <50BEF27E.3060205@tagnet.ru>

Hello!

     Yes, I use ignore_early_media
> Hi There,
>
> I noticed a similar problem using 1.2.5.2.  In my dial plans I had:
>
> <action application="set" data="ignore_early_media=true"/>
>
> By removing the ignore_early_media line completely from my dialplans the problem was fixed.  Do you have ignore_early_media set anywhere in the Dialplans it passes through?
>
>
>
> - Ash.
>
>
>
> On 05/12/2012, at 5:37 PM, Boris Kovalenko wrote:
>
>> Hello!
>>
>> What version of FS are You useing? I have the same troubles with 1.2.5
>> branch. 1.2.3 works fine.
>>> My best advice would be to to a packet capture, or turn on sip traces between the gateway,
>>>    FS, and the phones, then try a test.
>>>
>>> When you look through the call, take a look at the difference in the time stamps.
>>>
>>> You should see a progress or a ringing back from the phone, then a 200 OK when they pick up.
>>>
>>> After the call is picked up, see if there is additional traffic required to setup media,
>>> or any other entries in the log that might suggest that some additional actions are
>>> happening after the answer.
>>>
>>> Also, if you can capture with wireshark, you can play back the media stream.  Check to see
>>> if the first hello is there coming out of the phone, and then going to the gateway.
>>> Wireshark will put it all in a time line perspective for you, so you can determine if this
>>> is a problem with the phone, your configuration, or the gateway.
>>>
>>> Are you using late negotiation, bypass media, or any other options like that?
>>>
>>> -Tim
>>>
>>> Blake Priddy wrote:
>>>> I have some secretaries here at our school district that when
>>>> they receive a call they have to give their spill 2-3 times before the
>>>> party on the other end will hear them.. I have the Epygi Gateway and
>>>> FreeSwitch box in the same network. Any thoughts would be greatly
>>>> appreciated! :)
>>>>
>>>>
>>>> -- 
>>>>
>>>> *Blakelund Priddy*
>>>> Network Systems Engineer
>>>> Bryant Public School District
>>>> Bryant, Arkansas 72022
>>>> http://www.bryantschools.org <http://www.bryantschools.org/>
>>>> p 501-653-5038
>>>> f 501-847-5656
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> -- 
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    ???. +7 (3435) 230001
>>    ???? +7 (3435) 230005
>>    http://www.tagnet.ru
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005
   http://www.tagnet.ru



From a.venugopan at mundio.com  Wed Dec  5 12:35:03 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Wed, 5 Dec 2012 09:35:03 +0000
Subject: [Freeswitch-users] sip registration
In-Reply-To: <CAFiqYu=cpMkTxdE99smQBM6NgitNQ5ZpePwfSYr_R23dpj7Hng@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF2339A49@Mail-Kilo.squay.com>
	<CAFiqYu=cpMkTxdE99smQBM6NgitNQ5ZpePwfSYr_R23dpj7Hng@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2339B39@Mail-Kilo.squay.com>

Hi,
In that case can I have 1 more column say e-mail and can this e-mail be checked in DB instead of checking reg_user('100')? Is that feasible?
Also which code should be changed any idea please?

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 04 December 2012 19:51
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

You can have a user 'ana' in the domain 'gmail.com<http://gmail.com>'. Though using someone else's domain as local in your FS setup may not be a good idea.
You can't have a @ in the username itself (per the SIP standard, not limited to FreeSWITCH).

On 4 December 2012 18:00, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

Currently we register authentication name as say '100' in sip registration, this comes to freeswitch and it will check in our DB for 100 and if its present then registrations would be successful.

freeswitch at internal> show registrations
reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
100,fsfailover.uk01.com<http://fsfailover.uk01.com>,e4969067f9a8c098,sofia/internal/sip:100 at 192.168.2.234:5060;transport=udp,1354638871,192.168.2.234,5060,udp,squay-laptop-1.squay.com<http://squay-laptop-1.squay.com>

I want to change this 100 to some e-mail address, so instead of 100 it will be something like 'ana at gmail.com<mailto:ana at gmail.com>'. Can we do this? While coming to freeswitch whether there would be any issues?

Regards,
Archana


_________________________________________________________________________
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

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From mickstevens at yahoo.com  Wed Dec  5 13:54:25 2012
From: mickstevens at yahoo.com (Mick Stevens)
Date: Wed, 5 Dec 2012 02:54:25 -0800 (PST)
Subject: [Freeswitch-users] Explanation of rtp_audio_in / out fields in
	XML CDR's?
References: <1354102130.98970.YahooMailNeo@web160802.mail.bf1.yahoo.com>
	<CAKzWOxUcpgaS29KWmSHg8FmMAraOVAa75i71sXdpbBC0S7EC_w@mail.gmail.com>
	<1354221331.77886.YahooMailNeo@web160803.mail.bf1.yahoo.com> 
Message-ID: <1354704865.29602.YahooMailNeo@web160804.mail.bf1.yahoo.com>

Hey Team,

Sorry to go on about this, but really keen to understand these RTP fields in the XML CDR's better...

1) At the risk of sounding too desperate... Please, please, please can someone explain to me what the?<rtp_audio_in_skip_packet_count> &?<rtp_audio_out_skip_packet_count> fields indicates?

2) Also, for example in the extract below, why are the rtp_audio_in & rtp_audio_out bytes / packet counts not the same? Is this a red herring based on some codec interworking issue or similar, an indication of packet loss or expected/normal behaviour?

? ? <rtp_audio_in_raw_bytes>436848</rtp_audio_in_raw_bytes>
? ? <rtp_audio_in_media_bytes>436536</rtp_audio_in_media_bytes>
? ? <rtp_audio_in_packet_count>2562</rtp_audio_in_packet_count>
? ? <rtp_audio_in_media_packet_count>2538</rtp_audio_in_media_packet_count>
? ? <rtp_audio_in_skip_packet_count>1904</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>24</rtp_audio_in_cng_packet_count>
? ? <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
? ? <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
? ? <rtp_audio_out_raw_bytes>624570</rtp_audio_out_raw_bytes>
? ? <rtp_audio_out_media_bytes>624360</rtp_audio_out_media_bytes>
? ? <rtp_audio_out_packet_count>3645</rtp_audio_out_packet_count>
? ? <rtp_audio_out_media_packet_count>3630</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>15</rtp_audio_out_cng_packet_count>

Please help!?
?
?
Rgds, Mick
Tel/SMS. +44(0)7967 594432
Fax. +44(0)7053 452429

Email/IM. mickstevens at yahoo.com
Skype: mick_stevens
www.facebook.com/mickstevens
www.twitter.com/mickstevens



________________________________
 From: Mick Stevens <mickstevens at yahoo.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Thursday, 29 November 2012, 20:35
Subject: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?
 

Hi Michael & ?Co,

Many thanks for the prompt & positive response! Yes, more than happy to collate feedback from the global team & update the wiki accordingly (I'd like to be able to document what the field values between the > < indicate as well as just explain the field names if possible...) +anything else I can do to to contribute to the project...?

Yes, now the wiki is back up I've managed to work out that cng_packet = comfort noise generation! Also from the wiki, possibly that the jb in?rtp_audio_in_jb_packet_count &?rtp_audio_in_largest_jb_size = jitter buffer? (apologies if everybody else already knows this!).

To provide the background to my original enquiry, I'm trying to identify if the rtp_audio_in_skip_packet_count in the following XML CDR extract is indicative of packet loss, or perhaps more accurately noticeable audio loss (as the packets haven't been lost, just ignored/"skipped"?: If anybody knows please speak up!?

? ? <rtp_audio_in_raw_bytes>1565066</rtp_audio_in_raw_bytes>
? ? <rtp_audio_in_media_bytes>1564856</rtp_audio_in_media_bytes>
? ? <rtp_audio_in_packet_count>9113</rtp_audio_in_packet_count>
? ? <rtp_audio_in_media_packet_count>9098</rtp_audio_in_media_packet_count>
? ? <rtp_audio_in_skip_packet_count>1609</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>15</rtp_audio_in_cng_packet_count>
? ? <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
? ? <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>

Thank you in anticipation of enlightenment!

#FreeSWITCH
?
Rgds, Mick
Tel/SMS. +44(0)7967 594432
Fax. +44(0)7053 452429

Email/IM. mickstevens at yahoo.com
Skype: mick_stevens
www.facebook.com/mickstevens
www.twitter.com/mickstevens



________________________________
 From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Thursday, 29 November 2012, 19:14
Subject: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?
 

Mick,

This data is definitely not on the wiki - or anywhere else that I can see. I think we can crowdsource this to get the info collected and then add it to the mod_xml_curl wiki page. For the record, here's a quick dump of the fields that I took from an XML CDR:

??? <rtp_audio_in_raw_bytes>10664</rtp_audio_in_raw_bytes>
??? <rtp_audio_in_media_bytes>8772</rtp_audio_in_media_bytes>
??? <rtp_audio_in_packet_count>62</rtp_audio_in_packet_count>
??? <rtp_audio_in_media_packet_count>51</rtp_audio_in_media_packet_count>
??? <rtp_audio_in_skip_packet_count>18</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>11</rtp_audio_in_flush_packet_count>
??? <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
??? <rtp_audio_out_raw_bytes>11524</rtp_audio_out_raw_bytes>
??? <rtp_audio_out_media_bytes>11524</rtp_audio_out_media_bytes>
??? <rtp_audio_out_packet_count>67</rtp_audio_out_packet_count>
??? <rtp_audio_out_media_packet_count>67</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>

I think raw_bytes, media_bytes, packet_count, and media_packet_count are self-explanatory. I think cng_packet_count is probably self-explanatory too. My question on dtmf_packet_count would be whether it's only for RFC2833 packets (I suspect yes, but would like confirmation). 

If anyone knows these please reply to this email and Mick and I will get them documented on the wiki (right Mick? ;)

rtp_audio_in_skip_packet_count
rtp_audio_out_skip_packet_count
rtp_audio_in_jb_packet_count
rtp_audio_in_flush_packet_count
rtp_audio_in_largest_jb_size

Thanks all!

-MC


On Wed, Nov 28, 2012 at 3:28 AM, Mick Stevens <mickstevens at yahoo.com> wrote:

Hi Folks,
>
>
>I'm trying to use FS XML CDR's to diagnose historic audio problems. I think I can work out some of the rtp_audio_in / out fields (raw bytes & media bytes being nearly equal looks like a good sign) but am wondering about the skip, cng & flush fields for example?
>
>
>I have tried Googling this & can find evidence of other people having asked this question but not of the answer. I have also checked my FS 106 & Cookbook book's without success. The wiki appears to be down at the moment so my apologies if the answer lies there.
>
>
>I know how to do this in real time using wireshark etc but am interested in being able to do some analysis on historic problems reported by customers that aren't willing/able to replicate the problem in order for a protocol trace to be captured.
>
>
>Any help much appreciated!
>?
>Rgds, Mick
>Tel/SMS. +44(0)7967 594432
>Fax. +44(0)7053 452429
>
>Email/IM. mickstevens at yahoo.com
>Skype: mick_stevens
>www.facebook.com/mickstevens
>www.twitter.com/mickstevens
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org



_________________________________________________________________________
Professional FreeSWITCH Consulting 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 peter.olsson at visionutveckling.se  Wed Dec  5 14:12:43 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 5 Dec 2012 11:12:43 +0000
Subject: [Freeswitch-users] Explanation of rtp_audio_in / out fields
	in	XML CDR's?
Message-ID: <1FFF97C269757C458224B7C895F35F151DD0DF@cantor.std.visionutv.se>

1)       I guess this is for how many packets that has been skipped for incoming and/or outgoing traffic. For instance, if you get the same packet twice with the same timestamp, this would increase the skipped_packet_count.

2)       There is no reason that in and out is always the same. For instance, if you mute your phone, it might stop sending RTP frames, but you might still receive frames.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Mick Stevens
Skickat: den 5 december 2012 11:54
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?

Hey Team,


Sorry to go on about this, but really keen to understand these RTP fields in the XML CDR's better...


1) At the risk of sounding too desperate... Please, please, please can someone explain to me what the <rtp_audio_in_skip_packet_count> & <rtp_audio_out_skip_packet_count> fields indicates?


2) Also, for example in the extract below, why are the rtp_audio_in & rtp_audio_out bytes / packet counts not the same? Is this a red herring based on some codec interworking issue or similar, an indication of packet loss or expected/normal behaviour?

    <rtp_audio_in_raw_bytes>436848</rtp_audio_in_raw_bytes>
    <rtp_audio_in_media_bytes>436536</rtp_audio_in_media_bytes>
    <rtp_audio_in_packet_count>2562</rtp_audio_in_packet_count>
    <rtp_audio_in_media_packet_count>2538</rtp_audio_in_media_packet_count>
    <rtp_audio_in_skip_packet_count>1904</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>24</rtp_audio_in_cng_packet_count>
    <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
    <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
    <rtp_audio_out_raw_bytes>624570</rtp_audio_out_raw_bytes>
    <rtp_audio_out_media_bytes>624360</rtp_audio_out_media_bytes>
    <rtp_audio_out_packet_count>3645</rtp_audio_out_packet_count>
    <rtp_audio_out_media_packet_count>3630</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>15</rtp_audio_out_cng_packet_count>

Please help! [:) happy]


Rgds, Mick
Tel/SMS. +44(0)7967 594432
Fax. +44(0)7053 452429
Email/IM. mickstevens at yahoo.com
Skype: mick_stevens
www.facebook.com/mickstevens
www.twitter.com/mickstevens

________________________________
From: Mick Stevens <mickstevens at yahoo.com<mailto:mickstevens at yahoo.com>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Thursday, 29 November 2012, 20:35
Subject: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?

Hi Michael &  Co,

Many thanks for the prompt & positive response! Yes, more than happy to collate feedback from the global team & update the wiki accordingly (I'd like to be able to document what the field values between the > < indicate as well as just explain the field names if possible...) +anything else I can do to to contribute to the project...?

Yes, now the wiki is back up I've managed to work out that cng_packet = comfort noise generation! Also from the wiki, possibly that the jb in rtp_audio_in_jb_packet_count & rtp_audio_in_largest_jb_size = jitter buffer? (apologies if everybody else already knows this!).

To provide the background to my original enquiry, I'm trying to identify if the rtp_audio_in_skip_packet_count in the following XML CDR extract is indicative of packet loss, or perhaps more accurately noticeable audio loss (as the packets haven't been lost, just ignored/"skipped"?: If anybody knows please speak up! [:) happy]

    <rtp_audio_in_raw_bytes>1565066</rtp_audio_in_raw_bytes>
    <rtp_audio_in_media_bytes>1564856</rtp_audio_in_media_bytes>
    <rtp_audio_in_packet_count>9113</rtp_audio_in_packet_count>
    <rtp_audio_in_media_packet_count>9098</rtp_audio_in_media_packet_count>
    <rtp_audio_in_skip_packet_count>1609</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>15</rtp_audio_in_cng_packet_count>
    <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
    <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>

Thank you in anticipation of enlightenment!

#[:x lovestruck]FreeSWITCH

Rgds, Mick
Tel/SMS. +44(0)7967 594432
Fax. +44(0)7053 452429
Email/IM. mickstevens at yahoo.com
Skype: mick_stevens
www.facebook.com/mickstevens
www.twitter.com/mickstevens

________________________________
From: Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Thursday, 29 November 2012, 19:14
Subject: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?

Mick,

This data is definitely not on the wiki - or anywhere else that I can see. I think we can crowdsource this to get the info collected and then add it to the mod_xml_curl wiki page<http://wiki.freeswitch.org/wiki/Mod_xml_cdr>. For the record, here's a quick dump of the fields that I took from an XML CDR:

    <rtp_audio_in_raw_bytes>10664</rtp_audio_in_raw_bytes>
    <rtp_audio_in_media_bytes>8772</rtp_audio_in_media_bytes>
    <rtp_audio_in_packet_count>62</rtp_audio_in_packet_count>
    <rtp_audio_in_media_packet_count>51</rtp_audio_in_media_packet_count>
    <rtp_audio_in_skip_packet_count>18</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>11</rtp_audio_in_flush_packet_count>
    <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
    <rtp_audio_out_raw_bytes>11524</rtp_audio_out_raw_bytes>
    <rtp_audio_out_media_bytes>11524</rtp_audio_out_media_bytes>
    <rtp_audio_out_packet_count>67</rtp_audio_out_packet_count>
    <rtp_audio_out_media_packet_count>67</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>

I think raw_bytes, media_bytes, packet_count, and media_packet_count are self-explanatory. I think cng_packet_count is probably self-explanatory too. My question on dtmf_packet_count would be whether it's only for RFC2833 packets (I suspect yes, but would like confirmation).

If anyone knows these please reply to this email and Mick and I will get them documented on the wiki (right Mick? ;)

rtp_audio_in_skip_packet_count
rtp_audio_out_skip_packet_count
rtp_audio_in_jb_packet_count
rtp_audio_in_flush_packet_count
rtp_audio_in_largest_jb_size

Thanks all!

-MC
On Wed, Nov 28, 2012 at 3:28 AM, Mick Stevens <mickstevens at yahoo.com<mailto:mickstevens at yahoo.com>> wrote:
Hi Folks,

I'm trying to use FS XML CDR's to diagnose historic audio problems. I think I can work out some of the rtp_audio_in / out fields (raw bytes & media bytes being nearly equal looks like a good sign) but am wondering about the skip, cng & flush fields for example?

I have tried Googling this & can find evidence of other people having asked this question but not of the answer. I have also checked my FS 106 & Cookbook book's without success. The wiki appears to be down at the moment so my apologies if the answer lies there.

I know how to do this in real time using wireshark etc but am interested in being able to do some analysis on historic problems reported by customers that aren't willing/able to replicate the problem in order for a protocol trace to be captured.

Any help much appreciated!

Rgds, Mick
Tel/SMS. +44(0)7967 594432
Fax. +44(0)7053 452429
Email/IM. mickstevens at yahoo.com
Skype: mick_stevens
www.facebook.com/mickstevens
www.twitter.com/mickstevens

_________________________________________________________________________
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/>

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/>



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org<http://www.freeswitch.org/>
http://www.ClueCon.com<http://www.cluecon.com/>
http://www.OSTAG.org<http://www.ostag.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/>

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:50bf25f932762027524863!
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From mickstevens at yahoo.com  Wed Dec  5 14:34:05 2012
From: mickstevens at yahoo.com (Mick Stevens)
Date: Wed, 5 Dec 2012 03:34:05 -0800 (PST)
Subject: [Freeswitch-users] Explanation of rtp_audio_in / out fields in
	XML CDR's?
In-Reply-To: <1FFF97C269757C458224B7C895F35F151DD0DF@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F151DD0DF@cantor.std.visionutv.se>
Message-ID: <1354707245.68534.YahooMailNeo@web160804.mail.bf1.yahoo.com>

Thank you Peter!

Additional replies still welcome (there is no such thing as too much information!)! :-)

Rgds, Mick
?


________________________________
 From: Peter Olsson <peter.olsson at visionutveckling.se>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Wednesday, 5 December 2012, 11:12
Subject: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?
 

 
1)?????? I guess this is for how many packets that has been skipped for incoming and/or outgoing traffic. For instance, if you get the same packet twice with the same timestamp, this would increase the skipped_packet_count.
2)?????? There is no reason that in and out is always the same. For instance, if you mute your phone, it might stop sending RTP frames, but you might still receive frames.
?
/Peter
?
?
Fr?n:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Mick Stevens
Skickat: den 5 december 2012 11:54
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?
?
Hey Team,


Sorry to go on about this, but really keen to understand these RTP fields in the XML CDR's better...


1) At the risk of sounding too desperate... Please, please, please can someone explain to me what the?<rtp_audio_in_skip_packet_count> &?<rtp_audio_out_skip_packet_count> fields indicates?


2) Also, for example in the extract below, why are the rtp_audio_in & rtp_audio_out bytes / packet counts not the same? Is this a red herring based on some codec interworking issue or similar, an indication of packet loss or expected/normal behaviour?
?
? ? <rtp_audio_in_raw_bytes>436848</rtp_audio_in_raw_bytes>
? ? <rtp_audio_in_media_bytes>436536</rtp_audio_in_media_bytes>
? ? <rtp_audio_in_packet_count>2562</rtp_audio_in_packet_count>
? ? <rtp_audio_in_media_packet_count>2538</rtp_audio_in_media_packet_count>
? ? <rtp_audio_in_skip_packet_count>1904</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>24</rtp_audio_in_cng_packet_count>
? ? <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
? ? <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
? ? <rtp_audio_out_raw_bytes>624570</rtp_audio_out_raw_bytes>
? ? <rtp_audio_out_media_bytes>624360</rtp_audio_out_media_bytes>
? ? <rtp_audio_out_packet_count>3645</rtp_audio_out_packet_count>
? ? <rtp_audio_out_media_packet_count>3630</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>15</rtp_audio_out_cng_packet_count>
?
Please help!?
?
?
Rgds, Mick
Tel/SMS. +44(0)7967 594432
Fax. +44(0)7053 452429
Email/IM. mickstevens at yahoo.com
Skype: mick_stevens
www.facebook.com/mickstevens
www.twitter.com/mickstevens
?

________________________________
 
From:Mick Stevens <mickstevens at yahoo.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Thursday, 29 November 2012, 20:35
Subject: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?
?
Hi Michael & ?Co,
?
Many thanks for the prompt & positive response! Yes, more than happy to collate feedback from the global team & update the wiki accordingly (I'd like to be able to document what the field values between the > < indicate as well as just explain the field names if possible...) +anything else I can do to to contribute to the project...?
?
Yes, now the wiki is back up I've managed to work out that cng_packet = comfort noise generation! Also from the wiki, possibly that the jb in?rtp_audio_in_jb_packet_count &?rtp_audio_in_largest_jb_size = jitter buffer? (apologies if everybody else already knows this!).
?
To provide the background to my original enquiry, I'm trying to identify if the rtp_audio_in_skip_packet_count in the following XML CDR extract is indicative of packet loss, or perhaps more accurately noticeable audio loss (as the packets haven't been lost, just ignored/"skipped"?: If anybody knows please speak up!?
?
? ? <rtp_audio_in_raw_bytes>1565066</rtp_audio_in_raw_bytes>
? ? <rtp_audio_in_media_bytes>1564856</rtp_audio_in_media_bytes>
? ? <rtp_audio_in_packet_count>9113</rtp_audio_in_packet_count>
? ? <rtp_audio_in_media_packet_count>9098</rtp_audio_in_media_packet_count>
? ?  <rtp_audio_in_skip_packet_count>1609</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>15</rtp_audio_in_cng_packet_count>
? ? <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
? ? <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
?
Thank you in anticipation of enlightenment!
?
#FreeSWITCH
?
Rgds, Mick
Tel/SMS. +44(0)7967 594432
Fax. +44(0)7053 452429
Email/IM. mickstevens at yahoo.com
Skype: mick_stevens
www.facebook.com/mickstevens
www.twitter.com/mickstevens
?

________________________________
 
From:Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Thursday, 29 November 2012, 19:14
Subject: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?
?
Mick,

This data is definitely not on the wiki - or anywhere else that I can see. I think we can crowdsource this to get the info collected and then add it to the mod_xml_curl wiki page. For the record, here's a quick dump of the fields that I took from an XML CDR:

??? <rtp_audio_in_raw_bytes>10664</rtp_audio_in_raw_bytes>
??? <rtp_audio_in_media_bytes>8772</rtp_audio_in_media_bytes>
??? <rtp_audio_in_packet_count>62</rtp_audio_in_packet_count>
??? <rtp_audio_in_media_packet_count>51</rtp_audio_in_media_packet_count>
??? <rtp_audio_in_skip_packet_count>18</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>11</rtp_audio_in_flush_packet_count>
??? <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
??? <rtp_audio_out_raw_bytes>11524</rtp_audio_out_raw_bytes>
??? <rtp_audio_out_media_bytes>11524</rtp_audio_out_media_bytes>
??? <rtp_audio_out_packet_count>67</rtp_audio_out_packet_count>
??? <rtp_audio_out_media_packet_count>67</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>

I think raw_bytes, media_bytes, packet_count, and media_packet_count are self-explanatory. I think cng_packet_count is probably self-explanatory too. My question on dtmf_packet_count would be whether it's only for RFC2833 packets (I suspect yes, but would like
 confirmation). 

If anyone knows these please reply to this email and Mick and I will get them documented on the wiki (right Mick? ;)

rtp_audio_in_skip_packet_count
rtp_audio_out_skip_packet_count
rtp_audio_in_jb_packet_count
rtp_audio_in_flush_packet_count
rtp_audio_in_largest_jb_size

Thanks all!

-MC
On Wed, Nov 28, 2012 at 3:28 AM, Mick Stevens <mickstevens at yahoo.com> wrote:
Hi Folks,
?
I'm trying to use FS XML CDR's to diagnose historic audio problems. I think I can work out some of the rtp_audio_in / out fields (raw bytes & media bytes being nearly equal looks like a good sign) but am wondering about the skip, cng & flush fields for example?
?
I have tried Googling this & can find evidence of other people having asked this question but not of the answer. I have also checked my FS 106 & Cookbook book's without success. The wiki appears to be down at the moment so my apologies if the answer lies there.
?
I know how to do this in real time using wireshark etc but am interested in being able to do some analysis on historic problems reported by customers that aren't willing/able to replicate the problem in order for a protocol trace to be captured.
?
Any help much appreciated!
?
Rgds, Mick
Tel/SMS. +44(0)7967 594432
Fax. +44(0)7053 452429
Email/IM. mickstevens at yahoo.com
Skype: mick_stevens
www.facebook.com/mickstevens
www.twitter.com/mickstevens

_________________________________________________________________________
Professional FreeSWITCH Consulting 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



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org



_________________________________________________________________________
Professional FreeSWITCH Consulting 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


?
!DSPAM:50bf25f932762027524863! 
_________________________________________________________________________
Professional FreeSWITCH Consulting 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
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From 8f27e956 at gmail.com  Wed Dec  5 14:43:46 2012
From: 8f27e956 at gmail.com (Scott)
Date: Wed, 5 Dec 2012 06:43:46 -0500
Subject: [Freeswitch-users] NATIVE SQL ERROR [no such table: fifo_outbound]
Message-ID: <CA+Y7Lf7GzFpacZ_spvJg+jK8+s+4OoeeH0szMmRP2FtHfeOcsw@mail.gmail.com>

Just did a 'make current' to update to 1.2.5.2 (from 1.2.5.1).  Now, upon
restarting freeswitch, it logs the following,

...skipping...
SQL [Enabled]
2012-12-05 06:29:58.746676 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR [no
such table: fifo_outbound]
update fifo_outbound set
start_time=0,stop_time=0,ring_count=0,use_count=0,outbound_call_count=0,outbound_fail_count=0
where static=0
2012-12-05 06:29:58.887575 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR [no
such table: fifo_outbound]
delete from fifo_outbound where static=1 and hostname='fs1.soho.local'


In so far as I could see, the 'make' finished successfully without apparent
errors.

Thoughts & suggestions welcome.

With thanks,
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From ntomer at newgen.co.in  Wed Dec  5 14:50:49 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Wed, 5 Dec 2012 17:20:49 +0530
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <CAKzWOxUBVqEnXtRa=f7TJjBup47v0_2e3eTnUn-rWx=LBe6K+Q@mail.gmail.com>
References: <cdo25qex5xplrot14oad7fr7.1354501431119@email.android.com>	<CAKzWOxVLewGaOzAPoHT7dtEV9odJFSc-4JKXD4tBLsg+nSnPoQ@mail.gmail.com>	<00aa01cdd1d7$6bcea110$436be330$@co.in>
	<CAKzWOxUBVqEnXtRa=f7TJjBup47v0_2e3eTnUn-rWx=LBe6K+Q@mail.gmail.com>
Message-ID: <012f01cdd2de$c5867b30$50937190$@co.in>

I need to check, whether a call is parked at that extension.

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, December 04, 2012 9:51 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] valet_park help needed

 

 

On Mon, Dec 3, 2012 at 8:25 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:

Accepted J I should have looked that up.

 

Need another help, is there is way to check from Lua whether an extension is
available?

What do you mean by "extension is available"?
-MC
 

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, December 04, 2012 4:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] valet_park help needed

 

 

On Sun, Dec 2, 2012 at 6:23 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:

Hi Brian,

I've posted the contents of dialplan. Please tell me what I am doing wrong.

 

What he means is that you are doing a blind transfer to an extension that is
designed for attended transfers. We already discussed this earlier in the
thread. In the meantime kudos to Abaci for pointing out a channel variable
that is very clearly mentioned on the valet_park wiki page:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park#Channel_Vari
ables

Shame on us both for not looking that up. Well, mostly shame on you because
you're supposed to look at the wiki before you post here. :P

-MC

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
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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
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FreeSWITCH-users mailing list
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-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org




Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From shaik.bawajan at gmail.com  Wed Dec  5 15:23:40 2012
From: shaik.bawajan at gmail.com (shaik bawajan)
Date: Wed, 5 Dec 2012 17:53:40 +0530
Subject: [Freeswitch-users] having problem using java with esl library
Message-ID: <CAB3HJytcfi9bs8gNxLx0t4g5C4OveFUyS+1OqTMfgx=4kFEt3A@mail.gmail.com>

Hi,

I am new to freeswitch and am using java app with esl library to make
outbound calls and am getting the error is "Create additional event
dispatch thread 2".

I saw in the freeswitch logs is that - Event 0 Blocking and its keeps
printing.

Kindly let me know where am doing mistake and how to resolve it.

Thanks in Advance,
Bawajan
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From bigx333 at gmail.com  Wed Dec  5 16:50:06 2012
From: bigx333 at gmail.com (Nelson Camargo)
Date: Wed, 5 Dec 2012 15:50:06 +0200
Subject: [Freeswitch-users] NATIVE SQL ERROR [no such table:
	fifo_outbound]
In-Reply-To: <CA+Y7Lf7GzFpacZ_spvJg+jK8+s+4OoeeH0szMmRP2FtHfeOcsw@mail.gmail.com>
References: <CA+Y7Lf7GzFpacZ_spvJg+jK8+s+4OoeeH0szMmRP2FtHfeOcsw@mail.gmail.com>
Message-ID: <924695E4-6690-4264-8E42-CB4573698E8F@gmail.com>

Having same issue, git aa08157 2012-11-30 04:53:03Z)

On 05 Dec 2012, at 1:43 PM, Scott wrote:

> Just did a 'make current' to update to 1.2.5.2 (from 1.2.5.1).  Now, upon restarting freeswitch, it logs the following,
> 
> ...skipping...
> SQL [Enabled]
> 2012-12-05 06:29:58.746676 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR [no such table: fifo_outbound]
> update fifo_outbound set start_time=0,stop_time=0,ring_count=0,use_count=0,outbound_call_count=0,outbound_fail_count=0 where static=0
> 2012-12-05 06:29:58.887575 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR [no such table: fifo_outbound]
> delete from fifo_outbound where static=1 and hostname='fs1.soho.local'
>  
> 
> In so far as I could see, the 'make' finished successfully without apparent errors.
> 
> Thoughts & suggestions welcome.
> 
> With 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From andres.souto at quobis.es  Wed Dec  5 15:04:37 2012
From: andres.souto at quobis.es (=?ISO-8859-1?Q?Andr=E9s_Souto_Vidal?=)
Date: Wed, 5 Dec 2012 13:04:37 +0100
Subject: [Freeswitch-users] Can not use Python ESL with Django and
 apache web server
In-Reply-To: <1353729183075-7584885.post@n2.nabble.com>
References: <1353487101708-7584800.post@n2.nabble.com>
	<CAMbW9oJpkmfrsZAh8QOLaRydZibP+YjhHEJYJrnfDS9SPbqp9Q@mail.gmail.com>
	<1353729183075-7584885.post@n2.nabble.com>
Message-ID: <CANpea9tpGxKLwB8oimCe8T4BHOEzSUVNbUHBD+0RikWm=zLTTg@mail.gmail.com>

I'm having the same issue when connecting to the ESL from Django through
Apache WSGI.
I increased the number of processes with the processes=n option in the
WSGIDaemonProcess
directive and it first seems to work but then I realized that it works only
sometimes and I also notice it slows down all the web server.

When ESLconnection is called, the Apache process/thread freezes and doesn't
respond anything to the request so I couldn't obtained any debug trace of
what is really happening. Using the Django built-in server all works OK so
it seems a problem between WSGI and Freeswitch Python ESL.

Any help would be really appreciated!


2012/11/24 chaiyawut.so <chaiyawut.so at gmail.com>

> That solved the problem, thank you
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Can-not-use-Python-ESL-with-Django-and-apache-web-server-tp7584800p7584885.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Dec  5 18:17:41 2012
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 5 Dec 2012 23:17:41 +0800
Subject: [Freeswitch-users] NATIVE SQL ERROR [no such table:
	fifo_outbound]
In-Reply-To: <924695E4-6690-4264-8E42-CB4573698E8F@gmail.com>
References: <CA+Y7Lf7GzFpacZ_spvJg+jK8+s+4OoeeH0szMmRP2FtHfeOcsw@mail.gmail.com>
	<924695E4-6690-4264-8E42-CB4573698E8F@gmail.com>
Message-ID: <CAJs+-5p35CSzXBpCKyLfyzrCGEp_au=NPAUru39EgWLAOdeM=g@mail.gmail.com>

update to git head, it should be fixed.

On Wed, Dec 5, 2012 at 9:50 PM, Nelson Camargo <bigx333 at gmail.com> wrote:
> Having same issue, git aa08157 2012-11-30 04:53:03Z)
>
> On 05 Dec 2012, at 1:43 PM, Scott wrote:
>
> Just did a 'make current' to update to 1.2.5.2 (from 1.2.5.1).  Now, upon
> restarting freeswitch, it logs the following,
>
> ...skipping...
> SQL [Enabled]
> 2012-12-05 06:29:58.746676 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR [no
> such table: fifo_outbound]
> update fifo_outbound set
> start_time=0,stop_time=0,ring_count=0,use_count=0,outbound_call_count=0,outbound_fail_count=0
> where static=0
> 2012-12-05 06:29:58.887575 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR [no
> such table: fifo_outbound]
> delete from fifo_outbound where static=1 and hostname='fs1.soho.local'
>
>
> In so far as I could see, the 'make' finished successfully without apparent
> errors.
>
> Thoughts & suggestions welcome.
>
> With 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn


From steveayre at gmail.com  Wed Dec  5 18:20:10 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 5 Dec 2012 15:20:10 +0000
Subject: [Freeswitch-users] Explanation of rtp_audio_in / out fields in
 XML CDR's?
In-Reply-To: <1354704865.29602.YahooMailNeo@web160804.mail.bf1.yahoo.com>
References: <1354102130.98970.YahooMailNeo@web160802.mail.bf1.yahoo.com>
	<CAKzWOxUcpgaS29KWmSHg8FmMAraOVAa75i71sXdpbBC0S7EC_w@mail.gmail.com>
	<1354221331.77886.YahooMailNeo@web160803.mail.bf1.yahoo.com>
	<1354704865.29602.YahooMailNeo@web160804.mail.bf1.yahoo.com>
Message-ID: <CAFiqYu=sujtS0ZB91JPBMsSRJx5tUG8vn7JkMMPiJAVxtHH61A@mail.gmail.com>

In vs Out are the 'listening' and 'talking' directions of the channel.
They're not the same audio, and as such there's no reason they would match
up. Couple that with sending/receiving before the other side, and network
jitter/packet loss so that some packets arrive at different times or not at
all.

As you point out using codecs can also affect them, as different
codecs/ptimes will use different amounts of data/packets for the same time
segment. Especially if the codec uses a VBR.

Even using the 'echo' app and matching codec settings they will differ
because of timing issues.


On 5 December 2012 10:54, Mick Stevens <mickstevens at yahoo.com> wrote:

> Hey Team,
>
> Sorry to go on about this, but really keen to understand these RTP fields
> in the XML CDR's better...
>
> 1) At the risk of sounding too desperate... Please, please, please can
> someone explain to me what the <rtp_audio_in_skip_packet_count>
> & <rtp_audio_out_skip_packet_count> fields indicates?
>
> 2) Also, for example in the extract below, why are the rtp_audio_in &
> rtp_audio_out bytes / packet counts not the same? Is this a red herring
> based on some codec interworking issue or similar, an indication of packet
> loss or expected/normal behaviour?
>
>     <rtp_audio_in_raw_bytes>436848</rtp_audio_in_raw_bytes>
>     <rtp_audio_in_media_bytes>436536</rtp_audio_in_media_bytes>
>     <rtp_audio_in_packet_count>2562</rtp_audio_in_packet_count>
>     <rtp_audio_in_media_packet_count>2538</rtp_audio_in_media_packet_count>
>     <rtp_audio_in_skip_packet_count>1904</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>24</rtp_audio_in_cng_packet_count>
>     <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
>     <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
>     <rtp_audio_out_raw_bytes>624570</rtp_audio_out_raw_bytes>
>     <rtp_audio_out_media_bytes>624360</rtp_audio_out_media_bytes>
>     <rtp_audio_out_packet_count>3645</rtp_audio_out_packet_count>
>
> <rtp_audio_out_media_packet_count>3630</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>15</rtp_audio_out_cng_packet_count>
>
> Please help! [image: :) happy]
>
>
> Rgds, Mick
> Tel/SMS. +44(0)7967 594432
> Fax. +44(0)7053 452429
> Email/IM. mickstevens at yahoo.com
> Skype: mick_stevens
> www.facebook.com/mickstevens
> www.twitter.com/mickstevens
>
>   ------------------------------
> *From:* Mick Stevens <mickstevens at yahoo.com>
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, 29 November 2012, 20:35
>
> *Subject:* Re: [Freeswitch-users] Explanation of rtp_audio_in / out
> fields in XML CDR's?
>
> Hi Michael &  Co,
>
> Many thanks for the prompt & positive response! Yes, more than happy to
> collate feedback from the global team & update the wiki accordingly (I'd
> like to be able to document what the field values between the > < indicate
> as well as just explain the field names if possible...) +anything else I
> can do to to contribute to the project...?
>
> Yes, now the wiki is back up I've managed to work out that cng_packet =
> comfort noise generation! Also from the wiki, possibly that the jb in rtp_audio_in_jb_packet_count
> & rtp_audio_in_largest_jb_size = jitter buffer? (apologies if everybody
> else already knows this!).
>
> To provide the background to my original enquiry, I'm trying to identify
> if the rtp_audio_in_skip_packet_count in the following XML CDR extract is
> indicative of packet loss, or perhaps more accurately noticeable audio loss
> (as the packets haven't been lost, just ignored/"skipped"?: If anybody
> knows please speak up! [image: :) happy]
>
>     <rtp_audio_in_raw_bytes>1565066</rtp_audio_in_raw_bytes>
>     <rtp_audio_in_media_bytes>1564856</rtp_audio_in_media_bytes>
>     <rtp_audio_in_packet_count>9113</rtp_audio_in_packet_count>
>     <rtp_audio_in_media_packet_count>9098</rtp_audio_in_media_packet_count>
>     <rtp_audio_in_skip_packet_count>1609</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>15</rtp_audio_in_cng_packet_count>
>     <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
>     <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
>
> Thank you in anticipation of enlightenment!
>
> #[image: :x lovestruck]FreeSWITCH
>
> Rgds, Mick
> Tel/SMS. +44(0)7967 594432
> Fax. +44(0)7053 452429
> Email/IM. mickstevens at yahoo.com
> Skype: mick_stevens
> www.facebook.com/mickstevens
> www.twitter.com/mickstevens
>
>   ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, 29 November 2012, 19:14
> *Subject:* Re: [Freeswitch-users] Explanation of rtp_audio_in / out
> fields in XML CDR's?
>
> Mick,
>
> This data is definitely not on the wiki - or anywhere else that I can see.
> I think we can crowdsource this to get the info collected and then add it
> to the mod_xml_curl wiki page<http://wiki.freeswitch.org/wiki/Mod_xml_cdr>.
> For the record, here's a quick dump of the fields that I took from an XML
> CDR:
>
>     <rtp_audio_in_raw_bytes>10664</rtp_audio_in_raw_bytes>
>     <rtp_audio_in_media_bytes>8772</rtp_audio_in_media_bytes>
>     <rtp_audio_in_packet_count>62</rtp_audio_in_packet_count>
>     <rtp_audio_in_media_packet_count>51</rtp_audio_in_media_packet_count>
>     <rtp_audio_in_skip_packet_count>18</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>11</rtp_audio_in_flush_packet_count>
>     <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
>     <rtp_audio_out_raw_bytes>11524</rtp_audio_out_raw_bytes>
>     <rtp_audio_out_media_bytes>11524</rtp_audio_out_media_bytes>
>     <rtp_audio_out_packet_count>67</rtp_audio_out_packet_count>
>     <rtp_audio_out_media_packet_count>67</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>
>
> I think raw_bytes, media_bytes, packet_count, and media_packet_count are
> self-explanatory. I think cng_packet_count is probably self-explanatory
> too. My question on dtmf_packet_count would be whether it's only for
> RFC2833 packets (I suspect yes, but would like confirmation).
>
> If anyone knows these please reply to this email and Mick and I will get
> them documented on the wiki (right Mick? ;)
>
> rtp_audio_in_skip_packet_count
> rtp_audio_out_skip_packet_count
> rtp_audio_in_jb_packet_count
> rtp_audio_in_flush_packet_count
> rtp_audio_in_largest_jb_size
>
> Thanks all!
>
> -MC
>
> On Wed, Nov 28, 2012 at 3:28 AM, Mick Stevens <mickstevens at yahoo.com>wrote:
>
> Hi Folks,
>
> I'm trying to use FS XML CDR's to diagnose historic audio problems. I
> think I can work out some of the rtp_audio_in / out fields (raw bytes &
> media bytes being nearly equal looks like a good sign) but am wondering
> about the skip, cng & flush fields for example?
>
> I have tried Googling this & can find evidence of other people having
> asked this question but not of the answer. I have also checked my FS 106 &
> Cookbook book's without success. The wiki appears to be down at the moment
> so my apologies if the answer lies there.
>
> I know how to do this in real time using wireshark etc but am interested
> in being able to do some analysis on historic problems reported by
> customers that aren't willing/able to replicate the problem in order for a
> protocol trace to be captured.
>
> Any help much appreciated!
>
> Rgds, Mick
> Tel/SMS. +44(0)7967 594432
> Fax. +44(0)7053 452429
> Email/IM. mickstevens at yahoo.com
> Skype: mick_stevens
> www.facebook.com/mickstevens
> www.twitter.com/mickstevens
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
> http://www.ClueCon.com <http://www.cluecon.com/>
> http://www.OSTAG.org <http://www.ostag.org/>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
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> http://wiki.freeswitch.org
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>
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From abaci64 at gmail.com  Wed Dec  5 18:22:30 2012
From: abaci64 at gmail.com (Abaci)
Date: Wed, 05 Dec 2012 10:22:30 -0500
Subject: [Freeswitch-users] valet_park help needed
In-Reply-To: <012f01cdd2de$c5867b30$50937190$@co.in>
References: <cdo25qex5xplrot14oad7fr7.1354501431119@email.android.com>	<CAKzWOxVLewGaOzAPoHT7dtEV9odJFSc-4JKXD4tBLsg+nSnPoQ@mail.gmail.com>	<00aa01cdd1d7$6bcea110$436be330$@co.in>
	<CAKzWOxUBVqEnXtRa=f7TJjBup47v0_2e3eTnUn-rWx=LBe6K+Q@mail.gmail.com>
	<012f01cdd2de$c5867b30$50937190$@co.in>
Message-ID: <50BF66B6.2020202@gmail.com>

you can use the 'valet_info' to get a list of parked calls and parse the 
XML (in lua etc.) if you have the c skills you may look into expanding 
this api function (valet_info_function in mod_valet_parking.c) to be 
able to give you back status only on a specific parking slot, that would 
probable be a useful feature for others as well. the reason it was 
designed to give back the entire list is probably because it's a stored 
in a hash (not sql) and has to iterate the entire table.

On 12/5/2012 6:50 AM, Nitin Tomer wrote:
>
> I need to check, whether a call is parked at that extension.
>
> Regards
>
> Nitin
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Michael Collins
> *Sent:* Tuesday, December 04, 2012 9:51 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] valet_park help needed
>
> On Mon, Dec 3, 2012 at 8:25 PM, Nitin Tomer <ntomer at newgen.co.in 
> <mailto:ntomer at newgen.co.in>> wrote:
>
> Accepted JI should have looked that up...
>
> Need another help, is there is way to check from Lua whether an 
> extension is available?
>
> What do you mean by "extension is available"?
> -MC
>
>     Regards
>
>     Nitin
>
>     *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:* Tuesday, December 04, 2012 4:08 AM
>     *To:* FreeSWITCH Users Help
>     *Subject:* Re: [Freeswitch-users] valet_park help needed
>
>     On Sun, Dec 2, 2012 at 6:23 PM, Nitin Tomer <ntomer at newgen.co.in
>     <mailto:ntomer at newgen.co.in>> wrote:
>
>     Hi Brian,
>
>     I've posted the contents of dialplan. Please tell me what I am
>     doing wrong.
>
>     What he means is that you are doing a blind transfer to an
>     extension that is designed for attended transfers. We already
>     discussed this earlier in the thread. In the meantime kudos to
>     Abaci for pointing out a channel variable that is very clearly
>     mentioned on the valet_park wiki page:
>
>     http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park#Channel_Variables
>
>     Shame on us both for not looking that up. Well, mostly shame on
>     you because you're supposed to look at the wiki before you post
>     here. :P
>
>     -MC
>
>
>                 Disclaimer :- This e-mail and any attachment may
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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
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
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>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>             Disclaimer :- This e-mail and any attachment may contain
>             confidential, proprietary or legally privileged
>             information. If you are not the original intended
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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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>
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From NuwanW at unifybusiness.co.uk  Wed Dec  5 19:16:04 2012
From: NuwanW at unifybusiness.co.uk (Nuwan Wijerathne)
Date: Wed, 5 Dec 2012 16:16:04 +0000
Subject: [Freeswitch-users] [Confidential] - uuid_broadcast
Message-ID: <78990CE7CC964442A7C2CA5F4689695EA89D0615@BARXB0003.UnifyBusiness.local>

Hi all,

*         I have added further comments to this issue (case FS-4884). Please close the case as it's resolved now (Please see comments on the JIRA). Thank you all for your help.
Nuwan.
This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.
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From mickstevens at yahoo.com  Wed Dec  5 19:22:31 2012
From: mickstevens at yahoo.com (Mick Stevens)
Date: Wed, 5 Dec 2012 08:22:31 -0800 (PST)
Subject: [Freeswitch-users] Explanation of rtp_audio_in / out fields in
	XML CDR's?
In-Reply-To: <CAFiqYu=sujtS0ZB91JPBMsSRJx5tUG8vn7JkMMPiJAVxtHH61A@mail.gmail.com>
References: <1354102130.98970.YahooMailNeo@web160802.mail.bf1.yahoo.com>
	<CAKzWOxUcpgaS29KWmSHg8FmMAraOVAa75i71sXdpbBC0S7EC_w@mail.gmail.com>
	<1354221331.77886.YahooMailNeo@web160803.mail.bf1.yahoo.com>
	<1354704865.29602.YahooMailNeo@web160804.mail.bf1.yahoo.com>
	<CAFiqYu=sujtS0ZB91JPBMsSRJx5tUG8vn7JkMMPiJAVxtHH61A@mail.gmail.com>
Message-ID: <1354724551.52202.YahooMailNeo@web160802.mail.bf1.yahoo.com>

Thank you Steve!

Additional replies still welcome...

Rgds, Mick
?


________________________________
 From: Steven Ayre <steveayre at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Wednesday, 5 December 2012, 15:20
Subject: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?
 

In vs Out are the 'listening' and 'talking' directions of the channel. They're not the same audio, and as such there's no reason they would match up. Couple that with sending/receiving before the other side, and network jitter/packet loss so that some packets arrive at different times or not at all.

As you point out using codecs can also affect them, as different codecs/ptimes will use different amounts of data/packets for the same time segment. Especially if the codec uses a VBR.

Even using the 'echo' app and matching codec settings they will differ because of timing issues.



On 5 December 2012 10:54, Mick Stevens <mickstevens at yahoo.com> wrote:

Hey Team,
>
>
>Sorry to go on about this, but really keen to understand these RTP fields in the XML CDR's better...
>
>
>1) At the risk of sounding too desperate... Please, please, please can someone explain to me what the?<rtp_audio_in_skip_packet_count> &?<rtp_audio_out_skip_packet_count> fields indicates?
>
>
>2) Also, for example in the extract below, why are the rtp_audio_in & rtp_audio_out bytes / packet counts not the same? Is this a red herring based on some codec interworking issue or similar, an indication of packet loss or expected/normal behaviour?
>
>
>? ? <rtp_audio_in_raw_bytes>436848</rtp_audio_in_raw_bytes>
>? ? <rtp_audio_in_media_bytes>436536</rtp_audio_in_media_bytes>
>? ? <rtp_audio_in_packet_count>2562</rtp_audio_in_packet_count>
>? ? <rtp_audio_in_media_packet_count>2538</rtp_audio_in_media_packet_count>
>? ? <rtp_audio_in_skip_packet_count>1904</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>24</rtp_audio_in_cng_packet_count>
>? ? <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
>? ? <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
>? ? <rtp_audio_out_raw_bytes>624570</rtp_audio_out_raw_bytes>
>? ? <rtp_audio_out_media_bytes>624360</rtp_audio_out_media_bytes>
>? ? <rtp_audio_out_packet_count>3645</rtp_audio_out_packet_count>
>? ? <rtp_audio_out_media_packet_count>3630</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>15</rtp_audio_out_cng_packet_count>
>
>
>Please help!?
>?
>?
>Rgds, Mick
>Tel/SMS. +44(0)7967 594432
>Fax. +44(0)7053 452429
>
>Email/IM. mickstevens at yahoo.com
>Skype: mick_stevens
>www.facebook.com/mickstevens
>www.twitter.com/mickstevens
>
>
>
>
>________________________________
> From: Mick Stevens <mickstevens at yahoo.com>
>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Thursday, 29 November 2012, 20:35
>
>Subject: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?
> 
>
>
>Hi Michael & ?Co,
>
>
>Many thanks for the prompt & positive response! Yes, more than happy to collate feedback from the global team & update the wiki accordingly (I'd like to be able to document what the field values between the > < indicate as well as just explain the field names if possible...) +anything else I can do to to contribute to the project...?
>
>
>Yes, now the wiki is back up I've managed to work out that cng_packet = comfort noise generation! Also from the wiki, possibly that the jb in?rtp_audio_in_jb_packet_count &?rtp_audio_in_largest_jb_size = jitter buffer? (apologies if everybody else already knows this!).
>
>
>To provide the background to my original enquiry, I'm trying to identify if the rtp_audio_in_skip_packet_count in the following XML CDR extract is indicative of packet loss, or perhaps more accurately noticeable audio loss (as the packets haven't been lost, just ignored/"skipped"?: If anybody knows please speak up!?
>
>
>? ? <rtp_audio_in_raw_bytes>1565066</rtp_audio_in_raw_bytes>
>? ? <rtp_audio_in_media_bytes>1564856</rtp_audio_in_media_bytes>
>? ? <rtp_audio_in_packet_count>9113</rtp_audio_in_packet_count>
>? ? <rtp_audio_in_media_packet_count>9098</rtp_audio_in_media_packet_count>
>? ? <rtp_audio_in_skip_packet_count>1609</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>15</rtp_audio_in_cng_packet_count>
>? ? <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count>
>? ? <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
>
>
>Thank you in anticipation of enlightenment!
>
>
>#FreeSWITCH
>?
>Rgds, Mick
>Tel/SMS. +44(0)7967 594432
>Fax. +44(0)7053 452429
>
>Email/IM. mickstevens at yahoo.com
>Skype: mick_stevens
>www.facebook.com/mickstevens
>www.twitter.com/mickstevens
>
>
>
>
>________________________________
> From: Michael Collins <msc at freeswitch.org>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Thursday, 29 November 2012, 19:14
>Subject: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?
> 
>
>Mick,
>
>This data is definitely not on the wiki - or anywhere else that I can see. I think we can crowdsource this to get the info collected and then add it to the mod_xml_curl wiki page. For the record, here's a quick dump of the fields that I took from an XML CDR:
>
>??? <rtp_audio_in_raw_bytes>10664</rtp_audio_in_raw_bytes>
>??? <rtp_audio_in_media_bytes>8772</rtp_audio_in_media_bytes>
>??? <rtp_audio_in_packet_count>62</rtp_audio_in_packet_count>
>??? <rtp_audio_in_media_packet_count>51</rtp_audio_in_media_packet_count>
>??? <rtp_audio_in_skip_packet_count>18</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>11</rtp_audio_in_flush_packet_count>
>??? <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size>
>??? <rtp_audio_out_raw_bytes>11524</rtp_audio_out_raw_bytes>
>??? <rtp_audio_out_media_bytes>11524</rtp_audio_out_media_bytes>
>??? <rtp_audio_out_packet_count>67</rtp_audio_out_packet_count>
>??? <rtp_audio_out_media_packet_count>67</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>
>
>I think raw_bytes, media_bytes, packet_count, and media_packet_count are self-explanatory. I think cng_packet_count is probably self-explanatory too. My question on dtmf_packet_count would be whether it's only for RFC2833 packets (I suspect yes, but would like confirmation). 
>
>If anyone knows these please reply to this email and Mick and I will get them documented on the wiki (right Mick? ;)
>
>rtp_audio_in_skip_packet_count
>rtp_audio_out_skip_packet_count
>rtp_audio_in_jb_packet_count
>rtp_audio_in_flush_packet_count
>rtp_audio_in_largest_jb_size
>
>Thanks all!
>
>-MC
>
>
>On Wed, Nov 28, 2012 at 3:28 AM, Mick Stevens <mickstevens at yahoo.com> wrote:
>
>Hi Folks,
>>
>>
>>I'm trying to use FS XML CDR's to diagnose historic audio problems. I think I can work out some of the rtp_audio_in / out fields (raw bytes & media bytes being nearly equal looks like a good sign) but am wondering about the skip, cng & flush fields for example?
>>
>>
>>I have tried Googling this & can find evidence of other people having asked this question but not of the answer. I have also checked my FS 106 & Cookbook book's without success. The wiki appears to be down at the moment so my apologies if the answer lies there.
>>
>>
>>I know how to do this in real time using wireshark etc but am interested in being able to do some analysis on historic problems reported by customers that aren't willing/able to replicate the problem in order for a protocol trace to be captured.
>>
>>
>>Any help much appreciated!
>>?
>>Rgds, Mick
>>Tel/SMS. +44(0)7967 594432
>>Fax. +44(0)7053 452429
>>
>>Email/IM. mickstevens at yahoo.com
>>Skype: mick_stevens
>>www.facebook.com/mickstevens
>>www.twitter.com/mickstevens
>>
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting 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
>>
>>
>
>
>-- 
>Michael S Collins
>Twitter: @mercutioviz
>http://www.FreeSWITCH.org
>http://www.ClueCon.com
>http://www.OSTAG.org
>
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting 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
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>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>FreeSWITCH-users mailing list
>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
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
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From bdfoster at endigotech.com  Wed Dec  5 19:30:46 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 5 Dec 2012 11:30:46 -0500
Subject: [Freeswitch-users] Call disconnect upon re-invite - FS tears
	down call
In-Reply-To: <CAKzWOxUg_dvt4V84q+DVTiqJV0F2mCv=8U-z5qheC7CX-oMiKQ@mail.gmail.com>
References: <017f01cdb138$e00325c0$a0097140$@com>
	<CAFVUaV8TwiNqean_gs1EbNPRXjE=Cv+8Us7-c3yWG_n00NduGg@mail.gmail.com>
	<15AD3C94-1640-486D-B1C5-B961EB30F35B@freeswitch.org>
	<CAMbW9o+9SpGpTzj5JVQajx-UO3WER4gxqCd+CG0aydOLr2UPUQ@mail.gmail.com>
	<EB6C0153-B6EC-4B42-A3E8-893324C07BAB@endigotech.com>
	<CAKzWOxUg_dvt4V84q+DVTiqJV0F2mCv=8U-z5qheC7CX-oMiKQ@mail.gmail.com>
Message-ID: <679B6A38-7152-4BFD-A369-979C20AE2839@endigotech.com>

Wanna do something with this?



Sent from my iPhone

On Dec 4, 2012, at 4:59 PM, Michael Collins <msc at freeswitch.org> wrote:

> Okay, this is now on the wiki: http://wiki.freeswitch.org/wiki/File:Yunfj.jpeg
> We just need a good place for it. :P
> -MC
> 
> On Tue, Dec 4, 2012 at 1:34 PM, Brian Foster <bdfoster at endigotech.com> wrote:
>> <image.jpeg>
>> 
>> Sent from my iPhone
>> 
>> On Dec 4, 2012, at 10:50 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk> wrote:
>> 
>>> I was half expecting the meme image that goes with that...
>>> 
>>> On 4 December 2012 15:15, Brian West <brian at freeswitch.org> wrote:
>>>> Y U NO FILE JIRA?
>>>> --
>>>> 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
>>>> ISN: 410*543
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> On Oct 23, 2012, at 11:32 AM, Yiftach Golan <yiftah at choochee.com> wrote:
>>>> 
>>>> >
>>>> > You did not send the initial invite in the Jira so I do not know if the initial request constructed correctly
>>>> > but judging by the code it looks like your request is still pending when the second invite arrives
>>>> >
>>>> > Thanks,
>>>> > Yiftach.
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>> 
>>> 
>>> 
>>> -- 
>>> 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
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 msc at freeswitch.org  Wed Dec  5 20:05:12 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Dec 2012 09:05:12 -0800
Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today
Message-ID: <CAKzWOxUeDLWVF23J_9RZR+KfE0EQmY4koDw2DpNhWKz9F1G0-g@mail.gmail.com>

Hello folks,

We'll be having our conference call in just about one hour. The agenda page
is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_05

We'll be discussing some interesting facts about the FreeSWITCH XML
pre-processor and also an object lesson in how to use the source code to
answer a question and then put it on the wiki. Oh, and we have an update on
the ClueCon videos!

Talk to you soon.

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec  5 20:08:17 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Dec 2012 09:08:17 -0800
Subject: [Freeswitch-users] Displacing ringback audio during bridge -
 only first displacement heard
In-Reply-To: <50BEA565.1070508@gmail.com>
References: <50BEA565.1070508@gmail.com>
Message-ID: <CAKzWOxVM3KUeq6CbsBUJguE1ibnyDebC5cB1UeveYAc9sR8Z7g@mail.gmail.com>

Scott,

If you manually try to do displacements instead of using the sched_api,
does it exhibit the same symptom? Either way it sounds like a possible but.
Please file a ticket at jira.freeswitch.org so that the devs can have a
look. Also, be sure to test this on latest git HEAD to confirm that it
hasn't already been fixed.

-MC

On Tue, Dec 4, 2012 at 5:37 PM, Scott Beil <scobei001 at gmail.com> wrote:

> I am attempting to interrupt the ringback audio at regular intervals
> with announcements while a bridge is in progress. Everything is going
> fine - the far end is ringing, the ringback audio is playing, the first
> displacement is heard, but subsequent displacements are not.
>
> I am using an outbound ESL connection, FreeSWITCH version 1.2.3.
>
> First, the ringback audio is set:
>
>          esl_execute(handle, "set",
>
> "ringback=file_string://'../sounds/music/8000/suite-espanola-op-47-leyenda.wav'",NULL);
>
> Next, the bridge is initiated:
>
>          esl_execute(handle, "bridge", "user/1001",NULL);
>
> Now, a displacement is scheduled for 5  seconds in the future:
>
>          esl_send_recv(handle, "api sched_api +5 none uuid_displace
> <uuid> start digits/1.wav");
>
> After each MEDIA_BUG_STOP event is received, another displacement is
> scheduled.
>
> The log file shows successful attempts are being made to play the
> displacement audio:
>
> 66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:39.161496 [DEBUG]
> switch_core_media_bug.c:506 Attaching BUG to
> sofia/internal/1000 at 192.168.1.136
> 2012-12-04 18:25:39.161496 [DEBUG] mod_commands.c:3894 Command
> uuid_displace(66017ce6-23c8-4df1-accc-6275eed8fc45 start digits/1.wav):
> +OK Success
>
> 2012-12-04 18:25:39.161496 [DEBUG] switch_scheduler.c:138 Deleting task
> 17 sched_api_function (none)
> 66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:39.641496 [DEBUG]
> switch_core_media_bug.c:724 Removing BUG from
> sofia/internal/1000 at 192.168.1.136
> 2012-12-04 18:25:39.761496 [DEBUG] switch_scheduler.c:214 Added task 18
> sched_api_function (none) to run at 1354667144
> 66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:44.161496 [DEBUG]
> switch_core_media_bug.c:506 Attaching BUG to
> sofia/internal/1000 at 192.168.1.136
> 2012-12-04 18:25:44.161496 [DEBUG] mod_commands.c:3894 Command
> uuid_displace(66017ce6-23c8-4df1-accc-6275eed8fc45 start digits/1.wav):
> +OK Success
>
> 2012-12-04 18:25:44.161496 [DEBUG] switch_scheduler.c:138 Deleting task
> 18 sched_api_function (none)
> 66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:44.181496 [DEBUG]
> switch_core_media_bug.c:724 Removing BUG from
> sofia/internal/1000 at 192.168.1.136
> 2012-12-04 18:25:44.341496 [DEBUG] switch_scheduler.c:214 Added task 19
> sched_api_function (none) to run at 1354667149
> 66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:49.161496 [DEBUG]
> switch_core_media_bug.c:506 Attaching BUG to
> sofia/internal/1000 at 192.168.1.136
> 2012-12-04 18:25:49.161496 [DEBUG] mod_commands.c:3894 Command
> uuid_displace(66017ce6-23c8-4df1-accc-6275eed8fc45 start digits/1.wav):
> +OK Success
>
> 2012-12-04 18:25:49.161496 [DEBUG] switch_scheduler.c:138 Deleting task
> 19 sched_api_function (none)
> 66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:49.181496 [DEBUG]
> switch_core_media_bug.c:724 Removing BUG from
> sofia/internal/1000 at 192.168.1.136
> 2012-12-04 18:25:49.341496 [DEBUG] switch_scheduler.c:214 Added task 20
> sched_api_function (none) to run at 1354667154
> 66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:54.161496 [DEBUG]
> switch_core_media_bug.c:506 Attaching BUG to
> sofia/internal/1000 at 192.168.1.136
> 2012-12-04 18:25:54.161496 [DEBUG] mod_commands.c:3894 Command
> uuid_displace(66017ce6-23c8-4df1-accc-6275eed8fc45 start digits/1.wav):
> +OK Success
>
> Any guidance would be appreciated.
>
> Thanks,
>
> Scott
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec  5 20:12:50 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Dec 2012 09:12:50 -0800
Subject: [Freeswitch-users] sip registration
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2339B39@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2339A49@Mail-Kilo.squay.com>
	<CAFiqYu=cpMkTxdE99smQBM6NgitNQ5ZpePwfSYr_R23dpj7Hng@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339B39@Mail-Kilo.squay.com>
Message-ID: <CAKzWOxUW21XZte0AoVDqTtC6vkJ4FmA3sbP5LgJVK5bj4+1Sbw@mail.gmail.com>

If you're talking about the user configuration then yes, you could create
an "email" parameter or variable and access it with the user_data API
command.
-MC

On Wed, Dec 5, 2012 at 1:35 AM, Archana Venugopan <a.venugopan at mundio.com>wrote:

>  Hi,****
>
> In that case can I have 1 more column say e-mail and can this e-mail be
> checked in DB instead of checking reg_user(?100?)? Is that feasible?****
>
> Also which code should be changed any idea please?****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 04 December 2012 19:51
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
> ** **
>
> You can have a user 'ana' in the domain 'gmail.com'. Though using someone
> else's domain as local in your FS setup may not be a good idea.****
>
> You can't have a @ in the username itself (per the SIP standard, not
> limited to FreeSWITCH).****
>
> ** **
>
> On 4 December 2012 18:00, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
>  ****
>
> Currently we register authentication name as say ?100? in sip
> registration, this comes to freeswitch and it will check in our DB for 100
> and if its present then registrations would be successful. ****
>
>  ****
>
> freeswitch at internal> show registrations****
>
>
> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
> ****
>
> 100,fsfailover.uk01.com
> ,e4969067f9a8c098,sofia/internal/sip:100 at 192.168.2.234:5060
> ;transport=udp,1354638871,192.168.2.234,5060,udp,squay-laptop-1.squay.com*
> ***
>
>  ****
>
> I want to change this 100 to some e-mail address, so instead of 100 it
> will be something like ?ana at gmail.com?. Can we do this? While coming to
> freeswitch whether there would be any issues?****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From a.venugopan at mundio.com  Wed Dec  5 20:30:34 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Wed, 5 Dec 2012 17:30:34 +0000
Subject: [Freeswitch-users] sip registration
In-Reply-To: <CAKzWOxUW21XZte0AoVDqTtC6vkJ4FmA3sbP5LgJVK5bj4+1Sbw@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF2339A49@Mail-Kilo.squay.com>
	<CAFiqYu=cpMkTxdE99smQBM6NgitNQ5ZpePwfSYr_R23dpj7Hng@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339B39@Mail-Kilo.squay.com>
	<CAKzWOxUW21XZte0AoVDqTtC6vkJ4FmA3sbP5LgJVK5bj4+1Sbw@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF2339D42@Mail-Kilo.squay.com>

Hi,
Thanks for the information. But sorry, how to access user_data API command.

Am not clear on the flow. Once we register domain and usernumber in sip what exactly happens? Which script picks up this domain and username and validates with our database?
Could you please provide me with an overview.

Many thanks

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 05 December 2012 17:13
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

If you're talking about the user configuration then yes, you could create an "email" parameter or variable and access it with the user_data API command.
-MC
On Wed, Dec 5, 2012 at 1:35 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
In that case can I have 1 more column say e-mail and can this e-mail be checked in DB instead of checking reg_user('100')? Is that feasible?
Also which code should be changed any idea please?

Regards,
Archana

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 Steven Ayre
Sent: 04 December 2012 19:51
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

You can have a user 'ana' in the domain 'gmail.com<http://gmail.com>'. Though using someone else's domain as local in your FS setup may not be a good idea.
You can't have a @ in the username itself (per the SIP standard, not limited to FreeSWITCH).

On 4 December 2012 18:00, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

Currently we register authentication name as say '100' in sip registration, this comes to freeswitch and it will check in our DB for 100 and if its present then registrations would be successful.

freeswitch at internal> show registrations
reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
100,fsfailover.uk01.com<http://fsfailover.uk01.com>,e4969067f9a8c098,sofia/internal/sip:100 at 192.168.2.234:5060;transport=udp,1354638871,192.168.2.234,5060,udp,squay-laptop-1.squay.com<http://squay-laptop-1.squay.com>

I want to change this 100 to some e-mail address, so instead of 100 it will be something like 'ana at gmail.com<mailto:ana at gmail.com>'. Can we do this? While coming to freeswitch whether there would be any issues?

Regards,
Archana


_________________________________________________________________________
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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

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From bdfoster at endigotech.com  Wed Dec  5 20:59:39 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 5 Dec 2012 12:59:39 -0500
Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today
In-Reply-To: <CAKzWOxUeDLWVF23J_9RZR+KfE0EQmY4koDw2DpNhWKz9F1G0-g@mail.gmail.com>
References: <CAKzWOxUeDLWVF23J_9RZR+KfE0EQmY4koDw2DpNhWKz9F1G0-g@mail.gmail.com>
Message-ID: <9BDCA7E1-9C70-4131-958C-A2A159E3BC0D@endigotech.com>

Can't unmute and I am not able to get in front of IRC...

Sent from my iPhone

On Dec 5, 2012, at 12:05 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hello folks,
> 
> We'll be having our conference call in just about one hour. The agenda page is here:
> 
> http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_05
> 
> We'll be discussing some interesting facts about the FreeSWITCH XML pre-processor and also an object lesson in how to use the source code to answer a question and then put it on the wiki. Oh, and we have an update on the ClueCon videos!
> 
> Talk to you soon.
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 msc at freeswitch.org  Wed Dec  5 21:06:59 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Dec 2012 10:06:59 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Subject: Windows
	debugging tools
Message-ID: <CAKzWOxVKEBkaACC2q=SV9izD=ymaoUcTQxM0CyvCKiWTNUxSLg@mail.gmail.com>

Hey all!

I'm trying to gauge the interest level in this subject. We have an
experienced Windows user who is willing to share with us a lot of his
hard-earned knowledge when it comes to debugging crashes and such in a
Windows environment. If you are interested in hearing about this subject
please respond. (Only respond if you are interested - we don't need to hear
from those who are not interested.)

Thanks!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From 8f27e956 at gmail.com  Wed Dec  5 21:12:01 2012
From: 8f27e956 at gmail.com (S. Scott)
Date: Wed, 5 Dec 2012 13:12:01 -0500
Subject: [Freeswitch-users] NATIVE SQL ERROR [no such table:
	fifo_outbound]
In-Reply-To: <CAJs+-5p35CSzXBpCKyLfyzrCGEp_au=NPAUru39EgWLAOdeM=g@mail.gmail.com>
References: <CA+Y7Lf7GzFpacZ_spvJg+jK8+s+4OoeeH0szMmRP2FtHfeOcsw@mail.gmail.com>
	<924695E4-6690-4264-8E42-CB4573698E8F@gmail.com>
	<CAJs+-5p35CSzXBpCKyLfyzrCGEp_au=NPAUru39EgWLAOdeM=g@mail.gmail.com>
Message-ID: <-6411533123604268944@unknownmsgid>

Sorry if i misunderstand the git branches, but are you suggesting that
i have to leave the -stable branch to remedy a regression-type -stable
issue.

Thanks,

On 2012-12-05, at 10:21, Seven Du <dujinfang at gmail.com> wrote:

> update to git head, it should be fixed.
>
> On Wed, Dec 5, 2012 at 9:50 PM, Nelson Camargo <bigx333 at gmail.com> wrote:
>> Having same issue, git aa08157 2012-11-30 04:53:03Z)
>>
>> On 05 Dec 2012, at 1:43 PM, Scott wrote:
>>
>> Just did a 'make current' to update to 1.2.5.2 (from 1.2.5.1).  Now, upon
>> restarting freeswitch, it logs the following,
>>
>> ...skipping...
>> SQL [Enabled]
>> 2012-12-05 06:29:58.746676 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR [no
>> such table: fifo_outbound]
>> update fifo_outbound set
>> start_time=0,stop_time=0,ring_count=0,use_count=0,outbound_call_count=0,outbound_fail_count=0
>> where static=0
>> 2012-12-05 06:29:58.887575 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR [no
>> such table: fifo_outbound]
>> delete from fifo_outbound where static=1 and hostname='fs1.soho.local'
>>
>>
>> In so far as I could see, the 'make' finished successfully without apparent
>> errors.
>>
>> Thoughts & suggestions welcome.
>>
>> With 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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


From 8f27e956 at gmail.com  Wed Dec  5 21:42:03 2012
From: 8f27e956 at gmail.com (S. Scott)
Date: Wed, 5 Dec 2012 13:42:03 -0500
Subject: [Freeswitch-users] NATIVE SQL ERROR [no such table:
	fifo_outbound]
In-Reply-To: <CAJs+-5p35CSzXBpCKyLfyzrCGEp_au=NPAUru39EgWLAOdeM=g@mail.gmail.com>
References: <CA+Y7Lf7GzFpacZ_spvJg+jK8+s+4OoeeH0szMmRP2FtHfeOcsw@mail.gmail.com>
	<924695E4-6690-4264-8E42-CB4573698E8F@gmail.com>
	<CAJs+-5p35CSzXBpCKyLfyzrCGEp_au=NPAUru39EgWLAOdeM=g@mail.gmail.com>
Message-ID: <-5406475373508000333@unknownmsgid>

Overheard In the Wednesday teleconference, that a payload of changes
to 1.2-stable branch is on a less-than-24-hour to be released window,
that bumps 1.2.5 and that includes SQL fixes.

Is this sql error issue covered in this imminent bump tp 1.2-stable.


On 2012-12-05, at 10:21, Seven Du <dujinfang at gmail.com> wrote:

> update to git head, it should be fixed.
>
> On Wed, Dec 5, 2012 at 9:50 PM, Nelson Camargo <bigx333 at gmail.com> wrote:
>> Having same issue, git aa08157 2012-11-30 04:53:03Z)
>>
>> On 05 Dec 2012, at 1:43 PM, Scott wrote:
>>
>> Just did a 'make current' to update to 1.2.5.2 (from 1.2.5.1).  Now, upon
>> restarting freeswitch, it logs the following,
>>
>> ...skipping...
>> SQL [Enabled]
>> 2012-12-05 06:29:58.746676 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR [no
>> such table: fifo_outbound]
>> update fifo_outbound set
>> start_time=0,stop_time=0,ring_count=0,use_count=0,outbound_call_count=0,outbound_fail_count=0
>> where static=0
>> 2012-12-05 06:29:58.887575 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR [no
>> such table: fifo_outbound]
>> delete from fifo_outbound where static=1 and hostname='fs1.soho.local'
>>
>>
>> In so far as I could see, the 'make' finished successfully without apparent
>> errors.
>>
>> Thoughts & suggestions welcome.
>>
>> With 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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


From steveayre at gmail.com  Wed Dec  5 23:18:06 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 5 Dec 2012 20:18:06 +0000
Subject: [Freeswitch-users] Release tarball problems
Message-ID: <CAFiqYuk3rAog5fweKp7bZNKh-Rf6hupMpu-JsRPAgf8J2PAx+Q@mail.gmail.com>

The freeswitch-1.2.5.2.tar.bz2 tarball is missing from files.freeswitch.org,
are the developers aware of this?

The versions of debian packages built from the packages are also incorrect
- debian/changelog's last entry is 1.2~rc2-1.

The advice in debian/README.source suggests adding a new changelog entry to
set custom versions for builds from Git. For git versions that seems fine
(after all we wouldn't want to have to update the file for every commit),
but it seems that for every stable version release it would be better to
set the version within the tarball's debian/changelog?
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From regis.freeswitch.org at tornad.net  Wed Dec  5 22:28:48 2012
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Wed, 5 Dec 2012 20:28:48 +0100
Subject: [Freeswitch-users] how to Ignore DTMF shorter than Fixed duration ?
Message-ID: <CAHWtr1TbtvoxrxbCEY6y8r_rKgtsh2zHhXmeZYRA0+6P0dKGUQ@mail.gmail.com>

Hi,

We have a bugged provider that send us throw RTP wrong DTMF during call. It
seems that wrong DTMF are shorter than 1200ms (about 90%) so I want to try
to ignore them.
For the moment, FS catch them and send it back to the bridged side, boring
user.

switch_rtp.c:3410 RTP RECV DTMF C:552



Does min-dtmf-duration will make FS ignore them and not RECV them ?
I saw the DTMF in "normal" RTP packet, but no digit was pressed in user
side.

It's commented in my FS switch.conf.xml, how can I see the current value ?
<!--<param name="min-dtmf-duration" value="400"/>-->


Any other idea ?

thanks
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From msc at freeswitch.org  Wed Dec  5 23:34:18 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Dec 2012 12:34:18 -0800
Subject: [Freeswitch-users] sip registration
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF2339D42@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2339A49@Mail-Kilo.squay.com>
	<CAFiqYu=cpMkTxdE99smQBM6NgitNQ5ZpePwfSYr_R23dpj7Hng@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339B39@Mail-Kilo.squay.com>
	<CAKzWOxUW21XZte0AoVDqTtC6vkJ4FmA3sbP5LgJVK5bj4+1Sbw@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339D42@Mail-Kilo.squay.com>
Message-ID: <CAKzWOxVMWfR6vk8ODKkND1Tjxj-d5VFYn9U+dDoNyB_t2dE05w@mail.gmail.com>

When an event that requires a user lookup takes place then the system will
look in the XML user directory unless it has been configured to look
somewhere else. The other places to look are usually:
mod_xml_curl
One of the language like Lua, Perl, Python

If it's xml_curl then FS will do a POST to your web server in hopes of
receiving back the necessary XML for the given user. It would be up to you
to have your web server handle the request, poll the database, then format
and return the XML data. See this wiki
page<http://wiki.freeswitch.org/wiki/Xml_curl#How_do_I_configure_mod_xml_curl.3F>for
more info on xml curl.

If it's a language then you'll have a "binding" in the conf file for the
language that will handle the lookup. Again, your script will need to
handle the communication with your database. See this wiki
page<http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration>for
more information.

Hope this helps.
-MC

On Wed, Dec 5, 2012 at 9:30 AM, Archana Venugopan <a.venugopan at mundio.com>wrote:

>  Hi,****
>
> Thanks for the information. But sorry, how to access user_data API command.
> ****
>
> ** **
>
> Am not clear on the flow. Once we register domain and usernumber in sip
> what exactly happens? Which script picks up this domain and username and
> validates with our database?****
>
> Could you please provide me with an overview. ****
>
> ** **
>
> Many thanks****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 05 December 2012 17:13
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
> ** **
>
> If you're talking about the user configuration then yes, you could create
> an "email" parameter or variable and access it with the user_data API
> command.
> -MC****
>
> On Wed, Dec 5, 2012 at 1:35 AM, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
> In that case can I have 1 more column say e-mail and can this e-mail be
> checked in DB instead of checking reg_user(?100?)? Is that feasible?****
>
> Also which code should be changed any idea please?****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 04 December 2012 19:51
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
>  ****
>
> You can have a user 'ana' in the domain 'gmail.com'. Though using someone
> else's domain as local in your FS setup may not be a good idea.****
>
> You can't have a @ in the username itself (per the SIP standard, not
> limited to FreeSWITCH).****
>
>  ****
>
> On 4 December 2012 18:00, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
>  ****
>
> Currently we register authentication name as say ?100? in sip
> registration, this comes to freeswitch and it will check in our DB for 100
> and if its present then registrations would be successful. ****
>
>  ****
>
> freeswitch at internal> show registrations****
>
>
> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
> ****
>
> 100,fsfailover.uk01.com
> ,e4969067f9a8c098,sofia/internal/sip:100 at 192.168.2.234:5060
> ;transport=udp,1354638871,192.168.2.234,5060,udp,squay-laptop-1.squay.com*
> ***
>
>  ****
>
> I want to change this 100 to some e-mail address, so instead of 100 it
> will be something like ?ana at gmail.com?. Can we do this? While coming to
> freeswitch whether there would be any issues?****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
> ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec  5 23:35:52 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Dec 2012 12:35:52 -0800
Subject: [Freeswitch-users] Release tarball problems
In-Reply-To: <CAFiqYuk3rAog5fweKp7bZNKh-Rf6hupMpu-JsRPAgf8J2PAx+Q@mail.gmail.com>
References: <CAFiqYuk3rAog5fweKp7bZNKh-Rf6hupMpu-JsRPAgf8J2PAx+Q@mail.gmail.com>
Message-ID: <CAKzWOxVPRqs++hcjPkRryuN2nggRtQvE+cdtUu+b9Rg9LqzNxw@mail.gmail.com>

Thanks for the heads up. We'll have Ken & co. take care of the tarball.
I'll have to defer on the Deb question.
-MC

On Wed, Dec 5, 2012 at 12:18 PM, Steven Ayre <steveayre at gmail.com> wrote:

> The freeswitch-1.2.5.2.tar.bz2 tarball is missing from
> files.freeswitch.org, are the developers aware of this?
>
> The versions of debian packages built from the packages are also incorrect
> - debian/changelog's last entry is 1.2~rc2-1.
>
> The advice in debian/README.source suggests adding a new changelog entry
> to set custom versions for builds from Git. For git versions that seems
> fine (after all we wouldn't want to have to update the file for every
> commit), but it seems that for every stable version release it would be
> better to set the version within the tarball's debian/changelog?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jpablolorenzetti at hotmail.com  Wed Dec  5 23:36:42 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Wed, 5 Dec 2012 20:36:42 +0000
Subject: [Freeswitch-users] get a variable at hunting phase that is set at
	executing phase
Message-ID: <DUB002-W564CE266C94AB4E76E0A98D9460@phx.gbl>

Guys, i have a script the sets some variables the i m supposed to use to decide if a condition meets, i have tried to use the execute_extension app but it does not work, this is the dialplan:

<extension name="huawei_csoft_gw_outbound_call_international">        <condition field="${int}" expression="1"/>        <condition field="destination_number" expression="^00[\d]{7,}?$">                <action application="set" data="effective_caller_id_number=280${caller_id_number}"/>                <action application="bridge" data="sofia/gateway/huawei_csoft/${destination_number}"/>        </condition>   </extension>
the script that sets the var ${int} is called "get_user_profile.lua", i have placed it all over the place and cant make this work.i have used "execute_extension" in the following way:
<extension name="is_international_allowed">        <condition field="destination_number" expression="^iia$">                <action application="lua" data="get_user_profile.lua" />         </condition>    </extension>
    <extension name="huawei_csoft_gw_outbound_call_international">        <condition field="destination_number" expression="^00[\d]{7,}?$">                <action application="execute_extension" data="iia XML smart_private"/> -->        </condition>        <condition field="${int}" expression="1"/>        <condition field="destination_number" expression="^00[\d]{7,}?$">                <action application="set" data="effective_caller_id_number=280${caller_id_number}"/>                <action application="bridge" data="sofia/gateway/huawei_csoft/${destination_number}"/>        </condition>    </extension>
i know this is simple but i m missing something here, i appreciate any ideas. thanks!!
 		 	   		  
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From msc at freeswitch.org  Wed Dec  5 23:37:23 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Dec 2012 12:37:23 -0800
Subject: [Freeswitch-users] how to Ignore DTMF shorter than Fixed
	duration ?
In-Reply-To: <CAHWtr1TbtvoxrxbCEY6y8r_rKgtsh2zHhXmeZYRA0+6P0dKGUQ@mail.gmail.com>
References: <CAHWtr1TbtvoxrxbCEY6y8r_rKgtsh2zHhXmeZYRA0+6P0dKGUQ@mail.gmail.com>
Message-ID: <CAKzWOxXQSj9hPSoSmH9pQ19J=+eU5qM+8aNO51BoWzzN_gKj_w@mail.gmail.com>

Are they sending those digits inband or with RFC2833?
-MC

On Wed, Dec 5, 2012 at 11:28 AM, Regis M <regis.freeswitch.org at tornad.net>wrote:

> Hi,
>
> We have a bugged provider that send us throw RTP wrong DTMF during call.
> It seems that wrong DTMF are shorter than 1200ms (about 90%) so I want to
> try to ignore them.
> For the moment, FS catch them and send it back to the bridged side, boring
> user.
>
> switch_rtp.c:3410 RTP RECV DTMF C:552
>
>
>
> Does min-dtmf-duration will make FS ignore them and not RECV them ?
> I saw the DTMF in "normal" RTP packet, but no digit was pressed in user
> side.
>
> It's commented in my FS switch.conf.xml, how can I see the current value ?
> <!--<param name="min-dtmf-duration" value="400"/>-->
>
>
> Any other idea ?
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From krice at freeswitch.org  Wed Dec  5 23:43:25 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 05 Dec 2012 14:43:25 -0600
Subject: [Freeswitch-users] Release tarball problems
In-Reply-To: <CAFiqYuk3rAog5fweKp7bZNKh-Rf6hupMpu-JsRPAgf8J2PAx+Q@mail.gmail.com>
Message-ID: <CCE50E0D.616EC%krice@freeswitch.org>

Hey Steven it should be there let me find out why its not there

K


On 12/5/12 2:18 PM, "Steven Ayre" <steveayre at gmail.com> wrote:

> The freeswitch-1.2.5.2.tar.bz2 tarball is missing from files.freeswitch.org
> <http://files.freeswitch.org> , are the developers aware of this?
> 
> The versions of debian packages built from the packages are also incorrect -
> debian/changelog's last entry is?1.2~rc2-1.
> 
> The advice in debian/README.source suggests adding a new changelog entry to
> set custom versions for builds from Git. For git versions that seems fine
> (after all we wouldn't want to have to update the file for every commit), but
> it seems that for every stable?version?release it would be better to set the
> version within the tarball's debian/changelog?
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From avi at avimarcus.net  Wed Dec  5 23:50:05 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 5 Dec 2012 22:50:05 +0200
Subject: [Freeswitch-users] get a variable at hunting phase that is set
 at executing phase
In-Reply-To: <DUB002-W564CE266C94AB4E76E0A98D9460@phx.gbl>
References: <DUB002-W564CE266C94AB4E76E0A98D9460@phx.gbl>
Message-ID: <CALNZuw=8LV36NNZ=0_YKcd+RmYOjM7cnp_VuiYYrGyF=FzF3qA@mail.gmail.com>

Two options:
1) Transfer after running your script. It starts the dialplan over again.
2) inline="true" - that actually runs it right away. However, you can't
touch media at this point or you break the call routing. So I don't think
this option is allowed for lua. Or it might have been, give it a try.
If so, update the wiki here:
http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions

-Avi

On Wed, Dec 5, 2012 at 10:36 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com
> wrote:

> Guys, i have a script the sets some variables the i m supposed to use to
> decide if a condition meets, i have tried to use the execute_extension app
> but it does not work, this is the dialplan:
>
>
> <extension name="huawei_csoft_gw_outbound_call_international">
>         <condition field="${int}" expression="1"/>
>         <condition field="destination_number" expression="^00[\d]{7,}?$">
>                 <action application="set"
> data="effective_caller_id_number=280${caller_id_number}"/>
>                 <action application="bridge"
> data="sofia/gateway/huawei_csoft/${destination_number}"/>
>         </condition>
>    </extension>
>
> the script that sets the var ${int} is called "get_user_profile.lua", i
> have placed it all over the place and cant make this work.
> i have used "execute_extension" in the following way:
>
> <extension name="is_international_allowed">
>         <condition field="destination_number" expression="^iia$">
>                 <action application="lua" data="get_user_profile.lua" />
>         </condition>
>     </extension>
>
>     <extension name="huawei_csoft_gw_outbound_call_international">
>         <condition field="destination_number" expression="^00[\d]{7,}?$">
>                 <action application="execute_extension" data="iia XML
> smart_private"/> -->
>         </condition>
>         <condition field="${int}" expression="1"/>
>         <condition field="destination_number" expression="^00[\d]{7,}?$">
>                 <action application="set"
> data="effective_caller_id_number=280${caller_id_number}"/>
>                 <action application="bridge"
> data="sofia/gateway/huawei_csoft/${destination_number}"/>
>         </condition>
>     </extension>
>
> i know this is simple but i m missing something here, i appreciate any
> ideas. 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec  6 00:06:35 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Dec 2012 13:06:35 -0800
Subject: [Freeswitch-users] get a variable at hunting phase that is set
 at executing phase
In-Reply-To: <DUB002-W564CE266C94AB4E76E0A98D9460@phx.gbl>
References: <DUB002-W564CE266C94AB4E76E0A98D9460@phx.gbl>
Message-ID: <CAKzWOxWN+jY5NP8rNTa0m5nbW2NCW1bmCW6K_O-Xgr4Nv31qDw@mail.gmail.com>

It might be good for you to pastebin the debug log of the call flow. Use
pastebin.freeswitch.org and set the syntax highlight to FreeSWITCH Log.
Once that's there we will take a look and help you diagnose what is (and is
not) happening. Looking at the log is the best way to be sure that you know
what's really going on.

-MC

On Wed, Dec 5, 2012 at 12:36 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com
> wrote:

> Guys, i have a script the sets some variables the i m supposed to use to
> decide if a condition meets, i have tried to use the execute_extension app
> but it does not work, this is the dialplan:
>
>
> <extension name="huawei_csoft_gw_outbound_call_international">
>         <condition field="${int}" expression="1"/>
>         <condition field="destination_number" expression="^00[\d]{7,}?$">
>                 <action application="set"
> data="effective_caller_id_number=280${caller_id_number}"/>
>                 <action application="bridge"
> data="sofia/gateway/huawei_csoft/${destination_number}"/>
>         </condition>
>    </extension>
>
> the script that sets the var ${int} is called "get_user_profile.lua", i
> have placed it all over the place and cant make this work.
> i have used "execute_extension" in the following way:
>
> <extension name="is_international_allowed">
>         <condition field="destination_number" expression="^iia$">
>                 <action application="lua" data="get_user_profile.lua" />
>         </condition>
>     </extension>
>
>     <extension name="huawei_csoft_gw_outbound_call_international">
>         <condition field="destination_number" expression="^00[\d]{7,}?$">
>                 <action application="execute_extension" data="iia XML
> smart_private"/> -->
>         </condition>
>         <condition field="${int}" expression="1"/>
>         <condition field="destination_number" expression="^00[\d]{7,}?$">
>                 <action application="set"
> data="effective_caller_id_number=280${caller_id_number}"/>
>                 <action application="bridge"
> data="sofia/gateway/huawei_csoft/${destination_number}"/>
>         </condition>
>     </extension>
>
> i know this is simple but i m missing something here, i appreciate any
> ideas. 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From freeswitch at orresta.no-ip.com  Thu Dec  6 00:14:52 2012
From: freeswitch at orresta.no-ip.com (Jakob)
Date: Wed, 05 Dec 2012 22:14:52 +0100
Subject: [Freeswitch-users] Single sndcard multiple extensions?
In-Reply-To: <CAKzWOxXiEn+Ox=eJYfj-6ooEse48x8Q7YPsyi=oU_4YoN4KvNA@mail.gmail.com>
References: <50BE8980.3050900@orresta.no-ip.com>
	<CAKzWOxXiEn+Ox=eJYfj-6ooEse48x8Q7YPsyi=oU_4YoN4KvNA@mail.gmail.com>
Message-ID: <50BFB94C.4060803@orresta.no-ip.com>

After googeling for a day trying to accomplish this using alsa or jack i 
found a sample implementation using streams in freeswitch that does the 
exact right thing.

http://fisheye.freeswitch.org/browse/~raw,r=65b231f5a4450b903c0975ab1c8c6d470c0ab67f/freeswitch.git/conf/autoload_configs/portaudio.conf.xml 
<http://fisheye.freeswitch.org/browse/%7Eraw,r=65b231f5a4450b903c0975ab1c8c6d470c0ab67f/freeswitch.git/conf/autoload_configs/portaudio.conf.xml>

Thought it might be of intresst to someone else as well.

12/05/12 00:50, Michael Collins skrev:
> As far as I know mod_portaudio does not support multiple channels...
> -MC
>
> On Tue, Dec 4, 2012 at 3:38 PM, Jakob <freeswitch at orresta.no-ip.com 
> <mailto:freeswitch at orresta.no-ip.com>> wrote:
>
>     Hi,
>
>     I'm trying to get right and left channels from a soundcard into
>     separate
>     alsa or portaudio endpoints.
>
>     right channel into ext 3001 and left channel into ext 3002
>
>     Any hints?
>
>     Regards Jakob
>
>     _________________________________________________________________________
>     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
>
>
>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 jpablolorenzetti at hotmail.com  Thu Dec  6 00:19:22 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Wed, 5 Dec 2012 21:19:22 +0000
Subject: [Freeswitch-users] get a variable at hunting phase that is set
 at executing phase
In-Reply-To: <CAKzWOxWN+jY5NP8rNTa0m5nbW2NCW1bmCW6K_O-Xgr4Nv31qDw@mail.gmail.com>
References: <DUB002-W564CE266C94AB4E76E0A98D9460@phx.gbl>,
	<CAKzWOxWN+jY5NP8rNTa0m5nbW2NCW1bmCW6K_O-Xgr4Nv31qDw@mail.gmail.com>
Message-ID: <DUB002-W1FD5F4B80221F2C412F50D9460@phx.gbl>

Thanks Avi for you answer, i m going to try that (i already have and the ${int} got lost somewhere but i ll try again)
Michael, thanks for your suggestion, i have put it in pastebin: http://pastebin.freeswitch.org/20293

thanks!

Date: Wed, 5 Dec 2012 13:06:35 -0800
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] get a variable at hunting phase that is set at executing phase

It might be good for you to pastebin the debug log of the call flow. Use pastebin.freeswitch.org and set the syntax highlight to FreeSWITCH Log. Once that's there we will take a look and help you diagnose what is (and is not) happening. Looking at the log is the best way to be sure that you know what's really going on.


-MC

On Wed, Dec 5, 2012 at 12:36 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:




Guys, i have a script the sets some variables the i m supposed to use to decide if a condition meets, i have tried to use the execute_extension app but it does not work, this is the dialplan:


<extension name="huawei_csoft_gw_outbound_call_international">
        <condition field="${int}" expression="1"/>        <condition field="destination_number" expression="^00[\d]{7,}?$">
                <action application="set" data="effective_caller_id_number=280${caller_id_number}"/>                <action application="bridge" data="sofia/gateway/huawei_csoft/${destination_number}"/>
        </condition>   </extension>
the script that sets the var ${int} is called "get_user_profile.lua", i have placed it all over the place and cant make this work.
i have used "execute_extension" in the following way:
<extension name="is_international_allowed">
        <condition field="destination_number" expression="^iia$">                <action application="lua" data="get_user_profile.lua" /> 
        </condition>    </extension>
    <extension name="huawei_csoft_gw_outbound_call_international">
        <condition field="destination_number" expression="^00[\d]{7,}?$">                <action application="execute_extension" data="iia XML smart_private"/> -->
        </condition>        <condition field="${int}" expression="1"/>
        <condition field="destination_number" expression="^00[\d]{7,}?$">                <action application="set" data="effective_caller_id_number=280${caller_id_number}"/>
                <action application="bridge" data="sofia/gateway/huawei_csoft/${destination_number}"/>        </condition>
    </extension>
i know this is simple but i m missing something here, i appreciate any ideas. 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



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

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

http://www.freeswitch.org




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com

http://www.OSTAG.org




_________________________________________________________________________
Professional FreeSWITCH Consulting 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 jpablolorenzetti at hotmail.com  Thu Dec  6 00:20:41 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Wed, 5 Dec 2012 21:20:41 +0000
Subject: [Freeswitch-users] get a variable at hunting phase that is set
 at executing phase
In-Reply-To: <CAKzWOxWN+jY5NP8rNTa0m5nbW2NCW1bmCW6K_O-Xgr4Nv31qDw@mail.gmail.com>
References: <DUB002-W564CE266C94AB4E76E0A98D9460@phx.gbl>,
	<CAKzWOxWN+jY5NP8rNTa0m5nbW2NCW1bmCW6K_O-Xgr4Nv31qDw@mail.gmail.com>
Message-ID: <DUB002-W1012DAC003806804CE12A86D9460@phx.gbl>

Thanks Avi for you answer, i m going to try that (i already have and the ${int} got lost somewhere but i ll try again)
Michael, thanks for your suggestion, i have put it in pastebin: http://pastebin.freeswitch.org/20293

thanks!

Date: Wed, 5 Dec 2012 13:06:35 -0800
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] get a variable at hunting phase that is set at executing phase

It might be good for you to pastebin the debug log of the call flow. Use pastebin.freeswitch.org and set the syntax highlight to FreeSWITCH Log. Once that's there we will take a look and help you diagnose what is (and is not) happening. Looking at the log is the best way to be sure that you know what's really going on.


-MC

On Wed, Dec 5, 2012 at 12:36 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:




Guys, i have a script the sets some variables the i m supposed to use to decide if a condition meets, i have tried to use the execute_extension app but it does not work, this is the dialplan:


<extension name="huawei_csoft_gw_outbound_call_international">
        <condition field="${int}" expression="1"/>        <condition field="destination_number" expression="^00[\d]{7,}?$">
                <action application="set" data="effective_caller_id_number=280${caller_id_number}"/>                <action application="bridge" data="sofia/gateway/huawei_csoft/${destination_number}"/>
        </condition>   </extension>
the script that sets the var ${int} is called "get_user_profile.lua", i have placed it all over the place and cant make this work.
i have used "execute_extension" in the following way:
<extension name="is_international_allowed">
        <condition field="destination_number" expression="^iia$">                <action application="lua" data="get_user_profile.lua" /> 
        </condition>    </extension>
    <extension name="huawei_csoft_gw_outbound_call_international">
        <condition field="destination_number" expression="^00[\d]{7,}?$">                <action application="execute_extension" data="iia XML smart_private"/> -->
        </condition>        <condition field="${int}" expression="1"/>
        <condition field="destination_number" expression="^00[\d]{7,}?$">                <action application="set" data="effective_caller_id_number=280${caller_id_number}"/>
                <action application="bridge" data="sofia/gateway/huawei_csoft/${destination_number}"/>        </condition>
    </extension>
i know this is simple but i m missing something here, i appreciate any ideas. 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



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

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

http://www.freeswitch.org




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com

http://www.OSTAG.org




_________________________________________________________________________
Professional FreeSWITCH Consulting 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 jpablolorenzetti at hotmail.com  Thu Dec  6 01:13:36 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Wed, 5 Dec 2012 22:13:36 +0000
Subject: [Freeswitch-users] get a variable at hunting phase that is set
 at executing phase
In-Reply-To: <CALNZuw=8LV36NNZ=0_YKcd+RmYOjM7cnp_VuiYYrGyF=FzF3qA@mail.gmail.com>
References: <DUB002-W564CE266C94AB4E76E0A98D9460@phx.gbl>,
	<CALNZuw=8LV36NNZ=0_YKcd+RmYOjM7cnp_VuiYYrGyF=FzF3qA@mail.gmail.com>
Message-ID: <DUB002-W780C942BE5F1409292EB6ED9460@phx.gbl>

Hi Avi, i followed idea 1, i modified the dial plan to look like this:
<extension name="huawei_csoft_gw_outbound_call_international">                <condition field="destination_number" expression="^(00[\d]{7,}?)$">                        <action application="lua" data="get_user_profile.lua" />                        <action application="set" data="number_to_dial=$1" />                        <action application="execute_extension" data="ldi"/>                </condition>        </extension>
        <extension name="check_and_send_international_call">                <condition field="destination_number" expression="^ldi$" />                <condition field="${int}" expression="1">                        <action application="set" data="effective_caller_id_number=280${caller_id_number}"/>                        <action application="bridge" data="sofia/gateway/huawei_csoft/${number_to_dial}"/>                </condition>        </extension>

but i dont understand why the previous alternative did not work, is there anyways to avoid the dial plan to be parsed twice ? thanks!


From: avi at avimarcus.net
Date: Wed, 5 Dec 2012 22:50:05 +0200
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] get a variable at hunting phase that is set at executing phase

Two options:1) Transfer after running your script. It starts the dialplan over again.2) inline="true" - that actually runs it right away. However, you can't touch media at this point or you break the call routing. So I don't think this option is allowed for lua. Or it might have been, give it a try.

If so, update the wiki here: http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions


-Avi
On Wed, Dec 5, 2012 at 10:36 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:





Guys, i have a script the sets some variables the i m supposed to use to decide if a condition meets, i have tried to use the execute_extension app but it does not work, this is the dialplan:



<extension name="huawei_csoft_gw_outbound_call_international">

        <condition field="${int}" expression="1"/>        <condition field="destination_number" expression="^00[\d]{7,}?$">

                <action application="set" data="effective_caller_id_number=280${caller_id_number}"/>                <action application="bridge" data="sofia/gateway/huawei_csoft/${destination_number}"/>

        </condition>   </extension>
the script that sets the var ${int} is called "get_user_profile.lua", i have placed it all over the place and cant make this work.

i have used "execute_extension" in the following way:
<extension name="is_international_allowed">

        <condition field="destination_number" expression="^iia$">                <action application="lua" data="get_user_profile.lua" /> 

        </condition>    </extension>
    <extension name="huawei_csoft_gw_outbound_call_international">

        <condition field="destination_number" expression="^00[\d]{7,}?$">                <action application="execute_extension" data="iia XML smart_private"/> -->

        </condition>        <condition field="${int}" expression="1"/>

        <condition field="destination_number" expression="^00[\d]{7,}?$">                <action application="set" data="effective_caller_id_number=280${caller_id_number}"/>

                <action application="bridge" data="sofia/gateway/huawei_csoft/${destination_number}"/>        </condition>

    </extension>
i know this is simple but i m missing something here, i appreciate any ideas. 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



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FreeSWITCH-users at lists.freeswitch.org

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

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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

FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From daniel.eiland at gmail.com  Thu Dec  6 01:16:46 2012
From: daniel.eiland at gmail.com (Daniel Eiland)
Date: Wed, 5 Dec 2012 17:16:46 -0500
Subject: [Freeswitch-users] Distributed Conference Room
Message-ID: <CAK9Kg4HzR9zwiBQXEEkVFwz5rGEWh84U0RkoW8f00aMKJ--wMQ@mail.gmail.com>

Hi guys,

I'm trying to deploy a conferencing solution using FreeSWITCH and running
into a small issue with fail over / hot-standbys.

In my environment, I've got multiple FreeSWITCH/Conference endpoints
registered with an OpenSIPS proxy.  When calls come into OpenSIPS they are
routed to the FreeSWITCH endpoints based on their q-values.  If a
FreeSWITCH instance fails (namely the one with the highest q-value), the
call is simply routed to the next instance.  This works great in most
situations, however in some cases (namely network congestion) the
FreeSWITCH w/highest priority is simply temporarily unavailable and callers
to the same conference endpoint land on different servers.

I'm wondering if there is a mechanism for distributing (or sharing) a
conference room across multiple FreeSWITCH instances.  Namely, if a user
lands in a conference hosted on server A while another lands in the same
conference on server B, is there a mechanism in FreeSWITCH to connect the
two servers/conferences (Presumably some "static" connection between the
servers/rooms) so they can still talk with each over?

Thanks,
Daniel
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From jpablolorenzetti at hotmail.com  Thu Dec  6 01:22:21 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Wed, 5 Dec 2012 22:22:21 +0000
Subject: [Freeswitch-users] get a variable at hunting phase that is set
 at executing phase
In-Reply-To: <CALNZuw=8LV36NNZ=0_YKcd+RmYOjM7cnp_VuiYYrGyF=FzF3qA@mail.gmail.com>
References: <DUB002-W564CE266C94AB4E76E0A98D9460@phx.gbl>,
	<CALNZuw=8LV36NNZ=0_YKcd+RmYOjM7cnp_VuiYYrGyF=FzF3qA@mail.gmail.com>
Message-ID: <DUB002-W1186190A492920376DED748D9460@phx.gbl>

Hi Avi, i followed idea 1, i modified the dial plan to look like this:
<extension name="huawei_csoft_gw_outbound_call_international">                <condition field="destination_number" expression="^(00[\d]{7,}?)$">                        <action application="lua" data="get_user_profile.lua" />                        <action application="set" data="number_to_dial=$1" />                        <action application="execute_extension" data="ldi"/>                </condition>        </extension>
        <extension name="check_and_send_international_call">                <condition field="destination_number" expression="^ldi$" />                <condition field="${int}" expression="1">                        <action application="set" data="effective_caller_id_number=280${caller_id_number}"/>                        <action application="bridge" data="sofia/gateway/huawei_csoft/${number_to_dial}"/>                </condition>        </extension>

but i dont understand why the previous alternative did not work, is there anyways to avoid the dial plan to be parsed twice ? thanks!


From: avi at avimarcus.net
Date: Wed, 5 Dec 2012 22:50:05 +0200
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] get a variable at hunting phase that is set at executing phase

Two options:1) Transfer after running your script. It starts the dialplan over again.2) inline="true" - that actually runs it right away. However, you can't touch media at this point or you break the call routing. So I don't think this option is allowed for lua. Or it might have been, give it a try.

If so, update the wiki here: http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions


-Avi
On Wed, Dec 5, 2012 at 10:36 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:





Guys, i have a script the sets some variables the i m supposed to use to decide if a condition meets, i have tried to use the execute_extension app but it does not work, this is the dialplan:



<extension name="huawei_csoft_gw_outbound_call_international">

        <condition field="${int}" expression="1"/>        <condition field="destination_number" expression="^00[\d]{7,}?$">

                <action application="set" data="effective_caller_id_number=280${caller_id_number}"/>                <action application="bridge" data="sofia/gateway/huawei_csoft/${destination_number}"/>

        </condition>   </extension>
the script that sets the var ${int} is called "get_user_profile.lua", i have placed it all over the place and cant make this work.

i have used "execute_extension" in the following way:
<extension name="is_international_allowed">

        <condition field="destination_number" expression="^iia$">                <action application="lua" data="get_user_profile.lua" /> 

        </condition>    </extension>
    <extension name="huawei_csoft_gw_outbound_call_international">

        <condition field="destination_number" expression="^00[\d]{7,}?$">                <action application="execute_extension" data="iia XML smart_private"/> -->

        </condition>        <condition field="${int}" expression="1"/>

        <condition field="destination_number" expression="^00[\d]{7,}?$">                <action application="set" data="effective_caller_id_number=280${caller_id_number}"/>

                <action application="bridge" data="sofia/gateway/huawei_csoft/${destination_number}"/>        </condition>

    </extension>
i know this is simple but i m missing something here, i appreciate any ideas. 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



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

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

http://www.freeswitch.org





_________________________________________________________________________
Professional FreeSWITCH Consulting 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
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From anthony.minessale at gmail.com  Thu Dec  6 01:58:55 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 5 Dec 2012 16:58:55 -0600
Subject: [Freeswitch-users] 10 second delay
In-Reply-To: <50BEF27E.3060205@tagnet.ru>
References: <CADK-BjuB7ugJMPLSYByJPa0jQmnRywvWLgPdwDUCkKF2e7eFbA@mail.gmail.com>
	<50BED517.8060900@communicatefreely.net>
	<50BEEBA3.9040006@tagnet.ru>
	<83CE1052-D007-40D9-A5DB-1540C1A7D982@url.net.au>
	<50BEF27E.3060205@tagnet.ru>
Message-ID: <CAKbxfG94fSTiUsMwkWxfqDY0zFRAVh_4Y-0rKzKtERPCPBJ0pg@mail.gmail.com>

Try out latest GIT


On Wed, Dec 5, 2012 at 1:06 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>      Yes, I use ignore_early_media
> > Hi There,
> >
> > I noticed a similar problem using 1.2.5.2.  In my dial plans I had:
> >
> > <action application="set" data="ignore_early_media=true"/>
> >
> > By removing the ignore_early_media line completely from my dialplans the
> problem was fixed.  Do you have ignore_early_media set anywhere in the
> Dialplans it passes through?
> >
> >
> >
> > - Ash.
> >
> >
> >
> > On 05/12/2012, at 5:37 PM, Boris Kovalenko wrote:
> >
> >> Hello!
> >>
> >> What version of FS are You useing? I have the same troubles with 1.2.5
> >> branch. 1.2.3 works fine.
> >>> My best advice would be to to a packet capture, or turn on sip traces
> between the gateway,
> >>>    FS, and the phones, then try a test.
> >>>
> >>> When you look through the call, take a look at the difference in the
> time stamps.
> >>>
> >>> You should see a progress or a ringing back from the phone, then a 200
> OK when they pick up.
> >>>
> >>> After the call is picked up, see if there is additional traffic
> required to setup media,
> >>> or any other entries in the log that might suggest that some
> additional actions are
> >>> happening after the answer.
> >>>
> >>> Also, if you can capture with wireshark, you can play back the media
> stream.  Check to see
> >>> if the first hello is there coming out of the phone, and then going to
> the gateway.
> >>> Wireshark will put it all in a time line perspective for you, so you
> can determine if this
> >>> is a problem with the phone, your configuration, or the gateway.
> >>>
> >>> Are you using late negotiation, bypass media, or any other options
> like that?
> >>>
> >>> -Tim
> >>>
> >>> Blake Priddy wrote:
> >>>> I have some secretaries here at our school district that when
> >>>> they receive a call they have to give their spill 2-3 times before the
> >>>> party on the other end will hear them.. I have the Epygi Gateway and
> >>>> FreeSwitch box in the same network. Any thoughts would be greatly
> >>>> appreciated! :)
> >>>>
> >>>>
> >>>> --
> >>>>
> >>>> *Blakelund Priddy*
> >>>> Network Systems Engineer
> >>>> Bryant Public School District
> >>>> Bryant, Arkansas 72022
> >>>> http://www.bryantschools.org <http://www.bryantschools.org/>
> >>>> p 501-653-5038
> >>>> f 501-847-5656
> >>>>
> >>>>
> >>>>
> ------------------------------------------------------------------------
> >>>>
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting 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
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >> --
> >> ? ?????????,
> >>    ????? ?????????
> >>    ??? "??????"
> >>    ???. +7 (3435) 230001
> >>    ???? +7 (3435) 230005
> >>    http://www.tagnet.ru
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> --
> ? ?????????,
>    ????? ?????????
>    ??? "??????"
>    ???. +7 (3435) 230001
>    ???? +7 (3435) 230005
>    http://www.tagnet.ru
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Anthony Minessale II

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
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From anthony.minessale at gmail.com  Thu Dec  6 02:17:06 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 5 Dec 2012 17:17:06 -0600
Subject: [Freeswitch-users] NATIVE SQL ERROR [no such table:
	fifo_outbound]
In-Reply-To: <-5406475373508000333@unknownmsgid>
References: <CA+Y7Lf7GzFpacZ_spvJg+jK8+s+4OoeeH0szMmRP2FtHfeOcsw@mail.gmail.com>
	<924695E4-6690-4264-8E42-CB4573698E8F@gmail.com>
	<CAJs+-5p35CSzXBpCKyLfyzrCGEp_au=NPAUru39EgWLAOdeM=g@mail.gmail.com>
	<-5406475373508000333@unknownmsgid>
Message-ID: <CAKbxfG-VU_2UfKtLWp-c-53HFosaJxM1MQdFTNYt6N_fvUZkBg@mail.gmail.com>

The policy with stable and head is that if you have a problem in stable,
you need to work on fixing it in HEAD to wait for it to make it to stable.
So when he tells you to try it on HEAD that is how you can verify its fixed
in advance.



On Wed, Dec 5, 2012 at 12:42 PM, S. Scott <8f27e956 at gmail.com> wrote:

> Overheard In the Wednesday teleconference, that a payload of changes
> to 1.2-stable branch is on a less-than-24-hour to be released window,
> that bumps 1.2.5 and that includes SQL fixes.
>
> Is this sql error issue covered in this imminent bump tp 1.2-stable.
>
>
> On 2012-12-05, at 10:21, Seven Du <dujinfang at gmail.com> wrote:
>
> > update to git head, it should be fixed.
> >
> > On Wed, Dec 5, 2012 at 9:50 PM, Nelson Camargo <bigx333 at gmail.com>
> wrote:
> >> Having same issue, git aa08157 2012-11-30 04:53:03Z)
> >>
> >> On 05 Dec 2012, at 1:43 PM, Scott wrote:
> >>
> >> Just did a 'make current' to update to 1.2.5.2 (from 1.2.5.1).  Now,
> upon
> >> restarting freeswitch, it logs the following,
> >>
> >> ...skipping...
> >> SQL [Enabled]
> >> 2012-12-05 06:29:58.746676 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
> [no
> >> such table: fifo_outbound]
> >> update fifo_outbound set
> >>
> start_time=0,stop_time=0,ring_count=0,use_count=0,outbound_call_count=0,outbound_fail_count=0
> >> where static=0
> >> 2012-12-05 06:29:58.887575 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
> [no
> >> such table: fifo_outbound]
> >> delete from fifo_outbound where static=1 and hostname='fs1.soho.local'
> >>
> >>
> >> In so far as I could see, the 'make' finished successfully without
> apparent
> >> errors.
> >>
> >> Thoughts & suggestions welcome.
> >>
> >> With 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
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting 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
> >>
> >
> >
> >
> > --
> > About: http://about.me/dujinfang
> > Blog: http://www.dujinfang.com
> > Proj:  http://www.freeswitch.org.cn
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
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From rob at corp.coastside.net  Thu Dec  6 02:15:29 2012
From: rob at corp.coastside.net (Rob Genovesi)
Date: Wed, 5 Dec 2012 15:15:29 -0800
Subject: [Freeswitch-users] installing on Amazon EC2
Message-ID: <CA+x1V=PYYNVRZZ642e8AHSw0RM5opJkegbSmAR4oT=HxjiJ8tw@mail.gmail.com>

Hi all,

I'm trying to build/install Free Switch on an Amazon EC2 server
(Amazon Linux AMI 64-bit) and "./configure" keeps exiting with the
following error :

./configure
<snip>
checking for the version of libcurl... 7.24.0
checking for libcurl >= version 7.13.0... yes
checking whether libcurl is usable... no
no
checking for tgetent in -lncurses... no
checking for tgetent in -lcurses... no
configure: error: libtermcap, libcurses or libncurses are required!

(notice the extra "no" on a line by itself)

....

I have installed all the ncurses packages I can find, to no avail :

rpm -qa | grep -E 'curses|ncurses|term|curl'
ncurses-term-5.7-3.20090208.9.amzn1.x86_64
ncurses-libs-5.7-3.20090208.9.amzn1.i686
python-pycurl-7.19.0-8.7.amzn1.x86_64
libcurl-7.24.0-5.25.amzn1.x86_64
ncurses-base-5.7-3.20090208.9.amzn1.x86_64
ncurses-libs-5.7-3.20090208.9.amzn1.x86_64
ncurses-5.7-3.20090208.9.amzn1.x86_64
curl-7.24.0-5.25.amzn1.x86_64
ncurses-devel-5.7-3.20090208.9.amzn1.x86_64
libcurl-devel-7.24.0-5.25.amzn1.x86_64

....

I have been able to install succcessfully on a local CentOS machine
and Rackspace CentOS 6.3 VM, I believe the issue is something specific
to Amazon EC2 but I haven't been able to figure it out.

Any suggestions for a fix/workaround would be appreciated.



Thanks,

Rob


From Tim.Meade at Millicorp.com  Thu Dec  6 02:30:54 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Wed, 5 Dec 2012 23:30:54 +0000
Subject: [Freeswitch-users] installing on Amazon EC2
In-Reply-To: <CA+x1V=PYYNVRZZ642e8AHSw0RM5opJkegbSmAR4oT=HxjiJ8tw@mail.gmail.com>
References: <CA+x1V=PYYNVRZZ642e8AHSw0RM5opJkegbSmAR4oT=HxjiJ8tw@mail.gmail.com>
Message-ID: <804D48104511D4468F0D60DF9D3100350AE3B177@MAIL.millicorp.com>

I had this issue using the base amazon version of Linux.  I don't think we ever got past it.   I ended up using a centos instance for the base.

If you do figure it out please let us know!

Tim

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Genovesi
Sent: Wednesday, December 05, 2012 6:15 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] installing on Amazon EC2

Hi all,

I'm trying to build/install Free Switch on an Amazon EC2 server (Amazon Linux AMI 64-bit) and "./configure" keeps exiting with the following error :

./configure
<snip>
checking for the version of libcurl... 7.24.0 checking for libcurl >= version 7.13.0... yes checking whether libcurl is usable... no no checking for tgetent in -lncurses... no checking for tgetent in -lcurses... no
configure: error: libtermcap, libcurses or libncurses are required!

(notice the extra "no" on a line by itself)

....

I have installed all the ncurses packages I can find, to no avail :

rpm -qa | grep -E 'curses|ncurses|term|curl'
ncurses-term-5.7-3.20090208.9.amzn1.x86_64
ncurses-libs-5.7-3.20090208.9.amzn1.i686
python-pycurl-7.19.0-8.7.amzn1.x86_64
libcurl-7.24.0-5.25.amzn1.x86_64
ncurses-base-5.7-3.20090208.9.amzn1.x86_64
ncurses-libs-5.7-3.20090208.9.amzn1.x86_64
ncurses-5.7-3.20090208.9.amzn1.x86_64
curl-7.24.0-5.25.amzn1.x86_64
ncurses-devel-5.7-3.20090208.9.amzn1.x86_64
libcurl-devel-7.24.0-5.25.amzn1.x86_64

....

I have been able to install succcessfully on a local CentOS machine and Rackspace CentOS 6.3 VM, I believe the issue is something specific to Amazon EC2 but I haven't been able to figure it out.

Any suggestions for a fix/workaround would be appreciated.



Thanks,

Rob

_________________________________________________________________________
Professional FreeSWITCH Consulting 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


From msc at freeswitch.org  Thu Dec  6 02:41:45 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Dec 2012 15:41:45 -0800
Subject: [Freeswitch-users] get a variable at hunting phase that is set
 at executing phase
In-Reply-To: <DUB002-W1FD5F4B80221F2C412F50D9460@phx.gbl>
References: <DUB002-W564CE266C94AB4E76E0A98D9460@phx.gbl>
	<CAKzWOxWN+jY5NP8rNTa0m5nbW2NCW1bmCW6K_O-Xgr4Nv31qDw@mail.gmail.com>
	<DUB002-W1FD5F4B80221F2C412F50D9460@phx.gbl>
Message-ID: <CAKzWOxWTnyutxJivj5xZT8tKSCk0=qGM7JcHoS_LsYtCS2haiw@mail.gmail.com>

On Wed, Dec 5, 2012 at 1:19 PM, Juan Pablo L.
<jpablolorenzetti at hotmail.com>wrote:

> Thanks Avi for you answer, i m going to try that (i already have and the
> ${int} got lost somewhere but i ll try again)
>
> Michael, thanks for your suggestion, i have put it in pastebin:
> http://pastebin.freeswitch.org/20293
>
>
Okay, this is why PB is good. It's just as you and Avi suspected - the
variable is being tested prior to the Lua script running. If you want to
see how to determine that yourself then look at your pastebin entry. Lines
48-72 are the dialplan hunting. The executing starts at line 80 with the
Lua script. The Lua script is being executed after the variable is being
check, which you'll see on line # 66.

Either of Avi's suggestions should help.

-MC
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From msc at freeswitch.org  Thu Dec  6 02:53:39 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Dec 2012 15:53:39 -0800
Subject: [Freeswitch-users] Single sndcard multiple extensions?
In-Reply-To: <50BFB94C.4060803@orresta.no-ip.com>
References: <50BE8980.3050900@orresta.no-ip.com>
	<CAKzWOxXiEn+Ox=eJYfj-6ooEse48x8Q7YPsyi=oU_4YoN4KvNA@mail.gmail.com>
	<50BFB94C.4060803@orresta.no-ip.com>
Message-ID: <CAKzWOxXPuWd8dbPkf9uR9=vZNYfsLXRqK0kWNYPUM_pkULGvzw@mail.gmail.com>

Nice. Maybe you could add this to the wiki to help the next person find it
more easily?
-MC

On Wed, Dec 5, 2012 at 1:14 PM, Jakob <freeswitch at orresta.no-ip.com> wrote:

>  After googeling for a day trying to accomplish this using alsa or jack i
> found a sample implementation using streams in freeswitch that does the
> exact right thing.
>
>
> http://fisheye.freeswitch.org/browse/~raw,r=65b231f5a4450b903c0975ab1c8c6d470c0ab67f/freeswitch.git/conf/autoload_configs/portaudio.conf.xml
>
> Thought it might be of intresst to someone else as well.
>
> 12/05/12 00:50, Michael Collins skrev:
>
> As far as I know mod_portaudio does not support multiple channels...
> -MC
>
> On Tue, Dec 4, 2012 at 3:38 PM, Jakob <freeswitch at orresta.no-ip.com>wrote:
>
>> Hi,
>>
>> I'm trying to get right and left channels from a soundcard into separate
>> alsa or portaudio endpoints.
>>
>> right channel into ext 3001 and left channel into ext 3002
>>
>> Any hints?
>>
>> Regards Jakob
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From 8f27e956 at gmail.com  Thu Dec  6 03:11:03 2012
From: 8f27e956 at gmail.com (Scott)
Date: Wed, 5 Dec 2012 19:11:03 -0500
Subject: [Freeswitch-users] NATIVE SQL ERROR [no such table:
	fifo_outbound]
In-Reply-To: <CAKbxfG-VU_2UfKtLWp-c-53HFosaJxM1MQdFTNYt6N_fvUZkBg@mail.gmail.com>
References: <CA+Y7Lf7GzFpacZ_spvJg+jK8+s+4OoeeH0szMmRP2FtHfeOcsw@mail.gmail.com>
	<924695E4-6690-4264-8E42-CB4573698E8F@gmail.com>
	<CAJs+-5p35CSzXBpCKyLfyzrCGEp_au=NPAUru39EgWLAOdeM=g@mail.gmail.com>
	<-5406475373508000333@unknownmsgid>
	<CAKbxfG-VU_2UfKtLWp-c-53HFosaJxM1MQdFTNYt6N_fvUZkBg@mail.gmail.com>
Message-ID: <CA+Y7Lf4Cvpo1fhjphk8o_cFO3sum5a1VYAPGQppXnZHnStFHng@mail.gmail.com>

Acknowledged.

That said, though, is the likely fix for this in the 1.2.5-n bump that was
forecast in today's (WED) teleconference and, therefore, by waiting the 24
hours or so, I can should be able to draw down from the 1.2-stable branch?

Thanks!

On 5 December 2012 18:17, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> The policy with stable and head is that if you have a problem in stable,
> you need to work on fixing it in HEAD to wait for it to make it to stable.
> So when he tells you to try it on HEAD that is how you can verify its fixed
> in advance.
>
>
>
> On Wed, Dec 5, 2012 at 12:42 PM, S. Scott <8f27e956 at gmail.com> wrote:
>
>> Overheard In the Wednesday teleconference, that a payload of changes
>> to 1.2-stable branch is on a less-than-24-hour to be released window,
>> that bumps 1.2.5 and that includes SQL fixes.
>>
>> Is this sql error issue covered in this imminent bump tp 1.2-stable.
>>
>>
>> On 2012-12-05, at 10:21, Seven Du <dujinfang at gmail.com> wrote:
>>
>> > update to git head, it should be fixed.
>> >
>> > On Wed, Dec 5, 2012 at 9:50 PM, Nelson Camargo <bigx333 at gmail.com>
>> wrote:
>> >> Having same issue, git aa08157 2012-11-30 04:53:03Z)
>> >>
>> >> On 05 Dec 2012, at 1:43 PM, Scott wrote:
>> >>
>> >> Just did a 'make current' to update to 1.2.5.2 (from 1.2.5.1).  Now,
>> upon
>> >> restarting freeswitch, it logs the following,
>> >>
>> >> ...skipping...
>> >> SQL [Enabled]
>> >> 2012-12-05 06:29:58.746676 [ERR] switch_core_sqldb.c:583 NATIVE SQL
>> ERR [no
>> >> such table: fifo_outbound]
>> >> update fifo_outbound set
>> >>
>> start_time=0,stop_time=0,ring_count=0,use_count=0,outbound_call_count=0,outbound_fail_count=0
>> >> where static=0
>> >> 2012-12-05 06:29:58.887575 [ERR] switch_core_sqldb.c:583 NATIVE SQL
>> ERR [no
>> >> such table: fifo_outbound]
>> >> delete from fifo_outbound where static=1 and hostname='fs1.soho.local'
>> >>
>> >>
>> >> In so far as I could see, the 'make' finished successfully without
>> apparent
>> >> errors.
>> >>
>> >> Thoughts & suggestions welcome.
>> >>
>> >> With 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
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting 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
>> >>
>> >
>> >
>> >
>> > --
>> > About: http://about.me/dujinfang
>> > Blog: http://www.dujinfang.com
>> > Proj:  http://www.freeswitch.org.cn
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Dec  6 04:29:52 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 5 Dec 2012 19:29:52 -0600
Subject: [Freeswitch-users] NATIVE SQL ERROR [no such table:
	fifo_outbound]
In-Reply-To: <CA+Y7Lf4Cvpo1fhjphk8o_cFO3sum5a1VYAPGQppXnZHnStFHng@mail.gmail.com>
References: <CA+Y7Lf7GzFpacZ_spvJg+jK8+s+4OoeeH0szMmRP2FtHfeOcsw@mail.gmail.com>
	<924695E4-6690-4264-8E42-CB4573698E8F@gmail.com>
	<CAJs+-5p35CSzXBpCKyLfyzrCGEp_au=NPAUru39EgWLAOdeM=g@mail.gmail.com>
	<-5406475373508000333@unknownmsgid>
	<CAKbxfG-VU_2UfKtLWp-c-53HFosaJxM1MQdFTNYt6N_fvUZkBg@mail.gmail.com>
	<CA+Y7Lf4Cvpo1fhjphk8o_cFO3sum5a1VYAPGQppXnZHnStFHng@mail.gmail.com>
Message-ID: <CAKbxfG9EvaFj9xhdPp5jhK-SOfVCVrHLOBRhFGf4BEpicX24ow@mail.gmail.com>

I would say that's a safe bet...
On Dec 5, 2012 6:14 PM, "Scott" <8f27e956 at gmail.com> wrote:

> Acknowledged.
>
> That said, though, is the likely fix for this in the 1.2.5-n bump that was
> forecast in today's (WED) teleconference and, therefore, by waiting the 24
> hours or so, I can should be able to draw down from the 1.2-stable branch?
>
> Thanks!
>
> On 5 December 2012 18:17, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>
>> The policy with stable and head is that if you have a problem in stable,
>> you need to work on fixing it in HEAD to wait for it to make it to stable.
>> So when he tells you to try it on HEAD that is how you can verify its fixed
>> in advance.
>>
>>
>>
>> On Wed, Dec 5, 2012 at 12:42 PM, S. Scott <8f27e956 at gmail.com> wrote:
>>
>>> Overheard In the Wednesday teleconference, that a payload of changes
>>> to 1.2-stable branch is on a less-than-24-hour to be released window,
>>> that bumps 1.2.5 and that includes SQL fixes.
>>>
>>> Is this sql error issue covered in this imminent bump tp 1.2-stable.
>>>
>>>
>>> On 2012-12-05, at 10:21, Seven Du <dujinfang at gmail.com> wrote:
>>>
>>> > update to git head, it should be fixed.
>>> >
>>> > On Wed, Dec 5, 2012 at 9:50 PM, Nelson Camargo <bigx333 at gmail.com>
>>> wrote:
>>> >> Having same issue, git aa08157 2012-11-30 04:53:03Z)
>>> >>
>>> >> On 05 Dec 2012, at 1:43 PM, Scott wrote:
>>> >>
>>> >> Just did a 'make current' to update to 1.2.5.2 (from 1.2.5.1).  Now,
>>> upon
>>> >> restarting freeswitch, it logs the following,
>>> >>
>>> >> ...skipping...
>>> >> SQL [Enabled]
>>> >> 2012-12-05 06:29:58.746676 [ERR] switch_core_sqldb.c:583 NATIVE SQL
>>> ERR [no
>>> >> such table: fifo_outbound]
>>> >> update fifo_outbound set
>>> >>
>>> start_time=0,stop_time=0,ring_count=0,use_count=0,outbound_call_count=0,outbound_fail_count=0
>>> >> where static=0
>>> >> 2012-12-05 06:29:58.887575 [ERR] switch_core_sqldb.c:583 NATIVE SQL
>>> ERR [no
>>> >> such table: fifo_outbound]
>>> >> delete from fifo_outbound where static=1 and hostname='fs1.soho.local'
>>> >>
>>> >>
>>> >> In so far as I could see, the 'make' finished successfully without
>>> apparent
>>> >> errors.
>>> >>
>>> >> Thoughts & suggestions welcome.
>>> >>
>>> >> With 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
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting 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
>>> >>
>>> >
>>> >
>>> >
>>> > --
>>> > About: http://about.me/dujinfang
>>> > Blog: http://www.dujinfang.com
>>> > Proj:  http://www.freeswitch.org.cn
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 8f27e956 at gmail.com  Thu Dec  6 04:41:09 2012
From: 8f27e956 at gmail.com (S. Scott)
Date: Wed, 5 Dec 2012 20:41:09 -0500
Subject: [Freeswitch-users] NATIVE SQL ERROR [no such table:
	fifo_outbound]
In-Reply-To: <CAKbxfG9EvaFj9xhdPp5jhK-SOfVCVrHLOBRhFGf4BEpicX24ow@mail.gmail.com>
References: <CA+Y7Lf7GzFpacZ_spvJg+jK8+s+4OoeeH0szMmRP2FtHfeOcsw@mail.gmail.com>
	<924695E4-6690-4264-8E42-CB4573698E8F@gmail.com>
	<CAJs+-5p35CSzXBpCKyLfyzrCGEp_au=NPAUru39EgWLAOdeM=g@mail.gmail.com>
	<-5406475373508000333@unknownmsgid>
	<CAKbxfG-VU_2UfKtLWp-c-53HFosaJxM1MQdFTNYt6N_fvUZkBg@mail.gmail.com>
	<CA+Y7Lf4Cvpo1fhjphk8o_cFO3sum5a1VYAPGQppXnZHnStFHng@mail.gmail.com>
	<CAKbxfG9EvaFj9xhdPp5jhK-SOfVCVrHLOBRhFGf4BEpicX24ow@mail.gmail.com>
Message-ID: <2308359706982212747@unknownmsgid>

Thanks Anthony, et al.

You guys are the great!

/Scott

?????
iThing:  Big thumbs & little keys. Please excuse typo, spelling and grammar
errors ? Last night I played a blank CD at full blast. The Mime next door
went nuts.


On 2012-12-05, at 20:33, Anthony Minessale <anthony.minessale at gmail.com>
wrote:

I would say that's a safe bet...
On Dec 5, 2012 6:14 PM, "Scott" <8f27e956 at gmail.com> wrote:

> Acknowledged.
>
> That said, though, is the likely fix for this in the 1.2.5-n bump that was
> forecast in today's (WED) teleconference and, therefore, by waiting the 24
> hours or so, I can should be able to draw down from the 1.2-stable branch?
>
> Thanks!
>
> On 5 December 2012 18:17, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>
>> The policy with stable and head is that if you have a problem in stable,
>> you need to work on fixing it in HEAD to wait for it to make it to stable.
>> So when he tells you to try it on HEAD that is how you can verify its fixed
>> in advance.
>>
>>
>>
>> On Wed, Dec 5, 2012 at 12:42 PM, S. Scott <8f27e956 at gmail.com> wrote:
>>
>>> Overheard In the Wednesday teleconference, that a payload of changes
>>> to 1.2-stable branch is on a less-than-24-hour to be released window,
>>> that bumps 1.2.5 and that includes SQL fixes.
>>>
>>> Is this sql error issue covered in this imminent bump tp 1.2-stable.
>>>
>>>
>>> On 2012-12-05, at 10:21, Seven Du <dujinfang at gmail.com> wrote:
>>>
>>> > update to git head, it should be fixed.
>>> >
>>> > On Wed, Dec 5, 2012 at 9:50 PM, Nelson Camargo <bigx333 at gmail.com>
>>> wrote:
>>> >> Having same issue, git aa08157 2012-11-30 04:53:03Z)
>>> >>
>>> >> On 05 Dec 2012, at 1:43 PM, Scott wrote:
>>> >>
>>> >> Just did a 'make current' to update to 1.2.5.2 (from 1.2.5.1).  Now,
>>> upon
>>> >> restarting freeswitch, it logs the following,
>>> >>
>>> >> ...skipping...
>>> >> SQL [Enabled]
>>> >> 2012-12-05 06:29:58.746676 [ERR] switch_core_sqldb.c:583 NATIVE SQL
>>> ERR [no
>>> >> such table: fifo_outbound]
>>> >> update fifo_outbound set
>>> >>
>>> start_time=0,stop_time=0,ring_count=0,use_count=0,outbound_call_count=0,outbound_fail_count=0
>>> >> where static=0
>>> >> 2012-12-05 06:29:58.887575 [ERR] switch_core_sqldb.c:583 NATIVE SQL
>>> ERR [no
>>> >> such table: fifo_outbound]
>>> >> delete from fifo_outbound where static=1 and hostname='fs1.soho.local'
>>> >>
>>> >>
>>> >> In so far as I could see, the 'make' finished successfully without
>>> apparent
>>> >> errors.
>>> >>
>>> >> Thoughts & suggestions welcome.
>>> >>
>>> >> With 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
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting 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
>>> >>
>>> >
>>> >
>>> >
>>> > --
>>> > About: http://about.me/dujinfang
>>> > Blog: http://www.dujinfang.com
>>> > Proj:  http://www.freeswitch.org.cn
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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  Thu Dec  6 04:46:48 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Thu, 6 Dec 2012 01:46:48 +0000
Subject: [Freeswitch-users] get a variable at hunting phase that is set
 at executing phase
In-Reply-To: <CAKzWOxWTnyutxJivj5xZT8tKSCk0=qGM7JcHoS_LsYtCS2haiw@mail.gmail.com>
References: <DUB002-W564CE266C94AB4E76E0A98D9460@phx.gbl>,
	<CAKzWOxWN+jY5NP8rNTa0m5nbW2NCW1bmCW6K_O-Xgr4Nv31qDw@mail.gmail.com>,
	<DUB002-W1FD5F4B80221F2C412F50D9460@phx.gbl>,
	<CAKzWOxWTnyutxJivj5xZT8tKSCk0=qGM7JcHoS_LsYtCS2haiw@mail.gmail.com>
Message-ID: <DUB002-W1078E4F82563CB2F9E0A859D9450@phx.gbl>

Michael, thank you for showing me, it is clear in the logs now, i did not have that clear before but i see it now ... i implemented Avi's suggestion 1 and it works now but i noticed that execute_extension parses the dialplan again, is there anyway to avoid reading the dialplan again ? regards!

Date: Wed, 5 Dec 2012 15:41:45 -0800
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] get a variable at hunting phase that is set at executing phase



On Wed, Dec 5, 2012 at 1:19 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:




Thanks Avi for you answer, i m going to try that (i already have and the ${int} got lost somewhere but i ll try again)
Michael, thanks for your suggestion, i have put it in pastebin: http://pastebin.freeswitch.org/20293



Okay, this is why PB is good. It's just as you and Avi suspected - the variable is being tested prior to the Lua script running. If you want to see how to determine that yourself then look at your pastebin entry. Lines 48-72 are the dialplan hunting. The executing starts at line 80 with the Lua script. The Lua script is being executed after the variable is being check, which you'll see on line # 66.


Either of Avi's suggestions should help.

-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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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  Thu Dec  6 04:48:10 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Thu, 6 Dec 2012 01:48:10 +0000
Subject: [Freeswitch-users] get a variable at hunting phase that is set
 at executing phase
In-Reply-To: <CAKzWOxWTnyutxJivj5xZT8tKSCk0=qGM7JcHoS_LsYtCS2haiw@mail.gmail.com>
References: <DUB002-W564CE266C94AB4E76E0A98D9460@phx.gbl>,
	<CAKzWOxWN+jY5NP8rNTa0m5nbW2NCW1bmCW6K_O-Xgr4Nv31qDw@mail.gmail.com>,
	<DUB002-W1FD5F4B80221F2C412F50D9460@phx.gbl>,
	<CAKzWOxWTnyutxJivj5xZT8tKSCk0=qGM7JcHoS_LsYtCS2haiw@mail.gmail.com>
Message-ID: <DUB002-W569C79A46DACE1685A270CD9450@phx.gbl>

Michael, thank you for showing me, it is clear in the logs now, i did not have that clear before but i see it now ... i implemented Avi's suggestion 1 and it works now but i noticed that execute_extension parses the dialplan again, is there anyway to avoid reading the dialplan again ? regards!

Date: Wed, 5 Dec 2012 15:41:45 -0800
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] get a variable at hunting phase that is set at executing phase



On Wed, Dec 5, 2012 at 1:19 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:




Thanks Avi for you answer, i m going to try that (i already have and the ${int} got lost somewhere but i ll try again)
Michael, thanks for your suggestion, i have put it in pastebin: http://pastebin.freeswitch.org/20293



Okay, this is why PB is good. It's just as you and Avi suspected - the variable is being tested prior to the Lua script running. If you want to see how to determine that yourself then look at your pastebin entry. Lines 48-72 are the dialplan hunting. The executing starts at line 80 with the Lua script. The Lua script is being executed after the variable is being check, which you'll see on line # 66.


Either of Avi's suggestions should help.

-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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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  Thu Dec  6 05:27:32 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 5 Dec 2012 19:27:32 -0700
Subject: [Freeswitch-users] installing on Amazon EC2
In-Reply-To: <804D48104511D4468F0D60DF9D3100350AE3B177@MAIL.millicorp.com>
References: <CA+x1V=PYYNVRZZ642e8AHSw0RM5opJkegbSmAR4oT=HxjiJ8tw@mail.gmail.com>
	<804D48104511D4468F0D60DF9D3100350AE3B177@MAIL.millicorp.com>
Message-ID: <CAMwZ-tJOyHm7Z8tS-ZDMNSSajib3F1y_xi38cchuFDAYZ=kDjA@mail.gmail.com>

On Wed, Dec 5, 2012 at 4:30 PM, Tim Meade <Tim.Meade at millicorp.com> wrote:
> I ended up using a centos instance for the base.

What's your voice quality like?


Gabe


From covici at ccs.covici.com  Thu Dec  6 06:14:53 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 05 Dec 2012 22:14:53 -0500
Subject: [Freeswitch-users] segfault in mod_conference
Message-ID: <5758.1354763693@ccs.covici.com>

Hi.  I have gotten a seg fault in mod_conference.  I used the fscorepb
script and so the data is at 
http://pastebin.freeswitch.org/20294  .

I was also getting a strange error, not in the log, but in what I guess
would be sysout as follows:
2012-12-06_02:05:40.71336 Error in my_thread_global_end(): 20 threads
didn't exit
I was getting lots of these with various numbers.  My fs version is
FreeSWITCH Version 1.3.4b+git~20121114T021840Z~8f0b7e69de (git 8f0b7e6
2012-11-14 02:18:40Z)

Should I bother filing a jira, or just update to latest?

-- 
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 krice at freeswitch.org  Thu Dec  6 06:27:15 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 05 Dec 2012 21:27:15 -0600
Subject: [Freeswitch-users] segfault in mod_conference
In-Reply-To: <5758.1354763693@ccs.covici.com>
Message-ID: <CCE56CB3.61728%krice@freeswitch.org>

Hey Covici,

That backtrace is sorta useless, I would try to update to latest and see if
that makes a difference.

Also when using fscorepb make sure you still have the source where it was
compiled from so that gdb can match up, that pastebin has no info on any of
the threads 


On 12/5/12 9:14 PM, "covici at ccs.covici.com" <covici at ccs.covici.com> wrote:

> Hi.  I have gotten a seg fault in mod_conference.  I used the fscorepb
> script and so the data is at
> http://pastebin.freeswitch.org/20294  .
> 
> I was also getting a strange error, not in the log, but in what I guess
> would be sysout as follows:
> 2012-12-06_02:05:40.71336 Error in my_thread_global_end(): 20 threads
> didn't exit
> I was getting lots of these with various numbers.  My fs version is
> FreeSWITCH Version 1.3.4b+git~20121114T021840Z~8f0b7e69de (git 8f0b7e6
> 2012-11-14 02:18:40Z)
> 
> Should I bother filing a jira, or just update to latest?

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From anthony.minessale at gmail.com  Thu Dec  6 06:50:15 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 5 Dec 2012 21:50:15 -0600
Subject: [Freeswitch-users] segfault in mod_conference
In-Reply-To: <5758.1354763693@ccs.covici.com>
References: <5758.1354763693@ccs.covici.com>
Message-ID: <CAKbxfG93PvKFEqOq83etFDVXzkvfwWEPAU8HJ0GMCsGKH4qvOg@mail.gmail.com>

That looks like MySQL not using the threadsafe version of the lib in odbc
with the _r variant in osbcinst.ini also threading=0 consult wiki on odbc
in the core....
On Dec 5, 2012 9:18 PM, <covici at ccs.covici.com> wrote:

> Hi.  I have gotten a seg fault in mod_conference.  I used the fscorepb
> script and so the data is at
> http://pastebin.freeswitch.org/20294  .
>
> I was also getting a strange error, not in the log, but in what I guess
> would be sysout as follows:
> 2012-12-06_02:05:40.71336 Error in my_thread_global_end(): 20 threads
> didn't exit
> I was getting lots of these with various numbers.  My fs version is
> FreeSWITCH Version 1.3.4b+git~20121114T021840Z~8f0b7e69de (git 8f0b7e6
> 2012-11-14 02:18:40Z)
>
> Should I bother filing a jira, or just update to latest?
>
> --
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anton.jugatsu at gmail.com  Thu Dec  6 07:43:30 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Thu, 6 Dec 2012 08:43:30 +0400
Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today
In-Reply-To: <9BDCA7E1-9C70-4131-958C-A2A159E3BC0D@endigotech.com>
References: <CAKzWOxUeDLWVF23J_9RZR+KfE0EQmY4koDw2DpNhWKz9F1G0-g@mail.gmail.com>
	<9BDCA7E1-9C70-4131-958C-A2A159E3BC0D@endigotech.com>
Message-ID: <CANBhyPApMOtUqKGtwa_CvyHLfSe=Yt=MCYbj0rxopd=g0dxzZA@mail.gmail.com>

Michael, I don't see any presentation or videos at cluecon.com.


2012/12/5 Brian Foster <bdfoster at endigotech.com>

> Can't unmute and I am not able to get in front of IRC...
>
> Sent from my iPhone
>
> On Dec 5, 2012, at 12:05 PM, Michael Collins <msc at freeswitch.org> wrote:
>
> Hello folks,
>
> We'll be having our conference call in just about one hour. The agenda
> page is here:
>
> http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_05
>
> We'll be discussing some interesting facts about the FreeSWITCH XML
> pre-processor and also an object lesson in how to use the source code to
> answer a question and then put it on the wiki. Oh, and we have an update on
> the ClueCon videos!
>
> Talk to you soon.
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From 8f27e956 at gmail.com  Thu Dec  6 09:05:21 2012
From: 8f27e956 at gmail.com (Scott)
Date: Thu, 6 Dec 2012 01:05:21 -0500
Subject: [Freeswitch-users] RESOLVED: Re: NATIVE SQL ERROR [no such table:
	fifo_outbound]
Message-ID: <CA+Y7Lf5tCbp4AbA41im7k=7pb8Oc_E-rLSMg4aAaxzDZYkOozw@mail.gmail.com>

For closure (mark RESOLVED), no more 'native sql error.'

Just did a *'make current'* and received the fresh-out-of-the-oven bump to
1.2.5.3 (from 1.2.5.2), where baseline original pull was the wiki
perscribed,

'git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git'

Again, my thanks to all!

/Scott


On 5 December 2012 20:29, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> I would say that's a safe bet...
> On Dec 5, 2012 6:14 PM, "Scott" <8f27e956 at gmail.com> wrote:
>
>> Acknowledged.
>>
>> That said, though, is the likely fix for this in the 1.2.5-n bump that
>> was forecast in today's (WED) teleconference and, therefore, by waiting the
>> 24 hours or so, I can should be able to draw down from the 1.2-stable
>> branch?
>>
>> Thanks!
>>
>>
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From covici at ccs.covici.com  Thu Dec  6 09:50:32 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 06 Dec 2012 01:50:32 -0500
Subject: [Freeswitch-users] segfault in mod_conference
In-Reply-To: <CCE56CB3.61728%krice@freeswitch.org>
References: <CCE56CB3.61728%krice@freeswitch.org>
Message-ID: <2738.1354776632@ccs.covici.com>

This is strange, I did have the source, or rather still do, but I will
update and see  what happens.

Ken Rice <krice at freeswitch.org> wrote:

> Hey Covici,
> 
> That backtrace is sorta useless, I would try to update to latest and see if
> that makes a difference.
> 
> Also when using fscorepb make sure you still have the source where it was
> compiled from so that gdb can match up, that pastebin has no info on any of
> the threads 
> 
> 
> On 12/5/12 9:14 PM, "covici at ccs.covici.com" <covici at ccs.covici.com> wrote:
> 
> > Hi.  I have gotten a seg fault in mod_conference.  I used the fscorepb
> > script and so the data is at
> > http://pastebin.freeswitch.org/20294  .
> > 
> > I was also getting a strange error, not in the log, but in what I guess
> > would be sysout as follows:
> > 2012-12-06_02:05:40.71336 Error in my_thread_global_end(): 20 threads
> > didn't exit
> > I was getting lots of these with various numbers.  My fs version is
> > FreeSWITCH Version 1.3.4b+git~20121114T021840Z~8f0b7e69de (git 8f0b7e6
> > 2012-11-14 02:18:40Z)
> > 
> > Should I bother filing a jira, or just update to latest?
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 covici at ccs.covici.com  Thu Dec  6 09:53:39 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 06 Dec 2012 01:53:39 -0500
Subject: [Freeswitch-users] segfault in mod_conference
In-Reply-To: <CAKbxfG93PvKFEqOq83etFDVXzkvfwWEPAU8HJ0GMCsGKH4qvOg@mail.gmail.com>
References: <5758.1354763693@ccs.covici.com>
	<CAKbxfG93PvKFEqOq83etFDVXzkvfwWEPAU8HJ0GMCsGKH4qvOg@mail.gmail.com>
Message-ID: <3168.1354776819@ccs.covici.com>

I have odbc enabled, but I did not know I was actually using it -- I
will have to check the configs.

Anthony Minessale <anthony.minessale at gmail.com> wrote:

> That looks like MySQL not using the threadsafe version of the lib in odbc
> with the _r variant in osbcinst.ini also threading=0 consult wiki on odbc
> in the core....
> On Dec 5, 2012 9:18 PM, <covici at ccs.covici.com> wrote:
> 
> > Hi.  I have gotten a seg fault in mod_conference.  I used the fscorepb
> > script and so the data is at
> > http://pastebin.freeswitch.org/20294  .
> >
> > I was also getting a strange error, not in the log, but in what I guess
> > would be sysout as follows:
> > 2012-12-06_02:05:40.71336 Error in my_thread_global_end(): 20 threads
> > didn't exit
> > I was getting lots of these with various numbers.  My fs version is
> > FreeSWITCH Version 1.3.4b+git~20121114T021840Z~8f0b7e69de (git 8f0b7e6
> > 2012-11-14 02:18:40Z)
> >
> > Should I bother filing a jira, or just update to latest?
> >
> > --
> > 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 covici at ccs.covici.com  Thu Dec  6 10:30:45 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 06 Dec 2012 02:30:45 -0500
Subject: [Freeswitch-users] getting error on sysout not in log
Message-ID: <26956.1354779045@ccs.covici.com>

Hi.  I am getting the following error in sysout rather than the .log
file:
2012-12-06_07:19:57.65099 Error in my_thread_global_end(): 1 threads
didn't exit

This is with latest git.  The only thing happening at the same time as
the listed error was a sofia registration.

How can I proceed?  It looks I am using sqlite for the databases in the
core, if that makes any difference.

Thanks in advance for any help.

-- 
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 peter.olsson at visionutveckling.se  Thu Dec  6 10:41:03 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 6 Dec 2012 07:41:03 +0000
Subject: [Freeswitch-users] getting error on sysout not in log
Message-ID: <1FFF97C269757C458224B7C895F35F151DE3AE@cantor.std.visionutv.se>

That's from the MySQL libraries.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r covici at ccs.covici.com
Skickat: den 6 december 2012 08:31
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] getting error on sysout not in log

Hi.  I am getting the following error in sysout rather than the .log
file:
2012-12-06_07:19:57.65099 Error in my_thread_global_end(): 1 threads didn't exit

This is with latest git.  The only thing happening at the same time as the listed error was a sofia registration.

How can I proceed?  It looks I am using sqlite for the databases in the core, if that makes any difference.

Thanks in advance for any help.

--
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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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:50c0471f32761351719099!



From avi at avimarcus.net  Thu Dec  6 11:00:37 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 6 Dec 2012 10:00:37 +0200
Subject: [Freeswitch-users] get a variable at hunting phase that is set
 at executing phase
In-Reply-To: <DUB002-W1078E4F82563CB2F9E0A859D9450@phx.gbl>
References: <DUB002-W564CE266C94AB4E76E0A98D9460@phx.gbl>
	<CAKzWOxWN+jY5NP8rNTa0m5nbW2NCW1bmCW6K_O-Xgr4Nv31qDw@mail.gmail.com>
	<DUB002-W1FD5F4B80221F2C412F50D9460@phx.gbl>
	<CAKzWOxWTnyutxJivj5xZT8tKSCk0=qGM7JcHoS_LsYtCS2haiw@mail.gmail.com>
	<DUB002-W1078E4F82563CB2F9E0A859D9450@phx.gbl>
Message-ID: <CALNZuwmq8-gjaE4BW5rNV28899apGjbrdSEqHKdy_yyr4hT8eg@mail.gmail.com>

If you execute_extension to a new context, you can create very few
extensions in that context so that it goes quicker.

-Avi


On Thu, Dec 6, 2012 at 3:46 AM, Juan Pablo L.
<jpablolorenzetti at hotmail.com>wrote:

> Michael, thank you for showing me, it is clear in the logs now, i did not
> have that clear before but i see it now ... i implemented Avi's suggestion
> 1 and it works now but i noticed that execute_extension parses the dialplan
> again, is there anyway to avoid reading the dialplan again ? regards!
>
> ------------------------------
> Date: Wed, 5 Dec 2012 15:41:45 -0800
>
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] get a variable at hunting phase that is
> set at executing phase
>
>
>
> On Wed, Dec 5, 2012 at 1:19 PM, Juan Pablo L. <
> jpablolorenzetti at hotmail.com> wrote:
>
> Thanks Avi for you answer, i m going to try that (i already have and the
> ${int} got lost somewhere but i ll try again)
>
> Michael, thanks for your suggestion, i have put it in pastebin:
> http://pastebin.freeswitch.org/20293
>
>
> Okay, this is why PB is good. It's just as you and Avi suspected - the
> variable is being tested prior to the Lua script running. If you want to
> see how to determine that yourself then look at your pastebin entry. Lines
> 48-72 are the dialplan hunting. The executing starts at line 80 with the
> Lua script. The Lua script is being executed after the variable is being
> check, which you'll see on line # 66.
>
> Either of Avi's suggestions should help.
>
> -MC
>
> _________________________________________________________________________
> 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 FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 regis.freeswitch.org at tornad.net  Thu Dec  6 11:59:51 2012
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Thu, 6 Dec 2012 09:59:51 +0100
Subject: [Freeswitch-users] how to Ignore DTMF shorter than Fixed
	duration ?
In-Reply-To: <CAKzWOxXQSj9hPSoSmH9pQ19J=+eU5qM+8aNO51BoWzzN_gKj_w@mail.gmail.com>
References: <CAHWtr1TbtvoxrxbCEY6y8r_rKgtsh2zHhXmeZYRA0+6P0dKGUQ@mail.gmail.com>
	<CAKzWOxXQSj9hPSoSmH9pQ19J=+eU5qM+8aNO51BoWzzN_gKj_w@mail.gmail.com>
Message-ID: <CAHWtr1SfKpkXL7BeB0995sjN9k8yLwKEXRcxkXvZyvQfqqEjUg@mail.gmail.com>

Hi,

It's RFC2833 DTMF, not inband.

Regards,


2012/12/5 Michael Collins <msc at freeswitch.org>

> Are they sending those digits inband or with RFC2833?
> -MC
>
> On Wed, Dec 5, 2012 at 11:28 AM, Regis M <regis.freeswitch.org at tornad.net>wrote:
>
>> Hi,
>>
>> We have a bugged provider that send us throw RTP wrong DTMF during call.
>> It seems that wrong DTMF are shorter than 1200ms (about 90%) so I want to
>> try to ignore them.
>> For the moment, FS catch them and send it back to the bridged side,
>> boring user.
>>
>> switch_rtp.c:3410 RTP RECV DTMF C:552
>>
>>
>>
>> Does min-dtmf-duration will make FS ignore them and not RECV them ?
>> I saw the DTMF in "normal" RTP packet, but no digit was pressed in user
>> side.
>>
>> It's commented in my FS switch.conf.xml, how can I see the current value ?
>> <!--<param name="min-dtmf-duration" value="400"/>-->
>>
>>
>> Any other idea ?
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 th982a at googlemail.com  Thu Dec  6 13:17:05 2012
From: th982a at googlemail.com (Tamer Higazi)
Date: Thu, 06 Dec 2012 11:17:05 +0100
Subject: [Freeswitch-users] hylafax + mod_spandsp hangsup immediatly
Message-ID: <50C070A1.4000101@googlemail.com>

Hi people!
I've set up Freeswitch with mod_spandsp and hylafax and the modems are
running (initialized) and great so far. Now, when I want to send a fax,
hylafax aborts all the time and I am not getting smart.

http://permalink.gmane.org/gmane.comp.telephony.fax.hylafax.user/36639

Any ideas what it could be?!



Here is the log from the fs_console:


2012-12-06 11:13:49.936819 [DEBUG] mod_spandsp_modem.c:1070 Modem
/dev/FS0 [ONHOOK] - Hanging up
2012-12-06 11:13:49.936819 [DEBUG] mod_spandsp_modem.c:1076 Modem
/dev/FS0 [ONHOOK] - Changing state to HANGUP
2012-12-06 11:13:49.936819 [DEBUG] mod_spandsp_modem.c:1095 Modem
/dev/FS0 [HANGUP] - Changing state to ONHOOK
2012-12-06 11:13:54.456814 [DEBUG] mod_spandsp_modem.c:1044 Modem
/dev/FS0 [ONHOOK] - Dialing '1xxxxxxxxxxxxx'
2012-12-06 11:13:54.456814 [DEBUG] mod_spandsp_modem.c:1046 Modem
/dev/FS0 [ONHOOK] - Changing state to DIALING
2012-12-06 11:13:54.456814 [NOTICE] switch_channel.c:941 New Channel
modem/0/1xxxxxxxxxxxxx [585f9ebe-85fb-4152-b055-99fb72bf22ed]
2012-12-06 11:13:54.456814 [DEBUG] mod_spandsp_modem.c:769
modem/0/1xxxxxxxxxxxxx setup codec L16/8000/20
2012-12-06 11:13:54.456814 [DEBUG] mod_spandsp_modem.c:1003
(modem/0/1xxxxxxxxxxxxx) State Change CS_NEW -> CS_INIT
2012-12-06 11:13:54.456814 [DEBUG] switch_core_session.c:1229 Send
signal modem/0/1xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:54.456814 [DEBUG] mod_spandsp_modem.c:581
modem/0/1xxxxxxxxxxxxx CHANNEL KILL
2012-12-06 11:13:54.456814 [DEBUG] switch_channel.c:1108 EXPORT
(export_vars) [rtp_autoflush_during_bridge]=[false]
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:385
(modem/0/1xxxxxxxxxxxxx) Running State Change CS_INIT
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:424
(modem/0/1xxxxxxxxxxxxx) State INIT
2012-12-06 11:13:54.456814 [DEBUG] mod_spandsp_modem.c:484
(modem/0/1xxxxxxxxxxxxx) State Change CS_INIT -> CS_ROUTING
2012-12-06 11:13:54.456814 [DEBUG] switch_core_session.c:1229 Send
signal modem/0/1xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:54.456814 [DEBUG] mod_spandsp_modem.c:581
modem/0/1xxxxxxxxxxxxx CHANNEL KILL
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:424
(modem/0/1xxxxxxxxxxxxx) State INIT going to sleep
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:385
(modem/0/1xxxxxxxxxxxxx) Running State Change CS_ROUTING
2012-12-06 11:13:54.456814 [DEBUG] switch_channel.c:1934
(modem/0/1xxxxxxxxxxxxx) Callstate Change DOWN -> RINGING
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:433
(modem/0/1xxxxxxxxxxxxx) State ROUTING
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:104
modem/0/1xxxxxxxxxxxxx Standard ROUTING
2012-12-06 11:13:54.456814 [INFO] mod_dialplan_xml.c:485 Processing
FSModem <FS0>->1xxxxxxxxxxxxx in context default
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->unloop] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->tod_example]
continue=true
Dialplan: modem/0/1xxxxxxxxxxxxx Date/Time Match (PASS) [tod_example]
break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx Action set(open=true)
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->holiday_example]
continue=true
Dialplan: modem/0/1xxxxxxxxxxxxx Date/TimeMatch (FAIL) [holiday_example]
break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->global-intercept]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [global-intercept]
destination_number(1xxxxxxxxxxxxx) =~ /^886$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->group-intercept]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [group-intercept]
destination_number(1xxxxxxxxxxxxx) =~ /^\*8$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->intercept-ext]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [intercept-ext]
destination_number(1xxxxxxxxxxxxx) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->redial] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [redial]
destination_number(1xxxxxxxxxxxxx) =~ /^(redial|870)$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->global] continue=true
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [global]
${call_debug}(false) =~ /^true$/ break=never
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [global]
${sip_has_crypto}() =~
/^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: modem/0/1xxxxxxxxxxxxx Absolute Condition [global]
Dialplan: modem/0/1xxxxxxxxxxxxx Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: modem/0/1xxxxxxxxxxxxx Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})

Dialplan: modem/0/1xxxxxxxxxxxxx Action
hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: modem/0/1xxxxxxxxxxxxx Action
export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->snom-demo-2]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [snom-demo-2]
destination_number(1xxxxxxxxxxxxx) =~ /^9001$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->snom-demo-1]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [snom-demo-1]
destination_number(1xxxxxxxxxxxxx) =~ /^9000$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->eavesdrop] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [eavesdrop]
destination_number(1xxxxxxxxxxxxx) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->eavesdrop] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [eavesdrop]
destination_number(1xxxxxxxxxxxxx) =~ /^779$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->call_return]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [call_return]
destination_number(1xxxxxxxxxxxxx) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->del-group] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [del-group]
destination_number(1xxxxxxxxxxxxx) =~ /^80(\d{2})$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->add-group] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [add-group]
destination_number(1xxxxxxxxxxxxx) =~ /^81(\d{2})$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->call-group-simo]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [call-group-simo]
destination_number(1xxxxxxxxxxxxx) =~ /^82(\d{2})$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->call-group-order]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [call-group-order]
destination_number(1xxxxxxxxxxxxx) =~ /^83(\d{2})$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->extension-intercom]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [extension-intercom]
destination_number(1xxxxxxxxxxxxx) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->Local_Extension]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [Local_Extension]
destination_number(1xxxxxxxxxxxxx) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing
[default->Local_Extension_Skinny] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [Local_Extension_Skinny]
destination_number(1xxxxxxxxxxxxx) =~ /^(11[01][0-9])$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->group_dial_sales]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [group_dial_sales]
destination_number(1xxxxxxxxxxxxx) =~ /^2000$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->group_dial_support]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [group_dial_support]
destination_number(1xxxxxxxxxxxxx) =~ /^2001$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->group_dial_billing]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [group_dial_billing]
destination_number(1xxxxxxxxxxxxx) =~ /^2002$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->operator] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [operator]
destination_number(1xxxxxxxxxxxxx) =~ /^(operator|0)$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->vmain] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [vmain]
destination_number(1xxxxxxxxxxxxx) =~ /^vmain$|^4000$|^\*98$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->sip_uri] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [sip_uri]
destination_number(1xxxxxxxxxxxxx) =~ /^sip:(.*)$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->nb_conferences]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [nb_conferences]
destination_number(1xxxxxxxxxxxxx) =~ /^(30\d{2})$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->wb_conferences]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [wb_conferences]
destination_number(1xxxxxxxxxxxxx) =~ /^(31\d{2})$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->uwb_conferences]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [uwb_conferences]
destination_number(1xxxxxxxxxxxxx) =~ /^(32\d{2})$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing
[default->cdquality_conferences] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [cdquality_conferences]
destination_number(1xxxxxxxxxxxxx) =~ /^(33\d{2})$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing
[default->freeswitch_public_conf_via_sip] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL)
[freeswitch_public_conf_via_sip] destination_number(1xxxxxxxxxxxxx) =~
/^9(888|8888|1616|3232)$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->mad_boss_intercom]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [mad_boss_intercom]
destination_number(1xxxxxxxxxxxxx) =~ /^0911$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->mad_boss_intercom]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [mad_boss_intercom]
destination_number(1xxxxxxxxxxxxx) =~ /^0912$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->mad_boss] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [mad_boss]
destination_number(1xxxxxxxxxxxxx) =~ /^0913$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->ivr_demo] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [ivr_demo]
destination_number(1xxxxxxxxxxxxx) =~ /^5000$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->dynamic_conference]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [dynamic_conference]
destination_number(1xxxxxxxxxxxxx) =~ /^5001$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->rtp_multicast_page]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [rtp_multicast_page]
destination_number(1xxxxxxxxxxxxx) =~ /^pagegroup$|^7243$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->park] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [park]
destination_number(1xxxxxxxxxxxxx) =~ /^5900$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->unpark] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [unpark]
destination_number(1xxxxxxxxxxxxx) =~ /^5901$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->valet_park]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [valet_park]
destination_number(1xxxxxxxxxxxxx) =~ /^(6000)$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->valet_park]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [valet_park]
destination_number(1xxxxxxxxxxxxx) =~ /^(60\d[1-9])$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->park] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [park] source(mod_spandsp)
=~ /mod_sofia/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->unpark] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [unpark]
source(mod_spandsp) =~ /mod_sofia/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->park] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [park] source(mod_spandsp)
=~ /mod_sofia/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->unpark] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [unpark]
source(mod_spandsp) =~ /mod_sofia/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->wait] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [wait]
destination_number(1xxxxxxxxxxxxx) =~ /^wait$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->fax_receive]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [fax_receive]
destination_number(1xxxxxxxxxxxxx) =~ /^9178$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->fax_transmit]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [fax_transmit]
destination_number(1xxxxxxxxxxxxx) =~ /^9179$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->ringback_180]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [ringback_180]
destination_number(1xxxxxxxxxxxxx) =~ /^9180$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->ringback_183_uk_ring]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [ringback_183_uk_ring]
destination_number(1xxxxxxxxxxxxx) =~ /^9181$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing
[default->ringback_183_music_ring] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [ringback_183_music_ring]
destination_number(1xxxxxxxxxxxxx) =~ /^9182$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing
[default->ringback_post_answer_uk_ring] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL)
[ringback_post_answer_uk_ring] destination_number(1xxxxxxxxxxxxx) =~
/^9183$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing
[default->ringback_post_answer_music] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL)
[ringback_post_answer_music] destination_number(1xxxxxxxxxxxxx) =~
/^9184$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->ClueCon] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [ClueCon]
destination_number(1xxxxxxxxxxxxx) =~ /^9191$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->show_info] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [show_info]
destination_number(1xxxxxxxxxxxxx) =~ /^9192$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->video_record]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [video_record]
destination_number(1xxxxxxxxxxxxx) =~ /^9193$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->video_playback]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [video_playback]
destination_number(1xxxxxxxxxxxxx) =~ /^9194$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->delay_echo]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [delay_echo]
destination_number(1xxxxxxxxxxxxx) =~ /^9195$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->echo] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [echo]
destination_number(1xxxxxxxxxxxxx) =~ /^9196$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->milliwatt] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [milliwatt]
destination_number(1xxxxxxxxxxxxx) =~ /^9197$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->tone_stream]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [tone_stream]
destination_number(1xxxxxxxxxxxxx) =~ /^9198$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->zrtp_enrollement]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [zrtp_enrollement]
destination_number(1xxxxxxxxxxxxx) =~ /^9787$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->hold_music]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [hold_music]
destination_number(1xxxxxxxxxxxxx) =~ /^9664$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->laugh break]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [laugh break]
destination_number(1xxxxxxxxxxxxx) =~ /^9386$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->Talking Clock Time]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [Talking Clock Time]
destination_number(1xxxxxxxxxxxxx) =~ /^9170$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->Talking Clock Date]
continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [Talking Clock Date]
destination_number(1xxxxxxxxxxxxx) =~ /^9171$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->Talking Clock Date
and Time] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [Talking Clock Date and
Time] destination_number(1xxxxxxxxxxxxx) =~ /^9172$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->sipgate0] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [sipgate0]
destination_number(1xxxxxxxxxxxxx) =~ /^2(\d{3,})/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->sipgate1] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [sipgate1]
destination_number(1xxxxxxxxxxxxx) =~ /^3(\d{3,})/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->ABX] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [ABX]
destination_number(1xxxxxxxxxxxxx) =~ /^(90)$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->ABX2] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [ABX2]
destination_number(1xxxxxxxxxxxxx) =~ /^(91)$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->FSX] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [FSX]
destination_number(1xxxxxxxxxxxxx) =~ /^(92)$/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->easybell0] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (FAIL) [easybell0]
destination_number(1xxxxxxxxxxxxx) =~ /^0(\d{3,})/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx parsing [default->easybell1] continue=false
Dialplan: modem/0/1xxxxxxxxxxxxx Regex (PASS) [easybell1]
destination_number(1xxxxxxxxxxxxx) =~ /^1(\d{3,})/ break=on-false
Dialplan: modem/0/1xxxxxxxxxxxxx Action
set(effective_caller_id_number=${outbound_caller_id_number})
Dialplan: modem/0/1xxxxxxxxxxxxx Action set(absolute_codec_string=PCMA)
Dialplan: modem/0/1xxxxxxxxxxxxx Action set(fax_enable_t38=true)
Dialplan: modem/0/1xxxxxxxxxxxxx Action set(fax_enable_t38_request=true)
Dialplan: modem/0/1xxxxxxxxxxxxx Action
set(execute_on_answer=t38_gateway self)
Dialplan: modem/0/1xxxxxxxxxxxxx Action
bridge(sofia/gateway/easybell1/xxxxxxxxxxxxx)
Dialplan: modem/0/1xxxxxxxxxxxxx Action hangup()
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:154
(modem/0/1xxxxxxxxxxxxx) State Change CS_ROUTING -> CS_EXECUTE
2012-12-06 11:13:54.456814 [DEBUG] switch_core_session.c:1229 Send
signal modem/0/1xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:54.456814 [DEBUG] mod_spandsp_modem.c:581
modem/0/1xxxxxxxxxxxxx CHANNEL KILL
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:433
(modem/0/1xxxxxxxxxxxxx) State ROUTING going to sleep
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:385
(modem/0/1xxxxxxxxxxxxx) Running State Change CS_EXECUTE
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:440
(modem/0/1xxxxxxxxxxxxx) State EXECUTE
2012-12-06 11:13:54.456814 [DEBUG] mod_spandsp_modem.c:514
modem/0/1xxxxxxxxxxxxx CHANNEL EXECUTE
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:196
modem/0/1xxxxxxxxxxxxx Standard EXECUTE
EXECUTE modem/0/1xxxxxxxxxxxxx set(open=true)
2012-12-06 11:13:54.456814 [DEBUG] mod_dptools.c:1319
modem/0/1xxxxxxxxxxxxx SET [open]=[true]
EXECUTE modem/0/1xxxxxxxxxxxxx
hash(insert/212.255.30.34-spymap/FS0/585f9ebe-85fb-4152-b055-99fb72bf22ed)
EXECUTE modem/0/1xxxxxxxxxxxxx
hash(insert/212.255.30.34-last_dial/FS0/1xxxxxxxxxxxxx)
EXECUTE modem/0/1xxxxxxxxxxxxx
hash(insert/212.255.30.34-last_dial/global/585f9ebe-85fb-4152-b055-99fb72bf22ed)
EXECUTE modem/0/1xxxxxxxxxxxxx export(RFC2822_DATE=Thu, 06 Dec 2012
11:13:54 +0100)
2012-12-06 11:13:54.456814 [DEBUG] switch_channel.c:1108 EXPORT
(export_vars) [RFC2822_DATE]=[Thu, 06 Dec 2012 11:13:54 +0100]
EXECUTE modem/0/1xxxxxxxxxxxxx set(effective_caller_id_number=)
2012-12-06 11:13:54.456814 [DEBUG] mod_dptools.c:1319
modem/0/1xxxxxxxxxxxxx SET [effective_caller_id_number]=[UNDEF]
EXECUTE modem/0/1xxxxxxxxxxxxx set(absolute_codec_string=PCMA)
2012-12-06 11:13:54.456814 [DEBUG] mod_dptools.c:1319
modem/0/1xxxxxxxxxxxxx SET [absolute_codec_string]=[PCMA]
EXECUTE modem/0/1xxxxxxxxxxxxx set(fax_enable_t38=true)
2012-12-06 11:13:54.456814 [DEBUG] mod_dptools.c:1319
modem/0/1xxxxxxxxxxxxx SET [fax_enable_t38]=[true]
EXECUTE modem/0/1xxxxxxxxxxxxx set(fax_enable_t38_request=true)
2012-12-06 11:13:54.456814 [DEBUG] mod_dptools.c:1319
modem/0/1xxxxxxxxxxxxx SET [fax_enable_t38_request]=[true]
EXECUTE modem/0/1xxxxxxxxxxxxx set(execute_on_answer=t38_gateway self)
2012-12-06 11:13:54.456814 [DEBUG] mod_dptools.c:1319
modem/0/1xxxxxxxxxxxxx SET [execute_on_answer]=[t38_gateway self]
EXECUTE modem/0/1xxxxxxxxxxxxx bridge(sofia/gateway/easybell1/xxxxxxxxxxxxx)
2012-12-06 11:13:54.456814 [DEBUG] switch_channel.c:1062
modem/0/1xxxxxxxxxxxxx EXPORTING[export_vars]
[rtp_autoflush_during_bridge]=[false] to event
2012-12-06 11:13:54.456814 [DEBUG] switch_channel.c:1062
modem/0/1xxxxxxxxxxxxx EXPORTING[export_vars] [RFC2822_DATE]=[Thu, 06
Dec 2012 11:13:54 +0100] to event
2012-12-06 11:13:54.456814 [DEBUG] switch_ivr_originate.c:1961 Parsing
global variables
2012-12-06 11:13:54.456814 [NOTICE] switch_channel.c:941 New Channel
sofia/external/xxxxxxxxxxxxx [b0ac6763-2068-4574-849c-f443d99e978a]
2012-12-06 11:13:54.456814 [DEBUG] mod_sofia.c:4796
(sofia/external/xxxxxxxxxxxxx) State Change CS_NEW -> CS_INIT
2012-12-06 11:13:54.456814 [DEBUG] switch_core_session.c:1229 Send
signal sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:385
(sofia/external/xxxxxxxxxxxxx) Running State Change CS_INIT
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:424
(sofia/external/xxxxxxxxxxxxx) State INIT
2012-12-06 11:13:54.456814 [DEBUG] mod_sofia.c:85
sofia/external/xxxxxxxxxxxxx SOFIA INIT
2012-12-06 11:13:54.456814 [DEBUG] sofia_glue.c:2609 Local SDP:
v=0
o=FreeSWITCH 1354769940 1354769941 IN IP4 212.255.30.34
s=FreeSWITCH
c=IN IP4 212.255.30.34
t=0 0
m=audio 18894 RTP/AVP 98 0 8 3 101 13
a=rtpmap:98 L16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2012-12-06 11:13:54.456814 [DEBUG] mod_sofia.c:125
(sofia/external/xxxxxxxxxxxxx) State Change CS_INIT -> CS_ROUTING
2012-12-06 11:13:54.456814 [DEBUG] switch_core_session.c:1229 Send
signal sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:424
(sofia/external/xxxxxxxxxxxxx) State INIT going to sleep
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:385
(sofia/external/xxxxxxxxxxxxx) Running State Change CS_ROUTING
2012-12-06 11:13:54.456814 [DEBUG] switch_channel.c:1934
(sofia/external/xxxxxxxxxxxxx) Callstate Change DOWN -> RINGING
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:433
(sofia/external/xxxxxxxxxxxxx) State ROUTING
2012-12-06 11:13:54.456814 [DEBUG] mod_sofia.c:148
sofia/external/xxxxxxxxxxxxx SOFIA ROUTING
2012-12-06 11:13:54.456814 [DEBUG] switch_ivr_originate.c:67
(sofia/external/xxxxxxxxxxxxx) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2012-12-06 11:13:54.456814 [DEBUG] switch_core_session.c:1229 Send
signal sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:433
(sofia/external/xxxxxxxxxxxxx) State ROUTING going to sleep
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:385
(sofia/external/xxxxxxxxxxxxx) Running State Change CS_CONSUME_MEDIA
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:452
(sofia/external/xxxxxxxxxxxxx) State CONSUME_MEDIA
2012-12-06 11:13:54.456814 [DEBUG] switch_core_state_machine.c:452
(sofia/external/xxxxxxxxxxxxx) State CONSUME_MEDIA going to sleep
2012-12-06 11:13:54.456814 [DEBUG] switch_core_session.c:924 Send signal
sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:54.456814 [DEBUG] sofia.c:6055 Channel
sofia/external/xxxxxxxxxxxxx entering state [calling][0]
2012-12-06 11:13:54.496801 [DEBUG] switch_core_session.c:924 Send signal
sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:54.496801 [DEBUG] switch_core_session.c:924 Send signal
sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:54.496801 [DEBUG] sofia.c:6055 Channel
sofia/external/xxxxxxxxxxxxx entering state [calling][0]
2012-12-06 11:13:58.656800 [DEBUG] switch_core_session.c:924 Send signal
sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:58.656800 [DEBUG] switch_core_session.c:924 Send signal
sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:58.656800 [DEBUG] sofia.c:6055 Channel
sofia/external/xxxxxxxxxxxxx entering state [proceeding][183]
2012-12-06 11:13:58.656800 [DEBUG] sofia.c:6066 Remote SDP:
v=0
o=- 7632 1354788832 IN IP4 212.172.97.124
s=-
c=IN IP4 212.172.97.124
t=0 0
m=audio 32392 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=ptime:20
a=nortpproxy:yes

2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash
2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:3926 Deciding whether to
pass zrtp-hash between legs
2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:3928
CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash
2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:3926 Deciding whether to
pass zrtp-hash between legs
2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:3928
CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:5044 Audio Codec Compare
[PCMA:8:8000:20:64000]/[L16:70:8000:20:128000]
2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:5044 Audio Codec Compare
[PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:5044 Audio Codec Compare
[PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:3027 Set Codec
sofia/external/xxxxxxxxxxxxx PCMA/8000 20 ms 160 samples 64000 bits
2012-12-06 11:13:58.656800 [DEBUG] switch_core_codec.c:111
sofia/external/xxxxxxxxxxxxx Original read codec set to PCMA:8
2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:5158 Set 2833 dtmf send
payload to 101
2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:3276 AUDIO RTP
[sofia/external/xxxxxxxxxxxxx] 212.255.30.34 port 18894 ->
212.172.97.124 port 32392 codec: 8 ms: 20
2012-12-06 11:13:58.656800 [DEBUG] switch_rtp.c:1927 Starting timer
[soft] 160 bytes per 20ms
2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:3540 Set 2833 dtmf send
payload to 101
2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:3546 Set 2833 dtmf
receive payload to 101
2012-12-06 11:13:58.656800 [DEBUG] sofia_glue.c:3573
sofia/external/xxxxxxxxxxxxx Set rtp dtmf delay to 40
2012-12-06 11:13:58.656800 [NOTICE] sofia_glue.c:4176 Pre-Answer
sofia/external/xxxxxxxxxxxxx!
2012-12-06 11:13:58.656800 [DEBUG] switch_channel.c:3057
(sofia/external/xxxxxxxxxxxxx) Callstate Change RINGING -> EARLY
2012-12-06 11:13:58.656800 [DEBUG] switch_channel.c:3099 Send signal
modem/0/1xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:58.656800 [DEBUG] mod_spandsp_modem.c:581
modem/0/1xxxxxxxxxxxxx CHANNEL KILL
2012-12-06 11:13:58.656800 [DEBUG] mod_spandsp_modem.c:1131 Modem
/dev/FS0 [DIALING] - RNG 0
2012-12-06 11:13:58.656800 [DEBUG] mod_spandsp_modem.c:738 Modem
/dev/FS0 [DIALING] - Changing state to CONNECTED
2012-12-06 11:13:58.656800 [DEBUG] switch_core_session.c:840 Send signal
sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:58.656800 [DEBUG] switch_core_session.c:778 Send signal
modem/0/1xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:58.656800 [DEBUG] mod_spandsp_modem.c:581
modem/0/1xxxxxxxxxxxxx CHANNEL KILL
2012-12-06 11:13:58.656800 [NOTICE] switch_ivr_originate.c:3309
Pre-Answer modem/0/1xxxxxxxxxxxxx!
2012-12-06 11:13:58.656800 [DEBUG] switch_channel.c:3057
(modem/0/1xxxxxxxxxxxxx) Callstate Change RINGING -> EARLY
2012-12-06 11:13:58.656800 [DEBUG] switch_ivr_originate.c:3360 Originate
Resulted in Success: [sofia/external/xxxxxxxxxxxxx]
2012-12-06 11:13:58.656800 [DEBUG] switch_core_session.c:778 Send signal
sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:58.656800 [DEBUG] switch_core_session.c:840 Send signal
sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:58.656800 [DEBUG] switch_core_session.c:778 Send signal
modem/0/1xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:58.656800 [DEBUG] mod_spandsp_modem.c:581
modem/0/1xxxxxxxxxxxxx CHANNEL KILL
2012-12-06 11:13:58.656800 [DEBUG] switch_ivr_bridge.c:1359
(sofia/external/xxxxxxxxxxxxx) State Change CS_CONSUME_MEDIA ->
CS_EXCHANGE_MEDIA
2012-12-06 11:13:58.656800 [DEBUG] switch_core_session.c:1229 Send
signal sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:58.656800 [DEBUG] switch_core_state_machine.c:385
(sofia/external/xxxxxxxxxxxxx) Running State Change CS_EXCHANGE_MEDIA
2012-12-06 11:13:58.656800 [DEBUG] switch_core_state_machine.c:443
(sofia/external/xxxxxxxxxxxxx) State EXCHANGE_MEDIA
2012-12-06 11:13:58.656800 [DEBUG] mod_sofia.c:652 SOFIA EXCHANGE_MEDIA
2012-12-06 11:13:58.696829 [DEBUG] switch_rtp.c:3594 Correct ip/port
confirmed.
2012-12-06 11:13:58.936865 [DEBUG] switch_core_session.c:924 Send signal
sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:58.936865 [DEBUG] switch_core_session.c:924 Send signal
sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:58.956952 [DEBUG] sofia.c:6048 Channel
sofia/external/xxxxxxxxxxxxx skipping state [proceeding][180]
2012-12-06 11:13:59.956847 [DEBUG] switch_core_session.c:924 Send signal
sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:59.956847 [DEBUG] switch_core_session.c:924 Send signal
sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:59.956847 [DEBUG] switch_core_session.c:924 Send signal
sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:59.976824 [DEBUG] sofia.c:6055 Channel
sofia/external/xxxxxxxxxxxxx entering state [terminated][486]
2012-12-06 11:13:59.976824 [DEBUG] switch_channel.c:2914
(sofia/external/xxxxxxxxxxxxx) Callstate Change EARLY -> HANGUP
2012-12-06 11:13:59.976824 [NOTICE] sofia.c:6847 Hangup
sofia/external/xxxxxxxxxxxxx [CS_EXCHANGE_MEDIA] [USER_BUSY]
2012-12-06 11:13:59.976824 [DEBUG] switch_channel.c:2937 Send signal
sofia/external/xxxxxxxxxxxxx [KILL]
2012-12-06 11:13:59.976824 [DEBUG] switch_core_session.c:1229 Send
signal sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:59.976824 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD
DONE [sofia/external/xxxxxxxxxxxxx]
2012-12-06 11:13:59.976824 [DEBUG] switch_ivr_bridge.c:613 Send signal
modem/0/1xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:59.976824 [DEBUG] mod_spandsp_modem.c:581
modem/0/1xxxxxxxxxxxxx CHANNEL KILL
2012-12-06 11:13:59.976824 [DEBUG] switch_core_state_machine.c:443
(sofia/external/xxxxxxxxxxxxx) State EXCHANGE_MEDIA going to sleep
2012-12-06 11:13:59.976824 [DEBUG] switch_core_state_machine.c:385
(sofia/external/xxxxxxxxxxxxx) Running State Change CS_HANGUP
2012-12-06 11:13:59.976824 [DEBUG] switch_core_state_machine.c:625
(sofia/external/xxxxxxxxxxxxx) State HANGUP
2012-12-06 11:13:59.976824 [DEBUG] mod_sofia.c:474 Channel
sofia/external/xxxxxxxxxxxxx hanging up, cause: USER_BUSY
2012-12-06 11:13:59.976824 [DEBUG] switch_core_state_machine.c:47
sofia/external/xxxxxxxxxxxxx Standard HANGUP, cause: USER_BUSY
2012-12-06 11:13:59.976824 [DEBUG] switch_core_state_machine.c:625
(sofia/external/xxxxxxxxxxxxx) State HANGUP going to sleep
2012-12-06 11:13:59.976824 [DEBUG] switch_core_state_machine.c:416
(sofia/external/xxxxxxxxxxxxx) State Change CS_HANGUP -> CS_REPORTING
2012-12-06 11:13:59.976824 [DEBUG] switch_core_session.c:1229 Send
signal sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:59.976824 [DEBUG] switch_core_state_machine.c:385
(sofia/external/xxxxxxxxxxxxx) Running State Change CS_REPORTING
2012-12-06 11:13:59.976824 [DEBUG] switch_core_state_machine.c:685
(sofia/external/xxxxxxxxxxxxx) State REPORTING
2012-12-06 11:13:59.976824 [DEBUG] switch_core_state_machine.c:79
sofia/external/xxxxxxxxxxxxx Standard REPORTING, cause: USER_BUSY
2012-12-06 11:13:59.976824 [DEBUG] switch_core_state_machine.c:685
(sofia/external/xxxxxxxxxxxxx) State REPORTING going to sleep
2012-12-06 11:13:59.976824 [DEBUG] switch_core_state_machine.c:410
(sofia/external/xxxxxxxxxxxxx) State Change CS_REPORTING -> CS_DESTROY
2012-12-06 11:13:59.976824 [DEBUG] switch_core_session.c:1229 Send
signal sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:59.976824 [DEBUG] switch_core_session.c:1429 Session 24
(sofia/external/xxxxxxxxxxxxx) Locked, Waiting on external entities
2012-12-06 11:13:59.996802 [DEBUG] switch_ivr_bridge.c:501
sofia/external/xxxxxxxxxxxxx ending bridge by request from write function
2012-12-06 11:13:59.996802 [DEBUG] switch_ivr_bridge.c:588 BRIDGE THREAD
DONE [modem/0/1xxxxxxxxxxxxx]
2012-12-06 11:13:59.996802 [DEBUG] switch_ivr_bridge.c:613 Send signal
sofia/external/xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:59.996802 [DEBUG] switch_core_session.c:778 Send signal
modem/0/1xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:59.996802 [DEBUG] mod_spandsp_modem.c:581
modem/0/1xxxxxxxxxxxxx CHANNEL KILL
2012-12-06 11:13:59.996802 [NOTICE] switch_core_session.c:1447 Session
24 (sofia/external/xxxxxxxxxxxxx) Ended
2012-12-06 11:13:59.996802 [NOTICE] switch_core_session.c:1449 Close
Channel sofia/external/xxxxxxxxxxxxx [CS_DESTROY]
EXECUTE modem/0/1xxxxxxxxxxxxx hangup()
2012-12-06 11:13:59.996802 [DEBUG] switch_channel.c:2914
(modem/0/1xxxxxxxxxxxxx) Callstate Change EARLY -> HANGUP
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:514
(sofia/external/xxxxxxxxxxxxx) Callstate Change HANGUP -> DOWN
2012-12-06 11:13:59.996802 [NOTICE] mod_dptools.c:1134 Hangup
modem/0/1xxxxxxxxxxxxx [CS_EXECUTE] [NORMAL_CLEARING]
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:517
(sofia/external/xxxxxxxxxxxxx) Running State Change CS_DESTROY
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:527
(sofia/external/xxxxxxxxxxxxx) State DESTROY
2012-12-06 11:13:59.996802 [DEBUG] mod_sofia.c:374
sofia/external/xxxxxxxxxxxxx SOFIA DESTROY
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:86
sofia/external/xxxxxxxxxxxxx Standard DESTROY
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:527
(sofia/external/xxxxxxxxxxxxx) State DESTROY going to sleep
2012-12-06 11:13:59.996802 [DEBUG] switch_channel.c:2937 Send signal
modem/0/1xxxxxxxxxxxxx [KILL]
2012-12-06 11:13:59.996802 [DEBUG] mod_spandsp_modem.c:581
modem/0/1xxxxxxxxxxxxx CHANNEL KILL
2012-12-06 11:13:59.996802 [DEBUG] switch_core_session.c:1229 Send
signal modem/0/1xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:59.996802 [DEBUG] mod_spandsp_modem.c:581
modem/0/1xxxxxxxxxxxxx CHANNEL KILL
2012-12-06 11:13:59.996802 [DEBUG] switch_core_session.c:2345
modem/0/1xxxxxxxxxxxxx skip receive message [APPLICATION_EXEC_COMPLETE]
(channel is hungup already)
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:440
(modem/0/1xxxxxxxxxxxxx) State EXECUTE going to sleep
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:385
(modem/0/1xxxxxxxxxxxxx) Running State Change CS_HANGUP
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:625
(modem/0/1xxxxxxxxxxxxx) State HANGUP
2012-12-06 11:13:59.996802 [DEBUG] mod_spandsp_modem.c:553
modem/0/1xxxxxxxxxxxxx CHANNEL HANGUP
2012-12-06 11:13:59.996802 [DEBUG] mod_spandsp_modem.c:1070 Modem
/dev/FS0 [CONNECTED] - Hanging up
2012-12-06 11:13:59.996802 [DEBUG] mod_spandsp_modem.c:1076 Modem
/dev/FS0 [CONNECTED] - Changing state to HANGUP
2012-12-06 11:13:59.996802 [DEBUG] mod_spandsp_modem.c:1095 Modem
/dev/FS0 [HANGUP] - Changing state to ONHOOK
2012-12-06 11:13:59.996802 [DEBUG] mod_spandsp_modem.c:1131 Modem
/dev/FS0 [ONHOOK] - RNG 0
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:47
modem/0/1xxxxxxxxxxxxx Standard HANGUP, cause: NORMAL_CLEARING
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:625
(modem/0/1xxxxxxxxxxxxx) State HANGUP going to sleep
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:416
(modem/0/1xxxxxxxxxxxxx) State Change CS_HANGUP -> CS_REPORTING
2012-12-06 11:13:59.996802 [DEBUG] switch_core_session.c:1229 Send
signal modem/0/1xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:59.996802 [DEBUG] mod_spandsp_modem.c:581
modem/0/1xxxxxxxxxxxxx CHANNEL KILL
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:385
(modem/0/1xxxxxxxxxxxxx) Running State Change CS_REPORTING
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:685
(modem/0/1xxxxxxxxxxxxx) State REPORTING
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:79
modem/0/1xxxxxxxxxxxxx Standard REPORTING, cause: NORMAL_CLEARING
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:685
(modem/0/1xxxxxxxxxxxxx) State REPORTING going to sleep
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:410
(modem/0/1xxxxxxxxxxxxx) State Change CS_REPORTING -> CS_DESTROY
2012-12-06 11:13:59.996802 [DEBUG] switch_core_session.c:1229 Send
signal modem/0/1xxxxxxxxxxxxx [BREAK]
2012-12-06 11:13:59.996802 [DEBUG] mod_spandsp_modem.c:581
modem/0/1xxxxxxxxxxxxx CHANNEL KILL
2012-12-06 11:13:59.996802 [DEBUG] switch_core_session.c:1429 Session 23
(modem/0/1xxxxxxxxxxxxx) Locked, Waiting on external entities
2012-12-06 11:13:59.996802 [NOTICE] switch_core_session.c:1447 Session
23 (modem/0/1xxxxxxxxxxxxx) Ended
2012-12-06 11:13:59.996802 [NOTICE] switch_core_session.c:1449 Close
Channel modem/0/1xxxxxxxxxxxxx [CS_DESTROY]
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:514
(modem/0/1xxxxxxxxxxxxx) Callstate Change HANGUP -> DOWN
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:517
(modem/0/1xxxxxxxxxxxxx) Running State Change CS_DESTROY
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:527
(modem/0/1xxxxxxxxxxxxx) State DESTROY
2012-12-06 11:13:59.996802 [DEBUG] mod_spandsp_modem.c:534 Modem
/dev/FS0 [ONHOOK] - Changing state to ONHOOK
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:86
modem/0/1xxxxxxxxxxxxx Standard DESTROY
2012-12-06 11:13:59.996802 [DEBUG] switch_core_state_machine.c:527
(modem/0/1xxxxxxxxxxxxx) State DESTROY going to sleep
2012-12-06 11:14:00.996881 [DEBUG] mod_spandsp_modem.c:1070 Modem
/dev/FS0 [ONHOOK] - Hanging up
2012-12-06 11:14:00.996881 [DEBUG] mod_spandsp_modem.c:1076 Modem
/dev/FS0 [ONHOOK] - Changing state to HANGUP
2012-12-06 11:14:00.996881 [DEBUG] mod_spandsp_modem.c:1095 Modem
/dev/FS0 [HANGUP] - Changing state to ONHOOK
2012-12-06 11:14:19.676800 [DEBUG] mod_spandsp_modem.c:1070 Modem
/dev/FS0 [ONHOOK] - Hanging up
2012-12-06 11:14:19.676800 [DEBUG] mod_spandsp_modem.c:1076 Modem
/dev/FS0 [ONHOOK] - Changing state to HANGUP
2012-12-06 11:14:19.676800 [DEBUG] mod_spandsp_modem.c:1095 Modem
/dev/FS0 [HANGUP] - Changing state to ONHOOK
freeswitch at internal>



From shaik.bawajan at gmail.com  Thu Dec  6 13:29:53 2012
From: shaik.bawajan at gmail.com (bawajan)
Date: Thu, 6 Dec 2012 02:29:53 -0800 (PST)
Subject: [Freeswitch-users] having problem using java with esl library
Message-ID: <CAB3HJytM41noZgqhqsa492MC+nNEc4S7AvWmMi+8X4UcOdZA_w@mail.gmail.com>

I am new to freeswitch and am using java app with esl library to make
outbound calls and am getting the error is "Create additional event
dispatch thread 2".

After that in the freeswitch logs its printing - Event 0 Blocking and its
keeps printing.

2012-12-05 13:46:02.179956 [WARNING] switch_event.c:607 Create additional
event dispatch thread 2
2012-12-05 13:46:19.919951 [WARNING] switch_event.c:607 Create additional
event dispatch thread 3
2012-12-05 13:46:46.159954 [WARNING] switch_event.c:607 Create additional
event dispatch thread 4
2012-12-05 13:47:03.679956 [WARNING] switch_event.c:607 Create additional
event dispatch thread 5
2012-12-05 13:47:24.999950 [WARNING] switch_event.c:607 Create additional
event dispatch thread 6
2012-12-05 13:47:40.299950 [WARNING] switch_event.c:607 Create additional
event dispatch thread 7
2012-12-05 13:47:54.879950 [WARNING] switch_event.c:348 Event Thread 0 is
blocking
2012-12-05 13:47:55.079953 [WARNING] switch_event.c:348 Event Thread 0 is
blocking
2012-12-05 13:47:55.139957 [WARNING] switch_event.c:348 Event Thread 0 is
blocking
2012-12-05 13:47:55.639950 [WARNING] switch_event.c:348 Event Thread 0 is
blocking
2012-12-05 13:47:55.819960 [WARNING] switch_event.c:348 Event Thread 0 is
blocking
2012-12-05 13:47:55.859950 [WARNING] switch_event.c:348 Event Thread 0 is
blocking
2012-12-05 13:47:55.939953 [WARNING] switch_event.c:348 Event Thread 0 is
blocking



Kindly let me know where am doing mistake and how to resolve it.

Thanks in Advance,
Bawajan




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/having-problem-using-java-with-esl-library-tp7585250.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From rob at corp.coastside.net  Thu Dec  6 13:58:21 2012
From: rob at corp.coastside.net (Rob Genovesi)
Date: Thu, 6 Dec 2012 02:58:21 -0800
Subject: [Freeswitch-users] installing on Amazon EC2
In-Reply-To: <804D48104511D4468F0D60DF9D3100350AE3B177@MAIL.millicorp.com>
References: <CA+x1V=PYYNVRZZ642e8AHSw0RM5opJkegbSmAR4oT=HxjiJ8tw@mail.gmail.com>
	<804D48104511D4468F0D60DF9D3100350AE3B177@MAIL.millicorp.com>
Message-ID: <CA+x1V=O3Hzs-XYSuFBUN6oZcxV79bgLGV9xjpsim9cndqQ+6Bg@mail.gmail.com>

Tim,

I discovered that running the following command prior to running
"./bootstrap.sh" appears to have solved the problem:

export LDFLAGS="-ltinfo"

I'm not sure why configure doesn't detect it automatically but adding
the ld flag manually seems to have done the trick.


-Rob


On Wed, Dec 5, 2012 at 3:30 PM, Tim Meade <Tim.Meade at millicorp.com> wrote:
> I had this issue using the base amazon version of Linux.  I don't think we ever got past it.   I ended up using a centos instance for the base.
>
> If you do figure it out please let us know!
>
> Tim
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Genovesi
> Sent: Wednesday, December 05, 2012 6:15 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] installing on Amazon EC2
>
> Hi all,
>
> I'm trying to build/install Free Switch on an Amazon EC2 server (Amazon Linux AMI 64-bit) and "./configure" keeps exiting with the following error :
>
> ./configure
> <snip>
> checking for the version of libcurl... 7.24.0 checking for libcurl >= version 7.13.0... yes checking whether libcurl is usable... no no checking for tgetent in -lncurses... no checking for tgetent in -lcurses... no
> configure: error: libtermcap, libcurses or libncurses are required!
>
> (notice the extra "no" on a line by itself)
>
> ....
>
> I have installed all the ncurses packages I can find, to no avail :
>
> rpm -qa | grep -E 'curses|ncurses|term|curl'
> ncurses-term-5.7-3.20090208.9.amzn1.x86_64
> ncurses-libs-5.7-3.20090208.9.amzn1.i686
> python-pycurl-7.19.0-8.7.amzn1.x86_64
> libcurl-7.24.0-5.25.amzn1.x86_64
> ncurses-base-5.7-3.20090208.9.amzn1.x86_64
> ncurses-libs-5.7-3.20090208.9.amzn1.x86_64
> ncurses-5.7-3.20090208.9.amzn1.x86_64
> curl-7.24.0-5.25.amzn1.x86_64
> ncurses-devel-5.7-3.20090208.9.amzn1.x86_64
> libcurl-devel-7.24.0-5.25.amzn1.x86_64
>
> ....
>
> I have been able to install succcessfully on a local CentOS machine and Rackspace CentOS 6.3 VM, I believe the issue is something specific to Amazon EC2 but I haven't been able to figure it out.
>
> Any suggestions for a fix/workaround would be appreciated.
>
>
>
> Thanks,
>
> Rob
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Dec  6 15:10:18 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 6 Dec 2012 12:10:18 +0000
Subject: [Freeswitch-users] installing on Amazon EC2
In-Reply-To: <CAMwZ-tJOyHm7Z8tS-ZDMNSSajib3F1y_xi38cchuFDAYZ=kDjA@mail.gmail.com>
References: <CA+x1V=PYYNVRZZ642e8AHSw0RM5opJkegbSmAR4oT=HxjiJ8tw@mail.gmail.com>
	<804D48104511D4468F0D60DF9D3100350AE3B177@MAIL.millicorp.com>
	<CAMwZ-tJOyHm7Z8tS-ZDMNSSajib3F1y_xi38cchuFDAYZ=kDjA@mail.gmail.com>
Message-ID: <CAMbW9o+Z95K53s5KW3sO9-M-hWUJZrtN_G+TBAyxBvbAM6kPQw@mail.gmail.com>

Gabe, I used to use Rackspace, generally it was good, but had the odd
stutters. However, I was also testing it on 512MB RAM.

On 6 December 2012 02:27, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Wed, Dec 5, 2012 at 4:30 PM, Tim Meade <Tim.Meade at millicorp.com> wrote:
> > I ended up using a centos instance for the base.
>
> What's your voice quality like?
>
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Tim.Meade at Millicorp.com  Thu Dec  6 16:15:25 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Thu, 6 Dec 2012 13:15:25 +0000
Subject: [Freeswitch-users] installing on Amazon EC2
In-Reply-To: <CA+x1V=O3Hzs-XYSuFBUN6oZcxV79bgLGV9xjpsim9cndqQ+6Bg@mail.gmail.com>
References: <CA+x1V=PYYNVRZZ642e8AHSw0RM5opJkegbSmAR4oT=HxjiJ8tw@mail.gmail.com>
	<804D48104511D4468F0D60DF9D3100350AE3B177@MAIL.millicorp.com>
	<CA+x1V=O3Hzs-XYSuFBUN6oZcxV79bgLGV9xjpsim9cndqQ+6Bg@mail.gmail.com>
Message-ID: <804D48104511D4468F0D60DF9D3100350AE3F514@MAIL.millicorp.com>

Thanks Rob!

I'll spin up one of the Base Amazon Linux ones for testing then.  

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Genovesi
Sent: Thursday, December 06, 2012 5:58 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] installing on Amazon EC2

Tim,

I discovered that running the following command prior to running "./bootstrap.sh" appears to have solved the problem:

export LDFLAGS="-ltinfo"

I'm not sure why configure doesn't detect it automatically but adding the ld flag manually seems to have done the trick.


-Rob


On Wed, Dec 5, 2012 at 3:30 PM, Tim Meade <Tim.Meade at millicorp.com> wrote:
> I had this issue using the base amazon version of Linux.  I don't think we ever got past it.   I ended up using a centos instance for the base.
>
> If you do figure it out please let us know!
>
> Tim
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> Rob Genovesi
> Sent: Wednesday, December 05, 2012 6:15 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] installing on Amazon EC2
>
> Hi all,
>
> I'm trying to build/install Free Switch on an Amazon EC2 server (Amazon Linux AMI 64-bit) and "./configure" keeps exiting with the following error :
>
> ./configure
> <snip>
> checking for the version of libcurl... 7.24.0 checking for libcurl >= 
> version 7.13.0... yes checking whether libcurl is usable... no no 
> checking for tgetent in -lncurses... no checking for tgetent in 
> -lcurses... no
> configure: error: libtermcap, libcurses or libncurses are required!
>
> (notice the extra "no" on a line by itself)
>
> ....
>
> I have installed all the ncurses packages I can find, to no avail :
>
> rpm -qa | grep -E 'curses|ncurses|term|curl'
> ncurses-term-5.7-3.20090208.9.amzn1.x86_64
> ncurses-libs-5.7-3.20090208.9.amzn1.i686
> python-pycurl-7.19.0-8.7.amzn1.x86_64
> libcurl-7.24.0-5.25.amzn1.x86_64
> ncurses-base-5.7-3.20090208.9.amzn1.x86_64
> ncurses-libs-5.7-3.20090208.9.amzn1.x86_64
> ncurses-5.7-3.20090208.9.amzn1.x86_64
> curl-7.24.0-5.25.amzn1.x86_64
> ncurses-devel-5.7-3.20090208.9.amzn1.x86_64
> libcurl-devel-7.24.0-5.25.amzn1.x86_64
>
> ....
>
> I have been able to install succcessfully on a local CentOS machine and Rackspace CentOS 6.3 VM, I believe the issue is something specific to Amazon EC2 but I haven't been able to figure it out.
>
> Any suggestions for a fix/workaround would be appreciated.
>
>
>
> Thanks,
>
> Rob
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting 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-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
>
> FreeSWITCH-users 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

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


From jnvines at gmail.com  Thu Dec  6 16:27:09 2012
From: jnvines at gmail.com (Nick Vines)
Date: Thu, 6 Dec 2012 08:27:09 -0500
Subject: [Freeswitch-users] installing on Amazon EC2
In-Reply-To: <804D48104511D4468F0D60DF9D3100350AE3F514@MAIL.millicorp.com>
References: <CA+x1V=PYYNVRZZ642e8AHSw0RM5opJkegbSmAR4oT=HxjiJ8tw@mail.gmail.com>
	<804D48104511D4468F0D60DF9D3100350AE3B177@MAIL.millicorp.com>
	<CA+x1V=O3Hzs-XYSuFBUN6oZcxV79bgLGV9xjpsim9cndqQ+6Bg@mail.gmail.com>
	<804D48104511D4468F0D60DF9D3100350AE3F514@MAIL.millicorp.com>
Message-ID: <CA+NjZ2u0nzwpEOdqHf2555eNKNt+gDncL85Wy9pnAz8RMvL0Sg@mail.gmail.com>

Tim, since you are about to set it up, would you mind updating the wiki
page with how you get it to work, and perhaps the quality? I know there are
probably a few interested people out there, myself included.

http://wiki.freeswitch.org/wiki/Amazon_ec2



Thanks,
Nick




On Thu, Dec 6, 2012 at 8:15 AM, Tim Meade <Tim.Meade at millicorp.com> wrote:

> Thanks Rob!
>
> I'll spin up one of the Base Amazon Linux ones for testing then.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Genovesi
> Sent: Thursday, December 06, 2012 5:58 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] installing on Amazon EC2
>
> Tim,
>
> I discovered that running the following command prior to running
> "./bootstrap.sh" appears to have solved the problem:
>
> export LDFLAGS="-ltinfo"
>
> I'm not sure why configure doesn't detect it automatically but adding the
> ld flag manually seems to have done the trick.
>
>
> -Rob
>
>
> On Wed, Dec 5, 2012 at 3:30 PM, Tim Meade <Tim.Meade at millicorp.com> wrote:
> > I had this issue using the base amazon version of Linux.  I don't think
> we ever got past it.   I ended up using a centos instance for the base.
> >
> > If you do figure it out please let us know!
> >
> > Tim
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Rob Genovesi
> > Sent: Wednesday, December 05, 2012 6:15 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: [Freeswitch-users] installing on Amazon EC2
> >
> > Hi all,
> >
> > I'm trying to build/install Free Switch on an Amazon EC2 server (Amazon
> Linux AMI 64-bit) and "./configure" keeps exiting with the following error :
> >
> > ./configure
> > <snip>
> > checking for the version of libcurl... 7.24.0 checking for libcurl >=
> > version 7.13.0... yes checking whether libcurl is usable... no no
> > checking for tgetent in -lncurses... no checking for tgetent in
> > -lcurses... no
> > configure: error: libtermcap, libcurses or libncurses are required!
> >
> > (notice the extra "no" on a line by itself)
> >
> > ....
> >
> > I have installed all the ncurses packages I can find, to no avail :
> >
> > rpm -qa | grep -E 'curses|ncurses|term|curl'
> > ncurses-term-5.7-3.20090208.9.amzn1.x86_64
> > ncurses-libs-5.7-3.20090208.9.amzn1.i686
> > python-pycurl-7.19.0-8.7.amzn1.x86_64
> > libcurl-7.24.0-5.25.amzn1.x86_64
> > ncurses-base-5.7-3.20090208.9.amzn1.x86_64
> > ncurses-libs-5.7-3.20090208.9.amzn1.x86_64
> > ncurses-5.7-3.20090208.9.amzn1.x86_64
> > curl-7.24.0-5.25.amzn1.x86_64
> > ncurses-devel-5.7-3.20090208.9.amzn1.x86_64
> > libcurl-devel-7.24.0-5.25.amzn1.x86_64
> >
> > ....
> >
> > I have been able to install succcessfully on a local CentOS machine and
> Rackspace CentOS 6.3 VM, I believe the issue is something specific to
> Amazon EC2 but I haven't been able to figure it out.
> >
> > Any suggestions for a fix/workaround would be appreciated.
> >
> >
> >
> > Thanks,
> >
> > Rob
> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting 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-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
>
>
> >
> > FreeSWITCH-users 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
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 lists at kavun.ch  Thu Dec  6 16:57:31 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Dec 2012 08:57:31 -0500
Subject: [Freeswitch-users] Using bypass_media or proxy_media selectively
Message-ID: <343F3B15-35E9-4CCF-A2BB-2514C1167E4C@kavun.ch>

Hi there,

Can you please confirm that I need to either use proxy_media or bypass_media for a video call in H.264?
If so, how do I enable proxy_media or bypass_media conditionally by examining the codec prefs of both SIP endpoints? 

For a setup where I have 2 devices registered as 1000 at mydomain, one being audio only and the other video capable. Plus a video capable device registered as 1001 at mydomain.
How do I selectively use bypass_media or proxy_media for calls made by 1001 to 1000 and answered by the video capable terminal?

Thanks a million,
Emrah

From lists at kavun.ch  Thu Dec  6 17:00:04 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Dec 2012 09:00:04 -0500
Subject: [Freeswitch-users] Distributed Conference Room
In-Reply-To: <CAK9Kg4HzR9zwiBQXEEkVFwz5rGEWh84U0RkoW8f00aMKJ--wMQ@mail.gmail.com>
References: <CAK9Kg4HzR9zwiBQXEEkVFwz5rGEWh84U0RkoW8f00aMKJ--wMQ@mail.gmail.com>
Message-ID: <88380BD1-EBB7-401B-8D74-A7CF7D8266DF@kavun.ch>

This is most interesting and I am curious to know the answer as well.

Cheers,
E
On Dec 5, 2012, at 5:16 PM, Daniel Eiland <daniel.eiland at gmail.com> wrote:

> Hi guys,
> 
> I'm trying to deploy a conferencing solution using FreeSWITCH and running into a small issue with fail over / hot-standbys.
> 
> In my environment, I've got multiple FreeSWITCH/Conference endpoints registered with an OpenSIPS proxy.  When calls come into OpenSIPS they are routed to the FreeSWITCH endpoints based on their q-values.  If a FreeSWITCH instance fails (namely the one with the highest q-value), the call is simply routed to the next instance.  This works great in most situations, however in some cases (namely network congestion) the FreeSWITCH w/highest priority is simply temporarily unavailable and callers to the same conference endpoint land on different servers.
> 
> I'm wondering if there is a mechanism for distributing (or sharing) a conference room across multiple FreeSWITCH instances.  Namely, if a user lands in a conference hosted on server A while another lands in the same conference on server B, is there a mechanism in FreeSWITCH to connect the two servers/conferences (Presumably some "static" connection between the servers/rooms) so they can still talk with each over?
> 
> Thanks,
> Daniel
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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



From lists at kavun.ch  Thu Dec  6 17:09:13 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Dec 2012 09:09:13 -0500
Subject: [Freeswitch-users] Missed call notification
In-Reply-To: <BLU169-W2283D9F938CCE5C8E60C2D8C400@phx.gbl>
References: <BLU169-W120766FAB52400013BA1BB28C420@phx.gbl>,
	<8AE80082-5AAC-4C25-87E9-52FBD5EDB593@endigotech.com>,
	<CAAtWKeFdgZ_MyPBiNUsDYZZsBRJMCe=d6vTSynxDtZ+ifi+KTQ@mail.gmail.com>,
	<BLU169-W118507FC5529D5F80F26BA18C420@phx.gbl>,
	<CA+zebiYsNB0vhnDuPZo=d_uB_9tqE5W3fAzEnrdnsvVtVB4OPg@mail.gmail.com>,
	<CAAtWKeE9nFAFg1YWpb+B1BwLrTOKpOdMhJ9KEYE0E6KuGG3gNA@mail.gmail.com>
	<BLU169-W2283D9F938CCE5C8E60C2D8C400@phx.gbl>
Message-ID: <A4AE4A33-2735-4E89-A7F8-2CEB22CA6C09@kavun.ch>

I might be missing something, but this only works if the bridge fails and not if the caller hangs up before reaching voicemail.

My work around is to schedule the lua as a hangup hook and check a few things.
From the top of my head, it was the channel status or bridge status; and the voicemail recording duration should either be empty or 0. This way you get notified if someone lands in the VM and doesn't leave a message.

Emrah

On Dec 3, 2012, at 11:44 AM, ?? <itispip-qq at hotmail.com> wrote:

> Thanks Yehavi, it's really handy to implement this script to my environment :)
> 
> /Alex
> 
> Date: Sun, 2 Dec 2012 10:21:17 +0200
> From: yehavi.bourvine at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Missed call notification
> 
> Hi,
>  
>   I've created the following wiki page with example of how we do it:
> http://wiki.freeswitch.org/wiki/MailNoAsnwer
>  
> I would be greatfull if someone can add it to the examples page, as I don't have the edit authority on it...
>  
>                     Thanks, __Yehavi: 
> 
> 
> 2012/12/1 Komar, Jason <jkomar at jbox.ca>
> I would be interested to see that script too.
> 
> Thanks,
> Jason
> 
> 
> On Sat, Dec 1, 2012 at 11:34 AM, ?? <itispip-qq at hotmail.com> wrote:
> 
> Hi Yehavi, seems your solution is the best fit for the situation. Is that possible to share your script? Maybe can posted it to Freeswitch Wiki as part of the documentation.
> 
> Date: Sat, 1 Dec 2012 19:13:11 +0200
> From: yehavi.bourvine at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Missed call notification
> 
> We have a script to find the hangup cause and send Email if the user requested it. This way I know who is looking for me while I am on the phone :-)
>  
>                       __Yehavi: 
> 
> 
> 2012/12/1 Brian Foster <bdfoster at endigotech.com>
> Message Waiting Indicator (MWI) is the voicemail notification feature for IP phones. 
> 
> Missed call notifications are provided by the phone and not the switch.
> 
> Sent from my iPhone
> 
> On Dec 1, 2012, at 2:41 AM, ?? <itispip-qq at hotmail.com> wrote:
> 
> I heard that Freeswitch support missed call notification (i/o VM notification). But look around Wiki no document about this feature. Anyone knows where to setup it up?
> 
> /brgds, 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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.orghttp://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.orghttp://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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



From gozdal at gmail.com  Thu Dec  6 17:17:27 2012
From: gozdal at gmail.com (Marcin Gozdalik)
Date: Thu, 6 Dec 2012 15:17:27 +0100
Subject: [Freeswitch-users] SIP/RTP monitoring
Message-ID: <CADu1mRMpCuvvxpSY-WYLW1gfnJD4xWYOeV4bFUAfAGAcQGvR_A@mail.gmail.com>

Hi

I was wondering if any of you would be interested in
evaluating/using/ultimately paying for call quality monitoring software.

We've developed a call quality monitoring solution as a part of a bigger
project. The tool is not public yet and we are considering providing it on
commercial terms.
I am aware of VoIPmonitor but it seems it serves another purpose -
monitoring call quality at central point in the providers' network. Our
idea is to provide a tool for an unqualified end-user to download and run
on a Windows machine. The tool could be run in response to issues raised by
the end-user or even before the service is provided to end-user to assess
if end-user's network connection is sufficient to make VoIP calls.
Technically speaking, the tool sets up some SIP/RTP calls (either one-off
or continuously), monitors everything regarding SIP (detecting NATs, ALGs,
etc) and RTP (jitter, packet loss, etc.) and returns the data to central
server where it can be analyzed by VoIP provider using Web GUI.

I'd appreciate if subscribers of this list would speak up if they see a
need for such a tool. Please respond here or privately.

Best regards,

-- 
Marcin Gozdalik
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From lists at kavun.ch  Thu Dec  6 17:18:05 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Dec 2012 09:18:05 -0500
Subject: [Freeswitch-users] Domains and profiles
In-Reply-To: <8ED4CC3C-441A-4AA0-8DDB-1B57CE646A3C@kavun.ch>
References: <624B624D-6C0E-461C-A7E6-9D271502BBBD@insensate.co.uk>
	<1A6CD519-9A32-4CD3-9DBB-2C4FC11D9A0D@kavun.ch>
	<CAKbxfG-31nxkyo1niEzWvcv12SLr-pdHWm_tKvM4uWoi1G46bA@mail.gmail.com>
	<3FC39EED-0A60-4A87-866D-0AAA629F9560@kavun.ch>
	<8ED4CC3C-441A-4AA0-8DDB-1B57CE646A3C@kavun.ch>
Message-ID: <68AA779E-DC33-4F14-9720-1187A671280F@kavun.ch>

Hey guys,

Any further hints on this would be very appreciated. 

Thanks and all the best.
On Nov 15, 2012, at 4:46 PM, Emrah <lists at kavun.ch> wrote:

> I read and reread Anthony's explanation and am still not able to have multiple profile allow registering on the same domain.
> If I have 2 identical profiles, it looks like the first one that is up will take the ownership of the domain name. Anything registering on the secondary one will not be visible to a sofia_contact. 
> 
> If sip.example.com:5070 is up before sip.example.com:5060, only phones registered to sip.example.com:5070 will be visible to FS.
> 
> How do I fix this? I am not enforcing the domain in my configs.
> 
> Thanks,
> Emrah
> 
> On Nov 13, 2012, at 8:23 PM, Emrah <lists at kavun.ch> wrote:
> 
>> This is precious. I had figured out how the domain portion affects FS, I just didn't know how to declare my domains to my SIP profiles. Which I believe I now know and will experiment a bit.
>> 
>> Thanks!
>> On Nov 12, 2012, at 8:16 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> 
>>> The best thing to do is take a look at these things from a step back.
>>> 
>>> The domains inside the xml registry are completely different from the domains on the internet and again completely different from domains in sip packets.  The profiles are again entirely different from any of the above.  Its up to you to align them if you so choose.
>>> 
>>> 
>>> The default configuration distributed with FreeSWITCH sets up the scenario most likely to load on any machine and work out of the box.  That is the primary goal of that configuration, so, It sets the domain in both the directory, the global default domain variable and the name of the internal profile to be identical to the ip on the box that can reach the internet.  Then it sets the sip to force everything to that value.  When you want to detach from this behavior, you are probably on a venture to do some kind of multi-home setup.
>>> 
>>> 
>>> Aliases in the <aliases> tag are a list of keys you want to use to use that lead to the current profile your are configuring.  Think of it as the /etc/hosts file in unix only for profiles.   When you define aliases to match all of the possible domains hosted on a particular profile, then when you try to take a user at host.com notation and decide which profile it came from, you can use the aliases to find it providing you have added <alias name="host.com"/> to that profile.
>>> 
>>> The <domains> tag is an indicator telling the profile to open the xml registry in FreeSWITCH and run through any domains defined therein.
>>> The 2 key attributes are:
>>> 
>>> alias: [true/false]  (automatically create an alias for this domain as mentioned above)
>>> parse: [true/false] (scan the domain for gateway entries and include them into this profile)
>>> name: [<string>] (either the name of a specific domain or 'all' to denote parsing every domain in the directory)
>>> 
>>> As you showed in your question the default config has
>>> 
>>> <domain name="all" alias="true" parse="false"/>
>>> 
>>> If you apply what you have learned above, it will scan for every domain (there is only one by default) and add an alias for it and not parse it for gateways.  The default directory uses global config vars to set the domain to match the local ip on the box.  So now you will have a domain in your config that is your ip, and the internal profile will attach to it and add an alias so that value expands to match it.
>>> 
>>> 
>>> This is explained in a comment at the top of directory/default.xml
>>> 
>>>   FreeSWITCH works off the concept of users and domains just like email.                                                                                      
>>>   You have users that are in domains for example 1000 at domain.com.                                                                                             
>>> 
>>>   When freeswitch gets a register packet it looks for the user in the directory                                                                               
>>>   based on the from or to domain in the packet depending on how your sofia profile                                                                            
>>>   is configured.  Out of the box the default domain will be the IP address of the                                                                             
>>>   machine running FreeSWITCH.  This IP can be found by typing "sofia status" at the                                                                           
>>>   CLI.  You will register your phones to the IP and not the hostname by default.                                                                              
>>>   If you wish to register using the domain please open vars.xml in the root conf                                                                              
>>>   directory and set the default domain to the hostname you desire.  Then you would                                                                            
>>>   use the domain name in the client instead of the IP address to register                                                                                     
>>>   with FreeSWITCH.                                                                    
>>> 
>>> 
>>> 
>>> So having more than one profile with the default of 
>>> 
>>> <domain name="all" alias="true" parse="false"/>
>>> 
>>> is going to end up aliasing the same domains into all profiles who call it and cause an overwrite in the lookup table and probably an error in your logs somewhere.  If you had parse="true" on all of them, they would all try and register to the gateways in all of your domains.
>>> 
>>> 
>>> If you look at the stock config, external.xml is a good example of a secondary profile, it has
>>> 
>>> <domain name="all" alias="false" parse="true"/>
>>> 
>>> so no aliases, and yes parse ... the exact opposite of the internal so that all the gateways would register from external and internal would bind to the local ip.
>>> 
>>> So, you probably want to use separate <domain name="my.com"/> per domain per profile you want to bind it to in more complicated setups.
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> On Sun, Nov 11, 2012 at 9:09 PM, Emrah <lists at kavun.ch> wrote:
>>> Bless you!
>>> 
>>> Thanks for putting this together. You've beautifully summed up all my questions.
>>> On Nov 11, 2012, at 8:09 AM, Lawrence Conroy <lconroy at insensate.co.uk> wrote:
>>> 
>>>> Hi Folks,
>>>> I've started a new thread as it's not quite the same issue, and domains & profiles have confused the heck out of me every time I have developed a new setup for fS.
>>>> I have sometimes had to hack/hard-doce the dialstring to make multiple domains in one profile work, had hours of fun with presence, db and force register settings, and have still had some odd gotchas that have required extensive meditation.
>>>> [... and yes, I have read the 1.0.6 bridge book; I'm trying to abstract these elements ]
>>>> 
>>>> Coming at this from standards/specs and rolling my own SIP stacks, sofia/fS seems to use the term "domain" differently from sipdomain, and alias seems to be tied to the directory (and thus to the profile listed in a directory file), but I'm not sure.
>>>> so ...
>>>> Before I capture to the sofia conf xml wiki page, I have a couple of questions on the sip-profile XML setup;
>>>> 
>>>> Q: Is there a particular reason why there's a parameter called alias and an (entirely different) setting also called alias?
>>>> The sofia conf xml wiki's comment on the setting "alias" shows I'm not alone.
>>>> I agree that's what it appears to be doing, but can we nail this down please (and what happens if an external client uses this connection to register and call)?
>>>> 
>>>> In the current sofia conf xml wiki page, the domain setting is not exactly well documented :).
>>>> The current internal.xml vanilla example from git (as of time of writing) has the following lines:
>>>> -------------------------
>>>> ...
>>>> <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
>>>> <!--aliases are other names that will work as a valid profile name for this profile-->
>>>> <aliases>
>>>> <!--
>>>>      <alias name="default"/>
>>>>  -->
>>>> </aliases>
>>>> <!-- Outbound Registrations -->
>>>> <gateways>
>>>> <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
>>>> </gateways>
>>>> 
>>>> <domains>
>>>> <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
>>>> <!--<domain name="$${domain}" parse="true"/>-->
>>>> <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
>>>> <!--<domain name="all" alias="true" parse="true"/>-->
>>>> <domain name="all" alias="true" parse="false"/>
>>>> </domains>
>>>> ...
>>>> -------------------------
>>>> 
>>>> This stuff is entirely missing from the sofia.conf.xml wiki page, and it IS really important.
>>>> 
>>>> 
>>>> Q: what's the default value for the alias parameter in the domain element? -- it is missing from the first example.
>>>> Q: if there is more than one profile, what's the impact of setting parse = true in one (or all) of the profiles' XML files?
>>>> (or parse = false, or missing the parameter altogether)?
>>>> AFAICT, the gateways get pulled in via the pre-process directive just fine, regardless of the value of the parse parameter -- it works for me, at least.
>>>> 
>>>> Q: if there is more than one profile, what's the impact of putting domain name="all" into one (or all) of the profiles' XML files?
>>>> 
>>>> Ideally, having more than one sipdomain tied to one profile "would be good", but aliases doesn't do that -- as the git file says, these are aliases for the profile name.
>>>> 
>>>> Before I start scribbling, Answers on a postcard to this ML, please.
>>>> 
>>>> all the best,
>>>> Lawrence
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> -- 
>>> Anthony Minessale II
>>> 
>>> 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
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.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  Thu Dec  6 17:18:58 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 6 Dec 2012 14:18:58 +0000
Subject: [Freeswitch-users] Using bypass_media or proxy_media selectively
In-Reply-To: <343F3B15-35E9-4CCF-A2BB-2514C1167E4C@kavun.ch>
References: <343F3B15-35E9-4CCF-A2BB-2514C1167E4C@kavun.ch>
Message-ID: <CAFiqYunBGpiB=SkAuDH2oCsgg-YxG-nda4SOqHAhrwUMRWGtyw@mail.gmail.com>

If you have mod_h26x loaded and have added it to your codec prefs, you
should not need either.


On 6 December 2012 13:57, Emrah <lists at kavun.ch> wrote:

> Hi there,
>
> Can you please confirm that I need to either use proxy_media or
> bypass_media for a video call in H.264?
> If so, how do I enable proxy_media or bypass_media conditionally by
> examining the codec prefs of both SIP endpoints?
>
> For a setup where I have 2 devices registered as 1000 at mydomain, one being
> audio only and the other video capable. Plus a video capable device
> registered as 1001 at mydomain.
> How do I selectively use bypass_media or proxy_media for calls made by
> 1001 to 1000 and answered by the video capable terminal?
>
> Thanks a million,
> Emrah
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 govoiper at gmail.com  Thu Dec  6 17:19:58 2012
From: govoiper at gmail.com (SamyGo)
Date: Thu, 6 Dec 2012 19:19:58 +0500
Subject: [Freeswitch-users] Distributed Conference Room
In-Reply-To: <88380BD1-EBB7-401B-8D74-A7CF7D8266DF@kavun.ch>
References: <CAK9Kg4HzR9zwiBQXEEkVFwz5rGEWh84U0RkoW8f00aMKJ--wMQ@mail.gmail.com>
	<88380BD1-EBB7-401B-8D74-A7CF7D8266DF@kavun.ch>
Message-ID: <CAJUJwthjxWHdhcLmJ-smiF0kpB+v+d=C8f_0X2C=bWuekt3GSw@mail.gmail.com>

Hi,

If I were you, I'd prefer to have a memcached register of active conference
room on their server(s), if the primary server fails to accept my call or
it has reached a max.confernece.participant limit (N lets say) then I would
append custom header in the failure route which tells which server and
which conference room this call belongs to and route it to any End point
FreeSwitch.

Call gets accepted on newer server, add it into the previously stated
opensips registry. Then at the new FreeSwitch server add few scripts to
detect those custom headers..initiate a new conference and a new call to
dial to the server in header and add it into my current conference as
participant. This is really interesting to initiate a call between two
(freeswitches or more) and adding it as participant in local conference.

Thats what I always find more practical.

Any new call to same conference will again consult memcache from opensips
and hence gets routed to the last inserted server IP with participant
vacancy.

I hope I made some sense.

Thanks
Sammy
On Dec 6, 2012 7:05 PM, "Emrah" <lists at kavun.ch> wrote:

> This is most interesting and I am curious to know the answer as well.
>
> Cheers,
> E
> On Dec 5, 2012, at 5:16 PM, Daniel Eiland <daniel.eiland at gmail.com> wrote:
>
> > Hi guys,
> >
> > I'm trying to deploy a conferencing solution using FreeSWITCH and
> running into a small issue with fail over / hot-standbys.
> >
> > In my environment, I've got multiple FreeSWITCH/Conference endpoints
> registered with an OpenSIPS proxy.  When calls come into OpenSIPS they are
> routed to the FreeSWITCH endpoints based on their q-values.  If a
> FreeSWITCH instance fails (namely the one with the highest q-value), the
> call is simply routed to the next instance.  This works great in most
> situations, however in some cases (namely network congestion) the
> FreeSWITCH w/highest priority is simply temporarily unavailable and callers
> to the same conference endpoint land on different servers.
> >
> > I'm wondering if there is a mechanism for distributing (or sharing) a
> conference room across multiple FreeSWITCH instances.  Namely, if a user
> lands in a conference hosted on server A while another lands in the same
> conference on server B, is there a mechanism in FreeSWITCH to connect the
> two servers/conferences (Presumably some "static" connection between the
> servers/rooms) so they can still talk with each over?
> >
> > Thanks,
> > Daniel
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec  6 17:35:08 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 6 Dec 2012 14:35:08 +0000
Subject: [Freeswitch-users] Release tarball problems
In-Reply-To: <CCE50E0D.616EC%krice@freeswitch.org>
References: <CAFiqYuk3rAog5fweKp7bZNKh-Rf6hupMpu-JsRPAgf8J2PAx+Q@mail.gmail.com>
	<CCE50E0D.616EC%krice@freeswitch.org>
Message-ID: <CAFiqYu=Qvo1NGD4J7o+5uDNOD5ocwYQafo622Em=0RF-djjHWg@mail.gmail.com>

build/next-release.txt in v1.2.stable on latest git also seems to be
incorrect. It's currently 1.2.5, shouldn't that be 1.2.6?


On 5 December 2012 20:43, Ken Rice <krice at freeswitch.org> wrote:

>  Hey Steven it should be there let me find out why its not there
>
> K
>
>
> On 12/5/12 2:18 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>
> The freeswitch-1.2.5.2.tar.bz2 tarball is missing from
> files.freeswitch.org <http://files.freeswitch.org> , are the developers
> aware of this?
>
>
> The versions of debian packages built from the packages are also incorrect
> - debian/changelog's last entry is 1.2~rc2-1.
>
> The advice in debian/README.source suggests adding a new changelog entry
> to set custom versions for builds from Git. For git versions that seems
> fine (after all we wouldn't want to have to update the file for every
> commit), but it seems that for every stable version release it would be
> better to set the version within the tarball's debian/changelog?
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>
> --
> Ken
>
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *
> irc.freenode.net #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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cal.leeming at simplicitymedialtd.co.uk  Thu Dec  6 17:49:26 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Thu, 6 Dec 2012 14:49:26 +0000
Subject: [Freeswitch-users] Distributed Conference Room
In-Reply-To: <88380BD1-EBB7-401B-8D74-A7CF7D8266DF@kavun.ch>
References: <CAK9Kg4HzR9zwiBQXEEkVFwz5rGEWh84U0RkoW8f00aMKJ--wMQ@mail.gmail.com>
	<88380BD1-EBB7-401B-8D74-A7CF7D8266DF@kavun.ch>
Message-ID: <CAHKQagF3hz7i9RkViJdVEb-jBMt5yBnTFVVNxtg9OeLJu3cPKw@mail.gmail.com>

+1, would be very interesting to hear the different thoughts and approaches.

On Thu, Dec 6, 2012 at 2:00 PM, Emrah <lists at kavun.ch> wrote:

> This is most interesting and I am curious to know the answer as well.
>
> Cheers,
> E
> On Dec 5, 2012, at 5:16 PM, Daniel Eiland <daniel.eiland at gmail.com> wrote:
>
> > Hi guys,
> >
> > I'm trying to deploy a conferencing solution using FreeSWITCH and
> running into a small issue with fail over / hot-standbys.
> >
> > In my environment, I've got multiple FreeSWITCH/Conference endpoints
> registered with an OpenSIPS proxy.  When calls come into OpenSIPS they are
> routed to the FreeSWITCH endpoints based on their q-values.  If a
> FreeSWITCH instance fails (namely the one with the highest q-value), the
> call is simply routed to the next instance.  This works great in most
> situations, however in some cases (namely network congestion) the
> FreeSWITCH w/highest priority is simply temporarily unavailable and callers
> to the same conference endpoint land on different servers.
> >
> > I'm wondering if there is a mechanism for distributing (or sharing) a
> conference room across multiple FreeSWITCH instances.  Namely, if a user
> lands in a conference hosted on server A while another lands in the same
> conference on server B, is there a mechanism in FreeSWITCH to connect the
> two servers/conferences (Presumably some "static" connection between the
> servers/rooms) so they can still talk with each over?
> >
> > Thanks,
> > Daniel
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec  6 18:04:48 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 6 Dec 2012 10:04:48 -0500
Subject: [Freeswitch-users] Missed call notification
In-Reply-To: <A4AE4A33-2735-4E89-A7F8-2CEB22CA6C09@kavun.ch>
References: <BLU169-W120766FAB52400013BA1BB28C420@phx.gbl>
	<8AE80082-5AAC-4C25-87E9-52FBD5EDB593@endigotech.com>
	<CAAtWKeFdgZ_MyPBiNUsDYZZsBRJMCe=d6vTSynxDtZ+ifi+KTQ@mail.gmail.com>
	<BLU169-W118507FC5529D5F80F26BA18C420@phx.gbl>
	<CA+zebiYsNB0vhnDuPZo=d_uB_9tqE5W3fAzEnrdnsvVtVB4OPg@mail.gmail.com>
	<CAAtWKeE9nFAFg1YWpb+B1BwLrTOKpOdMhJ9KEYE0E6KuGG3gNA@mail.gmail.com>
	<BLU169-W2283D9F938CCE5C8E60C2D8C400@phx.gbl>
	<A4AE4A33-2735-4E89-A7F8-2CEB22CA6C09@kavun.ch>
Message-ID: <5AC80A54-A29F-43EB-B370-7DED9B073403@endigotech.com>

It can fail with ORIGINATOR_CANCEL

Sent from my iPhone

On Dec 6, 2012, at 9:09 AM, Emrah <lists at kavun.ch> wrote:

> I might be missing something, but this only works if the bridge fails and not if the caller hangs up before reaching voicemail.
> 
> My work around is to schedule the lua as a hangup hook and check a few things.
> From the top of my head, it was the channel status or bridge status; and the voicemail recording duration should either be empty or 0. This way you get notified if someone lands in the VM and doesn't leave a message.
> 
> Emrah
> 
> On Dec 3, 2012, at 11:44 AM, ?? <itispip-qq at hotmail.com> wrote:
> 
>> Thanks Yehavi, it's really handy to implement this script to my environment :)
>> 
>> /Alex
>> 
>> Date: Sun, 2 Dec 2012 10:21:17 +0200
>> From: yehavi.bourvine at gmail.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Missed call notification
>> 
>> Hi,
>> 
>>  I've created the following wiki page with example of how we do it:
>> http://wiki.freeswitch.org/wiki/MailNoAsnwer
>> 
>> I would be greatfull if someone can add it to the examples page, as I don't have the edit authority on it...
>> 
>>                    Thanks, __Yehavi: 
>> 
>> 
>> 2012/12/1 Komar, Jason <jkomar at jbox.ca>
>> I would be interested to see that script too.
>> 
>> Thanks,
>> Jason
>> 
>> 
>> On Sat, Dec 1, 2012 at 11:34 AM, ?? <itispip-qq at hotmail.com> wrote:
>> 
>> Hi Yehavi, seems your solution is the best fit for the situation. Is that possible to share your script? Maybe can posted it to Freeswitch Wiki as part of the documentation.
>> 
>> Date: Sat, 1 Dec 2012 19:13:11 +0200
>> From: yehavi.bourvine at gmail.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Missed call notification
>> 
>> We have a script to find the hangup cause and send Email if the user requested it. This way I know who is looking for me while I am on the phone :-)
>> 
>>                      __Yehavi: 
>> 
>> 
>> 2012/12/1 Brian Foster <bdfoster at endigotech.com>
>> Message Waiting Indicator (MWI) is the voicemail notification feature for IP phones. 
>> 
>> Missed call notifications are provided by the phone and not the switch.
>> 
>> Sent from my iPhone
>> 
>> On Dec 1, 2012, at 2:41 AM, ?? <itispip-qq at hotmail.com> wrote:
>> 
>> I heard that Freeswitch support missed call notification (i/o VM notification). But look around Wiki no document about this feature. Anyone knows where to setup it up?
>> 
>> /brgds, 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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.orghttp://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.orghttp://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Dec  6 18:07:54 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 6 Dec 2012 15:07:54 +0000
Subject: [Freeswitch-users] Distributed Conference Room
In-Reply-To: <CAHKQagF3hz7i9RkViJdVEb-jBMt5yBnTFVVNxtg9OeLJu3cPKw@mail.gmail.com>
References: <CAK9Kg4HzR9zwiBQXEEkVFwz5rGEWh84U0RkoW8f00aMKJ--wMQ@mail.gmail.com>
	<88380BD1-EBB7-401B-8D74-A7CF7D8266DF@kavun.ch>
	<CAHKQagF3hz7i9RkViJdVEb-jBMt5yBnTFVVNxtg9OeLJu3cPKw@mail.gmail.com>
Message-ID: <CAMbW9oKAkRz4-mcsE+xpQd6MdPn_43f_Wt6DjZ_5Riac1=XjSQ@mail.gmail.com>

I've also thought about these but not implemented any. A couple of my
suggestions, both require some location database to say where the
conference is:


   1. 302 to the correct server, OpenSIPS can handle this for you.
   2. Use the second style call to conference app, which then outbound
   dials into the original one if needs be.


On 6 December 2012 14:49, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> +1, would be very interesting to hear the different thoughts and
> approaches.
>
>
> On Thu, Dec 6, 2012 at 2:00 PM, Emrah <lists at kavun.ch> wrote:
>
>> This is most interesting and I am curious to know the answer as well.
>>
>> Cheers,
>> E
>> On Dec 5, 2012, at 5:16 PM, Daniel Eiland <daniel.eiland at gmail.com>
>> wrote:
>>
>> > Hi guys,
>> >
>> > I'm trying to deploy a conferencing solution using FreeSWITCH and
>> running into a small issue with fail over / hot-standbys.
>> >
>> > In my environment, I've got multiple FreeSWITCH/Conference endpoints
>> registered with an OpenSIPS proxy.  When calls come into OpenSIPS they are
>> routed to the FreeSWITCH endpoints based on their q-values.  If a
>> FreeSWITCH instance fails (namely the one with the highest q-value), the
>> call is simply routed to the next instance.  This works great in most
>> situations, however in some cases (namely network congestion) the
>> FreeSWITCH w/highest priority is simply temporarily unavailable and callers
>> to the same conference endpoint land on different servers.
>> >
>> > I'm wondering if there is a mechanism for distributing (or sharing) a
>> conference room across multiple FreeSWITCH instances.  Namely, if a user
>> lands in a conference hosted on server A while another lands in the same
>> conference on server B, is there a mechanism in FreeSWITCH to connect the
>> two servers/conferences (Presumably some "static" connection between the
>> servers/rooms) so they can still talk with each over?
>> >
>> > Thanks,
>> > Daniel
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
*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 Dec  6 18:09:27 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 6 Dec 2012 15:09:27 +0000
Subject: [Freeswitch-users] Distributed Conference Room
In-Reply-To: <CAMbW9oKAkRz4-mcsE+xpQd6MdPn_43f_Wt6DjZ_5Riac1=XjSQ@mail.gmail.com>
References: <CAK9Kg4HzR9zwiBQXEEkVFwz5rGEWh84U0RkoW8f00aMKJ--wMQ@mail.gmail.com>
	<88380BD1-EBB7-401B-8D74-A7CF7D8266DF@kavun.ch>
	<CAHKQagF3hz7i9RkViJdVEb-jBMt5yBnTFVVNxtg9OeLJu3cPKw@mail.gmail.com>
	<CAMbW9oKAkRz4-mcsE+xpQd6MdPn_43f_Wt6DjZ_5Riac1=XjSQ@mail.gmail.com>
Message-ID: <CAMbW9o+XpG_+kSb4Umj1FjnUNmRu0H0Q6D3MfVvrhYBtM_Qh5A@mail.gmail.com>

I forgot to add that the issue I anticipate with the second method is the
moderation controls not being propagated across the servers.

On 6 December 2012 15:07, Andrew Cassidy <andrew at cassidywebservices.co.uk>wrote:

> I've also thought about these but not implemented any. A couple of my
> suggestions, both require some location database to say where the
> conference is:
>
>
>    1. 302 to the correct server, OpenSIPS can handle this for you.
>    2. Use the second style call to conference app, which then outbound
>    dials into the original one if needs be.
>
>
> On 6 December 2012 14:49, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> +1, would be very interesting to hear the different thoughts and
>> approaches.
>>
>>
>> On Thu, Dec 6, 2012 at 2:00 PM, Emrah <lists at kavun.ch> wrote:
>>
>>> This is most interesting and I am curious to know the answer as well.
>>>
>>> Cheers,
>>> E
>>> On Dec 5, 2012, at 5:16 PM, Daniel Eiland <daniel.eiland at gmail.com>
>>> wrote:
>>>
>>> > Hi guys,
>>> >
>>> > I'm trying to deploy a conferencing solution using FreeSWITCH and
>>> running into a small issue with fail over / hot-standbys.
>>> >
>>> > In my environment, I've got multiple FreeSWITCH/Conference endpoints
>>> registered with an OpenSIPS proxy.  When calls come into OpenSIPS they are
>>> routed to the FreeSWITCH endpoints based on their q-values.  If a
>>> FreeSWITCH instance fails (namely the one with the highest q-value), the
>>> call is simply routed to the next instance.  This works great in most
>>> situations, however in some cases (namely network congestion) the
>>> FreeSWITCH w/highest priority is simply temporarily unavailable and callers
>>> to the same conference endpoint land on different servers.
>>> >
>>> > I'm wondering if there is a mechanism for distributing (or sharing) a
>>> conference room across multiple FreeSWITCH instances.  Namely, if a user
>>> lands in a conference hosted on server A while another lands in the same
>>> conference on server B, is there a mechanism in FreeSWITCH to connect the
>>> two servers/conferences (Presumably some "static" connection between the
>>> servers/rooms) so they can still talk with each over?
>>> >
>>> > Thanks,
>>> > Daniel
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> --
> *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 Tim.Meade at Millicorp.com  Thu Dec  6 18:10:43 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Thu, 6 Dec 2012 15:10:43 +0000
Subject: [Freeswitch-users] installing on Amazon EC2
In-Reply-To: <CA+NjZ2u0nzwpEOdqHf2555eNKNt+gDncL85Wy9pnAz8RMvL0Sg@mail.gmail.com>
References: <CA+x1V=PYYNVRZZ642e8AHSw0RM5opJkegbSmAR4oT=HxjiJ8tw@mail.gmail.com>
	<804D48104511D4468F0D60DF9D3100350AE3B177@MAIL.millicorp.com>
	<CA+x1V=O3Hzs-XYSuFBUN6oZcxV79bgLGV9xjpsim9cndqQ+6Bg@mail.gmail.com>
	<804D48104511D4468F0D60DF9D3100350AE3F514@MAIL.millicorp.com>
	<CA+NjZ2u0nzwpEOdqHf2555eNKNt+gDncL85Wy9pnAz8RMvL0Sg@mail.gmail.com>
Message-ID: <804D48104511D4468F0D60DF9D3100350AE407AD@MAIL.millicorp.com>

Sure I'll try to get to it over the weekend.

We did some testing using a large centos 5.8 instance in VA and QOS was excellent.   I wanted to do some further testing using the medium instances to see if it held up.

The issues we did have was connectivity with ODBC back to our database in our primary ATL datacenter.   But I do not think this was an AWS issues as I'm having the same issues from our LAX datacenter over a 10GB line to the ATL datacenter.

More on the WIKI but I will not be able to get to it until the weekend.

Tim


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nick Vines
Sent: Thursday, December 06, 2012 8:27 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] installing on Amazon EC2

Tim, since you are about to set it up, would you mind updating the wiki page with how you get it to work, and perhaps the quality? I know there are probably a few interested people out there, myself included.

http://wiki.freeswitch.org/wiki/Amazon_ec2



Thanks,
Nick



On Thu, Dec 6, 2012 at 8:15 AM, Tim Meade <Tim.Meade at millicorp.com<mailto:Tim.Meade at millicorp.com>> wrote:
Thanks Rob!

I'll spin up one of the Base Amazon Linux ones for testing then.

-----Original Message-----
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 Rob Genovesi
Sent: Thursday, December 06, 2012 5:58 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] installing on Amazon EC2

Tim,

I discovered that running the following command prior to running "./bootstrap.sh" appears to have solved the problem:

export LDFLAGS="-ltinfo"

I'm not sure why configure doesn't detect it automatically but adding the ld flag manually seems to have done the trick.


-Rob


On Wed, Dec 5, 2012 at 3:30 PM, Tim Meade <Tim.Meade at millicorp.com<mailto:Tim.Meade at millicorp.com>> wrote:
> I had this issue using the base amazon version of Linux.  I don't think we ever got past it.   I ended up using a centos instance for the base.
>
> If you do figure it out please let us know!
>
> Tim
>
> -----Original Message-----
> 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
> Rob Genovesi
> Sent: Wednesday, December 05, 2012 6:15 PM
> To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] installing on Amazon EC2
>
> Hi all,
>
> I'm trying to build/install Free Switch on an Amazon EC2 server (Amazon Linux AMI 64-bit) and "./configure" keeps exiting with the following error :
>
> ./configure
> <snip>
> checking for the version of libcurl... 7.24.0 checking for libcurl >=
> version 7.13.0... yes checking whether libcurl is usable... no no
> checking for tgetent in -lncurses... no checking for tgetent in
> -lcurses... no
> configure: error: libtermcap, libcurses or libncurses are required!
>
> (notice the extra "no" on a line by itself)
>
> ....
>
> I have installed all the ncurses packages I can find, to no avail :
>
> rpm -qa | grep -E 'curses|ncurses|term|curl'
> ncurses-term-5.7-3.20090208.9.amzn1.x86_64
> ncurses-libs-5.7-3.20090208.9.amzn1.i686
> python-pycurl-7.19.0-8.7.amzn1.x86_64
> libcurl-7.24.0-5.25.amzn1.x86_64
> ncurses-base-5.7-3.20090208.9.amzn1.x86_64
> ncurses-libs-5.7-3.20090208.9.amzn1.x86_64
> ncurses-5.7-3.20090208.9.amzn1.x86_64
> curl-7.24.0-5.25.amzn1.x86_64
> ncurses-devel-5.7-3.20090208.9.amzn1.x86_64
> libcurl-devel-7.24.0-5.25.amzn1.x86_64
>
> ....
>
> I have been able to install succcessfully on a local CentOS machine and Rackspace CentOS 6.3 VM, I believe the issue is something specific to Amazon EC2 but I haven't been able to figure it out.
>
> Any suggestions for a fix/workaround would be appreciated.
>
>
>
> Thanks,
>
> Rob
>
> ______________________________________________________________________
> ___ 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-use


> rs
> 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


>
> 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-use


> rs
> 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



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



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 krice at freeswitch.org  Thu Dec  6 18:38:23 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 06 Dec 2012 09:38:23 -0600
Subject: [Freeswitch-users] Release tarball problems
In-Reply-To: <CAFiqYu=Qvo1NGD4J7o+5uDNOD5ocwYQafo622Em=0RF-djjHWg@mail.gmail.com>
Message-ID: <CCE61810.61760%krice@freeswitch.org>

You can safely ignore that file... Its for scripted automation that?s not in
use


On 12/6/12 8:35 AM, "Steven Ayre" <steveayre at gmail.com> wrote:

> build/next-release.txt in v1.2.stable on latest git also seems to be
> incorrect. It's currently 1.2.5, shouldn't that be 1.2.6?
> 
> 
> On 5 December 2012 20:43, Ken Rice <krice at freeswitch.org> wrote:
>> Hey Steven it should be there let me find out why its not there
>> 
>> K
>> 
>> 
>> On 12/5/12 2:18 PM, "Steven Ayre" <steveayre at gmail.com
>> <http://steveayre at gmail.com> > wrote:
>> 
>>> The freeswitch-1.2.5.2.tar.bz2 tarball is missing from files.freeswitch.org
>>> <http://files.freeswitch.org>  <http://files.freeswitch.org> , are the
>>> developers aware of this?
>>> 
>>> 
>>> The versions of debian packages built from the packages are also incorrect -
>>> debian/changelog's last entry is?1.2~rc2-1.
>>> 
>>> The advice in debian/README.source suggests adding a new changelog entry to
>>> set custom versions for builds from Git. For git versions that seems fine
>>> (after all we wouldn't want to have to update the file for every commit),
>>> but it seems that for every stable?version?release it would be better to set
>>> the version within the tarball's debian/changelog?
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <http://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://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From steveayre at gmail.com  Thu Dec  6 18:54:39 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 6 Dec 2012 15:54:39 +0000
Subject: [Freeswitch-users] Release tarball problems
In-Reply-To: <CCE61810.61760%krice@freeswitch.org>
References: <CAFiqYu=Qvo1NGD4J7o+5uDNOD5ocwYQafo622Em=0RF-djjHWg@mail.gmail.com>
	<CCE61810.61760%krice@freeswitch.org>
Message-ID: <CAFiqYumOUCK394C0gHRdNJ7DOcbpd_zc_B0DCmbWR89P+XeEbA@mail.gmail.com>

It affects the autogeneration of debian version numbers for git builds

debian/README.source:

To build this package, I recommend running the following from the root
> directory of your FS git working tree:



  distro=sid
>   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 "unstable" "Custom build."
>   dpkg-buildpackage -b -us -uc -Zxz -z9
>   git reset --hard HEAD^





On 6 December 2012 15:38, Ken Rice <krice at freeswitch.org> wrote:

>  You can safely ignore that file... Its for scripted automation that?s
> not in use
>
>
>
> On 12/6/12 8:35 AM, "Steven Ayre" <steveayre at gmail.com> wrote:
>
> build/next-release.txt in v1.2.stable on latest git also seems to be
> incorrect. It's currently 1.2.5, shouldn't that be 1.2.6?
>
>
> On 5 December 2012 20:43, Ken Rice <krice at freeswitch.org> wrote:
>
> Hey Steven it should be there let me find out why its not there
>
> K
>
>
> On 12/5/12 2:18 PM, "Steven Ayre" <steveayre at gmail.com <
> http://steveayre at gmail.com> > wrote:
>
> The freeswitch-1.2.5.2.tar.bz2 tarball is missing from
> files.freeswitch.org <http://files.freeswitch.org>  <
> http://files.freeswitch.org> , are the developers aware of this?
>
>
>
> The versions of debian packages built from the packages are also incorrect
> - debian/changelog's last entry is 1.2~rc2-1.
>
> The advice in debian/README.source suggests adding a new changelog entry
> to set custom versions for builds from Git. For git versions that seems
> fine (after all we wouldn't want to have to update the file for every
> commit), but it seems that for every stable version release it would be
> better to set the version within the tarball's debian/changelog?
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://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://FreeSWITCH-users at lists.freeswitch.org>
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Alexander.Haugg at c4b.de  Thu Dec  6 16:34:04 2012
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Thu, 6 Dec 2012 13:34:04 +0000
Subject: [Freeswitch-users] mod_dingaling client registration sasl do not
	work plain
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC834751A@vc4bex2.corp.c4b.de>

Hi togehter,

i try to make a client regisistration on the OpenFire via mod_dingaling.
Is can see on the CLI that the mod_dingaling try the authentication with sasl md5, but configured i had sasle plain.

Here is my configuration in the client.xml:

<include>
  <!-- Client Profile (Original mode) -->
  <!-- to use this profile take the x- away from the open and close tags so its <profile> and </profile> -->
  <profile type="client">
    <param name="name" value="fsuser"/>
    <param name="login" value="fsuser"/>
    <param name="password" value="fsuser"/>
    <param name="server" value="OpenFireIP:5222"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="message" value="Jingle all the way"/>
    <param name="rtp-ip" value="auto"/>
    <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
    <param name="auto-login" value="false"/>
    <!-- SASL "plain" or "md5" -->
    <param name="sasl" value="plain"/>
    <!-- Enable TLS or not -->
    <param name="tls" value="false"/>
    <!-- disable to trade async for more calls -->
    <param name="use-rtp-timer" value="true"/>
    <!-- default extension (if one cannot be determined) -->
    <param name="exten" value="888"/>
<param name="local-network-acl" value="localnet.auto"/>
  </profile>
</include>

And here is the CLI output:

freeswitch at CC4BHAU1> dl_login profile=fsuser
2012-12-06 13:47:02.731665 [DEBUG] mod_dingaling.c:1998 Started Thread for fsuser at XML

OK

2012-12-06 13:47:02.731665 [DEBUG] libdingaling.c:1540 xmpp connecting


freeswitch at CC4BHAU1> 2012-12-06 13:47:03.331665 [NOTICE] libdingaling.c:1373 SEND:
-------------------------------------------------------------------------------
<stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:client" to="fsuser" version="1.0"></stream:stream>

2012-12-06 13:47:03.331665 [INFO] libdingaling.c:1371 RECV:
-------------------------------------------------------------------------------
<stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:client" from="XXX.XX.XXX.XX" id="c4f6a48e" xml:lang="en" version="1.0"></stream:stream>

2012-12-06 13:47:03.531665 [INFO] libdingaling.c:1371 RECV:
-------------------------------------------------------------------------------
<stream:features>
  <mechanisms xmlns="urn:ietf:params:xml:ns:xmpp-sasl">
    <mechanism>DIGEST-MD5</mechanism>
    <mechanism>PLAIN</mechanism>
    <mechanism>ANONYMOUS</mechanism>
    <mechanism>CRAM-MD5</mechanism>
  </mechanisms>
  <compression xmlns="http://jabber.org/features/compress">
    <method>zlib</method>
  </compression>
  <auth xmlns="http://jabber.org/features/iq-auth"></auth>
  <register xmlns="http://jabber.org/features/iq-register"></register>
</stream:features>

2012-12-06 13:47:03.531665 [NOTICE] libdingaling.c:1373 SEND:
-------------------------------------------------------------------------------
<auth xmlns="urn:ietf:params:xml:ns:xmpp-sasl" mechanism="DIGEST-MD5"></auth>

2012-12-06 13:47:03.551665 [INFO] libdingaling.c:1371 RECV:
-------------------------------------------------------------------------------
<challenge xmlns="urn:ietf:params:xml:ns:xmpp-sasl">cmVhbG09IjE3Mi4xNi4xMDMuMzMiLG5vbmNlPSI1RHFwTG9BODMzclFUaFlmSDE1cHNTUmt6cmdGa2NrUlQrMEFZcEVvIixxb3A9ImF1dGgiLGNoYXJzZXQ9dXRmLTgsYWxnb3JpdGhtPW1kNS1zZXNz</challenge>

2012-12-06 13:47:07.311665 [NOTICE] libdingaling.c:1373 SEND:
-------------------------------------------------------------------------------
<response xmlns="urn:ietf:params:xml:ns:xmpp-sasl">dXNlcm5hbWU9IihudWxsKSIscmVhbG09IjE3Mi4xNi4xMDMuMzMiLG5vbmNlPSI1RHFwTG9BODMzclFUaFlmSDE1cHNTUmt6cmdGa2NrUlQrMEFZcEVvIixjbm9uY2U9IjAwMDAwMDI5MDAwMDQ4MjMwMDAwMThiZTAwMDA2N
zg0IixuYz0wMDAwMDAwMSxxb3A9YXV0aCxkaWdlc3QtdXJpPSJ4bXBwL2ZzdXNlciIscmVzcG9uc2U9YTY5MDhhMzE0ODU3NmNkNGQ1NjhjNWYwNTE4NWFhMzYsY2hhcnNldD11dGYtOA==</response>

2012-12-06 13:47:07.311665 [INFO] libdingaling.c:1371 RECV:
-------------------------------------------------------------------------------
<failure xmlns="urn:ietf:params:xml:ns:xmpp-sasl">
  <not-authorized></not-authorized>
</failure>

2012-12-06 13:47:07.311665 [DEBUG] libdingaling.c:1294 sasl authentication failed


What can i do that the mod dingaling make the auth only with sasl plain???

Thanks for your help
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From scobei001 at gmail.com  Thu Dec  6 18:31:06 2012
From: scobei001 at gmail.com (Scott Beil)
Date: Thu, 06 Dec 2012 09:31:06 -0600
Subject: [Freeswitch-users] Displacing ringback audio during bridge -
 only first displacement heard
In-Reply-To: <CAKzWOxVM3KUeq6CbsBUJguE1ibnyDebC5cB1UeveYAc9sR8Z7g@mail.gmail.com>
References: <50BEA565.1070508@gmail.com>
	<CAKzWOxVM3KUeq6CbsBUJguE1ibnyDebC5cB1UeveYAc9sR8Z7g@mail.gmail.com>
Message-ID: <50C0BA3A.5040100@gmail.com>

I have filed a ticket (FS-4914) with proposed patch.

Thanks,

Scott


On 12/5/2012 11:08 AM, Michael Collins wrote:
> Scott,
>
> If you manually try to do displacements instead of using the 
> sched_api, does it exhibit the same symptom? Either way it sounds like 
> a possible but. Please file a ticket at jira.freeswitch.org 
> <http://jira.freeswitch.org> so that the devs can have a look. Also, 
> be sure to test this on latest git HEAD to confirm that it hasn't 
> already been fixed.
>
> -MC
>
> On Tue, Dec 4, 2012 at 5:37 PM, Scott Beil <scobei001 at gmail.com 
> <mailto:scobei001 at gmail.com>> wrote:
>
>     I am attempting to interrupt the ringback audio at regular intervals
>     with announcements while a bridge is in progress. Everything is going
>     fine - the far end is ringing, the ringback audio is playing, the
>     first
>     displacement is heard, but subsequent displacements are not.
>
>     I am using an outbound ESL connection, FreeSWITCH version 1.2.3.
>
>     First, the ringback audio is set:
>
>              esl_execute(handle, "set",
>     "ringback=file_string://'../sounds/music/8000/suite-espanola-op-47-leyenda.wav'",NULL);
>
>     Next, the bridge is initiated:
>
>              esl_execute(handle, "bridge", "user/1001",NULL);
>
>     Now, a displacement is scheduled for 5  seconds in the future:
>
>              esl_send_recv(handle, "api sched_api +5 none uuid_displace
>     <uuid> start digits/1.wav");
>
>     After each MEDIA_BUG_STOP event is received, another displacement is
>     scheduled.
>
>     The log file shows successful attempts are being made to play the
>     displacement audio:
>
>     66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:39.161496
>     [DEBUG]
>     switch_core_media_bug.c:506 Attaching BUG to
>     sofia/internal/1000 at 192.168.1.136 <mailto:1000 at 192.168.1.136>
>     2012-12-04 18:25:39.161496 [DEBUG] mod_commands.c:3894 Command
>     uuid_displace(66017ce6-23c8-4df1-accc-6275eed8fc45 start
>     digits/1.wav):
>     +OK Success
>
>     2012-12-04 18:25:39.161496 [DEBUG] switch_scheduler.c:138 Deleting
>     task
>     17 sched_api_function (none)
>     66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:39.641496
>     [DEBUG]
>     switch_core_media_bug.c:724 Removing BUG from
>     sofia/internal/1000 at 192.168.1.136 <mailto:1000 at 192.168.1.136>
>     2012-12-04 18:25:39.761496 [DEBUG] switch_scheduler.c:214 Added
>     task 18
>     sched_api_function (none) to run at 1354667144
>     66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:44.161496
>     [DEBUG]
>     switch_core_media_bug.c:506 Attaching BUG to
>     sofia/internal/1000 at 192.168.1.136 <mailto:1000 at 192.168.1.136>
>     2012-12-04 18:25:44.161496 [DEBUG] mod_commands.c:3894 Command
>     uuid_displace(66017ce6-23c8-4df1-accc-6275eed8fc45 start
>     digits/1.wav):
>     +OK Success
>
>     2012-12-04 18:25:44.161496 [DEBUG] switch_scheduler.c:138 Deleting
>     task
>     18 sched_api_function (none)
>     66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:44.181496
>     [DEBUG]
>     switch_core_media_bug.c:724 Removing BUG from
>     sofia/internal/1000 at 192.168.1.136 <mailto:1000 at 192.168.1.136>
>     2012-12-04 18:25:44.341496 [DEBUG] switch_scheduler.c:214 Added
>     task 19
>     sched_api_function (none) to run at 1354667149
>     66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:49.161496
>     [DEBUG]
>     switch_core_media_bug.c:506 Attaching BUG to
>     sofia/internal/1000 at 192.168.1.136 <mailto:1000 at 192.168.1.136>
>     2012-12-04 18:25:49.161496 [DEBUG] mod_commands.c:3894 Command
>     uuid_displace(66017ce6-23c8-4df1-accc-6275eed8fc45 start
>     digits/1.wav):
>     +OK Success
>
>     2012-12-04 18:25:49.161496 [DEBUG] switch_scheduler.c:138 Deleting
>     task
>     19 sched_api_function (none)
>     66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:49.181496
>     [DEBUG]
>     switch_core_media_bug.c:724 Removing BUG from
>     sofia/internal/1000 at 192.168.1.136 <mailto:1000 at 192.168.1.136>
>     2012-12-04 18:25:49.341496 [DEBUG] switch_scheduler.c:214 Added
>     task 20
>     sched_api_function (none) to run at 1354667154
>     66017ce6-23c8-4df1-accc-6275eed8fc45 2012-12-04 18:25:54.161496
>     [DEBUG]
>     switch_core_media_bug.c:506 Attaching BUG to
>     sofia/internal/1000 at 192.168.1.136 <mailto:1000 at 192.168.1.136>
>     2012-12-04 18:25:54.161496 [DEBUG] mod_commands.c:3894 Command
>     uuid_displace(66017ce6-23c8-4df1-accc-6275eed8fc45 start
>     digits/1.wav):
>     +OK Success
>
>     Any guidance would be appreciated.
>
>     Thanks,
>
>     Scott
>
>
>     _________________________________________________________________________
>     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
>
>
>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 bpriddy at bryantschools.org  Thu Dec  6 18:47:13 2012
From: bpriddy at bryantschools.org (Blake Priddy)
Date: Thu, 6 Dec 2012 09:47:13 -0600
Subject: [Freeswitch-users] 10 second delay
In-Reply-To: <83CE1052-D007-40D9-A5DB-1540C1A7D982@url.net.au>
References: <CADK-BjuB7ugJMPLSYByJPa0jQmnRywvWLgPdwDUCkKF2e7eFbA@mail.gmail.com>
	<50BED517.8060900@communicatefreely.net>
	<50BEEBA3.9040006@tagnet.ru>
	<83CE1052-D007-40D9-A5DB-1540C1A7D982@url.net.au>
Message-ID: <CADK-BjueJrqyH-XFCROhw2Z8vdOHT2Tt4QjAM9wWnNC20W+kmw@mail.gmail.com>

Where is that value at?
On Dec 5, 2012 1:01 AM, "Ashley Breeden" <ash at url.net.au> wrote:

> Hi There,
>
> I noticed a similar problem using 1.2.5.2.  In my dial plans I had:
>
> <action application="set" data="ignore_early_media=true"/>
>
> By removing the ignore_early_media line completely from my dialplans the
> problem was fixed.  Do you have ignore_early_media set anywhere in the
> Dialplans it passes through?
>
>
>
> - Ash.
>
>
>
> On 05/12/2012, at 5:37 PM, Boris Kovalenko wrote:
>
> > Hello!
> >
> > What version of FS are You useing? I have the same troubles with 1.2.5
> > branch. 1.2.3 works fine.
> >> My best advice would be to to a packet capture, or turn on sip traces
> between the gateway,
> >>   FS, and the phones, then try a test.
> >>
> >> When you look through the call, take a look at the difference in the
> time stamps.
> >>
> >> You should see a progress or a ringing back from the phone, then a 200
> OK when they pick up.
> >>
> >> After the call is picked up, see if there is additional traffic
> required to setup media,
> >> or any other entries in the log that might suggest that some additional
> actions are
> >> happening after the answer.
> >>
> >> Also, if you can capture with wireshark, you can play back the media
> stream.  Check to see
> >> if the first hello is there coming out of the phone, and then going to
> the gateway.
> >> Wireshark will put it all in a time line perspective for you, so you
> can determine if this
> >> is a problem with the phone, your configuration, or the gateway.
> >>
> >> Are you using late negotiation, bypass media, or any other options like
> that?
> >>
> >> -Tim
> >>
> >> Blake Priddy wrote:
> >>> I have some secretaries here at our school district that when
> >>> they receive a call they have to give their spill 2-3 times before the
> >>> party on the other end will hear them.. I have the Epygi Gateway and
> >>> FreeSwitch box in the same network. Any thoughts would be greatly
> >>> appreciated! :)
> >>>
> >>>
> >>> --
> >>>
> >>> *Blakelund Priddy*
> >>> Network Systems Engineer
> >>> Bryant Public School District
> >>> Bryant, Arkansas 72022
> >>> http://www.bryantschools.org <http://www.bryantschools.org/>
> >>> p 501-653-5038
> >>> f 501-847-5656
> >>>
> >>>
> >>>
> ------------------------------------------------------------------------
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting 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
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > --
> > ? ?????????,
> >   ????? ?????????
> >   ??? "??????"
> >   ???. +7 (3435) 230001
> >   ???? +7 (3435) 230005
> >   http://www.tagnet.ru
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.venugopan at mundio.com  Thu Dec  6 20:25:04 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Thu, 6 Dec 2012 17:25:04 +0000
Subject: [Freeswitch-users] sip registration
In-Reply-To: <CAKzWOxVMWfR6vk8ODKkND1Tjxj-d5VFYn9U+dDoNyB_t2dE05w@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF2339A49@Mail-Kilo.squay.com>
	<CAFiqYu=cpMkTxdE99smQBM6NgitNQ5ZpePwfSYr_R23dpj7Hng@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339B39@Mail-Kilo.squay.com>
	<CAKzWOxUW21XZte0AoVDqTtC6vkJ4FmA3sbP5LgJVK5bj4+1Sbw@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339D42@Mail-Kilo.squay.com>
	<CAKzWOxVMWfR6vk8ODKkND1Tjxj-d5VFYn9U+dDoNyB_t2dE05w@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233A02B@Mail-Kilo.squay.com>

Hi,
Thanks. I thought it will look in mod_sofia code. In the below screen I register the ID '100'. Now instead of '100' in "Authentication Name" I need to give some e-mail ID or name(Archana) which should validate in DB.
I tried giving a name in "Authentication Name" but the phone was not registered. Am not sure this authentication name is being looked in which column in table too.
Please let me know if this will be picked from any sofia code or any C script? Once we register in the below screen which script validates the Settings in freeswitch?

Sorry if am repeating the same question, but I could not get the exact code and am clueless.
Global SIP Settings
Top of Form
Basic SIP Authentication Settings


Screen Name


Screen Name 2


Phone Number


Caller ID


Authentication Name


Password


Bottom of Form


Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 05 December 2012 20:34
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

When an event that requires a user lookup takes place then the system will look in the XML user directory unless it has been configured to look somewhere else. The other places to look are usually:
mod_xml_curl
One of the language like Lua, Perl, Python

If it's xml_curl then FS will do a POST to your web server in hopes of receiving back the necessary XML for the given user. It would be up to you to have your web server handle the request, poll the database, then format and return the XML data. See this wiki page<http://wiki.freeswitch.org/wiki/Xml_curl#How_do_I_configure_mod_xml_curl.3F> for more info on xml curl.

If it's a language then you'll have a "binding" in the conf file for the language that will handle the lookup. Again, your script will need to handle the communication with your database. See this wiki page<http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration> for more information.

Hope this helps.
-MC
On Wed, Dec 5, 2012 at 9:30 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
Thanks for the information. But sorry, how to access user_data API command.

Am not clear on the flow. Once we register domain and usernumber in sip what exactly happens? Which script picks up this domain and username and validates with our database?
Could you please provide me with an overview.

Many thanks

Regards,
Archana

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: 05 December 2012 17:13
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

If you're talking about the user configuration then yes, you could create an "email" parameter or variable and access it with the user_data API command.
-MC
On Wed, Dec 5, 2012 at 1:35 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
In that case can I have 1 more column say e-mail and can this e-mail be checked in DB instead of checking reg_user('100')? Is that feasible?
Also which code should be changed any idea please?

Regards,
Archana

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 Steven Ayre
Sent: 04 December 2012 19:51
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

You can have a user 'ana' in the domain 'gmail.com<http://gmail.com>'. Though using someone else's domain as local in your FS setup may not be a good idea.
You can't have a @ in the username itself (per the SIP standard, not limited to FreeSWITCH).

On 4 December 2012 18:00, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

Currently we register authentication name as say '100' in sip registration, this comes to freeswitch and it will check in our DB for 100 and if its present then registrations would be successful.

freeswitch at internal> show registrations
reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
100,fsfailover.uk01.com<http://fsfailover.uk01.com>,e4969067f9a8c098,sofia/internal/sip:100 at 192.168.2.234:5060;transport=udp,1354638871,192.168.2.234,5060,udp,squay-laptop-1.squay.com<http://squay-laptop-1.squay.com>

I want to change this 100 to some e-mail address, so instead of 100 it will be something like 'ana at gmail.com<mailto:ana at gmail.com>'. Can we do this? While coming to freeswitch whether there would be any issues?

Regards,
Archana


_________________________________________________________________________
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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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
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Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

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From frank at carmickle.com  Thu Dec  6 23:11:21 2012
From: frank at carmickle.com (Frank Carmickle)
Date: Thu, 6 Dec 2012 15:11:21 -0500
Subject: [Freeswitch-users] pgsql as dsn
Message-ID: <DF2B131E-2C82-4D7E-B4CE-085432BD3C0F@carmickle.com>

Hi all

I'm trying to use the new pg support.  Here's what I get.

2012-12-06 20:00:50.078511 [CONSOLE] switch_loadable_module.c:1834 mod_db unloaded.
2012-12-06 20:00:50.078511 [CRIT] switch_core_sqldb.c:464 Failure! PGSQL NOT AVAILABLE! Can't connect to DSN host=my_hostname.tld dbname=freeswitch user=freeswitch password='stringofupperlowerandnumbers' options='-c client_min_messages=NOTICE' application_name='freeswitch'
2012-12-06 20:00:50.078511 [INFO] switch_time.c:1165 Timezone reloaded 530 definitions
2012-12-06 20:00:50.078511 [CONSOLE] switch_loadable_module.c:1348 Successfully
Loaded [mod_db]
2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:254 Adding Application 'db'
2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:254 Adding Application 'group'
2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:298 Adding API Function 'db'
2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:298 Adding API Function 'group'
2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:473 Adding Limit interface 'db'

Doing a tcpdump on the interface where this traffic should be showing up shows no attempts.  This connection will need to be ipv6.  Is there something special I need to do to make this work?

Thanks for any help you can give.
--FC




From jpablolorenzetti at hotmail.com  Fri Dec  7 00:04:17 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Thu, 6 Dec 2012 21:04:17 +0000
Subject: [Freeswitch-users] get a variable at hunting phase that is set
 at executing phase
In-Reply-To: <CALNZuwmq8-gjaE4BW5rNV28899apGjbrdSEqHKdy_yyr4hT8eg@mail.gmail.com>
References: <DUB002-W564CE266C94AB4E76E0A98D9460@phx.gbl>,
	<CAKzWOxWN+jY5NP8rNTa0m5nbW2NCW1bmCW6K_O-Xgr4Nv31qDw@mail.gmail.com>,
	<DUB002-W1FD5F4B80221F2C412F50D9460@phx.gbl>,
	<CAKzWOxWTnyutxJivj5xZT8tKSCk0=qGM7JcHoS_LsYtCS2haiw@mail.gmail.com>,
	<DUB002-W1078E4F82563CB2F9E0A859D9450@phx.gbl>,
	<CALNZuwmq8-gjaE4BW5rNV28899apGjbrdSEqHKdy_yyr4hT8eg@mail.gmail.com>
Message-ID: <DUB002-W96109A39DA124FDE49CEBED9450@phx.gbl>

Avi, yes that makes sense, i will try that is a good idea .. thanks!!

From: avi at avimarcus.net
Date: Thu, 6 Dec 2012 10:00:37 +0200
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] get a variable at hunting phase that is set at executing phase

If you execute_extension to a new context, you can create very few extensions in that context so that it goes quicker.
-Avi





On Thu, Dec 6, 2012 at 3:46 AM, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:





Michael, thank you for showing me, it is clear in the logs now, i did not have that clear before but i see it now ... i implemented Avi's suggestion 1 and it works now but i noticed that execute_extension parses the dialplan again, is there anyway to avoid reading the dialplan again ? regards!



Date: Wed, 5 Dec 2012 15:41:45 -0800
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org


Subject: Re: [Freeswitch-users] get a variable at hunting phase that is set at executing phase



On Wed, Dec 5, 2012 at 1:19 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:






Thanks Avi for you answer, i m going to try that (i already have and the ${int} got lost somewhere but i ll try again)
Michael, thanks for your suggestion, i have put it in pastebin: http://pastebin.freeswitch.org/20293





Okay, this is why PB is good. It's just as you and Avi suspected - the variable is being tested prior to the Lua script running. If you want to see how to determine that yourself then look at your pastebin entry. Lines 48-72 are the dialplan hunting. The executing starts at line 80 with the Lua script. The Lua script is being executed after the variable is being check, which you'll see on line # 66.




Either of Avi's suggestions should help.

-MC


_________________________________________________________________________
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Official FreeSWITCH Sites

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From anthony.minessale at gmail.com  Fri Dec  7 00:09:13 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 6 Dec 2012 15:09:13 -0600
Subject: [Freeswitch-users] pgsql as dsn
In-Reply-To: <DF2B131E-2C82-4D7E-B4CE-085432BD3C0F@carmickle.com>
References: <DF2B131E-2C82-4D7E-B4CE-085432BD3C0F@carmickle.com>
Message-ID: <CAKbxfG_iwHwBpF7kcNL=W36KvawJ2zy-0VSmH3VnXobmn1Voxg@mail.gmail.com>

try this from the build root

./configure --enable-core-pgsql-support --no-create --no-recursion

then

make install

and try again




On Thu, Dec 6, 2012 at 2:11 PM, Frank Carmickle <frank at carmickle.com> wrote:

> Hi all
>
> I'm trying to use the new pg support.  Here's what I get.
>
> 2012-12-06 20:00:50.078511 [CONSOLE] switch_loadable_module.c:1834 mod_db
> unloaded.
> 2012-12-06 20:00:50.078511 [CRIT] switch_core_sqldb.c:464 Failure! PGSQL
> NOT AVAILABLE! Can't connect to DSN host=my_hostname.tld dbname=freeswitch
> user=freeswitch password='stringofupperlowerandnumbers' options='-c
> client_min_messages=NOTICE' application_name='freeswitch'
> 2012-12-06 20:00:50.078511 [INFO] switch_time.c:1165 Timezone reloaded 530
> definitions
> 2012-12-06 20:00:50.078511 [CONSOLE] switch_loadable_module.c:1348
> Successfully
> Loaded [mod_db]
> 2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:254 Adding
> Application 'db'
> 2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:254 Adding
> Application 'group'
> 2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:298 Adding
> API Function 'db'
> 2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:298 Adding
> API Function 'group'
> 2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:473 Adding
> Limit interface 'db'
>
> Doing a tcpdump on the interface where this traffic should be showing up
> shows no attempts.  This connection will need to be ipv6.  Is there
> something special I need to do to make this work?
>
> Thanks for any help you can give.
> --FC
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
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From lists at kavun.ch  Fri Dec  7 00:14:47 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Dec 2012 16:14:47 -0500
Subject: [Freeswitch-users] Missed call notification
In-Reply-To: <5AC80A54-A29F-43EB-B370-7DED9B073403@endigotech.com>
References: <BLU169-W120766FAB52400013BA1BB28C420@phx.gbl>
	<8AE80082-5AAC-4C25-87E9-52FBD5EDB593@endigotech.com>
	<CAAtWKeFdgZ_MyPBiNUsDYZZsBRJMCe=d6vTSynxDtZ+ifi+KTQ@mail.gmail.com>
	<BLU169-W118507FC5529D5F80F26BA18C420@phx.gbl>
	<CA+zebiYsNB0vhnDuPZo=d_uB_9tqE5W3fAzEnrdnsvVtVB4OPg@mail.gmail.com>
	<CAAtWKeE9nFAFg1YWpb+B1BwLrTOKpOdMhJ9KEYE0E6KuGG3gNA@mail.gmail.com>
	<BLU169-W2283D9F938CCE5C8E60C2D8C400@phx.gbl>
	<A4AE4A33-2735-4E89-A7F8-2CEB22CA6C09@kavun.ch>
	<5AC80A54-A29F-43EB-B370-7DED9B073403@endigotech.com>
Message-ID: <FB15E716-E98E-4BBA-978A-AECE0167AD7F@kavun.ch>

It doesn't fail with a LUA script invoked in a hangup hook. It would fail with the script Yehava put up as example.
But I like the idea and it's a good start. I'll try to polish my script and post it on the Wiki as well.
On Dec 6, 2012, at 10:04 AM, Brian Foster <bdfoster at endigotech.com> wrote:

> It can fail with ORIGINATOR_CANCEL
> 
> Sent from my iPhone
> 
> On Dec 6, 2012, at 9:09 AM, Emrah <lists at kavun.ch> wrote:
> 
>> I might be missing something, but this only works if the bridge fails and not if the caller hangs up before reaching voicemail.
>> 
>> My work around is to schedule the lua as a hangup hook and check a few things.
>> From the top of my head, it was the channel status or bridge status; and the voicemail recording duration should either be empty or 0. This way you get notified if someone lands in the VM and doesn't leave a message.
>> 
>> Emrah
>> 
>> On Dec 3, 2012, at 11:44 AM, ?? <itispip-qq at hotmail.com> wrote:
>> 
>>> Thanks Yehavi, it's really handy to implement this script to my environment :)
>>> 
>>> /Alex
>>> 
>>> Date: Sun, 2 Dec 2012 10:21:17 +0200
>>> From: yehavi.bourvine at gmail.com
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Missed call notification
>>> 
>>> Hi,
>>> 
>>> I've created the following wiki page with example of how we do it:
>>> http://wiki.freeswitch.org/wiki/MailNoAsnwer
>>> 
>>> I would be greatfull if someone can add it to the examples page, as I don't have the edit authority on it...
>>> 
>>>                   Thanks, __Yehavi: 
>>> 
>>> 
>>> 2012/12/1 Komar, Jason <jkomar at jbox.ca>
>>> I would be interested to see that script too.
>>> 
>>> Thanks,
>>> Jason
>>> 
>>> 
>>> On Sat, Dec 1, 2012 at 11:34 AM, ?? <itispip-qq at hotmail.com> wrote:
>>> 
>>> Hi Yehavi, seems your solution is the best fit for the situation. Is that possible to share your script? Maybe can posted it to Freeswitch Wiki as part of the documentation.
>>> 
>>> Date: Sat, 1 Dec 2012 19:13:11 +0200
>>> From: yehavi.bourvine at gmail.com
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Missed call notification
>>> 
>>> We have a script to find the hangup cause and send Email if the user requested it. This way I know who is looking for me while I am on the phone :-)
>>> 
>>>                     __Yehavi: 
>>> 
>>> 
>>> 2012/12/1 Brian Foster <bdfoster at endigotech.com>
>>> Message Waiting Indicator (MWI) is the voicemail notification feature for IP phones. 
>>> 
>>> Missed call notifications are provided by the phone and not the switch.
>>> 
>>> Sent from my iPhone
>>> 
>>> On Dec 1, 2012, at 2:41 AM, ?? <itispip-qq at hotmail.com> wrote:
>>> 
>>> I heard that Freeswitch support missed call notification (i/o VM notification). But look around Wiki no document about this feature. Anyone knows where to setup it up?
>>> 
>>> /brgds, 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
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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.orghttp://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.orghttp://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: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
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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



From lists at kavun.ch  Fri Dec  7 00:17:42 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Dec 2012 16:17:42 -0500
Subject: [Freeswitch-users] Using bypass_media or proxy_media selectively
In-Reply-To: <CAFiqYunBGpiB=SkAuDH2oCsgg-YxG-nda4SOqHAhrwUMRWGtyw@mail.gmail.com>
References: <343F3B15-35E9-4CCF-A2BB-2514C1167E4C@kavun.ch>
	<CAFiqYunBGpiB=SkAuDH2oCsgg-YxG-nda4SOqHAhrwUMRWGtyw@mail.gmail.com>
Message-ID: <9B43674A-3C1D-44C3-9624-DA302B8AE52D@kavun.ch>

Thanks a million for your help, this is precious.
On Dec 6, 2012, at 9:18 AM, Steven Ayre <steveayre at gmail.com> wrote:

> If you have mod_h26x loaded and have added it to your codec prefs, you should not need either.
> 
> 
> On 6 December 2012 13:57, Emrah <lists at kavun.ch> wrote:
> Hi there,
> 
> Can you please confirm that I need to either use proxy_media or bypass_media for a video call in H.264?
> If so, how do I enable proxy_media or bypass_media conditionally by examining the codec prefs of both SIP endpoints?
> 
> For a setup where I have 2 devices registered as 1000 at mydomain, one being audio only and the other video capable. Plus a video capable device registered as 1001 at mydomain.
> How do I selectively use bypass_media or proxy_media for calls made by 1001 to 1000 and answered by the video capable terminal?
> 
> Thanks a million,
> Emrah
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From lists at kavun.ch  Fri Dec  7 00:36:15 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 6 Dec 2012 16:36:15 -0500
Subject: [Freeswitch-users] Using bypass_media or proxy_media selectively
In-Reply-To: <9B43674A-3C1D-44C3-9624-DA302B8AE52D@kavun.ch>
References: <343F3B15-35E9-4CCF-A2BB-2514C1167E4C@kavun.ch>
	<CAFiqYunBGpiB=SkAuDH2oCsgg-YxG-nda4SOqHAhrwUMRWGtyw@mail.gmail.com>
	<9B43674A-3C1D-44C3-9624-DA302B8AE52D@kavun.ch>
Message-ID: <618D242B-73A7-4DB9-BF3E-5E28140E6EB5@kavun.ch>

Got excited too quickly. Mod_h26x is already loaded. I can only use H264 in pass-through mode and that's either with bypass_media or proxy_media.
On Dec 6, 2012, at 4:17 PM, Emrah <lists at kavun.ch> wrote:

> Thanks a million for your help, this is precious.
> On Dec 6, 2012, at 9:18 AM, Steven Ayre <steveayre at gmail.com> wrote:
> 
>> If you have mod_h26x loaded and have added it to your codec prefs, you should not need either.
>> 
>> 
>> On 6 December 2012 13:57, Emrah <lists at kavun.ch> wrote:
>> Hi there,
>> 
>> Can you please confirm that I need to either use proxy_media or bypass_media for a video call in H.264?
>> If so, how do I enable proxy_media or bypass_media conditionally by examining the codec prefs of both SIP endpoints?
>> 
>> For a setup where I have 2 devices registered as 1000 at mydomain, one being audio only and the other video capable. Plus a video capable device registered as 1001 at mydomain.
>> How do I selectively use bypass_media or proxy_media for calls made by 1001 to 1000 and answered by the video capable terminal?
>> 
>> Thanks a million,
>> Emrah
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.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 Dec  7 00:41:53 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 6 Dec 2012 16:41:53 -0500
Subject: [Freeswitch-users] VIEW specific events for specific device (ip
	address)
Message-ID: <181201cdd3fa$822acc20$86806460$@bizfocused.com>

Hi,

 

I am trying to debug a sip registration problem.  To get the events I am
currently using an SSH log feature with "sofia global siptrace on".
Unfortunately, this server has many many many sip events and this produce a
huge log very quickly.

 

Is there another way to view/log SIP events and filter by device IP ADDRESS
or maybe by extension?

 

Thanks,

Sean

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From msc at freeswitch.org  Fri Dec  7 00:49:49 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Dec 2012 13:49:49 -0800
Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today
In-Reply-To: <CANBhyPApMOtUqKGtwa_CvyHLfSe=Yt=MCYbj0rxopd=g0dxzZA@mail.gmail.com>
References: <CAKzWOxUeDLWVF23J_9RZR+KfE0EQmY4koDw2DpNhWKz9F1G0-g@mail.gmail.com>
	<9BDCA7E1-9C70-4131-958C-A2A159E3BC0D@endigotech.com>
	<CANBhyPApMOtUqKGtwa_CvyHLfSe=Yt=MCYbj0rxopd=g0dxzZA@mail.gmail.com>
Message-ID: <CAKzWOxUTgc4-xMNjGLhmT2npng=epUosd+FmQuzLxLXMZjbJPA@mail.gmail.com>

On Wed, Dec 5, 2012 at 8:43 PM, Anton Kvashenkin <anton.jugatsu at gmail.com>wrote:

> Michael, I don't see any presentation or videos at cluecon.com.

Correct. They've not been uploaded yet. The announcement was that the
transcoding and cleanup were done and we have a volunteer (Jay Binks) who
is splicing/editing so that each presentation is just a single video file.
I'll let you know when he's done.

-MC
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From msc at freeswitch.org  Fri Dec  7 00:52:41 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Dec 2012 13:52:41 -0800
Subject: [Freeswitch-users] how to Ignore DTMF shorter than Fixed
	duration ?
In-Reply-To: <CAHWtr1SfKpkXL7BeB0995sjN9k8yLwKEXRcxkXvZyvQfqqEjUg@mail.gmail.com>
References: <CAHWtr1TbtvoxrxbCEY6y8r_rKgtsh2zHhXmeZYRA0+6P0dKGUQ@mail.gmail.com>
	<CAKzWOxXQSj9hPSoSmH9pQ19J=+eU5qM+8aNO51BoWzzN_gKj_w@mail.gmail.com>
	<CAHWtr1SfKpkXL7BeB0995sjN9k8yLwKEXRcxkXvZyvQfqqEjUg@mail.gmail.com>
Message-ID: <CAKzWOxUoXCMFOb-P8ZxOVJzaK4=o5Kxt=dF+EnMHEt_cu2qQqQ@mail.gmail.com>

Bummer. I'm not aware of any way to ignore those. Anyone else have ideas?
-MC

On Thu, Dec 6, 2012 at 12:59 AM, Regis M <regis.freeswitch.org at tornad.net>wrote:

> Hi,
>
> It's RFC2833 DTMF, not inband.
>
> Regards,
>
>
> 2012/12/5 Michael Collins <msc at freeswitch.org>
>
>> Are they sending those digits inband or with RFC2833?
>> -MC
>>
>> On Wed, Dec 5, 2012 at 11:28 AM, Regis M <regis.freeswitch.org at tornad.net
>> > wrote:
>>
>>> Hi,
>>>
>>> We have a bugged provider that send us throw RTP wrong DTMF during call.
>>> It seems that wrong DTMF are shorter than 1200ms (about 90%) so I want to
>>> try to ignore them.
>>> For the moment, FS catch them and send it back to the bridged side,
>>> boring user.
>>>
>>> switch_rtp.c:3410 RTP RECV DTMF C:552
>>>
>>>
>>>
>>> Does min-dtmf-duration will make FS ignore them and not RECV them ?
>>> I saw the DTMF in "normal" RTP packet, but no digit was pressed in user
>>> side.
>>>
>>> It's commented in my FS switch.conf.xml, how can I see the current value
>>> ?
>>> <!--<param name="min-dtmf-duration" value="400"/>-->
>>>
>>>
>>> Any other idea ?
>>>
>>> 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From gsamat at gmail.com  Thu Dec  6 22:38:13 2012
From: gsamat at gmail.com (Samat Galimov)
Date: Thu, 6 Dec 2012 23:38:13 +0400
Subject: [Freeswitch-users]  symbian s60 SIP over TLS
Message-ID: <A852B629-C8BB-4942-8602-CC5D40611CE7@gmail.com>

Hello Eric,

Have you finally managed to use symbian60 default SIP client to work via TLS or SRTP with Freeswitch?
I found you asked that back 2008 http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/034031.html

Maybe any other VoIP server?
I am trying to find any success story on using s60 devices with TLS/SRTP on the internet and the only is patched asterisk http://forums.digium.com/viewtopic.php?f=1&t=76233&start=0 , which I doesn't feel ok with.

Best,
 Samat Galimov.

From shishko69 at gmail.com  Thu Dec  6 23:23:03 2012
From: shishko69 at gmail.com (Shishko)
Date: Thu, 6 Dec 2012 21:23:03 +0100
Subject: [Freeswitch-users] Problems with T.38
Message-ID: <CAHT_yt_wfuG4+b8uZ7G+OSge+e7sVmLzE9hviUxg2Kq4Nn9b-Q@mail.gmail.com>

Hi people!

I've setup two Windows boxes with FS 1.3.8b and started testing T.38
functionality between them.
First FS box should send fax and second one should receive it, hopefully.

On receiving side I've setup following dialplan

<extension name="fax_receive">
	<condition field="destination_number" expression="^9178">
		<action application="set" data="fax_enable_t38=true"/>
		<action application="set" data="fax_enable_t38_request=true"/>
		<action application="set" data="absolute_codec_string=PCMA"/>
		<action application="set" data="proxy_media=true"/>
		<action application="set" data="bypass_media=false"/>
		<action application="set" data="use-ecm=true"/>
		<action application="answer" />
		<action application="playback" data="silence_stream://2000"/>
		<action application="rxfax" data="fax/rxfax.tif"/>
		<action application="hangup"/>
	</condition>
</extension>


On sending side I use command originate
{origination_caller_id_number=1234}sofia/gateway/fs2/9178
&txfax(fax.tif)

But I always get error "Fax processing not successful - result (49)
The call dropped prematurely." or "Fax processing not successful -
result (17) Received a DCN while waiting for a DIS".

Any hint for me?

Thanks

-------------------------------------------------------------------------------------------------------------------------
freeswitch at fs2> recv 1068 bytes from udp/[10.0.0.1]:5060 at 20:20:29.334875:
   ------------------------------------------------------------------------
   INVITE sip:9178 at 10.0.0.2 SIP/2.0
   Via: SIP/2.0/UDP 10.0.0.1;rport;branch=z9hG4bKry6g7U1QDer3g
   Max-Forwards: 70
   From: "" <sip:FreeSWITCH at 10.0.0.2>;tag=rN8rZtevXta7S
   To: <sip:9178 at 10.0.0.2>
   Call-ID: 3824cd3f-ba85-1230-1b9e-9b4137433975
   CSeq: 37076550 INVITE
   Contact: <sip:FreeSWITCH at 10.0.0.1:5060;transport=udp;gw=fs2>
   User-Agent: FreeSWITCH-mod_sofia/1.3.8b
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, r
efer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 193
   X-FS-Support: update_display,send_info
   Remote-Party-ID: <sip:1234 at 10.0.0.2>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1354798433 1354798434 IN IP4 10.0.0.1
   s=FreeSWITCH
   c=IN IP4 10.0.0.1
   t=0 0
   m=audio 26796 RTP/AVP 8 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
send 297 bytes to udp/[10.0.0.1]:5060 at 20:20:29.334875:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.0.0.1;rport=5060;branch=z9hG4bKry6g7U1QDer3g
   From: "" <sip:FreeSWITCH at 10.0.0.2>;tag=rN8rZtevXta7S
   To: <sip:9178 at 10.0.0.2>
   Call-ID: 3824cd3f-ba85-1230-1b9e-9b4137433975
   CSeq: 37076550 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.3.8b
   Content-Length: 0

   ------------------------------------------------------------------------
2012-12-06 21:20:29.334875 [NOTICE] switch_channel.c:968 New Channel
sofia/internal/FreeSWITCH at 10.0.0.2
[712d8bfe-f045-4665-8b9c-f610a4f1a438]
2012-12-06 21:20:29.350500 [DEBUG] switch_core_session.c:975 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:29.350500 [DEBUG] switch_core_session.c:975 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:29.350500 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/FreeSWITCH at 10.0.0.2) Running State Change CS_NEW
2012-12-06 21:20:29.350500 [DEBUG] switch_core_state_machine.c:433
(sofia/internal/FreeSWITCH at 10.0.0.2) State NEW
2012-12-06 21:20:29.350500 [DEBUG] sofia.c:7678 IP 10.0.0.1 Approved
by acl "domains[]". Access Granted.
2012-12-06 21:20:29.522375 [DEBUG] sofia.c:5603 Channel
sofia/internal/FreeSWITCH at 10.0.0.2 entering state [received][100]
2012-12-06 21:20:29.522375 [DEBUG] sofia.c:5614 Remote SDP:
v=0
o=FreeSWITCH 1354798433 1354798434 IN IP4 10.0.0.1
s=FreeSWITCH
c=IN IP4 10.0.0.1
t=0 0
m=audio 26796 RTP/AVP 8 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2012-12-06 21:20:29.522375 [DEBUG] sofia.c:5811
(sofia/internal/FreeSWITCH at 10.0.0.2) State Change CS_NEW -> CS_INIT
2012-12-06 21:20:29.522375 [DEBUG] switch_core_session.c:1283 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/FreeSWITCH at 10.0.0.2) Running State Change CS_INIT
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:454
(sofia/internal/FreeSWITCH at 10.0.0.2) State INIT
2012-12-06 21:20:29.522375 [DEBUG] mod_sofia.c:86
sofia/internal/FreeSWITCH at 10.0.0.2 SOFIA INIT
2012-12-06 21:20:29.522375 [DEBUG] mod_sofia.c:126
(sofia/internal/FreeSWITCH at 10.0.0.2) State Change CS_INIT ->
CS_ROUTING
2012-12-06 21:20:29.522375 [DEBUG] switch_core_session.c:1283 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:454
(sofia/internal/FreeSWITCH at 10.0.0.2) State INIT going to sleep
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/FreeSWITCH at 10.0.0.2) Running State Change CS_ROUTING
2012-12-06 21:20:29.522375 [DEBUG] switch_channel.c:2003
(sofia/internal/FreeSWITCH at 10.0.0.2) Callstate Change DOWN -> RINGING
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:470
(sofia/internal/FreeSWITCH at 10.0.0.2) State ROUTING
2012-12-06 21:20:29.522375 [DEBUG] mod_sofia.c:149
sofia/internal/FreeSWITCH at 10.0.0.2 SOFIA ROUTING
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:117
sofia/internal/FreeSWITCH at 10.0.0.2 Standard ROUTING
2012-12-06 21:20:29.522375 [INFO] mod_dialplan_xml.c:498 Processing
<1234>->9178 in context public
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing [public->unloop]
continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[public->outside_call] continue=true
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Absolute Condition [outside_call]
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action set(outside_call=true)
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action
export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[public->call_debug] continue=true
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [call_debug]
${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[public->public_extensions] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[public_extensions] destination_number(9178) =~ /^(10[01][0-9])$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[public->fax_extensions] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (PASS)
[fax_extensions] destination_number(9178) =~ /^9178/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action transfer(9178 XML default)
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:167
(sofia/internal/FreeSWITCH at 10.0.0.2) State Change CS_ROUTING ->
CS_EXECUTE
2012-12-06 21:20:29.522375 [DEBUG] switch_core_session.c:1283 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:470
(sofia/internal/FreeSWITCH at 10.0.0.2) State ROUTING going to sleep
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/FreeSWITCH at 10.0.0.2) Running State Change CS_EXECUTE
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:477
(sofia/internal/FreeSWITCH at 10.0.0.2) State EXECUTE
2012-12-06 21:20:29.522375 [DEBUG] mod_sofia.c:242
sofia/internal/FreeSWITCH at 10.0.0.2 SOFIA EXECUTE
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:209
sofia/internal/FreeSWITCH at 10.0.0.2 Standard EXECUTE
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2 set(outside_call=true)
2012-12-06 21:20:29.522375 [DEBUG] mod_dptools.c:1344
sofia/internal/FreeSWITCH at 10.0.0.2 SET [outside_call]=[true]
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2 export(RFC2822_DATE=Thu, 06
Dec 2012 21:20:29 Central European Standard Time)
2012-12-06 21:20:29.522375 [DEBUG] switch_channel.c:1135 EXPORT
(export_vars) [RFC2822_DATE]=[Thu, 06 Dec 2012 21:20:29 Central
European Standard Time
]
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2 transfer(9178 XML default)
2012-12-06 21:20:29.522375 [DEBUG] switch_ivr.c:1773
(sofia/internal/FreeSWITCH at 10.0.0.2) State Change CS_EXECUTE ->
CS_ROUTING
2012-12-06 21:20:29.522375 [DEBUG] switch_core_session.c:1283 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:29.522375 [DEBUG] switch_core_session.c:830 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:29.522375 [NOTICE] switch_ivr.c:1779 Transfer
sofia/internal/FreeSWITCH at 10.0.0.2 to XML[9178 at default]
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:477
(sofia/internal/FreeSWITCH at 10.0.0.2) State EXECUTE going to sleep
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/FreeSWITCH at 10.0.0.2) Running State Change CS_ROUTING
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:470
(sofia/internal/FreeSWITCH at 10.0.0.2) State ROUTING
2012-12-06 21:20:29.522375 [DEBUG] mod_sofia.c:149
sofia/internal/FreeSWITCH at 10.0.0.2 SOFIA ROUTING
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:117
sofia/internal/FreeSWITCH at 10.0.0.2 Standard ROUTING
2012-12-06 21:20:29.522375 [INFO] mod_dialplan_xml.c:498 Processing
<1234>->9178 in context default
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing [default->unloop]
continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->tod_example] continue=true
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Date/TimeMatch (FAIL)
[tod_example] break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->holiday_example] continue=true
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Date/TimeMatch (FAIL)
[holiday_example] break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->global-intercept] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[global-intercept] destination_number(9178) =~ /^886$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->group-intercept] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[group-intercept] destination_number(9178) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->intercept-ext] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[intercept-ext] destination_number(9178) =~ /^\*\*(\d+)$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing [default->redial]
continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [redial]
destination_number(9178) =~ /^(redial|870)$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing [default->global]
continue=true
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [global]
${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [global]
${sip_has_crypto}() =~
/^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=
never
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (PASS) [global]
${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED
NEGOTIATION)/ break=on-f
alse
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [global]
${switch_r_sdp}(v=0
o=FreeSWITCH 1354798433 1354798434 IN IP4 10.0.0.1
s=FreeSWITCH
c=IN IP4 10.0.0.1
t=0 0
m=audio 26796 RTP/AVP 8 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Absolute Condition [global]
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action
hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action
export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->snom-demo-2] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[snom-demo-2] destination_number(9178) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->snom-demo-1] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[snom-demo-1] destination_number(9178) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->eavesdrop] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [eavesdrop]
destination_number(9178) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->eavesdrop] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [eavesdrop]
destination_number(9178) =~ /^779$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->call_return] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[call_return] destination_number(9178) =~ /^\*69$|^869$|^lcr$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->del-group] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [del-group]
destination_number(9178) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->add-group] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [add-group]
destination_number(9178) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->call-group-simo] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[call-group-simo] destination_number(9178) =~ /^82(\d{2})$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->call-group-order] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[call-group-order] destination_number(9178) =~ /^83(\d{2})$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->extension-intercom] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[extension-intercom] destination_number(9178) =~ /^8(10[01][0-9])$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[Local_Extension] destination_number(9178) =~ /^(10[01][0-9])$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->Local_Extension_Skinny] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[Local_Extension_Skinny] destination_number(9178) =~ /^(11[01][0-9])$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->group_dial_sales] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[group_dial_sales] destination_number(9178) =~ /^2000$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->group_dial_support] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[group_dial_support] destination_number(9178) =~ /^2001$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->group_dial_billing] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[group_dial_billing] destination_number(9178) =~ /^2002$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->operator] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [operator]
destination_number(9178) =~ /^(operator|0)$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing [default->vmain]
continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [vmain]
destination_number(9178) =~ /^vmain$|^4000$|^\*98$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->sip_uri] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [sip_uri]
destination_number(9178) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->nb_conferences] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[nb_conferences] destination_number(9178) =~ /^(30\d{2})$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->wb_conferences] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[wb_conferences] destination_number(9178) =~ /^(31\d{2})$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->uwb_conferences] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[uwb_conferences] destination_number(9178) =~ /^(32\d{2})$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->cdquality_conferences] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[cdquality_conferences] destination_number(9178) =~ /^(33\d{2})$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->freeswitch_public_conf_via_sip] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[freeswitch_public_conf_via_sip] destination_number(9178) =~
/^9(888|8888|1616|3232)$/ break
=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->mad_boss_intercom] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[mad_boss_intercom] destination_number(9178) =~ /^0911$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->mad_boss_intercom] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[mad_boss_intercom] destination_number(9178) =~ /^0912$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->mad_boss] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [mad_boss]
destination_number(9178) =~ /^0913$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->ivr_demo] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [ivr_demo]
destination_number(9178) =~ /^5000$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->dynamic_conference] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[dynamic_conference] destination_number(9178) =~ /^5001$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->rtp_multicast_page] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL)
[rtp_multicast_page] destination_number(9178) =~ /^pagegroup$|^7243$/
break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing [default->park]
continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [park]
destination_number(9178) =~ /^5900$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [unpark]
destination_number(9178) =~ /^5901$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->valet_park] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [valet_park]
destination_number(9178) =~ /^(6000)$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->valet_park] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [valet_park]
destination_number(9178) =~ /^(60\d[1-9])$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing [default->park]
continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [park]
destination_number(9178) =~ /park\+(\d+)/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [unpark]
destination_number(9178) =~ /^parking$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing [default->park]
continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [park]
destination_number(9178) =~ /callpark/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [unpark]
destination_number(9178) =~ /pickup/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing [default->wait]
continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (FAIL) [wait]
destination_number(9178) =~ /^wait$/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 parsing
[default->fax_receive] continue=false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Regex (PASS)
[fax_receive] destination_number(9178) =~ /^9178/ break=on-false
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action set(fax_enable_t38=true)
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action
set(fax_enable_t38_request=true)
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action
set(absolute_codec_string=PCMA)
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action set(proxy_media=true)
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action set(bypass_media=false)
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action set(use-ecm=true)
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action answer()
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action
playback(silence_stream://2000)
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action rxfax(fax/rxfax.tif)
Dialplan: sofia/internal/FreeSWITCH at 10.0.0.2 Action hangup()
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:167
(sofia/internal/FreeSWITCH at 10.0.0.2) State Change CS_ROUTING ->
CS_EXECUTE
2012-12-06 21:20:29.522375 [DEBUG] switch_core_session.c:1283 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:470
(sofia/internal/FreeSWITCH at 10.0.0.2) State ROUTING going to sleep
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/FreeSWITCH at 10.0.0.2) Running State Change CS_EXECUTE
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:477
(sofia/internal/FreeSWITCH at 10.0.0.2) State EXECUTE
2012-12-06 21:20:29.522375 [DEBUG] mod_sofia.c:242
sofia/internal/FreeSWITCH at 10.0.0.2 SOFIA EXECUTE
2012-12-06 21:20:29.522375 [DEBUG] switch_core_state_machine.c:209
sofia/internal/FreeSWITCH at 10.0.0.2 Standard EXECUTE
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2
hash(insert/0.0.0.0-spymap/1234/712d8bfe-f045-4665-8b9c-f610a4f1a438)
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2
hash(insert/0.0.0.0-last_dial/1234/9178)
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2
hash(insert/0.0.0.0-last_dial/global/712d8bfe-f045-4665-8b9c-f610a4f1a438)
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2 export(RFC2822_DATE=Thu, 06
Dec 2012 21:20:29 Central European Standard Time)
2012-12-06 21:20:29.522375 [DEBUG] switch_channel.c:1135 EXPORT
(export_vars) [RFC2822_DATE]=[Thu, 06 Dec 2012 21:20:29 Central
European Standard Time
]
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2 set(fax_enable_t38=true)
2012-12-06 21:20:29.522375 [DEBUG] mod_dptools.c:1344
sofia/internal/FreeSWITCH at 10.0.0.2 SET [fax_enable_t38]=[true]
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2 set(fax_enable_t38_request=true)
2012-12-06 21:20:29.522375 [DEBUG] mod_dptools.c:1344
sofia/internal/FreeSWITCH at 10.0.0.2 SET [fax_enable_t38_request]=[true]
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2 set(absolute_codec_string=PCMA)
2012-12-06 21:20:29.522375 [DEBUG] mod_dptools.c:1344
sofia/internal/FreeSWITCH at 10.0.0.2 SET [absolute_codec_string]=[PCMA]
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2 set(proxy_media=true)
2012-12-06 21:20:29.522375 [DEBUG] mod_dptools.c:1344
sofia/internal/FreeSWITCH at 10.0.0.2 SET [proxy_media]=[true]
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2 set(bypass_media=false)
2012-12-06 21:20:29.522375 [DEBUG] mod_dptools.c:1344
sofia/internal/FreeSWITCH at 10.0.0.2 SET [bypass_media]=[false]
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2 set(use-ecm=true)
2012-12-06 21:20:29.522375 [DEBUG] mod_dptools.c:1344
sofia/internal/FreeSWITCH at 10.0.0.2 SET [use-ecm]=[true]
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2 answer()
2012-12-06 21:20:29.663000 [DEBUG] sofia_glue.c:5134 Audio Codec
Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
2012-12-06 21:20:29.663000 [DEBUG] sofia_glue.c:3093 Set Codec
sofia/internal/FreeSWITCH at 10.0.0.2 PCMA/8000 20 ms 160 samples 64000
bits
2012-12-06 21:20:29.663000 [DEBUG] switch_core_codec.c:111
sofia/internal/FreeSWITCH at 10.0.0.2 Original read codec set to PCMA:8
2012-12-06 21:20:29.663000 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf
send/recv payload to 101
2012-12-06 21:20:29.663000 [DEBUG] sofia_glue.c:3348 AUDIO RTP
[sofia/internal/FreeSWITCH at 10.0.0.2] 10.0.0.2 port 16666 -> 10.0.0.1
port 26796 codec:
8 ms: 20
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:1928 Starting timer
[soft] 160 bytes per 20ms
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
START SESSION INITIALIZATION. sID=43.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
ZID=303030303030303030393363.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
Loading User's profile:
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 allowclear: OFF
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 autosecure: ON
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
disclose_bit: OFF
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
signal. role: Unknown
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
        TTL: 4294967295
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
SAS schemes: 2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916 B256
2012-12-0
6 21:20:29.663000 [DEBUG] switch_rtp.c:916 B32  2012-12-06
21:20:29.663000 [DEBUG] switch_rtp.c:916
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
   Ciphers: 2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916 AES3
2012-12-06
 21:20:29.663000 [DEBUG] switch_rtp.c:916 AES1 2012-12-06
21:20:29.663000 [DEBUG] switch_rtp.c:916
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 PK schemes: 2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916 DH3k
2012-12-0
6 21:20:29.663000 [DEBUG] switch_rtp.c:916 DH2k 2012-12-06
21:20:29.663000 [DEBUG] switch_rtp.c:916 Mult 2012-12-06
21:20:29.663000 [DEBUG] switch_rtp
.c:916
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
        ATL: 2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916 HS32
2012-12-0
6 21:20:29.663000 [DEBUG] switch_rtp.c:916
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
    Hashes: 2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916 S256
2012-12-06
 21:20:29.663000 [DEBUG] switch_rtp.c:916
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
Session initialization - DONE. sID=43.

2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
ATTACH NEW STREAM to sID=43:
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [        zrtp]:
 Stream ID=0 UNKNOWN switching <NONE> ---> <ACTIVE>.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 Empty slot was found - initializing new stream with ID=43.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 Preparing ZRTP Hello according to the Session profile.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
ATTACH NEW STREAM - DONE.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
START STREAM ID=43 mode=CLEAR state=ACTIVE.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [        zrtp]:
 Stream ID=43 CLEAR switching <ACTIVE> ---> <START>.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22463 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:29.663000 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf
send payload to 101
2012-12-06 21:20:29.663000 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf
receive payload to 101
2012-12-06 21:20:29.663000 [DEBUG] sofia_glue.c:3645
sofia/internal/FreeSWITCH at 10.0.0.2 Set rtp dtmf delay to 40
2012-12-06 21:20:29.663000 [DEBUG] sofia_glue.c:3651 Set comfort noise
payload to 13
2012-12-06 21:20:29.663000 [NOTICE] sofia_glue.c:4256 Pre-Answer
sofia/internal/FreeSWITCH at 10.0.0.2!
2012-12-06 21:20:29.663000 [DEBUG] switch_channel.c:3136
(sofia/internal/FreeSWITCH at 10.0.0.2) Callstate Change RINGING -> EARLY
2012-12-06 21:20:29.663000 [DEBUG] sofia_glue.c:3348 AUDIO RTP
[sofia/internal/FreeSWITCH at 10.0.0.2] 10.0.0.2 port 16666 -> 10.0.0.1
port 26796 codec:
8 ms: 20
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:1928 Starting timer
[soft] 160 bytes per 20ms
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
START SESSION INITIALIZATION. sID=44.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
ZID=303030303030303030393363.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
Loading User's profile:
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 allowclear: OFF
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 autosecure: ON
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
disclose_bit: OFF
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
signal. role: Unknown
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
        TTL: 4294967295
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
SAS schemes: 2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916 B256
2012-12-0
6 21:20:29.663000 [DEBUG] switch_rtp.c:916 B32  2012-12-06
21:20:29.663000 [DEBUG] switch_rtp.c:916
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
   Ciphers: 2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916 AES3
2012-12-06
 21:20:29.663000 [DEBUG] switch_rtp.c:916 AES1 2012-12-06
21:20:29.663000 [DEBUG] switch_rtp.c:916
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 PK schemes: 2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916 DH3k
2012-12-0
6 21:20:29.663000 [DEBUG] switch_rtp.c:916 DH2k 2012-12-06
21:20:29.663000 [DEBUG] switch_rtp.c:916 Mult 2012-12-06
21:20:29.663000 [DEBUG] switch_rtp
.c:916
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
        ATL: 2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916 HS32
2012-12-0
6 21:20:29.663000 [DEBUG] switch_rtp.c:916
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
    Hashes: 2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916 S256
2012-12-06
 21:20:29.663000 [DEBUG] switch_rtp.c:916
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
Session initialization - DONE. sID=44.

2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
ATTACH NEW STREAM to sID=44:
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [        zrtp]:
 Stream ID=0 UNKNOWN switching <NONE> ---> <ACTIVE>.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 Empty slot was found - initializing new stream with ID=44.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 Preparing ZRTP Hello according to the Session profile.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [   zrtp main]:
ATTACH NEW STREAM - DONE.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
START STREAM ID=44 mode=CLEAR state=ACTIVE.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [        zrtp]:
 Stream ID=44 CLEAR switching <ACTIVE> ---> <START>.
2012-12-06 21:20:29.663000 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445208357 seq=9178 size=140. Stream 44:CLEAR:START
2012-12-06 21:20:29.663000 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf
send payload to 101
2012-12-06 21:20:29.663000 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf
receive payload to 101
2012-12-06 21:20:29.663000 [DEBUG] sofia_glue.c:3645
sofia/internal/FreeSWITCH at 10.0.0.2 Set rtp dtmf delay to 40
2012-12-06 21:20:29.663000 [DEBUG] sofia_glue.c:3651 Set comfort noise
payload to 13
2012-12-06 21:20:29.663000 [DEBUG] mod_sofia.c:856 Local SDP
sofia/internal/FreeSWITCH at 10.0.0.2:
v=0
o=FreeSWITCH 1354808563 1354808564 IN IP4 10.0.0.2
s=FreeSWITCH
c=IN IP4 10.0.0.2
t=0 0
m=audio 16666 RTP/AVP 8 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
a=sendrecv

2012-12-06 21:20:29.663000 [DEBUG] switch_core_session.c:830 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:29.663000 [DEBUG] switch_channel.c:3395
(sofia/internal/FreeSWITCH at 10.0.0.2) Callstate Change EARLY -> ACTIVE
2012-12-06 21:20:29.663000 [NOTICE] mod_dptools.c:1176 Channel
[sofia/internal/FreeSWITCH at 10.0.0.2] has been answered
send 1181 bytes to udp/[10.0.0.1]:5060 at 20:20:29.678625:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.0.0.1;rport=5060;branch=z9hG4bKry6g7U1QDer3g
   From: "" <sip:FreeSWITCH at 10.0.0.2>;tag=rN8rZtevXta7S
   To: <sip:9178 at 10.0.0.2>;tag=em42Xtm0UDeDD
   Call-ID: 3824cd3f-ba85-1230-1b9e-9b4137433975
   CSeq: 37076550 INVITE
   Contact: <sip:9178 at 10.0.0.2:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.3.8b
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, r
efer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 236
   X-FS-Display-Name: 9178
   X-FS-Display-Number: sip:9178 at 10.0.0.2
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "9178"
<sip:9178 at 10.0.0.2>;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 1354808563 1354808564 IN IP4 10.0.0.2
   s=FreeSWITCH
   c=IN IP4 10.0.0.2
   t=0 0
   m=audio 16666 RTP/AVP 8 101 13
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20
   ------------------------------------------------------------------------
2012-12-06 21:20:29.663000 [DEBUG] switch_core_session.c:975 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:29.663000 [DEBUG] sofia.c:5603 Channel
sofia/internal/FreeSWITCH at 10.0.0.2 entering state [completed][200]
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2 playback(silence_stream://2000)
2012-12-06 21:20:29.663000 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms
2012-12-06 21:20:29.741125 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445208357 seq=9179 size=140. Stream 44:CLEAR:START
2012-12-06 21:20:29.741125 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22464 size=140. Stream
43:CLEAR:START
recv 376 bytes from udp/[10.0.0.1]:5060 at 20:20:29.850500:
   ------------------------------------------------------------------------
   ACK sip:9178 at 10.0.0.2:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 10.0.0.1;rport;branch=z9hG4bKS7Z98pjUaQepc
   Max-Forwards: 70
   From: "" <sip:FreeSWITCH at 10.0.0.2>;tag=rN8rZtevXta7S
   To: <sip:9178 at 10.0.0.2>;tag=em42Xtm0UDeDD
   Call-ID: 3824cd3f-ba85-1230-1b9e-9b4137433975
   CSeq: 37076550 ACK
   Contact: <sip:FreeSWITCH at 10.0.0.1:5060;transport=udp;gw=fs2>
   Content-Length: 0

   ------------------------------------------------------------------------
2012-12-06 21:20:29.850500 [DEBUG] switch_core_session.c:975 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:29.850500 [DEBUG] switch_core_session.c:975 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:29.850500 [DEBUG] switch_core_session.c:975 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:29.850500 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445208357 seq=9180 size=140. Stream 44:CLEAR:START
2012-12-06 21:20:29.866125 [DEBUG] sofia.c:5603 Channel
sofia/internal/FreeSWITCH at 10.0.0.2 entering state [ready][200]
2012-12-06 21:20:29.881750 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22465 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:30.053625 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445208357 seq=9181 size=140. Stream 44:CLEAR:START
2012-12-06 21:20:30.084875 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22466 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:30.131750 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
Received <Hello   > packet with ssrc=1413688741 seq=4776/4776
size=140. Stream44:
CLEAR:START.
2012-12-06 21:20:30.131750 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
 Processing HELLO from FreeSWITCH V=1.10, P=0, M=1.
2012-12-06 21:20:30.131750 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
         ac=1 cc=2 sc=2 kc=3
2012-12-06 21:20:30.131750 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
         S256AES3AES1HS32DH3kDH2kMultB256B32
2012-12-06 21:20:30.131750 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
 Received HELLO had the same protocol V.
2012-12-06 21:20:30.131750 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
Received a ZRTP_HELLO packet with the same ZRTP ID that we have.
 This is likely due to a bug in the software. Ignoring the ZRTP_HELLO
 packet, therefore this call cannot be encrypted.
2012-12-06 21:20:30.131750 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
 Enter InitiatingError State with ERROR:<Equal ZIDs in Hello>,
notification Ena
bled. ID=44
2012-12-06 21:20:30.131750 [DEBUG] switch_rtp.c:916  [        zrtp]:
 Stream ID=44 CLEAR switching <START> ---> <INITERROR>.
2012-12-06 21:20:30.131750 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <ERROR> ssrc=1445208357 seq=9182 size=32. Stream
44:CLEAR:INITERROR
2012-12-06 21:20:30.131750 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
 ERROR! _zrtp_machine_process_hello() failed with status=1. ID=44
2012-12-06 21:20:30.147375 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
Received <Error   > packet with ssrc=1413688741 seq=4777/4777 size=32.
Stream44:C
LEAR:INITERROR.
2012-12-06 21:20:30.147375 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
 Enter PendingError State with ERROR:<Equal ZIDs in Hello>. ID=44
2012-12-06 21:20:30.147375 [DEBUG] switch_rtp.c:916  [        zrtp]:
 Stream ID=44 CLEAR switching <INITERROR> ---> <PENDERROR>.
2012-12-06 21:20:30.147375 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <ERRORACK> ssrc=1445208357 seq=9183 size=28. Stream
44:CLEAR:PENDERROR
2012-12-06 21:20:30.194250 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
Received <ErrorACK> packet with ssrc=1413688741 seq=4778/4778 size=28.
Stream44:C
LEAR:PENDERROR.
2012-12-06 21:20:30.288000 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <ERRORACK> ssrc=1445208357 seq=9184 size=28. Stream
44:CLEAR:PENDERROR
2012-12-06 21:20:30.303625 [DEBUG] switch_rtp.c:3622 Correct ip/port confirmed.
2012-12-06 21:20:30.303625 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22467 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:30.350500 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
Received <ErrorACK> packet with ssrc=1413688741 seq=4779/4779 size=28.
Stream44:C
LEAR:PENDERROR.
2012-12-06 21:20:30.444250 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <ERRORACK> ssrc=1445208357 seq=9185 size=28. Stream
44:CLEAR:PENDERROR
2012-12-06 21:20:30.522375 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
WARNING! HELLO have been resent 5 times without a response. Raising
ZRTP_EVENT_NO
_ZRTP_QUICK event. ID=43
2012-12-06 21:20:30.522375 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22468 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:30.553625 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
Received <ErrorACK> packet with ssrc=1413688741 seq=4780/4780 size=28.
Stream44:C
LEAR:PENDERROR.
2012-12-06 21:20:30.616125 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
 WARNING! ERRORACK Max retransmissions count reached. ID=44
2012-12-06 21:20:30.616125 [DEBUG] switch_rtp.c:916  [        zrtp]:
 Stream ID=44 CLEAR switching <PENDERROR> ---> <ERROR>.
2012-12-06 21:20:30.725500 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22469 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:30.928625 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22470 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:31.131750 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22471 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:31.350500 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22472 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:31.553625 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22473 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:31.741125 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22474 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:31.834875 [DEBUG] switch_ivr_play_say.c:1682 done
playing file silence_stream://2000
EXECUTE sofia/internal/FreeSWITCH at 10.0.0.2 rxfax(fax/rxfax.tif)
2012-12-06 21:20:31.834875 [DEBUG] mod_spandsp_fax.c:1363 Raw read
codec activation Success L16 20000
2012-12-06 21:20:31.834875 [DEBUG] switch_core_codec.c:219
sofia/internal/FreeSWITCH at 10.0.0.2 Push codec L16:70
2012-12-06 21:20:31.834875 [DEBUG] mod_spandsp_fax.c:1379 Raw write
codec activation Success L16
2012-12-06 21:20:31.944250 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22475 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:32.163000 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22476 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:32.366125 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22477 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:32.569250 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22478 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:32.772375 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22479 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:32.975500 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22480 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:33.084875 [DEBUG] sofia_glue.c:173
sofia/internal/FreeSWITCH at 10.0.0.2 image media sdp:
v=0
o=FreeSWITCH 1354808563 1354808565 IN IP4 10.0.0.2
s=FreeSWITCH
c=IN IP4 10.0.0.2
t=0 0
m=image 16666 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:2000
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

2012-12-06 21:20:33.084875 [DEBUG] sofia_glue.c:2647 Local SDP:
v=0
o=FreeSWITCH 1354808563 1354808565 IN IP4 10.0.0.2
s=FreeSWITCH
c=IN IP4 10.0.0.2
t=0 0
m=image 16666 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:2000
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

send 992 bytes to udp/[10.0.0.1]:5060 at 20:20:33.084875:
   ------------------------------------------------------------------------
   INVITE sip:FreeSWITCH at 10.0.0.1:5060;transport=udp;gw=fs2 SIP/2.0
   Via: SIP/2.0/UDP 10.0.0.2;rport;branch=z9hG4bKj8736jSrDv4ZB
   Max-Forwards: 69
   From: <sip:9178 at 10.0.0.2>;tag=em42Xtm0UDeDD
   To: "" <sip:FreeSWITCH at 10.0.0.2>;tag=rN8rZtevXta7S
   Call-ID: 3824cd3f-ba85-1230-1b9e-9b4137433975
   CSeq: 37076552 INVITE
   Contact: <sip:9178 at 10.0.0.2:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.3.8b
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 308
   X-FS-Support: update_display,send_info

   v=0
   o=FreeSWITCH 1354808563 1354808565 IN IP4 10.0.0.2
   s=FreeSWITCH
   c=IN IP4 10.0.0.2
   t=0 0
   m=image 16666 udptl t38
   a=T38FaxVersion:0
   a=T38MaxBitRate:14400
   a=T38FaxFillBitRemoval
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxMaxBuffer:2000
   a=T38FaxMaxDatagram:400
   a=T38FaxUdpEC:t38UDPRedundancy
   ------------------------------------------------------------------------
2012-12-06 21:20:33.084875 [DEBUG] switch_core_session.c:975 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
recv 315 bytes from udp/[10.0.0.1]:5060 at 20:20:33.084875:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.0.0.2;rport=5060;branch=z9hG4bKj8736jSrDv4ZB
   From: <sip:9178 at 10.0.0.2>;tag=em42Xtm0UDeDD
   To: "" <sip:FreeSWITCH at 10.0.0.2>;tag=rN8rZtevXta7S
   Call-ID: 3824cd3f-ba85-1230-1b9e-9b4137433975
   CSeq: 37076552 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.3.8b
   Content-Length: 0

   ------------------------------------------------------------------------
2012-12-06 21:20:33.084875 [DEBUG] sofia.c:5603 Channel
sofia/internal/FreeSWITCH at 10.0.0.2 entering state [calling][0]
recv 927 bytes from udp/[10.0.0.1]:5060 at 20:20:33.116125:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.0.0.2;rport=5060;branch=z9hG4bKj8736jSrDv4ZB
   From: <sip:9178 at 10.0.0.2>;tag=em42Xtm0UDeDD
   To: "" <sip:FreeSWITCH at 10.0.0.2>;tag=rN8rZtevXta7S
   Call-ID: 3824cd3f-ba85-1230-1b9e-9b4137433975
   CSeq: 37076552 INVITE
   Contact: <sip:FreeSWITCH at 10.0.0.1:5060;transport=udp;gw=fs2>
   User-Agent: FreeSWITCH-mod_sofia/1.3.8b
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 308

   v=0
   o=FreeSWITCH 1354798433 1354798435 IN IP4 10.0.0.1
   s=FreeSWITCH
   c=IN IP4 10.0.0.1
   t=0 0
   m=image 26796 udptl t38
   a=T38FaxVersion:0
   a=T38MaxBitRate:14400
   a=T38FaxFillBitRemoval
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxMaxBuffer:2000
   a=T38FaxMaxDatagram:400
   a=T38FaxUdpEC:t38UDPRedundancy
   ------------------------------------------------------------------------
2012-12-06 21:20:33.116125 [DEBUG] switch_core_session.c:975 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:33.116125 [DEBUG] switch_core_session.c:975 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:33.116125 [INFO] sofia.c:931
sofia/internal/FreeSWITCH at 10.0.0.2 Update Callee ID to "FreeSWITCH"
<FreeSWITCH>
2012-12-06 21:20:33.194250 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22481 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:33.272375 [DEBUG] sofia.c:5603 Channel
sofia/internal/FreeSWITCH at 10.0.0.2 entering state [completing][200]
2012-12-06 21:20:33.272375 [DEBUG] sofia.c:5614 Remote SDP:
v=0
o=FreeSWITCH 1354798433 1354798435 IN IP4 10.0.0.1
s=FreeSWITCH
c=IN IP4 10.0.0.1
t=0 0
m=image 26796 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:2000
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

send 327 bytes to udp/[10.0.0.1]:5060 at 20:20:33.288000:
   ------------------------------------------------------------------------
   ACK sip:FreeSWITCH at 10.0.0.1:5060;transport=udp;gw=fs2 SIP/2.0
   Via: SIP/2.0/UDP 10.0.0.2;rport;branch=z9hG4bKKH1v8Dava5tjQ
   Max-Forwards: 70
   From: <sip:9178 at 10.0.0.2>;tag=em42Xtm0UDeDD
   To: "" <sip:FreeSWITCH at 10.0.0.2>;tag=rN8rZtevXta7S
   Call-ID: 3824cd3f-ba85-1230-1b9e-9b4137433975
   CSeq: 37076552 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
2012-12-06 21:20:33.272375 [DEBUG] switch_core_session.c:975 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:33.272375 [DEBUG] switch_core_session.c:975 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:33.288000 [DEBUG] sofia.c:5603 Channel
sofia/internal/FreeSWITCH at 10.0.0.2 entering state [ready][200]
2012-12-06 21:20:33.366125 [WARNING] switch_rtp.c:3652 Ignoring
invalid RTP packet size of 6 bytes.
2012-12-06 21:20:33.381750 [WARNING] switch_rtp.c:3652 Ignoring
invalid RTP packet size of 6 bytes.
2012-12-06 21:20:33.381750 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1445101237 seq=22482 size=140. Stream
43:CLEAR:START
2012-12-06 21:20:33.413000 [WARNING] switch_rtp.c:3652 Ignoring
invalid RTP packet size of 6 bytes.
2012-12-06 21:20:33.428625 [WARNING] switch_rtp.c:3652 Ignoring
invalid RTP packet size of 8 bytes.
2012-12-06 21:20:33.444250 [WARNING] switch_rtp.c:3652 Ignoring
invalid RTP packet size of 8 bytes.
2012-12-06 21:20:33.444250 [DEBUG] sofia_glue.c:3348 AUDIO RTP
[sofia/internal/FreeSWITCH at 10.0.0.2] 10.0.0.2 port 16666 -> 10.0.0.1
port 26796 codec:
8 ms: 20
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:1928 Starting timer
[soft] 160 bytes per 20ms
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
START SESSION INITIALIZATION. sID=45.
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
ZID=303030303030303030393363.
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
Loading User's profile:
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 allowclear: OFF
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 autosecure: ON
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
disclose_bit: OFF
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
signal. role: Unknown
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
        TTL: 4294967295
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
SAS schemes: 2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916 B256
2012-12-0
6 21:20:33.444250 [DEBUG] switch_rtp.c:916 B32  2012-12-06
21:20:33.444250 [DEBUG] switch_rtp.c:916
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
   Ciphers: 2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916 AES3
2012-12-06
 21:20:33.444250 [DEBUG] switch_rtp.c:916 AES1 2012-12-06
21:20:33.444250 [DEBUG] switch_rtp.c:916
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 PK schemes: 2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916 DH3k
2012-12-0
6 21:20:33.444250 [DEBUG] switch_rtp.c:916 DH2k 2012-12-06
21:20:33.444250 [DEBUG] switch_rtp.c:916 Mult 2012-12-06
21:20:33.444250 [DEBUG] switch_rtp
.c:916
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
        ATL: 2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916 HS32
2012-12-0
6 21:20:33.444250 [DEBUG] switch_rtp.c:916
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
    Hashes: 2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916 S256
2012-12-06
 21:20:33.444250 [DEBUG] switch_rtp.c:916
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
Session initialization - DONE. sID=45.

2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
ATTACH NEW STREAM to sID=45:
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [        zrtp]:
 Stream ID=0 UNKNOWN switching <NONE> ---> <ACTIVE>.
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 Empty slot was found - initializing new stream with ID=45.
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
 Preparing ZRTP Hello according to the Session profile.
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [   zrtp main]:
ATTACH NEW STREAM - DONE.
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
START STREAM ID=45 mode=CLEAR state=ACTIVE.
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [        zrtp]:
 Stream ID=45 CLEAR switching <ACTIVE> ---> <START>.
2012-12-06 21:20:33.444250 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63418 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:33.444250 [ERR] sofia_glue.c:3549 Invalid
Jitterbuffer spec [0] must be between 20 and 10000
2012-12-06 21:20:33.444250 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf
send payload to 101
2012-12-06 21:20:33.444250 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf
receive payload to 101
2012-12-06 21:20:33.444250 [DEBUG] sofia_glue.c:3645
sofia/internal/FreeSWITCH at 10.0.0.2 Set rtp dtmf delay to 40
2012-12-06 21:20:33.475500 [WARNING] switch_rtp.c:3652 Ignoring
invalid RTP packet size of 8 bytes.
2012-12-06 21:20:33.491125 [DEBUG] mod_sofia.c:1487 Remote
address:port [10.0.0.1:26796] has not changed.
2012-12-06 21:20:33.491125 [DEBUG] mod_sofia.c:1487 Remote
address:port [10.0.0.1:26796] has not changed.
2012-12-06 21:20:33.491125 [DEBUG] sofia_glue.c:173
sofia/internal/FreeSWITCH at 10.0.0.2 image media sdp:
v=0
o=FreeSWITCH 1354808563 1354808566 IN IP4 10.0.0.2
s=FreeSWITCH
c=IN IP4 10.0.0.2
t=0 0
m=image 16666 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:2000
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

2012-12-06 21:20:33.491125 [DEBUG] mod_sofia.c:1487 Remote
address:port [10.0.0.1:26796] has not changed.
2012-12-06 21:20:33.491125 [DEBUG] mod_sofia.c:1487 Remote
address:port [10.0.0.1:26796] has not changed.
2012-12-06 21:20:33.491125 [DEBUG] sofia_glue.c:173
sofia/internal/FreeSWITCH at 10.0.0.2 image media sdp:
v=0
o=FreeSWITCH 1354808563 1354808567 IN IP4 10.0.0.2
s=FreeSWITCH
c=IN IP4 10.0.0.2
t=0 0
m=image 16666 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:2000
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

2012-12-06 21:20:33.491125 [DEBUG] switch_core_session.c:975 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:33.491125 [DEBUG] switch_core_session.c:975 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:33.522375 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63419 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:33.584875 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
WARNING! HELLO Max retransmissions count reached (20 retries). ID=43
2012-12-06 21:20:33.584875 [DEBUG] switch_rtp.c:916  [        zrtp]:
 Stream ID=43 CLEAR switching <START> ---> <NOZRTP>.
2012-12-06 21:20:33.631750 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63420 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:33.850500 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63421 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:34.053625 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63422 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:34.241125 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
WARNING! HELLO have been resent 5 times without a response. Raising
ZRTP_EVENT_NO
_ZRTP_QUICK event. ID=45
2012-12-06 21:20:34.241125 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63423 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:34.444250 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63424 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:34.663000 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63425 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:34.866125 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63426 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:35.069250 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63427 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:35.272375 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63428 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:35.475500 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63429 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:35.694250 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63430 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:35.881750 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63431 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:36.084875 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63432 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:36.303625 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63433 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:36.506750 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63434 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:36.709875 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63435 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:36.913000 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63436 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:37.131750 [DEBUG] switch_rtp.c:916  [  zrtp utils]:
 Send <HELLO> ssrc=1447881465 seq=63437 size=140. Stream
45:CLEAR:START
2012-12-06 21:20:37.334875 [DEBUG] switch_rtp.c:916  [ zrtp engine]:
WARNING! HELLO Max retransmissions count reached (20 retries). ID=45
2012-12-06 21:20:37.334875 [DEBUG] switch_rtp.c:916  [        zrtp]:
 Stream ID=45 CLEAR switching <START> ---> <NOZRTP>.
recv 622 bytes from udp/[10.0.0.1]:5060 at 20:20:42.600500:
   ------------------------------------------------------------------------
   BYE sip:9178 at 10.0.0.2:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 10.0.0.1;rport;branch=z9hG4bKtgS2aj3y7Z48Q
   Max-Forwards: 70
   From: "" <sip:FreeSWITCH at 10.0.0.2>;tag=rN8rZtevXta7S
   To: <sip:9178 at 10.0.0.2>;tag=em42Xtm0UDeDD
   Call-ID: 3824cd3f-ba85-1230-1b9e-9b4137433975
   CSeq: 37076551 BYE
   Contact: <sip:FreeSWITCH at 10.0.0.1:5060;transport=udp;gw=fs2>
   User-Agent: FreeSWITCH-mod_sofia/1.3.8b
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0

   ------------------------------------------------------------------------
2012-12-06 21:20:42.584875 [DEBUG] switch_core_session.c:975 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:43.491125 [ERR] mod_spandsp_fax.c:641 INVALID WRITE: 22:1
2012-12-06 21:20:43.491125 [ERR] mod_spandsp_fax.c:653 TERMINATING T30 STATE
2012-12-06 21:20:43.491125 [DEBUG] mod_spandsp_fax.c:487
==============================================================================
2012-12-06 21:20:43.491125 [DEBUG] mod_spandsp_fax.c:500 Fax
processing not successful - result (49) The call dropped prematurely.
2012-12-06 21:20:43.491125 [DEBUG] mod_spandsp_fax.c:505 Remote station id:
2012-12-06 21:20:43.491125 [DEBUG] mod_spandsp_fax.c:506 Local station
id:  SpanDSP Fax Ident
2012-12-06 21:20:43.491125 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0
2012-12-06 21:20:43.491125 [DEBUG] mod_spandsp_fax.c:509 Total fax pages:   0
2012-12-06 21:20:43.491125 [DEBUG] mod_spandsp_fax.c:510 Image resolution:  0x0
2012-12-06 21:20:43.491125 [DEBUG] mod_spandsp_fax.c:511 Transfer
Rate:     14400
2012-12-06 21:20:43.491125 [DEBUG] mod_spandsp_fax.c:513 ECM status         off
2012-12-06 21:20:43.491125 [DEBUG] mod_spandsp_fax.c:514 remote country:
2012-12-06 21:20:43.491125 [DEBUG] mod_spandsp_fax.c:515 remote vendor:
2012-12-06 21:20:43.491125 [DEBUG] mod_spandsp_fax.c:516 remote model:
2012-12-06 21:20:43.491125 [DEBUG] mod_spandsp_fax.c:518
==============================================================================
2012-12-06 21:20:43.491125 [DEBUG] switch_channel.c:2994
(sofia/internal/FreeSWITCH at 10.0.0.2) Callstate Change ACTIVE -> HANGUP
2012-12-06 21:20:43.491125 [NOTICE] sofia.c:711 Hangup
sofia/internal/FreeSWITCH at 10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING]
2012-12-06 21:20:43.491125 [DEBUG] switch_channel.c:3017 Send signal
sofia/internal/FreeSWITCH at 10.0.0.2 [KILL]
2012-12-06 21:20:43.491125 [DEBUG] switch_core_session.c:1283 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
send 466 bytes to udp/[10.0.0.1]:5060 at 20:20:43.491125:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.0.0.1;rport=5060;branch=z9hG4bKtgS2aj3y7Z48Q
   From: "" <sip:FreeSWITCH at 10.0.0.2>;tag=rN8rZtevXta7S
   To: <sip:9178 at 10.0.0.2>;tag=em42Xtm0UDeDD
   Call-ID: 3824cd3f-ba85-1230-1b9e-9b4137433975
   CSeq: 37076551 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.3.8b
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
2012-12-06 21:20:43.506750 [DEBUG] switch_core_codec.c:244
sofia/internal/FreeSWITCH at 10.0.0.2 Restore previous codec PCMA:8.
2012-12-06 21:20:43.506750 [DEBUG] switch_core_session.c:2687
sofia/internal/FreeSWITCH at 10.0.0.2 skip receive message
[APPLICATION_EXEC_COMPLETE] (cha
nnel is hungup already)
2012-12-06 21:20:43.506750 [DEBUG] switch_core_state_machine.c:477
(sofia/internal/FreeSWITCH at 10.0.0.2) State EXECUTE going to sleep
2012-12-06 21:20:43.506750 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/FreeSWITCH at 10.0.0.2) Running State Change CS_HANGUP
2012-12-06 21:20:43.506750 [DEBUG] switch_core_state_machine.c:667
(sofia/internal/FreeSWITCH at 10.0.0.2) State HANGUP
2012-12-06 21:20:43.506750 [DEBUG] mod_sofia.c:503 Channel
sofia/internal/FreeSWITCH at 10.0.0.2 hanging up, cause: NORMAL_CLEARING
2012-12-06 21:20:43.647375 [DEBUG] switch_core_state_machine.c:48
sofia/internal/FreeSWITCH at 10.0.0.2 Standard HANGUP, cause:
NORMAL_CLEARING
2012-12-06 21:20:43.647375 [DEBUG] switch_core_state_machine.c:667
(sofia/internal/FreeSWITCH at 10.0.0.2) State HANGUP going to sleep
2012-12-06 21:20:43.647375 [DEBUG] switch_core_state_machine.c:446
(sofia/internal/FreeSWITCH at 10.0.0.2) State Change CS_HANGUP ->
CS_REPORTING
2012-12-06 21:20:43.647375 [DEBUG] switch_core_session.c:1283 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:43.647375 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/FreeSWITCH at 10.0.0.2) Running State Change CS_REPORTING
2012-12-06 21:20:43.647375 [DEBUG] switch_core_state_machine.c:749
(sofia/internal/FreeSWITCH at 10.0.0.2) State REPORTING
2012-12-06 21:20:43.647375 [DEBUG] switch_core_state_machine.c:92
sofia/internal/FreeSWITCH at 10.0.0.2 Standard REPORTING, cause:
NORMAL_CLEARING
2012-12-06 21:20:43.647375 [DEBUG] switch_core_state_machine.c:749
(sofia/internal/FreeSWITCH at 10.0.0.2) State REPORTING going to sleep
2012-12-06 21:20:43.647375 [DEBUG] switch_core_state_machine.c:440
(sofia/internal/FreeSWITCH at 10.0.0.2) State Change CS_REPORTING ->
CS_DESTROY
2012-12-06 21:20:43.647375 [DEBUG] switch_core_session.c:1283 Send
signal sofia/internal/FreeSWITCH at 10.0.0.2 [BREAK]
2012-12-06 21:20:43.647375 [DEBUG] switch_core_session.c:1488 Session
19 (sofia/internal/FreeSWITCH at 10.0.0.2) Locked, Waiting on external
entities
2012-12-06 21:20:43.647375 [NOTICE] switch_core_session.c:1506 Session
19 (sofia/internal/FreeSWITCH at 10.0.0.2) Ended
2012-12-06 21:20:43.647375 [NOTICE] switch_core_session.c:1510 Close
Channel sofia/internal/FreeSWITCH at 10.0.0.2 [CS_DESTROY]
2012-12-06 21:20:43.647375 [DEBUG] switch_core_state_machine.c:556
(sofia/internal/FreeSWITCH at 10.0.0.2) Callstate Change HANGUP -> DOWN
2012-12-06 21:20:43.647375 [DEBUG] switch_core_state_machine.c:559
(sofia/internal/FreeSWITCH at 10.0.0.2) Running State Change CS_DESTROY
2012-12-06 21:20:43.647375 [DEBUG] switch_core_state_machine.c:569
(sofia/internal/FreeSWITCH at 10.0.0.2) State DESTROY
2012-12-06 21:20:43.647375 [DEBUG] mod_sofia.c:396
sofia/internal/FreeSWITCH at 10.0.0.2 SOFIA DESTROY
2012-12-06 21:20:43.647375 [DEBUG] switch_core_state_machine.c:99
sofia/internal/FreeSWITCH at 10.0.0.2 Standard DESTROY
2012-12-06 21:20:43.647375 [DEBUG] switch_core_state_machine.c:569
(sofia/internal/FreeSWITCH at 10.0.0.2) State DESTROY going to sleep


freeswitch at fs2>


From msc at freeswitch.org  Fri Dec  7 00:58:13 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Dec 2012 13:58:13 -0800
Subject: [Freeswitch-users] sip registration
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233A02B@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2339A49@Mail-Kilo.squay.com>
	<CAFiqYu=cpMkTxdE99smQBM6NgitNQ5ZpePwfSYr_R23dpj7Hng@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339B39@Mail-Kilo.squay.com>
	<CAKzWOxUW21XZte0AoVDqTtC6vkJ4FmA3sbP5LgJVK5bj4+1Sbw@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339D42@Mail-Kilo.squay.com>
	<CAKzWOxVMWfR6vk8ODKkND1Tjxj-d5VFYn9U+dDoNyB_t2dE05w@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233A02B@Mail-Kilo.squay.com>
Message-ID: <CAKzWOxVqn1tGWgHKFs+vCEDpXUg2rNyvpbw5-bBSPw846hg7Xw@mail.gmail.com>

Use "100" for the authentication name as well.
-MC

On Thu, Dec 6, 2012 at 9:25 AM, Archana Venugopan <a.venugopan at mundio.com>wrote:

>  Hi,****
>
> Thanks. I thought it will look in mod_sofia code. In the below screen I
> register the ID ?100?. Now instead of ?100? in ?Authentication Name? I need
> to give some e-mail ID or name(Archana) which should validate in DB.****
>
> I tried giving a name in ?Authentication Name? but the phone was not
> registered. Am not sure this authentication name is being looked in which
> column in table too.****
>
> Please let me know if this will be picked from any sofia code or any C
> script? Once we register in the below screen which script validates the
> Settings in freeswitch?****
>
>  ****
>
> Sorry if am repeating the same question, but I could not get the exact
> code and am clueless.****
>
> *Global SIP Settings*
>
> Top of Form****
>
> *Basic SIP Authentication Settings*
>
>
>
> Screen Name****
>
> ****
>
> Screen Name 2****
>
> ****
>
> Phone Number****
>
> ****
>
> Caller ID****
>
> ****
>
> Authentication Name****
>
> ****
>
> Password****
>
> ****
>
> Bottom of Form****
>
> ** **
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 05 December 2012 20:34
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
> ** **
>
> When an event that requires a user lookup takes place then the system will
> look in the XML user directory unless it has been configured to look
> somewhere else. The other places to look are usually:
> mod_xml_curl
> One of the language like Lua, Perl, Python
>
> If it's xml_curl then FS will do a POST to your web server in hopes of
> receiving back the necessary XML for the given user. It would be up to you
> to have your web server handle the request, poll the database, then format
> and return the XML data. See this wiki page<http://wiki.freeswitch.org/wiki/Xml_curl#How_do_I_configure_mod_xml_curl.3F>for more info on xml curl.
>
> If it's a language then you'll have a "binding" in the conf file for the
> language that will handle the lookup. Again, your script will need to
> handle the communication with your database. See this wiki page<http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration>for more information.
>
> Hope this helps.
> -MC****
>
> On Wed, Dec 5, 2012 at 9:30 AM, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
> Thanks for the information. But sorry, how to access user_data API command.
> ****
>
>  ****
>
> Am not clear on the flow. Once we register domain and usernumber in sip
> what exactly happens? Which script picks up this domain and username and
> validates with our database?****
>
> Could you please provide me with an overview. ****
>
>  ****
>
> Many thanks****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 05 December 2012 17:13
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
>  ****
>
> If you're talking about the user configuration then yes, you could create
> an "email" parameter or variable and access it with the user_data API
> command.
> -MC****
>
> On Wed, Dec 5, 2012 at 1:35 AM, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
> In that case can I have 1 more column say e-mail and can this e-mail be
> checked in DB instead of checking reg_user(?100?)? Is that feasible?****
>
> Also which code should be changed any idea please?****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 04 December 2012 19:51
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
>  ****
>
> You can have a user 'ana' in the domain 'gmail.com'. Though using someone
> else's domain as local in your FS setup may not be a good idea.****
>
> You can't have a @ in the username itself (per the SIP standard, not
> limited to FreeSWITCH).****
>
>  ****
>
> On 4 December 2012 18:00, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
>  ****
>
> Currently we register authentication name as say ?100? in sip
> registration, this comes to freeswitch and it will check in our DB for 100
> and if its present then registrations would be successful. ****
>
>  ****
>
> freeswitch at internal> show registrations****
>
>
> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
> ****
>
> 100,fsfailover.uk01.com
> ,e4969067f9a8c098,sofia/internal/sip:100 at 192.168.2.234:5060
> ;transport=udp,1354638871,192.168.2.234,5060,udp,squay-laptop-1.squay.com*
> ***
>
>  ****
>
> I want to change this 100 to some e-mail address, so instead of 100 it
> will be something like ?ana at gmail.com?. Can we do this? While coming to
> freeswitch whether there would be any issues?****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
> ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Fri Dec  7 01:01:54 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Dec 2012 14:01:54 -0800
Subject: [Freeswitch-users] VIEW specific events for specific device (ip
	address)
In-Reply-To: <181201cdd3fa$822acc20$86806460$@bizfocused.com>
References: <181201cdd3fa$822acc20$86806460$@bizfocused.com>
Message-ID: <CAKzWOxX29cVWvr3vzwi1fz93XN9Djvq=VK3bH-7f=UaniZzKRQ@mail.gmail.com>

The sofia siptrace feature does not allow for granular filtering like that.
The only way to get that would be to use one of the packet
capture<http://wiki.freeswitch.org/wiki/Packet_Capture>tools.

-MC

On Thu, Dec 6, 2012 at 1:41 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Hi,****
>
> ** **
>
> I am trying to debug a sip registration problem.  To get the events I am
> currently using an SSH log feature with ?sofia global siptrace on?.
> Unfortunately, this server has many many many sip events and this produce a
> huge log very quickly.****
>
> ** **
>
> Is there another way to view/log SIP events and filter by device IP
> ADDRESS or maybe by extension?****
>
> ** **
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From sdevoy at bizfocused.com  Fri Dec  7 01:25:00 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 6 Dec 2012 17:25:00 -0500
Subject: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch
Message-ID: <186801cdd400$87a246a0$96e6d3e0$@bizfocused.com>

I am new to Polycom phones.  We have always used CISCO 5xx Series.  I have
been trying to use the WEB interface.  The example in the WIKI uses tftp
through DHCP option 60.  MY home router/gateway's DHCP does not support
option 60.

 

I am unable to get the Polycom 335 to register, it keeps getting
"Unauthorized".  I have double checked by having my Cisco phone register
with this information and it works fine.  I must be missing something.

 

On the Polycom Web Interface I have the following values:

SIP TAB:

SERVERS:

Outbound Proxy is blank

SERVER 1:

Address: <my server's FQDN >

Port: 5060

Transport: UDPOnly

Expires:3600

Register: 1

Retry Timeout: 0

Retry Max Count: 3

Line Seize Timeout: 30

SERVER 2: <same as server 1 for now except port 5062)

LINE TAB:

LINE 1:

Display Name: 228

Address: 228

Auth UserID:228

Auth Password: <actual password here>

Label: 228

Type: Private

3rd Party Name: blank

Number of Line Keys: 1

Calls per line: 24

SERVER 1:

Address: <my server's FQDN >

Port: 5060

Transport: UDPOnly

Expires:3600

Register: 1

Retry Timeout: 0

Retry Max Count: 3

Line Seize Timeout: 30

SERVER 2: <same as server 1 for now)

Everything else is the defaults.

 

The resulting sofia trace messages:

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

recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.210138:

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

   REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0

   Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C

   From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B

   To: <sip:228 at fs_bfis.bizfocused.com>

   CSeq: 1 REGISTER

   Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47

   Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"

   User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069

   Accept-Language: en

   Max-Forwards: 70

   Expires: 3600

   Content-Length: 0

   

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

send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.211063:

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

   SIP/2.0 401 Unauthorized

   Via: SIP/2.0/UDP
10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57

   From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B

   To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre

   Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47

   CSeq: 1 REGISTER

   User-Agent:
FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120
712T101002Z

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: precondition, path, replaces

   WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com",
nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"

   Content-Length: 0

   

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

recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.712339:

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

   REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0

   Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C

   From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B

   To: <sip:228 at fs_bfis.bizfocused.com>

   CSeq: 1 REGISTER

   Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47

   Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"

   User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069

   Accept-Language: en

   Max-Forwards: 70

   Expires: 3600

   Content-Length: 0

 

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

send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.712479:

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

   SIP/2.0 401 Unauthorized

  Via: SIP/2.0/UDP
10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57

   From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B

  To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre

  Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47

   CSeq: 1 REGISTER

   User-Agent:
FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120
712T101002Z

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: precondition, path, replaces

   WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com",
nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"

   Content-Length: 0

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

 

 

Any ideas?

 

 

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From bdfoster at endigotech.com  Fri Dec  7 01:37:33 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 6 Dec 2012 17:37:33 -0500
Subject: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch
In-Reply-To: <186801cdd400$87a246a0$96e6d3e0$@bizfocused.com>
References: <186801cdd400$87a246a0$96e6d3e0$@bizfocused.com>
Message-ID: <949E214D-7B59-4D8E-BE39-DC5A3145C6D7@endigotech.com>

I think Address should be <extension>@<host name/ip>

Sent from my iPhone

On Dec 6, 2012, at 5:25 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> I am new to Polycom phones.  We have always used CISCO 5xx Series.  I have been trying to use the WEB interface.  The example in the WIKI uses tftp through DHCP option 60.  MY home router/gateway?s DHCP does not support option 60.
>  
> I am unable to get the Polycom 335 to register, it keeps getting ?Unauthorized?.  I have double checked by having my Cisco phone register with this information and it works fine.  I must be missing something.
>  
> On the Polycom Web Interface I have the following values:
> SIP TAB:
> SERVERS:
> Outbound Proxy is blank
> SERVER 1:
> Address: <my server?s FQDN >
> Port: 5060
> Transport: UDPOnly
> Expires:3600
> Register: 1
> Retry Timeout: 0
> Retry Max Count: 3
> Line Seize Timeout: 30
> SERVER 2: <same as server 1 for now except port 5062)
> LINE TAB:
> LINE 1:
> Display Name: 228
> Address: 228
> Auth UserID:228
> Auth Password: <actual password here>
> Label: 228
> Type: Private
> 3rd Party Name: blank
> Number of Line Keys: 1
> Calls per line: 24
> SERVER 1:
> Address: <my server?s FQDN >
> Port: 5060
> Transport: UDPOnly
> Expires:3600
> Register: 1
> Retry Timeout: 0
> Retry Max Count: 3
> Line Seize Timeout: 30
> SERVER 2: <same as server 1 for now)
> Everything else is the defaults.
>  
> The resulting sofia trace messages:
>    ------------------------------------------------------------------------
> recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.210138:
>    ------------------------------------------------------------------------
>    REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B
>    To: <sip:228 at fs_bfis.bizfocused.com>
>    CSeq: 1 REGISTER
>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47
>    Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
>    User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069
>    Accept-Language: en
>    Max-Forwards: 70
>    Expires: 3600
>    Content-Length: 0
>   
>    ------------------------------------------------------------------------
> send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.211063:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B
>    To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre
>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: precondition, path, replaces
>    WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"
>    Content-Length: 0
>   
>    ------------------------------------------------------------------------
> recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.712339:
>    ------------------------------------------------------------------------
>    REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B
>    To: <sip:228 at fs_bfis.bizfocused.com>
>    CSeq: 1 REGISTER
>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47
>    Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
>    User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069
>    Accept-Language: en
>    Max-Forwards: 70
>    Expires: 3600
>    Content-Length: 0
>  
>   ------------------------------------------------------------------------
> send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.712479:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>   Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B
>   To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre
>   Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: precondition, path, replaces
>    WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"
>    Content-Length: 0
>    ------------------------------------------------------------------------
>  
>  
> Any ideas?
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 anthony.minessale at gmail.com  Fri Dec  7 01:42:59 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 6 Dec 2012 16:42:59 -0600
Subject: [Freeswitch-users] Domains and profiles
In-Reply-To: <68AA779E-DC33-4F14-9720-1187A671280F@kavun.ch>
References: <624B624D-6C0E-461C-A7E6-9D271502BBBD@insensate.co.uk>
	<1A6CD519-9A32-4CD3-9DBB-2C4FC11D9A0D@kavun.ch>
	<CAKbxfG-31nxkyo1niEzWvcv12SLr-pdHWm_tKvM4uWoi1G46bA@mail.gmail.com>
	<3FC39EED-0A60-4A87-866D-0AAA629F9560@kavun.ch>
	<8ED4CC3C-441A-4AA0-8DDB-1B57CE646A3C@kavun.ch>
	<68AA779E-DC33-4F14-9720-1187A671280F@kavun.ch>
Message-ID: <CAKbxfG-2qU6q4v6hiJ3shQe0=PHkgu66ybexnSbdn=zY2eD8wA@mail.gmail.com>

did you try adding in the profile name like the explanation?

sofia_contact profile1/user at domain.com
sofia_contact profile2/user at domain.com




On Thu, Dec 6, 2012 at 8:18 AM, Emrah <lists at kavun.ch> wrote:

> Hey guys,
>
> Any further hints on this would be very appreciated.
>
> Thanks and all the best.
> On Nov 15, 2012, at 4:46 PM, Emrah <lists at kavun.ch> wrote:
>
> > I read and reread Anthony's explanation and am still not able to have
> multiple profile allow registering on the same domain.
> > If I have 2 identical profiles, it looks like the first one that is up
> will take the ownership of the domain name. Anything registering on the
> secondary one will not be visible to a sofia_contact.
> >
> > If sip.example.com:5070 is up before sip.example.com:5060, only phones
> registered to sip.example.com:5070 will be visible to FS.
> >
> > How do I fix this? I am not enforcing the domain in my configs.
> >
> > Thanks,
> > Emrah
> >
> > On Nov 13, 2012, at 8:23 PM, Emrah <lists at kavun.ch> wrote:
> >
> >> This is precious. I had figured out how the domain portion affects FS,
> I just didn't know how to declare my domains to my SIP profiles. Which I
> believe I now know and will experiment a bit.
> >>
> >> Thanks!
> >> On Nov 12, 2012, at 8:16 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
> >>
> >>> The best thing to do is take a look at these things from a step back.
> >>>
> >>> The domains inside the xml registry are completely different from the
> domains on the internet and again completely different from domains in sip
> packets.  The profiles are again entirely different from any of the above.
>  Its up to you to align them if you so choose.
> >>>
> >>>
> >>> The default configuration distributed with FreeSWITCH sets up the
> scenario most likely to load on any machine and work out of the box.  That
> is the primary goal of that configuration, so, It sets the domain in both
> the directory, the global default domain variable and the name of the
> internal profile to be identical to the ip on the box that can reach the
> internet.  Then it sets the sip to force everything to that value.  When
> you want to detach from this behavior, you are probably on a venture to do
> some kind of multi-home setup.
> >>>
> >>>
> >>> Aliases in the <aliases> tag are a list of keys you want to use to use
> that lead to the current profile your are configuring.  Think of it as the
> /etc/hosts file in unix only for profiles.   When you define aliases to
> match all of the possible domains hosted on a particular profile, then when
> you try to take a user at host.com notation and decide which profile it came
> from, you can use the aliases to find it providing you have added <alias
> name="host.com"/> to that profile.
> >>>
> >>> The <domains> tag is an indicator telling the profile to open the xml
> registry in FreeSWITCH and run through any domains defined therein.
> >>> The 2 key attributes are:
> >>>
> >>> alias: [true/false]  (automatically create an alias for this domain as
> mentioned above)
> >>> parse: [true/false] (scan the domain for gateway entries and include
> them into this profile)
> >>> name: [<string>] (either the name of a specific domain or 'all' to
> denote parsing every domain in the directory)
> >>>
> >>> As you showed in your question the default config has
> >>>
> >>> <domain name="all" alias="true" parse="false"/>
> >>>
> >>> If you apply what you have learned above, it will scan for every
> domain (there is only one by default) and add an alias for it and not parse
> it for gateways.  The default directory uses global config vars to set the
> domain to match the local ip on the box.  So now you will have a domain in
> your config that is your ip, and the internal profile will attach to it and
> add an alias so that value expands to match it.
> >>>
> >>>
> >>> This is explained in a comment at the top of directory/default.xml
> >>>
> >>>   FreeSWITCH works off the concept of users and domains just like
> email.
> >>>   You have users that are in domains for example 1000 at domain.com.
> >>>
> >>>   When freeswitch gets a register packet it looks for the user in the
> directory
> >>>   based on the from or to domain in the packet depending on how your
> sofia profile
> >>>   is configured.  Out of the box the default domain will be the IP
> address of the
> >>>   machine running FreeSWITCH.  This IP can be found by typing "sofia
> status" at the
> >>>   CLI.  You will register your phones to the IP and not the hostname
> by default.
> >>>   If you wish to register using the domain please open vars.xml in the
> root conf
> >>>   directory and set the default domain to the hostname you desire.
>  Then you would
> >>>   use the domain name in the client instead of the IP address to
> register
> >>>   with FreeSWITCH.
> >>>
> >>>
> >>>
> >>> So having more than one profile with the default of
> >>>
> >>> <domain name="all" alias="true" parse="false"/>
> >>>
> >>> is going to end up aliasing the same domains into all profiles who
> call it and cause an overwrite in the lookup table and probably an error in
> your logs somewhere.  If you had parse="true" on all of them, they would
> all try and register to the gateways in all of your domains.
> >>>
> >>>
> >>> If you look at the stock config, external.xml is a good example of a
> secondary profile, it has
> >>>
> >>> <domain name="all" alias="false" parse="true"/>
> >>>
> >>> so no aliases, and yes parse ... the exact opposite of the internal so
> that all the gateways would register from external and internal would bind
> to the local ip.
> >>>
> >>> So, you probably want to use separate <domain name="my.com"/> per
> domain per profile you want to bind it to in more complicated setups.
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> On Sun, Nov 11, 2012 at 9:09 PM, Emrah <lists at kavun.ch> wrote:
> >>> Bless you!
> >>>
> >>> Thanks for putting this together. You've beautifully summed up all my
> questions.
> >>> On Nov 11, 2012, at 8:09 AM, Lawrence Conroy <lconroy at insensate.co.uk>
> wrote:
> >>>
> >>>> Hi Folks,
> >>>> I've started a new thread as it's not quite the same issue, and
> domains & profiles have confused the heck out of me every time I have
> developed a new setup for fS.
> >>>> I have sometimes had to hack/hard-doce the dialstring to make
> multiple domains in one profile work, had hours of fun with presence, db
> and force register settings, and have still had some odd gotchas that have
> required extensive meditation.
> >>>> [... and yes, I have read the 1.0.6 bridge book; I'm trying to
> abstract these elements ]
> >>>>
> >>>> Coming at this from standards/specs and rolling my own SIP stacks,
> sofia/fS seems to use the term "domain" differently from sipdomain, and
> alias seems to be tied to the directory (and thus to the profile listed in
> a directory file), but I'm not sure.
> >>>> so ...
> >>>> Before I capture to the sofia conf xml wiki page, I have a couple of
> questions on the sip-profile XML setup;
> >>>>
> >>>> Q: Is there a particular reason why there's a parameter called alias
> and an (entirely different) setting also called alias?
> >>>> The sofia conf xml wiki's comment on the setting "alias" shows I'm
> not alone.
> >>>> I agree that's what it appears to be doing, but can we nail this down
> please (and what happens if an external client uses this connection to
> register and call)?
> >>>>
> >>>> In the current sofia conf xml wiki page, the domain setting is not
> exactly well documented :).
> >>>> The current internal.xml vanilla example from git (as of time of
> writing) has the following lines:
> >>>> -------------------------
> >>>> ...
> >>>> <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
> >>>> <!--aliases are other names that will work as a valid profile name
> for this profile-->
> >>>> <aliases>
> >>>> <!--
> >>>>      <alias name="default"/>
> >>>>  -->
> >>>> </aliases>
> >>>> <!-- Outbound Registrations -->
> >>>> <gateways>
> >>>> <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
> >>>> </gateways>
> >>>>
> >>>> <domains>
> >>>> <!-- indicator to parse the directory for domains with parse="true"
> to get gateways-->
> >>>> <!--<domain name="$${domain}" parse="true"/>-->
> >>>> <!-- indicator to parse the directory for domains with parse="true"
> to get gateways and alias every domain to this profile -->
> >>>> <!--<domain name="all" alias="true" parse="true"/>-->
> >>>> <domain name="all" alias="true" parse="false"/>
> >>>> </domains>
> >>>> ...
> >>>> -------------------------
> >>>>
> >>>> This stuff is entirely missing from the sofia.conf.xml wiki page, and
> it IS really important.
> >>>>
> >>>>
> >>>> Q: what's the default value for the alias parameter in the domain
> element? -- it is missing from the first example.
> >>>> Q: if there is more than one profile, what's the impact of setting
> parse = true in one (or all) of the profiles' XML files?
> >>>> (or parse = false, or missing the parameter altogether)?
> >>>> AFAICT, the gateways get pulled in via the pre-process directive just
> fine, regardless of the value of the parse parameter -- it works for me, at
> least.
> >>>>
> >>>> Q: if there is more than one profile, what's the impact of putting
> domain name="all" into one (or all) of the profiles' XML files?
> >>>>
> >>>> Ideally, having more than one sipdomain tied to one profile "would be
> good", but aliases doesn't do that -- as the git file says, these are
> aliases for the profile name.
> >>>>
> >>>> Before I start scribbling, Answers on a postcard to this ML, please.
> >>>>
> >>>> all the best,
> >>>> Lawrence
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting 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
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>> --
> >>> Anthony Minessale II
> >>>
> >>> 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
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Anthony Minessale II

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
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From anton.jugatsu at gmail.com  Fri Dec  7 07:21:33 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Fri, 7 Dec 2012 08:21:33 +0400
Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today
In-Reply-To: <CAKzWOxUTgc4-xMNjGLhmT2npng=epUosd+FmQuzLxLXMZjbJPA@mail.gmail.com>
References: <CAKzWOxUeDLWVF23J_9RZR+KfE0EQmY4koDw2DpNhWKz9F1G0-g@mail.gmail.com>
	<9BDCA7E1-9C70-4131-958C-A2A159E3BC0D@endigotech.com>
	<CANBhyPApMOtUqKGtwa_CvyHLfSe=Yt=MCYbj0rxopd=g0dxzZA@mail.gmail.com>
	<CAKzWOxUTgc4-xMNjGLhmT2npng=epUosd+FmQuzLxLXMZjbJPA@mail.gmail.com>
Message-ID: <CANBhyPC8mO0X9g--xfG4-+zjvm=XfGvZ0BBn3Ddgrh_Re3DY0A@mail.gmail.com>

Great. Thanks alot )


2012/12/7 Michael Collins <msc at freeswitch.org>

>
>
> On Wed, Dec 5, 2012 at 8:43 PM, Anton Kvashenkin <anton.jugatsu at gmail.com>wrote:
>
>> Michael, I don't see any presentation or videos at cluecon.com.
>
> Correct. They've not been uploaded yet. The announcement was that the
> transcoding and cleanup were done and we have a volunteer (Jay Binks) who
> is splicing/editing so that each presentation is just a single video file.
> I'll let you know when he's done.
>
> -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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec  7 07:41:13 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 6 Dec 2012 23:41:13 -0500
Subject: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch
In-Reply-To: <949E214D-7B59-4D8E-BE39-DC5A3145C6D7@endigotech.com>
References: <186801cdd400$87a246a0$96e6d3e0$@bizfocused.com>
	<949E214D-7B59-4D8E-BE39-DC5A3145C6D7@endigotech.com>
Message-ID: <1a1401cdd435$167f2940$437d7bc0$@bizfocused.com>

Thanks Brian.  I have looked up more info and that is correct.

However, I still get 401 Unauthorized.

 

Any other ideas anyone?

 

I am ready to return them and just by Cisco SPA504Gs.  I was hoping to add another choice for my customers but I am yet to come across any decent documentation from Polycom.  If I can?t even get them to log in, I am not hopeful about advanced features.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster
Sent: Thursday, December 06, 2012 5:38 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch

 

I think Address should be <extension>@<host name/ip>

Sent from my iPhone


On Dec 6, 2012, at 5:25 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

I am new to Polycom phones.  We have always used CISCO 5xx Series.  I have been trying to use the WEB interface.  The example in the WIKI uses tftp through DHCP option 60.  MY home router/gateway?s DHCP does not support option 60.

 

I am unable to get the Polycom 335 to register, it keeps getting ?Unauthorized?.  I have double checked by having my Cisco phone register with this information and it works fine.  I must be missing something.

 

On the Polycom Web Interface I have the following values:

SIP TAB:

SERVERS:

Outbound Proxy is blank

SERVER 1:

Address: <my server?s FQDN >

Port: 5060

Transport: UDPOnly

Expires:3600

Register: 1

Retry Timeout: 0

Retry Max Count: 3

Line Seize Timeout: 30

SERVER 2: <same as server 1 for now except port 5062)

LINE TAB:

LINE 1:

Display Name: 228

Address: 228

Auth UserID:228

Auth Password: <actual password here>

Label: 228

Type: Private

3rd Party Name: blank

Number of Line Keys: 1

Calls per line: 24

SERVER 1:

Address: <my server?s FQDN >

Port: 5060

Transport: UDPOnly

Expires:3600

Register: 1

Retry Timeout: 0

Retry Max Count: 3

Line Seize Timeout: 30

SERVER 2: <same as server 1 for now)

Everything else is the defaults.

 

The resulting sofia trace messages:

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

recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.210138:

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

   REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0

   Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C

   From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B

   To: <sip:228 at fs_bfis.bizfocused.com>

   CSeq: 1 REGISTER

   Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47

   Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"

   User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069

   Accept-Language: en

   Max-Forwards: 70

   Expires: 3600

   Content-Length: 0

   

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

send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.211063:

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

   SIP/2.0 401 Unauthorized

   Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57

   From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B

   To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre

   Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47

   CSeq: 1 REGISTER

   User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: precondition, path, replaces

   WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"

   Content-Length: 0

   

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

recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.712339:

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

   REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0

   Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C

   From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B

   To: <sip:228 at fs_bfis.bizfocused.com>

   CSeq: 1 REGISTER

   Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47

   Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"

   User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069

   Accept-Language: en

   Max-Forwards: 70

   Expires: 3600

   Content-Length: 0

 

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

send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.712479:

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

   SIP/2.0 401 Unauthorized

  Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57

   From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B

  To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre

  Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47

   CSeq: 1 REGISTER

   User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: precondition, path, replaces

   WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"

   Content-Length: 0

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

 

 

Any ideas?

 

 

_________________________________________________________________________
Professional FreeSWITCH Consulting 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
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From bdfoster at endigotech.com  Fri Dec  7 07:59:57 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 6 Dec 2012 23:59:57 -0500
Subject: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch
In-Reply-To: <1a1401cdd435$167f2940$437d7bc0$@bizfocused.com>
References: <186801cdd400$87a246a0$96e6d3e0$@bizfocused.com>
	<949E214D-7B59-4D8E-BE39-DC5A3145C6D7@endigotech.com>
	<1a1401cdd435$167f2940$437d7bc0$@bizfocused.com>
Message-ID: <EB14CF4C-8DFF-4BF6-A4F0-936525148763@endigotech.com>

We deploy these exact phones with Freeswitch and have no problems like this. There must be something you are missing and it's probably super simple. We deploy with UCS firmware if this makes a difference.

Decent documentation is certainly out there. There is plenty of documentation on Polycom's website. There's a pretty extensive manual with a few hundred pages of really awe-inspiring goodness.

Go back to the basics. Reset the phone and use the single page setup. 

Sent from my iPhone

On Dec 6, 2012, at 11:41 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> Thanks Brian.  I have looked up more info and that is correct.
> However, I still get 401 Unauthorized.
>  
> Any other ideas anyone?
>  
> I am ready to return them and just by Cisco SPA504Gs.  I was hoping to add another choice for my customers but I am yet to come across any decent documentation from Polycom.  If I can?t even get them to log in, I am not hopeful about advanced features.
>  
> Sean
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster
> Sent: Thursday, December 06, 2012 5:38 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch
>  
> I think Address should be <extension>@<host name/ip>
> 
> Sent from my iPhone
> 
> On Dec 6, 2012, at 5:25 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:
> 
> I am new to Polycom phones.  We have always used CISCO 5xx Series.  I have been trying to use the WEB interface.  The example in the WIKI uses tftp through DHCP option 60.  MY home router/gateway?s DHCP does not support option 60.
>  
> I am unable to get the Polycom 335 to register, it keeps getting ?Unauthorized?.  I have double checked by having my Cisco phone register with this information and it works fine.  I must be missing something.
>  
> On the Polycom Web Interface I have the following values:
> SIP TAB:
> SERVERS:
> Outbound Proxy is blank
> SERVER 1:
> Address: <my server?s FQDN >
> Port: 5060
> Transport: UDPOnly
> Expires:3600
> Register: 1
> Retry Timeout: 0
> Retry Max Count: 3
> Line Seize Timeout: 30
> SERVER 2: <same as server 1 for now except port 5062)
> LINE TAB:
> LINE 1:
> Display Name: 228
> Address: 228
> Auth UserID:228
> Auth Password: <actual password here>
> Label: 228
> Type: Private
> 3rd Party Name: blank
> Number of Line Keys: 1
> Calls per line: 24
> SERVER 1:
> Address: <my server?s FQDN >
> Port: 5060
> Transport: UDPOnly
> Expires:3600
> Register: 1
> Retry Timeout: 0
> Retry Max Count: 3
> Line Seize Timeout: 30
> SERVER 2: <same as server 1 for now)
> Everything else is the defaults.
>  
> The resulting sofia trace messages:
>    ------------------------------------------------------------------------
> recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.210138:
>    ------------------------------------------------------------------------
>    REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B
>    To: <sip:228 at fs_bfis.bizfocused.com>
>    CSeq: 1 REGISTER
>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47
>    Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
>    User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069
>    Accept-Language: en
>    Max-Forwards: 70
>    Expires: 3600
>    Content-Length: 0
>   
>    ------------------------------------------------------------------------
> send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.211063:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B
>    To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre
>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: precondition, path, replaces
>    WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"
>    Content-Length: 0
>   
>    ------------------------------------------------------------------------
> recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.712339:
>    ------------------------------------------------------------------------
>    REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B
>    To: <sip:228 at fs_bfis.bizfocused.com>
>    CSeq: 1 REGISTER
>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47
>    Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
>    User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069
>    Accept-Language: en
>    Max-Forwards: 70
>    Expires: 3600
>    Content-Length: 0
>  
>   ------------------------------------------------------------------------
> send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.712479:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>   Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B
>   To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre
>   Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: precondition, path, replaces
>    WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"
>    Content-Length: 0
>    ------------------------------------------------------------------------
>  
>  
> Any ideas?
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From virbhati at gmail.com  Fri Dec  7 11:52:42 2012
From: virbhati at gmail.com (virendra bhati)
Date: Fri, 7 Dec 2012 14:22:42 +0530
Subject: [Freeswitch-users] Nibblebill negative balance
Message-ID: <CANNhuhfyXQTao-4OriaS3Av_MC2=nm+GOLvfwJQAeJ76jRmVNg@mail.gmail.com>

Hi team,

I have configure nibblebill with my freeswitch and it's working. But I am
facing an issue with billing. Balance goes to -ve and after that calls also
throw as well....

Is that configuration issue or bug in nibblebill ?

-- 

Thanks and regards

 Virendra Bhati
+91-9250078532
Asterisk Developer
E-mail-: virbhati at gmail.com
Skype id:- virbhati2
New Delhi(India)
[image: View my profile on
LinkedIn]<http://in.linkedin.com/pub/virendra-bhati/6/a30/755>
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From a.venugopan at mundio.com  Fri Dec  7 12:21:30 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Fri, 7 Dec 2012 09:21:30 +0000
Subject: [Freeswitch-users] sip registration
In-Reply-To: <CAKzWOxVqn1tGWgHKFs+vCEDpXUg2rNyvpbw5-bBSPw846hg7Xw@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF2339A49@Mail-Kilo.squay.com>
	<CAFiqYu=cpMkTxdE99smQBM6NgitNQ5ZpePwfSYr_R23dpj7Hng@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339B39@Mail-Kilo.squay.com>
	<CAKzWOxUW21XZte0AoVDqTtC6vkJ4FmA3sbP5LgJVK5bj4+1Sbw@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339D42@Mail-Kilo.squay.com>
	<CAKzWOxVMWfR6vk8ODKkND1Tjxj-d5VFYn9U+dDoNyB_t2dE05w@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233A02B@Mail-Kilo.squay.com>
	<CAKzWOxVqn1tGWgHKFs+vCEDpXUg2rNyvpbw5-bBSPw846hg7Xw@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233A0D5@Mail-Kilo.squay.com>

Hi,

I want to give some alphabets instead of number. I want to know which script checks this authentication name to corresponding DB table. Please let me know.
Thanks

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 06 December 2012 21:58
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

Use "100" for the authentication name as well.
-MC
On Thu, Dec 6, 2012 at 9:25 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
Thanks. I thought it will look in mod_sofia code. In the below screen I register the ID '100'. Now instead of '100' in "Authentication Name" I need to give some e-mail ID or name(Archana) which should validate in DB.
I tried giving a name in "Authentication Name" but the phone was not registered. Am not sure this authentication name is being looked in which column in table too.
Please let me know if this will be picked from any sofia code or any C script? Once we register in the below screen which script validates the Settings in freeswitch?

Sorry if am repeating the same question, but I could not get the exact code and am clueless.
Global SIP Settings
Top of Form
Basic SIP Authentication Settings



Screen Name

Screen Name 2

Phone Number

Caller ID

Authentication Name

Password

Bottom of Form


Regards,
Archana

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: 05 December 2012 20:34
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

When an event that requires a user lookup takes place then the system will look in the XML user directory unless it has been configured to look somewhere else. The other places to look are usually:
mod_xml_curl
One of the language like Lua, Perl, Python

If it's xml_curl then FS will do a POST to your web server in hopes of receiving back the necessary XML for the given user. It would be up to you to have your web server handle the request, poll the database, then format and return the XML data. See this wiki page<http://wiki.freeswitch.org/wiki/Xml_curl#How_do_I_configure_mod_xml_curl.3F> for more info on xml curl.

If it's a language then you'll have a "binding" in the conf file for the language that will handle the lookup. Again, your script will need to handle the communication with your database. See this wiki page<http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration> for more information.

Hope this helps.
-MC
On Wed, Dec 5, 2012 at 9:30 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
Thanks for the information. But sorry, how to access user_data API command.

Am not clear on the flow. Once we register domain and usernumber in sip what exactly happens? Which script picks up this domain and username and validates with our database?
Could you please provide me with an overview.

Many thanks

Regards,
Archana

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: 05 December 2012 17:13
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

If you're talking about the user configuration then yes, you could create an "email" parameter or variable and access it with the user_data API command.
-MC
On Wed, Dec 5, 2012 at 1:35 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
In that case can I have 1 more column say e-mail and can this e-mail be checked in DB instead of checking reg_user('100')? Is that feasible?
Also which code should be changed any idea please?

Regards,
Archana

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 Steven Ayre
Sent: 04 December 2012 19:51
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

You can have a user 'ana' in the domain 'gmail.com<http://gmail.com>'. Though using someone else's domain as local in your FS setup may not be a good idea.
You can't have a @ in the username itself (per the SIP standard, not limited to FreeSWITCH).

On 4 December 2012 18:00, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

Currently we register authentication name as say '100' in sip registration, this comes to freeswitch and it will check in our DB for 100 and if its present then registrations would be successful.

freeswitch at internal> show registrations
reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
100,fsfailover.uk01.com<http://fsfailover.uk01.com>,e4969067f9a8c098,sofia/internal/sip:100 at 192.168.2.234:5060;transport=udp,1354638871,192.168.2.234,5060,udp,squay-laptop-1.squay.com<http://squay-laptop-1.squay.com>

I want to change this 100 to some e-mail address, so instead of 100 it will be something like 'ana at gmail.com<mailto:ana at gmail.com>'. Can we do this? While coming to freeswitch whether there would be any issues?

Regards,
Archana


_________________________________________________________________________
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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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

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
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--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

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From oej at edvina.net  Fri Dec  7 10:08:40 2012
From: oej at edvina.net (Olle E. Johansson)
Date: Fri, 7 Dec 2012 08:08:40 +0100
Subject: [Freeswitch-users] how to Ignore DTMF shorter than Fixed
	duration ?
In-Reply-To: <CAKzWOxUoXCMFOb-P8ZxOVJzaK4=o5Kxt=dF+EnMHEt_cu2qQqQ@mail.gmail.com>
References: <CAHWtr1TbtvoxrxbCEY6y8r_rKgtsh2zHhXmeZYRA0+6P0dKGUQ@mail.gmail.com>
	<CAKzWOxXQSj9hPSoSmH9pQ19J=+eU5qM+8aNO51BoWzzN_gKj_w@mail.gmail.com>
	<CAHWtr1SfKpkXL7BeB0995sjN9k8yLwKEXRcxkXvZyvQfqqEjUg@mail.gmail.com>
	<CAKzWOxUoXCMFOb-P8ZxOVJzaK4=o5Kxt=dF+EnMHEt_cu2qQqQ@mail.gmail.com>
Message-ID: <A9CAED77-5449-4C93-95DB-7C742F68CE06@edvina.net>


6 dec 2012 kl. 22:52 skrev Michael Collins <msc at freeswitch.org>:

> Bummer. I'm not aware of any way to ignore those. Anyone else have ideas?

The problem with ignoring DTMF in RTP is that it will delay audio, always. Let's say you want to delete all DTMF shorter than 1200 ms.
We receive a DTMF begin packet, then will have to wait to 1220 ms to see if the DTMF CONTINUE packets continue or if we actually get a DTMF END packet arriving in time. If it does, we'll delete the DTMF - but that
will produce a gap in the audio we will have to fill. If it is longer than 1200 ms we will start playing it out on the other side of the call bridge.

You seriously do not want to delay audio 1200 ms. Not even 100 ms.

There no simple solution to this. We've had issues with GENBAND servers and their DSPs that send short DTMF in calls with
female voices. 

In short. When we receive the start of the DTMF we have no idea about the duration. We can't make a decision until it's too late to make a decision... If you just focus on delaying DTMF, you will start playing DTMF on top of audio, cancelling out parts of the conversation.

/O

> -MC
> 
> On Thu, Dec 6, 2012 at 12:59 AM, Regis M <regis.freeswitch.org at tornad.net> wrote:
> Hi,
> 
> It's RFC2833 DTMF, not inband.
> 
> Regards,
> 
> 
> 2012/12/5 Michael Collins <msc at freeswitch.org>
> Are they sending those digits inband or with RFC2833? 
> -MC
> 
> On Wed, Dec 5, 2012 at 11:28 AM, Regis M <regis.freeswitch.org at tornad.net> wrote:
> Hi,
> 
> We have a bugged provider that send us throw RTP wrong DTMF during call. It seems that wrong DTMF are shorter than 1200ms (about  90%) so I want to try to ignore them.
> For the moment, FS catch them and send it back to the bridged side, boring user.
> 
> switch_rtp.c:3410 RTP RECV DTMF C:552
> 
> 
> 
> Does min-dtmf-duration will make FS ignore them and not RECV them ?
> I saw the DTMF in "normal" RTP packet, but no digit was pressed in user side.
> 
> It's commented in my FS switch.conf.xml, how can I see the current value ?
> <!--<param name="min-dtmf-duration" value="400"/>-->
> 
> 
> Any other idea ?
> 
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 asilva at wirelessmundi.com  Fri Dec  7 14:10:05 2012
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Fri, 07 Dec 2012 12:10:05 +0100
Subject: [Freeswitch-users] Compiling FSComm
Message-ID: <1354878605.5967.44.camel@vmmarces.vm.marces.com>

Hi,

i'm trying to compile fscomm but i have the following errors:

"
freeswitch-git/fscomm# qmake
freeswitch-git/fscomm# make
/usr/bin/uic-qt4 mainwindow.ui -o ui_mainwindow.h
/usr/bin/uic-qt4 preferences/prefdialog.ui -o ui_prefdialog.h
/usr/bin/uic-qt4 preferences/accountdialog.ui -o ui_accountdialog.h
/usr/bin/uic-qt4 widgets/codecwidget.ui -o ui_codecwidget.h
/usr/bin/uic-qt4 debugtools/consolewindow.ui -o ui_consolewindow.h
/usr/bin/uic-qt4 debugtools/statedebugdialog.ui -o ui_statedebugdialog.h
Warning: name layoutWidget is already used
Warning: name layoutWidget is already used
g++ -c -pipe -O2 -Wall -W -D_REENTRANT -DQT_NO_DEBUG -DQT_XML_LIB
-DQT_GUI_LIB -DQT_CORE_LIB -DQT_SHARED -I/usr/share/qt4/mkspecs/linux-g
++ -I. -I/usr/include/qt4/QtCore -I/usr/include/qt4/QtGui
-I/usr/include/qt4/QtXml -I/usr/include/qt4 -I../src/include
-I../libs/apr/include -I../libs/libteletone/src -I. -I. -o main.o
main.cpp
In file included from mainwindow.h:38,
                 from main.cpp:32:
../src/include/switch.h:110:18: error: stfu.h: No such file or directory
In file included from ../src/include/switch.h:121,
                 from mainwindow.h:38,
                 from main.cpp:32:
../src/include/switch_core.h:752: error: expected constructor,
destructor, or type conversion before ?*? token
In file included from ../src/include/switch_loadable_module.h:46,
                 from ../src/include/switch.h:122,
                 from mainwindow.h:38,
                 from main.cpp:32:
../src/include/switch_module_interfaces.h:121: error: expected
initializer before ?*? token
../src/include/switch_module_interfaces.h:162: error:
?switch_io_get_jb_t? does not name a type
In file included from ../src/include/switch.h:134,
                 from mainwindow.h:38,
                 from main.cpp:32:
../src/include/switch_rtp.h:243: error: expected constructor,
destructor, or type conversion before ?*? token
./fshost.h:43: warning: ?void eventHandlerCallback(switch_event_t*)?
declared ?static? but never defined
./fshost.h:44: warning: ?switch_status_t loggerHandler(const
switch_log_node_t*, switch_log_level_t)? declared ?static? but never
defined
make: *** [main.o] Error 1
"

watching this error: 
"../src/include/switch.h:110:18: error: stfu.h: No such file or
directory"

i manually change switch.h to fix the problem with the include by adding
the following:

"
diff --git a/src/include/switch.h b/src/include/switch.h
index c7ea7b0..2847112 100644
--- a/src/include/switch.h
+++ b/src/include/switch.h
@@ -107,7 +107,8 @@
 #include <errno.h>
 
 #ifndef WIN32
-#include "stfu.h"
+/* #include "stfu.h" */
+#include "../../../libs/stfu/stfu.h"
 #else
 #include "../../../libs/stfu/stfu.h"
 #endif

"

I could compile fscomm, but now i can't start it... i have the following
error:
"
Initializing core... 
Failed to initialize FreeSWITCH's core: Cannot Open log directory or XML
Root!
Everything OK, Entering runtime loop ... 
Segmentation fault
"
i had try the fix in the wiki: "chmod  644
~/.fscomm/conf/freeswitch.xml", and even "chmod -R 777 ~/.fscomm",  but
no luck...

Can you help me to go further...?

I'm trying it to install on a debian squeeze, i installed qt4-dev-tools.
Is it possible to install in debian squeeze our i should just  give
up... and try another distro?
The freeswitch-git is the lasted head. 

 
Thanks,
Ant?nio
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From regis.freeswitch.org at tornad.net  Fri Dec  7 15:59:02 2012
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Fri, 7 Dec 2012 13:59:02 +0100
Subject: [Freeswitch-users] how to Ignore DTMF shorter than Fixed
	duration ?
In-Reply-To: <A9CAED77-5449-4C93-95DB-7C742F68CE06@edvina.net>
References: <CAHWtr1TbtvoxrxbCEY6y8r_rKgtsh2zHhXmeZYRA0+6P0dKGUQ@mail.gmail.com>
	<CAKzWOxXQSj9hPSoSmH9pQ19J=+eU5qM+8aNO51BoWzzN_gKj_w@mail.gmail.com>
	<CAHWtr1SfKpkXL7BeB0995sjN9k8yLwKEXRcxkXvZyvQfqqEjUg@mail.gmail.com>
	<CAKzWOxUoXCMFOb-P8ZxOVJzaK4=o5Kxt=dF+EnMHEt_cu2qQqQ@mail.gmail.com>
	<A9CAED77-5449-4C93-95DB-7C742F68CE06@edvina.net>
Message-ID: <CAHWtr1Sr=ks6QB+EVsK=0Xj0+ppLCiNhQmSAcds67kCAdff22Q@mail.gmail.com>

Thanks.
But does FS start replaying it on other leg at the end or while it receive
info.
I had begin to reduce impact by binding digit action, but It seems the
problem is still here.

I know if i need dtmf or not in some calls, so, any other way to ignore ALL
dtmf ? :)
Regards


2012/12/7 Olle E. Johansson <oej at edvina.net>

>
> 6 dec 2012 kl. 22:52 skrev Michael Collins <msc at freeswitch.org>:
>
> Bummer. I'm not aware of any way to ignore those. Anyone else have ideas?
>
>
> The problem with ignoring DTMF in RTP is that it will delay audio, always.
> Let's say you want to delete all DTMF shorter than 1200 ms.
> We receive a DTMF begin packet, then will have to wait to 1220 ms to see
> if the DTMF CONTINUE packets continue or if we actually get a DTMF END
> packet arriving in time. If it does, we'll delete the DTMF - but that
> will produce a gap in the audio we will have to fill. If it is longer than
> 1200 ms we will start playing it out on the other side of the call bridge.
>
> You seriously do not want to delay audio 1200 ms. Not even 100 ms.
>
> There no simple solution to this. We've had issues with GENBAND servers
> and their DSPs that send short DTMF in calls with
> female voices.
>
> In short. When we receive the start of the DTMF we have no idea about the
> duration. We can't make a decision until it's too late to make a
> decision... If you just focus on delaying DTMF, you will start playing DTMF
> on top of audio, cancelling out parts of the conversation.
>
> /O
>
> -MC
>
> On Thu, Dec 6, 2012 at 12:59 AM, Regis M <regis.freeswitch.org at tornad.net>wrote:
>
>> Hi,
>>
>> It's RFC2833 DTMF, not inband.
>>
>> Regards,
>>
>>
>> 2012/12/5 Michael Collins <msc at freeswitch.org>
>>
>>> Are they sending those digits inband or with RFC2833?
>>> -MC
>>>
>>> On Wed, Dec 5, 2012 at 11:28 AM, Regis M <
>>> regis.freeswitch.org at tornad.net> wrote:
>>>
>>>> Hi,
>>>>
>>>> We have a bugged provider that send us throw RTP wrong DTMF during
>>>> call. It seems that wrong DTMF are shorter than 1200ms (about 90%) so I
>>>> want to try to ignore them.
>>>> For the moment, FS catch them and send it back to the bridged side,
>>>> boring user.
>>>>
>>>> switch_rtp.c:3410 RTP RECV DTMF C:552
>>>>
>>>>
>>>>
>>>> Does min-dtmf-duration will make FS ignore them and not RECV them ?
>>>> I saw the DTMF in "normal" RTP packet, but no digit was pressed in user
>>>> side.
>>>>
>>>> It's commented in my FS switch.conf.xml, how can I see the current
>>>> value ?
>>>> <!--<param name="min-dtmf-duration" value="400"/>-->
>>>>
>>>>
>>>> Any other idea ?
>>>>
>>>> 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
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Michael S Collins
>>> Twitter: @mercutioviz
>>> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
>>> http://www.ClueCon.com <http://www.cluecon.com/>
>>> http://www.OSTAG.org <http://www.ostag.org/>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
> http://www.ClueCon.com <http://www.cluecon.com/>
> http://www.OSTAG.org <http://www.ostag.org/>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From adrottenberg at gmail.com  Fri Dec  7 18:10:37 2012
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Fri, 7 Dec 2012 10:10:37 -0500
Subject: [Freeswitch-users] FreeSWITCH Conference Call Subject: Windows
 debugging tools
In-Reply-To: <CAKzWOxVKEBkaACC2q=SV9izD=ymaoUcTQxM0CyvCKiWTNUxSLg@mail.gmail.com>
References: <CAKzWOxVKEBkaACC2q=SV9izD=ymaoUcTQxM0CyvCKiWTNUxSLg@mail.gmail.com>
Message-ID: <CAC4JBEBOtebHJ-vBPPUvCxfvkbveg0hM3xdPusUmVa4W2cZ7Vw@mail.gmail.com>

I would be interested. I have 2 FS servers running on windows.

Thank You,
Duvid Rottenberg

On Wed, Dec 5, 2012 at 1:06 PM, Michael Collins <msc at freeswitch.org> wrote:

> nd such in a Windows environment. If you are interested in hearing about
> this subject please respond. (Only respond if you are interested - we don't
> need to hear f
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From steveayre at gmail.com  Fri Dec  7 18:25:28 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 7 Dec 2012 15:25:28 +0000
Subject: [Freeswitch-users] how to Ignore DTMF shorter than Fixed
	duration ?
In-Reply-To: <CAHWtr1Sr=ks6QB+EVsK=0Xj0+ppLCiNhQmSAcds67kCAdff22Q@mail.gmail.com>
References: <CAHWtr1TbtvoxrxbCEY6y8r_rKgtsh2zHhXmeZYRA0+6P0dKGUQ@mail.gmail.com>
	<CAKzWOxXQSj9hPSoSmH9pQ19J=+eU5qM+8aNO51BoWzzN_gKj_w@mail.gmail.com>
	<CAHWtr1SfKpkXL7BeB0995sjN9k8yLwKEXRcxkXvZyvQfqqEjUg@mail.gmail.com>
	<CAKzWOxUoXCMFOb-P8ZxOVJzaK4=o5Kxt=dF+EnMHEt_cu2qQqQ@mail.gmail.com>
	<A9CAED77-5449-4C93-95DB-7C742F68CE06@edvina.net>
	<CAHWtr1Sr=ks6QB+EVsK=0Xj0+ppLCiNhQmSAcds67kCAdff22Q@mail.gmail.com>
Message-ID: <CAFiqYunsXVM1cPW5dbzPa0=NoRLBskwHXkFDvh=Cj7RAFfia3Q@mail.gmail.com>

For the reasons RFC2833 Olle already mentioned, on a bridge FS *must* forward
the RFC2833 onto the other leg before it has received the end packet. If FS
waited for the end packet before deciding to forward the RTP then you'd get
a 1200ms+ lag on audio.


On 7 December 2012 12:59, Regis M <regis.freeswitch.org at tornad.net> wrote:

> Thanks.
> But does FS start replaying it on other leg at the end or while it receive
> info.
> I had begin to reduce impact by binding digit action, but It seems the
> problem is still here.
>
> I know if i need dtmf or not in some calls, so, any other way to ignore
> ALL dtmf ? :)
> Regards
>
>
> 2012/12/7 Olle E. Johansson <oej at edvina.net>
>
>
>> 6 dec 2012 kl. 22:52 skrev Michael Collins <msc at freeswitch.org>:
>>
>> Bummer. I'm not aware of any way to ignore those. Anyone else have ideas?
>>
>>
>> The problem with ignoring DTMF in RTP is that it will delay audio,
>> always. Let's say you want to delete all DTMF shorter than 1200 ms.
>> We receive a DTMF begin packet, then will have to wait to 1220 ms to see
>> if the DTMF CONTINUE packets continue or if we actually get a DTMF END
>> packet arriving in time. If it does, we'll delete the DTMF - but that
>> will produce a gap in the audio we will have to fill. If it is longer
>> than 1200 ms we will start playing it out on the other side of the call
>> bridge.
>>
>> You seriously do not want to delay audio 1200 ms. Not even 100 ms.
>>
>> There no simple solution to this. We've had issues with GENBAND servers
>> and their DSPs that send short DTMF in calls with
>> female voices.
>>
>> In short. When we receive the start of the DTMF we have no idea about the
>> duration. We can't make a decision until it's too late to make a
>> decision... If you just focus on delaying DTMF, you will start playing DTMF
>> on top of audio, cancelling out parts of the conversation.
>>
>> /O
>>
>> -MC
>>
>> On Thu, Dec 6, 2012 at 12:59 AM, Regis M <regis.freeswitch.org at tornad.net
>> > wrote:
>>
>>> Hi,
>>>
>>> It's RFC2833 DTMF, not inband.
>>>
>>> Regards,
>>>
>>>
>>> 2012/12/5 Michael Collins <msc at freeswitch.org>
>>>
>>>> Are they sending those digits inband or with RFC2833?
>>>> -MC
>>>>
>>>> On Wed, Dec 5, 2012 at 11:28 AM, Regis M <
>>>> regis.freeswitch.org at tornad.net> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> We have a bugged provider that send us throw RTP wrong DTMF during
>>>>> call. It seems that wrong DTMF are shorter than 1200ms (about 90%) so I
>>>>> want to try to ignore them.
>>>>> For the moment, FS catch them and send it back to the bridged side,
>>>>> boring user.
>>>>>
>>>>> switch_rtp.c:3410 RTP RECV DTMF C:552
>>>>>
>>>>>
>>>>>
>>>>> Does min-dtmf-duration will make FS ignore them and not RECV them ?
>>>>> I saw the DTMF in "normal" RTP packet, but no digit was pressed in
>>>>> user side.
>>>>>
>>>>> It's commented in my FS switch.conf.xml, how can I see the current
>>>>> value ?
>>>>> <!--<param name="min-dtmf-duration" value="400"/>-->
>>>>>
>>>>>
>>>>> Any other idea ?
>>>>>
>>>>> 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
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Michael S Collins
>>>> Twitter: @mercutioviz
>>>> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
>>>> http://www.ClueCon.com <http://www.cluecon.com/>
>>>> http://www.OSTAG.org <http://www.ostag.org/>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
>> http://www.ClueCon.com <http://www.cluecon.com/>
>> http://www.OSTAG.org <http://www.ostag.org/>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 Hector.Geraldino at ipsoft.com  Fri Dec  7 19:03:46 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Fri, 7 Dec 2012 16:03:46 +0000
Subject: [Freeswitch-users] having problem using java with esl library
In-Reply-To: <CAB3HJytM41noZgqhqsa492MC+nNEc4S7AvWmMi+8X4UcOdZA_w@mail.gmail.com>
References: <CAB3HJytM41noZgqhqsa492MC+nNEc4S7AvWmMi+8X4UcOdZA_w@mail.gmail.com>
Message-ID: <ADF32A4A0F985A428C85B277B8E1C9D80F4924EE@NY1-EXMB-02.ip-soft.net>

I haven't used the java esl library, I'm more familiar with the Java ESL Client [http://wiki.freeswitch.org/wiki/Java_ESL_Client] which I highly recommend.

As you're just starting, I think you should take a look on this java implementation, which has no dependencies on the native FreeSWITCH library and relies on the netty java.nio implementation.

Give it a try and let us know how it goes!

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of bawajan
Sent: Thursday, December 06, 2012 5:30 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] having problem using java with esl library

I am new to freeswitch and am using java app with esl library to make outbound calls and am getting the error is "Create additional event dispatch thread 2".

After that in the freeswitch logs its printing - Event 0 Blocking and its keeps printing.

2012-12-05 13:46:02.179956 [WARNING] switch_event.c:607 Create additional event dispatch thread 2
2012-12-05 13:46:19.919951 [WARNING] switch_event.c:607 Create additional event dispatch thread 3
2012-12-05 13:46:46.159954 [WARNING] switch_event.c:607 Create additional event dispatch thread 4
2012-12-05 13:47:03.679956 [WARNING] switch_event.c:607 Create additional event dispatch thread 5
2012-12-05 13:47:24.999950 [WARNING] switch_event.c:607 Create additional event dispatch thread 6
2012-12-05 13:47:40.299950 [WARNING] switch_event.c:607 Create additional event dispatch thread 7
2012-12-05 13:47:54.879950 [WARNING] switch_event.c:348 Event Thread 0 is blocking
2012-12-05 13:47:55.079953 [WARNING] switch_event.c:348 Event Thread 0 is blocking
2012-12-05 13:47:55.139957 [WARNING] switch_event.c:348 Event Thread 0 is blocking
2012-12-05 13:47:55.639950 [WARNING] switch_event.c:348 Event Thread 0 is blocking
2012-12-05 13:47:55.819960 [WARNING] switch_event.c:348 Event Thread 0 is blocking
2012-12-05 13:47:55.859950 [WARNING] switch_event.c:348 Event Thread 0 is blocking
2012-12-05 13:47:55.939953 [WARNING] switch_event.c:348 Event Thread 0 is blocking



Kindly let me know where am doing mistake and how to resolve it.

Thanks in Advance,
Bawajan
________________________________
View this message in context: having problem using java with esl library<http://freeswitch-users.2379917.n2.nabble.com/having-problem-using-java-with-esl-library-tp7585250.html>
Sent from the freeswitch-users mailing list archive<http://freeswitch-users.2379917.n2.nabble.com/> at Nabble.com.
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From cal.leeming at simplicitymedialtd.co.uk  Fri Dec  7 19:23:30 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Fri, 7 Dec 2012 16:23:30 +0000
Subject: [Freeswitch-users] IVR menu delay - 1 second between events
Message-ID: <CAHKQagF3C76QiybFNrS7GCdiGqaTbtPpQzD4wFivtJ+EK1CirQ@mail.gmail.com>

Hello,

There appears to be a delay of 1 second between menu events within an IVR
(i.e. play long, invalid option, play short).

We have the following IVR menu;

          <menu name="ivr.2"
              greet-long="phrase:demo_ivr_main_menu"
              greet-short="phrase:demo_ivr_main_menu"

              exit-sound="phrase:demo_ivr_main_menu"
              confirm-macro=""
              confirm-key=""
              confirm-attempts="3"
              timeout="10000"
              inter-digit-timeout="10"
              max-failures="3"
              max-timeouts="3"
              digit-len="1">

In the above example, any option will result in an invalid selection being
met.

However, FS seems to take its sweet time getting from the point of a user
pushing a button, to actually playing the next audio.

freeswitch at vded213> 2012-12-07 16:17:21.752141 [DEBUG] switch_rtp.c:3797
RTP RECV DTMF 2:2240
2012-12-07 16:17:21.752141 [DEBUG] switch_ivr_play_say.c:1682 done playing
file
/usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.wav
2012-12-07 16:17:21.872105 [DEBUG] switch_ivr_menu.c:395 digits '2'
2012-12-07 16:17:21.872105 [DEBUG] switch_ivr_menu.c:579 IVR menu 'ivr.2'
caught invalid input '2'
2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [en]
2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en)
2012-12-07 16:17:22.872155 [DEBUG] switch_core_file.c:180 File
/usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.wav
sample rate 48000 doesn't match requested rate 8000
2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms

In the above, you can see the IVR knew the menu was invalid at
"16:17:21.872105", but didn't start playing the next audio until
"16:17:22.872155".

The difference between the two is almost exactly 1 second, which makes me
think maybe there's a config option somewhere that isn't
documented/obvious, or a hard coded delay.

Any advice on this would be much appreciated, slow/laggy menus are a pet
hate of mine :/

Thanks

Cal
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From grcamauer at gmail.com  Fri Dec  7 19:29:22 2012
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Fri, 7 Dec 2012 13:29:22 -0300
Subject: [Freeswitch-users] IVR menu delay - 1 second between events
In-Reply-To: <CAHKQagF3C76QiybFNrS7GCdiGqaTbtPpQzD4wFivtJ+EK1CirQ@mail.gmail.com>
References: <CAHKQagF3C76QiybFNrS7GCdiGqaTbtPpQzD4wFivtJ+EK1CirQ@mail.gmail.com>
Message-ID: <CAPNTnnrVW4Vpj4LcaNyeQrcmYq99DDF26E3UXpFoJyf=-q11fg@mail.gmail.com>

Are you sure it's not timing out?  Change the timeout="10000" to
timeout="30000" and see if your lag is now 3 seconds.




On Fri, Dec 7, 2012 at 1:23 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Hello,
>
> There appears to be a delay of 1 second between menu events within an IVR
> (i.e. play long, invalid option, play short).
>
> We have the following IVR menu;
>
>           <menu name="ivr.2"
>               greet-long="phrase:demo_ivr_main_menu"
>               greet-short="phrase:demo_ivr_main_menu"
>
>               exit-sound="phrase:demo_ivr_main_menu"
>               confirm-macro=""
>               confirm-key=""
>               confirm-attempts="3"
>               timeout="10000"
>               inter-digit-timeout="10"
>               max-failures="3"
>               max-timeouts="3"
>               digit-len="1">
>
> In the above example, any option will result in an invalid selection being
> met.
>
> However, FS seems to take its sweet time getting from the point of a user
> pushing a button, to actually playing the next audio.
>
> freeswitch at vded213> 2012-12-07 16:17:21.752141 [DEBUG] switch_rtp.c:3797
> RTP RECV DTMF 2:2240
> 2012-12-07 16:17:21.752141 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file
> /usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.wav
> 2012-12-07 16:17:21.872105 [DEBUG] switch_ivr_menu.c:395 digits '2'
> 2012-12-07 16:17:21.872105 [DEBUG] switch_ivr_menu.c:579 IVR menu 'ivr.2'
> caught invalid input '2'
> 2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]
> 2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en)
> 2012-12-07 16:17:22.872155 [DEBUG] switch_core_file.c:180 File
> /usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.wav
> sample rate 48000 doesn't match requested rate 8000
> 2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> In the above, you can see the IVR knew the menu was invalid at
> "16:17:21.872105", but didn't start playing the next audio until
> "16:17:22.872155".
>
> The difference between the two is almost exactly 1 second, which makes me
> think maybe there's a config option somewhere that isn't
> documented/obvious, or a hard coded delay.
>
> Any advice on this would be much appreciated, slow/laggy menus are a pet
> hate of mine :/
>
> Thanks
>
> Cal
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Guillermo Ruiz Camauer
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From sdevoy at bizfocused.com  Fri Dec  7 19:39:36 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 7 Dec 2012 11:39:36 -0500
Subject: [Freeswitch-users] Polycom IP 335 - 401 Unauthorized .... NAT ISSUE?
Message-ID: <1dc901cdd499$721ed340$565c79c0$@bizfocused.com>

HI All,

 

I am still banging my head against the wall here try to get a Polycom 335 to
register w/ FS.  I have checked all the SERVER and USER/AUTH fields like
1000 times and 900 variations.  I think my problem may be NAT related.  I
know on my CISCO 504G I had to enable several NAT features to work behind
our firewall.  I am totally new to Polycom, so some very basic help is
needed.

 

The server is remote but not behind a NAT there.  The phones are NAT'ed to
the internet.  In the sofia sip trace I see this over and over:

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

recv 552 bytes from udp/[71.127.152.57]:1026 at 16:26:07.358892:

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

   REGISTER sip:fs_bfis.bizfocused.com SIP/2.0

   Via: SIP/2.0/UDP 10.10.40.47:5060;branch=z9hG4bKbf81dbdc8E687A5

   From: "228 Sean" <sip:228 at fs_bfis.bizfocused.com>;tag=3F42C046-B61A297

   To: <sip:228 at fs_bfis.bizfocused.com>

   CSeq: 1 REGISTER

   Call-ID: 2f482c2-2599cc43-1fb1a78 at 10.10.40.47

   Contact: <sip:228 at 10.10.40.47:5060>;methods="INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"

   User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069

   Accept-Language: en

   Max-Forwards: 70

   Expires: 600

   Content-Length: 0

 

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

send 710 bytes to udp/[71.127.152.57]:5060 at 16:26:07.359067:

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

   SIP/2.0 401 Unauthorized

   Via: SIP/2.0/UDP
10.10.40.47:5060;branch=z9hG4bKbf81dbdc8E687A5;received=71.127.152.57

  From: "228 Sean" <sip:228 at fs_bfis.bizfocused.com>;tag=3F42C046-B61A297

   To: <sip:228 at fs_bfis.bizfocused.com>;tag=t232me1NSD02S

   Call-ID: 2f482c2-2599cc43-1fb1a78 at 10.10.40.47

   CSeq: 1 REGISTER

   User-Agent:
FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120
712T101002Z

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: precondition, path, replaces

   WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com",
nonce="b9583359-0163-4bf2-9818-788f64c34207", algorithm=MD5, qop="auth"

   Content-Length: 0

If I understand correctly, the server should be sending back this 401
message with the nonce so the phone can re-attempt the registration with an
encrypted password.  If NAT is failing, the phone is never seeing the 401 w/
the nonce.

 

So what do I do in the WEB config interface to enable NAT on this phone?

 

Thanks,

Sean 

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From drk at drkngs.net  Fri Dec  7 20:55:32 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Fri, 07 Dec 2012 09:55:32 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Subject: Windows
 debugging tools
In-Reply-To: <CAC4JBEBOtebHJ-vBPPUvCxfvkbveg0hM3xdPusUmVa4W2cZ7Vw@mail.gmail.com>
Message-ID: <20121207175532.e5d77613@mail.tritonwest.net>

If there's not enough interest to do this on a weekly meeting, I would be willing to do it for a smaller group, out of band.  
   
--Dave
      _____  

  From: Duvid Rottenberg [mailto:adrottenberg at gmail.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Fri, 07 Dec 2012 07:10:37 -0800
Subject: Re: [Freeswitch-users] FreeSWITCH Conference Call Subject: Windows debugging tools

  
I would be interested. I have 2 FS servers running on windows.  

  
Thank You,  
Duvid Rottenberg

  
On Wed, Dec 5, 2012 at 1:06 PM, Michael Collins <msc at freeswitch.org> wrote:
  nd such in a Windows environment. If you are interested in hearing about this subject please respond. (Only respond if you are interested - we don't need to hear f
      
   
 
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From a.venugopan at mundio.com  Fri Dec  7 21:04:37 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Fri, 7 Dec 2012 18:04:37 +0000
Subject: [Freeswitch-users] sofia_reg.c
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233A2D2@Mail-Kilo.squay.com>

Hi,

In sofia_reg.c I find variables like 'sip_auth_username','sip_auth_password',etc. Can anyone please tell me from where it will get value for all these variables?
Is it from some query in lua script or is it from any sip URL?

Regards,
Archana

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From talk2ram at gmail.com  Fri Dec  7 21:10:23 2012
From: talk2ram at gmail.com (ram)
Date: Fri, 7 Dec 2012 23:40:23 +0530
Subject: [Freeswitch-users] sofia_reg.c
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233A2D2@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233A2D2@Mail-Kilo.squay.com>
Message-ID: <CA+XvL-Nnu+de=YscRGUz3+RU4q6RTrG5X07suBsAvW-fFKToxQ@mail.gmail.com>

Hi

hope this URL help you

http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration




On Fri, Dec 7, 2012 at 11:34 PM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi,****
>
> ** **
>
> In sofia_reg.c I find variables like
> ?sip_auth_username?,?sip_auth_password?,etc. Can anyone please tell me from
> where it will get value for all these variables?****
>
> Is it from some query in lua script or is it from any sip URL?****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 steveayre at gmail.com  Fri Dec  7 21:15:58 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 7 Dec 2012 18:15:58 +0000
Subject: [Freeswitch-users] sofia_reg.c
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233A2D2@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233A2D2@Mail-Kilo.squay.com>
Message-ID: <CAFiqYu=9pLsxG0Ji_EDu-47tmfVDwSfWa1FZamqcs99G_qyDJA@mail.gmail.com>

They're just normal channel variables
http://wiki.freeswitch.org/wiki/Channel_Variables#sip_auth_username
http://wiki.freeswitch.org/wiki/Channel_Variables#sip_auth_password

When set on an outbound leg they'll be used for authentication. You don't
need them if you're dialing a sofia/gateway/ prefix.

There're multiple ways to set them... you can set them on the aleg to
export to the bleg, or put them within the dialstring - for example:
<action application="bridge"
data="[sip_auth_username=user1,sip_auth_password=pass123]sofia/profile/1234 at a.b.c.d
"/>

How you actually get the value to assign is going to be down to your
specific setup. It could be static in dialplan, or fetched from somewhere
else (eg lua, db etc).





On 7 December 2012 18:04, Archana Venugopan <a.venugopan at mundio.com> wrote:

>  Hi,****
>
> ** **
>
> In sofia_reg.c I find variables like
> ?sip_auth_username?,?sip_auth_password?,etc. Can anyone please tell me from
> where it will get value for all these variables?****
>
> Is it from some query in lua script or is it from any sip URL?****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 a.venugopan at mundio.com  Fri Dec  7 21:19:27 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Fri, 7 Dec 2012 18:19:27 +0000
Subject: [Freeswitch-users] sofia_reg.c
In-Reply-To: <CA+XvL-Nnu+de=YscRGUz3+RU4q6RTrG5X07suBsAvW-fFKToxQ@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF233A2D2@Mail-Kilo.squay.com>
	<CA+XvL-Nnu+de=YscRGUz3+RU4q6RTrG5X07suBsAvW-fFKToxQ@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233A2F3@Mail-Kilo.squay.com>

Hi,
Thanks. Yes I checked this link and which says like the values are fetched based on the domain name from sip_profiles.
But if that is the case if I change the user_id to some name based on domain, its not picking up sip_auth_username and throwing error!! Correct me please if my understanding is wrong.


Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ram
Sent: 07 December 2012 18:10
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sofia_reg.c

Hi

hope this URL help you

http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration



On Fri, Dec 7, 2012 at 11:34 PM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

In sofia_reg.c I find variables like 'sip_auth_username','sip_auth_password',etc. Can anyone please tell me from where it will get value for all these variables?
Is it from some query in lua script or is it from any sip URL?

Regards,
Archana


_________________________________________________________________________
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
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From steveayre at gmail.com  Fri Dec  7 21:20:54 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 7 Dec 2012 18:20:54 +0000
Subject: [Freeswitch-users] Polycom IP 335 - 401 Unauthorized .... NAT
	ISSUE?
In-Reply-To: <1dc901cdd499$721ed340$565c79c0$@bizfocused.com>
References: <1dc901cdd499$721ed340$565c79c0$@bizfocused.com>
Message-ID: <CAFiqYu=wDhL2+SDcNmMJ+AjJ40prrmyvN56WQ6Ab4c30DeUgdQ@mail.gmail.com>

Try this parameter:
http://wiki.freeswitch.org/wiki/NDLB#NDLB-force-rport

or if that fails
http://wiki.freeswitch.org/wiki/NDLB#NDLB-connectile-dysfunction


On 7 December 2012 16:39, Sean Devoy <sdevoy at bizfocused.com> wrote:

> HI All,****
>
> ** **
>
> I am still banging my head against the wall here try to get a Polycom 335
> to register w/ FS.  I have checked all the SERVER and USER/AUTH fields like
> 1000 times and 900 variations.  I think my problem may be NAT related.  I
> know on my CISCO 504G I had to enable several NAT features to work behind
> our firewall.  I am totally new to Polycom, so some very basic help is
> needed.****
>
> ** **
>
> The server is remote but not behind a NAT there.  The phones are NAT?ed to
> the internet.  In the sofia sip trace I see this over and over:****
>
>    ------------------------------------------------------------------------
> ****
>
> recv 552 bytes from udp/[71.127.152.57]:1026 at 16:26:07.358892:****
>
>    ------------------------------------------------------------------------
> ****
>
>    REGISTER sip:fs_bfis.bizfocused.com SIP/2.0****
>
>    Via: SIP/2.0/UDP 10.10.40.47:5060;branch=z9hG4bKbf81dbdc8E687A5****
>
>    From: "228 Sean" <sip:228 at fs_bfis.bizfocused.com>;tag=3F42C046-B61A297*
> ***
>
>    To: <sip:228 at fs_bfis.bizfocused.com>****
>
>    CSeq: 1 REGISTER****
>
>    Call-ID: 2f482c2-2599cc43-1fb1a78 at 10.10.40.47****
>
>    Contact: <sip:228 at 10.10.40.47:5060>;methods="INVITE, ACK, BYE, CANCEL,
> OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"****
>
>    User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069****
>
>    Accept-Language: en****
>
>    Max-Forwards: 70****
>
>    Expires: 600****
>
>    Content-Length: 0****
>
> ** **
>
>    ------------------------------------------------------------------------
> ****
>
> send 710 bytes to udp/[71.127.152.57]:5060 at 16:26:07.359067:****
>
>    ------------------------------------------------------------------------
> ****
>
>    SIP/2.0 401 Unauthorized****
>
>    Via: SIP/2.0/UDP 10.10.40.47:5060
> ;branch=z9hG4bKbf81dbdc8E687A5;received=71.127.152.57****
>
>   From: "228 Sean" <sip:228 at fs_bfis.bizfocused.com>;tag=3F42C046-B61A297**
> **
>
>    To: <sip:228 at fs_bfis.bizfocused.com>;tag=t232me1NSD02S****
>
>    Call-ID: 2f482c2-2599cc43-1fb1a78 at 10.10.40.47****
>
>    CSeq: 1 REGISTER****
>
>    User-Agent:
> FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
> ****
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE****
>
>    Supported: precondition, path, replaces****
>
>    WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com",
> nonce="b9583359-0163-4bf2-9818-788f64c34207", algorithm=MD5, qop="auth"***
> *
>
>    Content-Length: 0****
>
> ****
>
> If I understand correctly, the server should be sending back this 401
> message with the nonce so the phone can re-attempt the registration with an
> encrypted password.  If NAT is failing, the phone is never seeing the 401
> w/ the nonce.****
>
> ** **
>
> So what do I do in the WEB config interface to enable NAT on this phone?**
> **
>
> ** **
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Dec  7 21:54:28 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 7 Dec 2012 13:54:28 -0500
Subject: [Freeswitch-users] Polycom IP 335 - 401 Unauthorized .... NAT
	ISSUE?
In-Reply-To: <CAFiqYu=wDhL2+SDcNmMJ+AjJ40prrmyvN56WQ6Ab4c30DeUgdQ@mail.gmail.com>
References: <1dc901cdd499$721ed340$565c79c0$@bizfocused.com>
	<CAFiqYu=wDhL2+SDcNmMJ+AjJ40prrmyvN56WQ6Ab4c30DeUgdQ@mail.gmail.com>
Message-ID: <CAKV-jFxZ1ovBxejR-xnHouCx=BY3wCnWM83Z=qOr-zCf1fZm_A@mail.gmail.com>

Let's do this step by step. First of all, my server is off site. We are
going through a NAT in order to get to FreeSWITCH. Looks like this is the
same setup you have.

I have the same phone as you do:
Phone InformationPhone ModelSoundPoint IP 335Part Number2345-12375-001 Rev:AMAC
Address00:04:F2:37:3D:C0IP Address10.0.0.39UC Software
Version4.0.2.11307BootROM
Software Version5.0.2.12692

Alright so now that we have that squared away, the next step is to set up
the phone.

Settings > SIP

Local Settings:

Local SIP Port: 0
Calls Per Line Key: 4
New SDP Type: Disable
Live Communication Server Support: Disable
Non-Standard Line Seize: Enable
Digitmap: Not relevent
Digitmap Timeout: 3|3|3|3|3|3
Remove End-of-Dial Marker: Enable
Digit Impossible Match: 0

Outbound Proxy:

Address: <Blank>
Port: 0
Transport: DNSnaptr

Server 1:

Address: pbx.endigovoip.com
Port: 0
Transport: DNSnaptr (You shouldn't have issues with UDPonly, might be worth
trying though.
Espires (s): 3600
Register: Yes
Retry Timeout (ms): 0
Retry Maximum Count: 3
Line Seize Timeout: 30

I do not have a second server.

Settings > Network > NAT

NAT
*** IP Address*** Signalling Port*** Media Port StartKeep-Alive Interval (s)

Settings > Lines

 *Identification*
Display NameAddressAuthentication User IDAuthentication PasswordLabelType
Private  SharedThird Party NameNumber of Line KeysCalls Per LineRing TypeLow
TrillLow Double TrillMedium TrillMedium Double TrillHigh TrillHigh Double
TrillHighest TrillHighest Double TrillBeebleTripletRingback-styleLow Trill
PrecedenceRing Splash

 *Outbound Proxy*
AddressPortTransport                   UDPOnly
TCPpreferred                   DNSnaptr                   TCPonly
          TLS

 *Server 1*
AddressPortTransport                   UDPOnly
TCPpreferred                   DNSnaptr                   TCPonly
          TLS                 Expires (s)Register Yes  NoRetry Timeout
(ms)Retry
Maximum CountLine Seize Timeout (s)

 *Server 2*
AddressPortTransport                   UDPOnly
TCPpreferred                   DNSnaptr                   TCPonly
          TLS                 Expires (s)Register Yes  NoRetry Timeout
(ms)Retry
Maximum CountLine Seize Timeout (s)

 *Call Diversion*
*** Always Forward Enable  Disable*** Always Forward To Contact*** If Busy,
Forward Enable  Disable*** If Busy, Forward To Contact*** On No Answer,
Forward Enable  Disable*** On No Answer, Forward To Contact*** No Answer
Timeout (seconds)*** On Do Not Disturb, Forward Enable  Disable*** On Do
Not Disturb, Forward To Contact*** Disable Forward For Shared Lines Yes  No*
** Forward Specific Caller Enable  Disable

 *Message Center*
Subscription AddressCallback Mode                   Registration
       Contact                   Disabled                 Callback Contact


Check those and let us know where you stand after that.

-BDF

On Fri, Dec 7, 2012 at 1:20 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Try this parameter:
> http://wiki.freeswitch.org/wiki/NDLB#NDLB-force-rport
>
> or if that fails
> http://wiki.freeswitch.org/wiki/NDLB#NDLB-connectile-dysfunction
>
>
> On 7 December 2012 16:39, Sean Devoy <sdevoy at bizfocused.com> wrote:
>
>> HI All,****
>>
>> ** **
>>
>> I am still banging my head against the wall here try to get a Polycom 335
>> to register w/ FS.  I have checked all the SERVER and USER/AUTH fields like
>> 1000 times and 900 variations.  I think my problem may be NAT related.  I
>> know on my CISCO 504G I had to enable several NAT features to work behind
>> our firewall.  I am totally new to Polycom, so some very basic help is
>> needed.****
>>
>> ** **
>>
>> The server is remote but not behind a NAT there.  The phones are NAT?ed
>> to the internet.  In the sofia sip trace I see this over and over:****
>>
>>
>> ------------------------------------------------------------------------*
>> ***
>>
>> recv 552 bytes from udp/[71.127.152.57]:1026 at 16:26:07.358892:****
>>
>>
>> ------------------------------------------------------------------------*
>> ***
>>
>>    REGISTER sip:fs_bfis.bizfocused.com SIP/2.0****
>>
>>    Via: SIP/2.0/UDP 10.10.40.47:5060;branch=z9hG4bKbf81dbdc8E687A5****
>>
>>    From: "228 Sean" <sip:228 at fs_bfis.bizfocused.com>;tag=3F42C046-B61A297
>> ****
>>
>>    To: <sip:228 at fs_bfis.bizfocused.com>****
>>
>>    CSeq: 1 REGISTER****
>>
>>    Call-ID: 2f482c2-2599cc43-1fb1a78 at 10.10.40.47****
>>
>>    Contact: <sip:228 at 10.10.40.47:5060>;methods="INVITE, ACK, BYE,
>> CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"*
>> ***
>>
>>    User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069****
>>
>>    Accept-Language: en****
>>
>>    Max-Forwards: 70****
>>
>>    Expires: 600****
>>
>>    Content-Length: 0****
>>
>> ** **
>>
>>
>> ------------------------------------------------------------------------*
>> ***
>>
>> send 710 bytes to udp/[71.127.152.57]:5060 at 16:26:07.359067:****
>>
>>
>> ------------------------------------------------------------------------*
>> ***
>>
>>    SIP/2.0 401 Unauthorized****
>>
>>    Via: SIP/2.0/UDP 10.10.40.47:5060
>> ;branch=z9hG4bKbf81dbdc8E687A5;received=71.127.152.57****
>>
>>   From: "228 Sean" <sip:228 at fs_bfis.bizfocused.com>;tag=3F42C046-B61A297*
>> ***
>>
>>    To: <sip:228 at fs_bfis.bizfocused.com>;tag=t232me1NSD02S****
>>
>>    Call-ID: 2f482c2-2599cc43-1fb1a78 at 10.10.40.47****
>>
>>    CSeq: 1 REGISTER****
>>
>>    User-Agent:
>> FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
>> ****
>>
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE****
>>
>>    Supported: precondition, path, replaces****
>>
>>    WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com",
>> nonce="b9583359-0163-4bf2-9818-788f64c34207", algorithm=MD5, qop="auth"**
>> **
>>
>>    Content-Length: 0****
>>
>> ****
>>
>> If I understand correctly, the server should be sending back this 401
>> message with the nonce so the phone can re-attempt the registration with an
>> encrypted password.  If NAT is failing, the phone is never seeing the 401
>> w/ the nonce.****
>>
>> ** **
>>
>> So what do I do in the WEB config interface to enable NAT on this phone?*
>> ***
>>
>> ** **
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
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
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From bdfoster at endigotech.com  Fri Dec  7 22:11:23 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 7 Dec 2012 14:11:23 -0500
Subject: [Freeswitch-users] Polycom IP 335 - 401 Unauthorized .... NAT
	ISSUE?
In-Reply-To: <CAKV-jFxZ1ovBxejR-xnHouCx=BY3wCnWM83Z=qOr-zCf1fZm_A@mail.gmail.com>
References: <1dc901cdd499$721ed340$565c79c0$@bizfocused.com>
	<CAFiqYu=wDhL2+SDcNmMJ+AjJ40prrmyvN56WQ6Ab4c30DeUgdQ@mail.gmail.com>
	<CAKV-jFxZ1ovBxejR-xnHouCx=BY3wCnWM83Z=qOr-zCf1fZm_A@mail.gmail.com>
Message-ID: <CAKV-jFz8_pJOOOddmkY0_Gb_JrSbAwxYnQKkTmLg8wzkX2g2dA@mail.gmail.com>

This might help you:

http://pastebin.freeswitch.org/20301

Try comparing the SIP messages to yours. Notice that it tries to register
twice, the first one is Unauthorized.


On Fri, Dec 7, 2012 at 1:54 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> Let's do this step by step. First of all, my server is off site. We are
> going through a NAT in order to get to FreeSWITCH. Looks like this is the
> same setup you have.
>
> I have the same phone as you do:
>  Phone InformationPhone Model SoundPoint IP 335Part Number2345-12375-001
> Rev:A MAC Address00:04:F2:37:3D:C0 IP Address10.0.0.39UC Software Version
> 4.0.2.11307BootROM Software Version5.0.2.12692
>
> Alright so now that we have that squared away, the next step is to set up
> the phone.
>
> Settings > SIP
>
> Local Settings:
>
> Local SIP Port: 0
> Calls Per Line Key: 4
> New SDP Type: Disable
> Live Communication Server Support: Disable
> Non-Standard Line Seize: Enable
> Digitmap: Not relevent
> Digitmap Timeout: 3|3|3|3|3|3
> Remove End-of-Dial Marker: Enable
> Digit Impossible Match: 0
>
> Outbound Proxy:
>
> Address: <Blank>
> Port: 0
> Transport: DNSnaptr
>
> Server 1:
>
> Address: pbx.endigovoip.com
> Port: 0
> Transport: DNSnaptr (You shouldn't have issues with UDPonly, might be
> worth trying though.
> Espires (s): 3600
> Register: Yes
> Retry Timeout (ms): 0
> Retry Maximum Count: 3
> Line Seize Timeout: 30
>
> I do not have a second server.
>
> Settings > Network > NAT
>
> NAT
> *** IP Address *** Signalling Port *** Media Port Start Keep-Alive
> Interval (s)
>
> Settings > Lines
>
>  *Identification*
> Display Name Address Authentication User ID Authentication Password Label
> Type  Private  Shared Third Party Name Number of Line Keys Calls Per Line Ring
> TypeLow TrillLow Double TrillMedium TrillMedium Double TrillHigh TrillHigh
> Double TrillHighest TrillHighest Double TrillBeebleTripletRingback-styleLow
> Trill PrecedenceRing Splash
>
>  *Outbound Proxy*
> Address Port Transport                    UDPOnly
> TCPpreferred                   DNSnaptr                   TCPonly
>           TLS
>
>  *Server 1*
> Address Port Transport                   UDPOnly
> TCPpreferred                   DNSnaptr                   TCPonly
>           TLS                  Expires (s) Register Yes  No Retry Timeout
> (ms) Retry Maximum Count Line Seize Timeout (s)
>
>  *Server 2*
> Address Port Transport                   UDPOnly
> TCPpreferred                   DNSnaptr                   TCPonly
>           TLS                  Expires (s) Register Yes  No Retry Timeout
> (ms) Retry Maximum Count Line Seize Timeout (s)
>
>  *Call Diversion*
> *** Always Forward  Enable  Disable *** Always Forward To Contact *** If
> Busy, Forward Enable  Disable *** If Busy, Forward To Contact *** On No
> Answer, Forward  Enable  Disable *** On No Answer, Forward To Contact *** No
> Answer Timeout (seconds) *** On Do Not Disturb, Forward  Enable  Disable *
> ** On Do Not Disturb, Forward To Contact *** Disable Forward For Shared
> Lines  Yes  No *** Forward Specific Caller  Enable  Disable
>
>  *Message Center*
> Subscription Address Callback Mode                    Registration
>              Contact                   Disabled                  Callback
> Contact
>
>
> Check those and let us know where you stand after that.
>
> -BDF
>
> On Fri, Dec 7, 2012 at 1:20 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Try this parameter:
>> http://wiki.freeswitch.org/wiki/NDLB#NDLB-force-rport
>>
>> or if that fails
>> http://wiki.freeswitch.org/wiki/NDLB#NDLB-connectile-dysfunction
>>
>>
>> On 7 December 2012 16:39, Sean Devoy <sdevoy at bizfocused.com> wrote:
>>
>>> HI All,****
>>>
>>> ** **
>>>
>>> I am still banging my head against the wall here try to get a Polycom
>>> 335 to register w/ FS.  I have checked all the SERVER and USER/AUTH fields
>>> like 1000 times and 900 variations.  I think my problem may be NAT
>>> related.  I know on my CISCO 504G I had to enable several NAT features to
>>> work behind our firewall.  I am totally new to Polycom, so some very basic
>>> help is needed.****
>>>
>>> ** **
>>>
>>> The server is remote but not behind a NAT there.  The phones are NAT?ed
>>> to the internet.  In the sofia sip trace I see this over and over:****
>>>
>>>
>>> ------------------------------------------------------------------------
>>> ****
>>>
>>> recv 552 bytes from udp/[71.127.152.57]:1026 at 16:26:07.358892:****
>>>
>>>
>>> ------------------------------------------------------------------------
>>> ****
>>>
>>>    REGISTER sip:fs_bfis.bizfocused.com SIP/2.0****
>>>
>>>    Via: SIP/2.0/UDP 10.10.40.47:5060;branch=z9hG4bKbf81dbdc8E687A5****
>>>
>>>    From: "228 Sean" <sip:228 at fs_bfis.bizfocused.com
>>> >;tag=3F42C046-B61A297****
>>>
>>>    To: <sip:228 at fs_bfis.bizfocused.com>****
>>>
>>>    CSeq: 1 REGISTER****
>>>
>>>    Call-ID: 2f482c2-2599cc43-1fb1a78 at 10.10.40.47****
>>>
>>>    Contact: <sip:228 at 10.10.40.47:5060>;methods="INVITE, ACK, BYE,
>>> CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
>>> ****
>>>
>>>    User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069****
>>>
>>>    Accept-Language: en****
>>>
>>>    Max-Forwards: 70****
>>>
>>>    Expires: 600****
>>>
>>>    Content-Length: 0****
>>>
>>> ** **
>>>
>>>
>>> ------------------------------------------------------------------------
>>> ****
>>>
>>> send 710 bytes to udp/[71.127.152.57]:5060 at 16:26:07.359067:****
>>>
>>>
>>> ------------------------------------------------------------------------
>>> ****
>>>
>>>    SIP/2.0 401 Unauthorized****
>>>
>>>    Via: SIP/2.0/UDP 10.10.40.47:5060
>>> ;branch=z9hG4bKbf81dbdc8E687A5;received=71.127.152.57****
>>>
>>>   From: "228 Sean" <sip:228 at fs_bfis.bizfocused.com>;tag=3F42C046-B61A297
>>> ****
>>>
>>>    To: <sip:228 at fs_bfis.bizfocused.com>;tag=t232me1NSD02S****
>>>
>>>    Call-ID: 2f482c2-2599cc43-1fb1a78 at 10.10.40.47****
>>>
>>>    CSeq: 1 REGISTER****
>>>
>>>    User-Agent:
>>> FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
>>> ****
>>>
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE****
>>>
>>>    Supported: precondition, path, replaces****
>>>
>>>    WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com",
>>> nonce="b9583359-0163-4bf2-9818-788f64c34207", algorithm=MD5, qop="auth"*
>>> ***
>>>
>>>    Content-Length: 0****
>>>
>>> ****
>>>
>>> If I understand correctly, the server should be sending back this 401
>>> message with the nonce so the phone can re-attempt the registration with an
>>> encrypted password.  If NAT is failing, the phone is never seeing the 401
>>> w/ the nonce.****
>>>
>>> ** **
>>>
>>> So what do I do in the WEB config interface to enable NAT on this phone?
>>> ****
>>>
>>> ** **
>>>
>>> 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> --
> 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.
>
>


-- 
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.
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From cal.leeming at simplicitymedialtd.co.uk  Fri Dec  7 22:32:10 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Fri, 7 Dec 2012 19:32:10 +0000
Subject: [Freeswitch-users] IVR menu delay - 1 second between events
In-Reply-To: <CAPNTnnrVW4Vpj4LcaNyeQrcmYq99DDF26E3UXpFoJyf=-q11fg@mail.gmail.com>
References: <CAHKQagF3C76QiybFNrS7GCdiGqaTbtPpQzD4wFivtJ+EK1CirQ@mail.gmail.com>
	<CAPNTnnrVW4Vpj4LcaNyeQrcmYq99DDF26E3UXpFoJyf=-q11fg@mail.gmail.com>
Message-ID: <CAHKQagG1dW+MUtLYgoJHN0mWa0xowKGo4-VFR4wV2nZqcMKWiw@mail.gmail.com>

Thanks for the response, although this sadly isn't the cause.

timeout uses milliseconds, of which 10000 which would equate to 10 seconds.

Plus, digit-len is set to 1, so it should automatically search for an entry
after the first button push.

Cal

On Fri, Dec 7, 2012 at 4:29 PM, Guillermo Ruiz Camauer
<grcamauer at gmail.com>wrote:

> Are you sure it's not timing out?  Change the timeout="10000" to
> timeout="30000" and see if your lag is now 3 seconds.
>
>
>
>
> On Fri, Dec 7, 2012 at 1:23 PM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> Hello,
>>
>> There appears to be a delay of 1 second between menu events within an IVR
>> (i.e. play long, invalid option, play short).
>>
>> We have the following IVR menu;
>>
>>           <menu name="ivr.2"
>>               greet-long="phrase:demo_ivr_main_menu"
>>               greet-short="phrase:demo_ivr_main_menu"
>>
>>               exit-sound="phrase:demo_ivr_main_menu"
>>               confirm-macro=""
>>               confirm-key=""
>>               confirm-attempts="3"
>>               timeout="10000"
>>               inter-digit-timeout="10"
>>               max-failures="3"
>>               max-timeouts="3"
>>               digit-len="1">
>>
>> In the above example, any option will result in an invalid selection
>> being met.
>>
>> However, FS seems to take its sweet time getting from the point of a user
>> pushing a button, to actually playing the next audio.
>>
>> freeswitch at vded213> 2012-12-07 16:17:21.752141 [DEBUG] switch_rtp.c:3797
>> RTP RECV DTMF 2:2240
>> 2012-12-07 16:17:21.752141 [DEBUG] switch_ivr_play_say.c:1682 done
>> playing file
>> /usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.wav
>> 2012-12-07 16:17:21.872105 [DEBUG] switch_ivr_menu.c:395 digits '2'
>> 2012-12-07 16:17:21.872105 [DEBUG] switch_ivr_menu.c:579 IVR menu 'ivr.2'
>> caught invalid input '2'
>> 2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:67 No language
>> specified - Using [en]
>> 2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:244 Handle
>> play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en)
>> 2012-12-07 16:17:22.872155 [DEBUG] switch_core_file.c:180 File
>> /usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.wav
>> sample rate 48000 doesn't match requested rate 8000
>> 2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:1309 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>>
>> In the above, you can see the IVR knew the menu was invalid at
>> "16:17:21.872105", but didn't start playing the next audio until
>> "16:17:22.872155".
>>
>> The difference between the two is almost exactly 1 second, which makes me
>> think maybe there's a config option somewhere that isn't
>> documented/obvious, or a hard coded delay.
>>
>> Any advice on this would be much appreciated, slow/laggy menus are a pet
>> hate of mine :/
>>
>> Thanks
>>
>> Cal
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> --
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From frank at carmickle.com  Fri Dec  7 22:57:34 2012
From: frank at carmickle.com (Frank Carmickle)
Date: Fri, 7 Dec 2012 14:57:34 -0500
Subject: [Freeswitch-users] pgsql as dsn
In-Reply-To: <CAKbxfG_iwHwBpF7kcNL=W36KvawJ2zy-0VSmH3VnXobmn1Voxg@mail.gmail.com>
References: <DF2B131E-2C82-4D7E-B4CE-085432BD3C0F@carmickle.com>
	<CAKbxfG_iwHwBpF7kcNL=W36KvawJ2zy-0VSmH3VnXobmn1Voxg@mail.gmail.com>
Message-ID: <E3AD9560-8E69-4588-B929-A2AD4B401C3D@carmickle.com>

Thank you Tony.  I saw that the deb packages had depends of libpq so I thought it was built.  That fixed it.

--FC


On Dec 6, 2012, at 4:09 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> try this from the build root
> 
> ./configure --enable-core-pgsql-support --no-create --no-recursion
> 
> then 
> 
> make install



From cal.leeming at simplicitymedialtd.co.uk  Fri Dec  7 23:08:07 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Fri, 7 Dec 2012 20:08:07 +0000
Subject: [Freeswitch-users] IVR menu delay - 1 second between events
In-Reply-To: <CAHKQagG1dW+MUtLYgoJHN0mWa0xowKGo4-VFR4wV2nZqcMKWiw@mail.gmail.com>
References: <CAHKQagF3C76QiybFNrS7GCdiGqaTbtPpQzD4wFivtJ+EK1CirQ@mail.gmail.com>
	<CAPNTnnrVW4Vpj4LcaNyeQrcmYq99DDF26E3UXpFoJyf=-q11fg@mail.gmail.com>
	<CAHKQagG1dW+MUtLYgoJHN0mWa0xowKGo4-VFR4wV2nZqcMKWiw@mail.gmail.com>
Message-ID: <CAHKQagH-5HwgRpP77bVSLkOusVzMH6bU68Ox=jR7fvS7X-hkFg@mail.gmail.com>

Just trawled through the source and found this;

src/switch_ivr_menu.c:
if (status == SWITCH_STATUS_SUCCESS) {
status = switch_ivr_sleep(session, 1000, SWITCH_FALSE, NULL);
}

So, I changed 1000 to 10, the delay has now disappeared and it seems to
work perfectly.

Could a core dev please explain why this sleep call is needed, and would
there be any unforeseen side effects by reducing it?

If there are no side effects, I'd like to make a suggestion to have this
reduced, or at least make it adjustable via config.

Use case for reducing? Slow menus are really, really annoying.

Thanks

Cal

On Fri, Dec 7, 2012 at 7:32 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Thanks for the response, although this sadly isn't the cause.
>
> timeout uses milliseconds, of which 10000 which would equate to 10 seconds.
>
> Plus, digit-len is set to 1, so it should automatically search for an
> entry after the first button push.
>
> Cal
>
> On Fri, Dec 7, 2012 at 4:29 PM, Guillermo Ruiz Camauer <
> grcamauer at gmail.com> wrote:
>
>> Are you sure it's not timing out?  Change the timeout="10000" to
>> timeout="30000" and see if your lag is now 3 seconds.
>>
>>
>>
>>
>> On Fri, Dec 7, 2012 at 1:23 PM, Cal Leeming [Simplicity Media Ltd] <
>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>
>>> Hello,
>>>
>>> There appears to be a delay of 1 second between menu events within an
>>> IVR (i.e. play long, invalid option, play short).
>>>
>>> We have the following IVR menu;
>>>
>>>           <menu name="ivr.2"
>>>               greet-long="phrase:demo_ivr_main_menu"
>>>               greet-short="phrase:demo_ivr_main_menu"
>>>
>>>               exit-sound="phrase:demo_ivr_main_menu"
>>>               confirm-macro=""
>>>               confirm-key=""
>>>               confirm-attempts="3"
>>>               timeout="10000"
>>>               inter-digit-timeout="10"
>>>               max-failures="3"
>>>               max-timeouts="3"
>>>               digit-len="1">
>>>
>>> In the above example, any option will result in an invalid selection
>>> being met.
>>>
>>> However, FS seems to take its sweet time getting from the point of a
>>> user pushing a button, to actually playing the next audio.
>>>
>>> freeswitch at vded213> 2012-12-07 16:17:21.752141 [DEBUG]
>>> switch_rtp.c:3797 RTP RECV DTMF 2:2240
>>> 2012-12-07 16:17:21.752141 [DEBUG] switch_ivr_play_say.c:1682 done
>>> playing file
>>> /usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.wav
>>> 2012-12-07 16:17:21.872105 [DEBUG] switch_ivr_menu.c:395 digits '2'
>>> 2012-12-07 16:17:21.872105 [DEBUG] switch_ivr_menu.c:579 IVR menu
>>> 'ivr.2' caught invalid input '2'
>>> 2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:67 No language
>>> specified - Using [en]
>>> 2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:244 Handle
>>> play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en)
>>> 2012-12-07 16:17:22.872155 [DEBUG] switch_core_file.c:180 File
>>> /usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.wav
>>> sample rate 48000 doesn't match requested rate 8000
>>> 2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:1309 Codec
>>> Activated L16 at 8000hz 1 channels 20ms
>>>
>>> In the above, you can see the IVR knew the menu was invalid at
>>> "16:17:21.872105", but didn't start playing the next audio until
>>> "16:17:22.872155".
>>>
>>> The difference between the two is almost exactly 1 second, which makes
>>> me think maybe there's a config option somewhere that isn't
>>> documented/obvious, or a hard coded delay.
>>>
>>> Any advice on this would be much appreciated, slow/laggy menus are a pet
>>> hate of mine :/
>>>
>>> Thanks
>>>
>>> Cal
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>>
>>>
>>
>>
>> --
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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  Fri Dec  7 23:14:10 2012
From: th982a at googlemail.com (Tamer Higazi)
Date: Fri, 07 Dec 2012 21:14:10 +0100
Subject: [Freeswitch-users] T.30 <---> T.38 howto ?!
Message-ID: <50C24E12.7020808@googlemail.com>

Hi people!
I am using hylafax with mod_spands and I have problems to setup the
dialplan.

Here is the case:

Hylafax (T.30) <--> FS (T.38) <--> VoIP Provider (T.38)

How is the diaplan successfully to setup to successfully make a T.30
<--> T.38 sending and receiving?!




I am not getting smart, for any help I would kindly thank you all.



Tamer


From sdevoy at bizfocused.com  Fri Dec  7 23:15:17 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 7 Dec 2012 15:15:17 -0500
Subject: [Freeswitch-users] Polycom IP 335 - 401 Unauthorized ....
	NAT	ISSUE?
In-Reply-To: <CAKV-jFz8_pJOOOddmkY0_Gb_JrSbAwxYnQKkTmLg8wzkX2g2dA@mail.gmail.com>
References: <1dc901cdd499$721ed340$565c79c0$@bizfocused.com>	<CAFiqYu=wDhL2+SDcNmMJ+AjJ40prrmyvN56WQ6Ab4c30DeUgdQ@mail.gmail.com>	<CAKV-jFxZ1ovBxejR-xnHouCx=BY3wCnWM83Z=qOr-zCf1fZm_A@mail.gmail.com>
	<CAKV-jFz8_pJOOOddmkY0_Gb_JrSbAwxYnQKkTmLg8wzkX2g2dA@mail.gmail.com>
Message-ID: <202e01cdd4b7$931f5420$b95dfc60$@bizfocused.com>

First Brian THANKS, it clearly took some time to put together such a
detailed response.

 

I reset to factory settings.

I have updated to a minor newer version than yours.

Yours 


UC Software Version

4.0.2.11307

Mine


UC Software Version

4.0.3.7562

 

I set everything EXACTLY as you specified in your post except my server,
extension, password, etc  where it applies.

It still fails.  There is an interesting difference in the SIP Messages
though.  From yours the phone sends CSeq: 1 Register, gets a response for
CSeq: 1 Register (with the nonce), then your phone sends CSeq: 2 Register .
>From mine the phone sends CSeq: 1 Register  again, and again, and again.  I
still think it is actually not receiving the 401 message with the nonce.

 

One other minor difference that may by important. On your FIRST 401
Unauthorized Message Via line says

Via: SIP/2.0/UDP
10.0.0.39;branch=z9hG4bK848ac3ba5589D827;received=76.238.166.184;rport=5060

Mine says:

Via: SIP/2.0/UDP
10.10.40.47;branch=z9hG4bKa9b37440268F7B2B;received=71.127.152.57

 

It does not have an rport.  I don't know if that matters.  It doesn't it
matter that other phones here are using port 5060, right?  Are their other
ports I can specify for rport?  How can I tell if the phone is actually
getting the 401?

 

The syslog from the phone says:

sip  |4|03|Registration failed User: 228, Error Code:480 Temporarily not
available 10.10.40.47   07/12 14:48:17.315  

 

Thanks again for your help.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
Foster
Sent: Friday, December 07, 2012 2:11 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Polycom IP 335 - 401 Unauthorized .... NAT
ISSUE?

 

This might help you:

 

http://pastebin.freeswitch.org/20301

 

Try comparing the SIP messages to yours. Notice that it tries to register
twice, the first one is Unauthorized.

 

On Fri, Dec 7, 2012 at 1:54 PM, Brian Foster <bdfoster at endigotech.com>
wrote:

Let's do this step by step. First of all, my server is off site. We are
going through a NAT in order to get to FreeSWITCH. Looks like this is the
same setup you have.

 

I have the same phone as you do:


Phone Information


Phone Model

SoundPoint IP 335


Part Number

2345-12375-001 Rev:A


MAC Address

00:04:F2:37:3D:C0


IP Address

10.0.0.39


UC Software Version

4.0.2.11307


BootROM Software Version

5.0.2.12692


Alright so now that we have that squared away, the next step is to set up
the phone.

 

Settings > SIP

 

Local Settings:

 

Local SIP Port: 0

Calls Per Line Key: 4

New SDP Type: Disable

Live Communication Server Support: Disable

Non-Standard Line Seize: Enable

Digitmap: Not relevent

Digitmap Timeout: 3|3|3|3|3|3

Remove End-of-Dial Marker: Enable

Digit Impossible Match: 0

 

Outbound Proxy:

 

Address: <Blank>

Port: 0

Transport: DNSnaptr

 

Server 1:

 

Address: pbx.endigovoip.com

Port: 0

Transport: DNSnaptr (You shouldn't have issues with UDPonly, might be worth
trying though.

Espires (s): 3600

Register: Yes

Retry Timeout (ms): 0

Retry Maximum Count: 3

Line Seize Timeout: 30

 

I do not have a second server.

 

Settings > Network > NAT

 


NAT


* IP Address

	

* Signalling Port

0


* Media Port Start

0


Keep-Alive Interval (s)

0

 

Settings > Lines

 

  <http://10.0.0.39/images/icon_minus.gif>  Identification


Display Name

Brian Foster


Address

2546 at pbx.endigovoip.com


Authentication User ID

2546


Authentication Password

[          ]


Label

2546


Type

(X) Private ( ) Shared


Third Party Name

	

Number of Line Keys

2


Calls Per Line

4


Ring Type

[Low Trill \/]

  <http://10.0.0.39/images/icon_minus.gif>  Outbound Proxy


Address

	

Port

0


Transport

[DNSnaptr \/]

  <http://10.0.0.39/images/icon_minus.gif>  Server 1


Address

	

Port

0


Transport

[DNSnaptr \/]


Expires (s)

3600


Register

(X) Yes ( ) No


Retry Timeout (ms)

0


Retry Maximum Count

3


Line Seize Timeout (s)

30

  <http://10.0.0.39/images/icon_minus.gif>  Server 2


Address

	

Port

0


Transport

[DNSnaptr \/]


Expires (s)

3600


Register

(X) Yes ( ) No


Retry Timeout (ms)

0


Retry Maximum Count

3


Line Seize Timeout (s)

30

  <http://10.0.0.39/images/icon_minus.gif>  Call Diversion


* Always Forward

(X) Enable ( ) Disable


* Always Forward To Contact

	
		

* If Busy, Forward

(X) Enable ( ) Disable


* If Busy, Forward To Contact

	
		

* On No Answer, Forward

(X) Enable ( ) Disable


* On No Answer, Forward To Contact

	

* No Answer Timeout (seconds)

55

		

* On Do Not Disturb, Forward

( ) Enable (X) Disable


* On Do Not Disturb, Forward To Contact

	
		

* Disable Forward For Shared Lines

(X) Yes ( ) No

		

* Forward Specific Caller

(X) Enable ( ) Disable

  <http://10.0.0.39/images/icon_minus.gif>  Message Center


Subscription Address

	

Callback Mode

[Registration \/]


Callback Contact

	

 

 

Check those and let us know where you stand after that.

 

-BDF

 

On Fri, Dec 7, 2012 at 1:20 PM, Steven Ayre <steveayre at gmail.com> wrote:

Try this parameter:

http://wiki.freeswitch.org/wiki/NDLB#NDLB-force-rport

 

or if that fails

http://wiki.freeswitch.org/wiki/NDLB#NDLB-connectile-dysfunction

 

On 7 December 2012 16:39, Sean Devoy <sdevoy at bizfocused.com> wrote:

HI All,

 

I am still banging my head against the wall here try to get a Polycom 335 to
register w/ FS.  I have checked all the SERVER and USER/AUTH fields like
1000 times and 900 variations.  I think my problem may be NAT related.  I
know on my CISCO 504G I had to enable several NAT features to work behind
our firewall.  I am totally new to Polycom, so some very basic help is
needed.

 

The server is remote but not behind a NAT there.  The phones are NAT'ed to
the internet.  In the sofia sip trace I see this over and over:

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

recv 552 bytes from udp/[71.127.152.57]:1026 at 16:26:07.358892:

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

   REGISTER sip:fs_bfis.bizfocused.com SIP/2.0

   Via: SIP/2.0/UDP 10.10.40.47:5060;branch=z9hG4bKbf81dbdc8E687A5

   From: "228 Sean" <sip:228 at fs_bfis.bizfocused.com
<mailto:sip%3A228 at fs_bfis.bizfocused.com> >;tag=3F42C046-B61A297

   To: <sip:228 at fs_bfis.bizfocused.com
<mailto:sip%3A228 at fs_bfis.bizfocused.com> >

   CSeq: 1 REGISTER

   Call-ID: 2f482c2-2599cc43-1fb1a78 at 10.10.40.47

   Contact: <sip:228 at 10.10.40.47:5060>;methods="INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"

   User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069

   Accept-Language: en

   Max-Forwards: 70

   Expires: 600

   Content-Length: 0

 

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

send 710 bytes to udp/[71.127.152.57]:5060 at 16:26:07.359067:

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

   SIP/2.0 401 Unauthorized

   Via: SIP/2.0/UDP
10.10.40.47:5060;branch=z9hG4bKbf81dbdc8E687A5;received=71.127.152.57

  From: "228 Sean" <sip:228 at fs_bfis.bizfocused.com
<mailto:sip%3A228 at fs_bfis.bizfocused.com> >;tag=3F42C046-B61A297

   To: <sip:228 at fs_bfis.bizfocused.com
<mailto:sip%3A228 at fs_bfis.bizfocused.com> >;tag=t232me1NSD02S

   Call-ID: 2f482c2-2599cc43-1fb1a78 at 10.10.40.47

   CSeq: 1 REGISTER

   User-Agent:
FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120
712T101002Z

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: precondition, path, replaces

   WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com",
nonce="b9583359-0163-4bf2-9818-788f64c34207", algorithm=MD5, qop="auth"

   Content-Length: 0

If I understand correctly, the server should be sending back this 401
message with the nonce so the phone can re-attempt the registration with an
encrypted password.  If NAT is failing, the phone is never seeing the 401 w/
the nonce.

 

So what do I do in the WEB config interface to enable NAT on this phone?

 

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

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

 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
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.





 

-- 
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.

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From sdevoy at bizfocused.com  Sat Dec  8 00:25:33 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 7 Dec 2012 16:25:33 -0500
Subject: [Freeswitch-users] Polycom IP 335 - 401 Unauthorized
	....	NAT	ISSUE?
In-Reply-To: <202e01cdd4b7$931f5420$b95dfc60$@bizfocused.com>
References: <1dc901cdd499$721ed340$565c79c0$@bizfocused.com>	<CAFiqYu=wDhL2+SDcNmMJ+AjJ40prrmyvN56WQ6Ab4c30DeUgdQ@mail.gmail.com>	<CAKV-jFxZ1ovBxejR-xnHouCx=BY3wCnWM83Z=qOr-zCf1fZm_A@mail.gmail.com>	<CAKV-jFz8_pJOOOddmkY0_Gb_JrSbAwxYnQKkTmLg8wzkX2g2dA@mail.gmail.com>
	<202e01cdd4b7$931f5420$b95dfc60$@bizfocused.com>
Message-ID: <20a301cdd4c1$63fcf030$2bf6d090$@bizfocused.com>

 

I had to set SETTINGS > SIP > LOCAL SETTINGS > Local SIP Port: 5062 (first
nonzero value I tired).

 

I got logged in!!!!  Can't call or be called, but I got REGISTERED!  The
rest is usually much easier.  I think I have NAT RTP issues now.

I will look at these next:

Try this parameter:

http://wiki.freeswitch.org/wiki/NDLB#NDLB-force-rport

 

or if that fails

http://wiki.freeswitch.org/wiki/NDLB#NDLB-connectile-dysfunction

 

THANKS AGAIN.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Devoy
Sent: Friday, December 07, 2012 3:15 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Polycom IP 335 - 401 Unauthorized .... NAT
ISSUE?

 

First Brian THANKS, it clearly took some time to put together such a
detailed response.

 

I reset to factory settings.

I have updated to a minor newer version than yours.

Yours 


UC Software Version

4.0.2.11307

Mine


UC Software Version

4.0.3.7562

 

I set everything EXACTLY as you specified in your post except my server,
extension, password, etc  where it applies.

It still fails.  There is an interesting difference in the SIP Messages
though.  From yours the phone sends CSeq: 1 Register, gets a response for
CSeq: 1 Register (with the nonce), then your phone sends CSeq: 2 Register .
>From mine the phone sends CSeq: 1 Register  again, and again, and again.  I
still think it is actually not receiving the 401 message with the nonce.

 

One other minor difference that may by important. On your FIRST 401
Unauthorized Message Via line says

Via: SIP/2.0/UDP
10.0.0.39;branch=z9hG4bK848ac3ba5589D827;received=76.238.166.184;rport=5060

Mine says:

Via: SIP/2.0/UDP
10.10.40.47;branch=z9hG4bKa9b37440268F7B2B;received=71.127.152.57

 

It does not have an rport.  I don't know if that matters.  It doesn't it
matter that other phones here are using port 5060, right?  Are their other
ports I can specify for rport?  How can I tell if the phone is actually
getting the 401?

 

The syslog from the phone says:

sip  |4|03|Registration failed User: 228, Error Code:480 Temporarily not
available 10.10.40.47   07/12 14:48:17.315  

 

Thanks again for your help.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
Foster
Sent: Friday, December 07, 2012 2:11 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Polycom IP 335 - 401 Unauthorized .... NAT
ISSUE?

 

This might help you:

 

http://pastebin.freeswitch.org/20301

 

Try comparing the SIP messages to yours. Notice that it tries to register
twice, the first one is Unauthorized.

 

On Fri, Dec 7, 2012 at 1:54 PM, Brian Foster <bdfoster at endigotech.com>
wrote:

Let's do this step by step. First of all, my server is off site. We are
going through a NAT in order to get to FreeSWITCH. Looks like this is the
same setup you have.

 

I have the same phone as you do:


Phone Information


Phone Model

SoundPoint IP 335


Part Number

2345-12375-001 Rev:A


MAC Address

00:04:F2:37:3D:C0


IP Address

10.0.0.39


UC Software Version

4.0.2.11307


BootROM Software Version

5.0.2.12692


Alright so now that we have that squared away, the next step is to set up
the phone.

 

Settings > SIP

 

Local Settings:

 

Local SIP Port: 0

Calls Per Line Key: 4

New SDP Type: Disable

Live Communication Server Support: Disable

Non-Standard Line Seize: Enable

Digitmap: Not relevent

Digitmap Timeout: 3|3|3|3|3|3

Remove End-of-Dial Marker: Enable

Digit Impossible Match: 0

 

Outbound Proxy:

 

Address: <Blank>

Port: 0

Transport: DNSnaptr

 

Server 1:

 

Address: pbx.endigovoip.com

Port: 0

Transport: DNSnaptr (You shouldn't have issues with UDPonly, might be worth
trying though.

Espires (s): 3600

Register: Yes

Retry Timeout (ms): 0

Retry Maximum Count: 3

Line Seize Timeout: 30

 

I do not have a second server.

 

Settings > Network > NAT

 


NAT


* IP Address

	

* Signalling Port

0


* Media Port Start

0


Keep-Alive Interval (s)

0

 

Settings > Lines

 

  <http://10.0.0.39/images/icon_minus.gif>  Identification


Display Name

Brian Foster


Address

2546 at pbx.endigovoip.com


Authentication User ID

2546


Authentication Password

[          ]


Label

2546


Type

(X) Private ( ) Shared


Third Party Name

	

Number of Line Keys

2


Calls Per Line

4


Ring Type

[Low Trill \/]

  <http://10.0.0.39/images/icon_minus.gif>  Outbound Proxy


Address

	

Port

0


Transport

[DNSnaptr \/]

  <http://10.0.0.39/images/icon_minus.gif>  Server 1


Address

	

Port

0


Transport

[DNSnaptr \/]


Expires (s)

3600


Register

(X) Yes ( ) No


Retry Timeout (ms)

0


Retry Maximum Count

3


Line Seize Timeout (s)

30

  <http://10.0.0.39/images/icon_minus.gif>  Server 2


Address

	

Port

0


Transport

[DNSnaptr \/]


Expires (s)

3600


Register

(X) Yes ( ) No


Retry Timeout (ms)

0


Retry Maximum Count

3


Line Seize Timeout (s)

30

  <http://10.0.0.39/images/icon_minus.gif>  Call Diversion


* Always Forward

(X) Enable ( ) Disable


* Always Forward To Contact

	
		

* If Busy, Forward

(X) Enable ( ) Disable


* If Busy, Forward To Contact

	
		

* On No Answer, Forward

(X) Enable ( ) Disable


* On No Answer, Forward To Contact

	

* No Answer Timeout (seconds)

55

		

* On Do Not Disturb, Forward

( ) Enable (X) Disable


* On Do Not Disturb, Forward To Contact

	
		

* Disable Forward For Shared Lines

(X) Yes ( ) No

		

* Forward Specific Caller

(X) Enable ( ) Disable

  <http://10.0.0.39/images/icon_minus.gif>  Message Center


Subscription Address

	

Callback Mode

[Registration \/]


Callback Contact

	

 

 

Check those and let us know where you stand after that.

 

-BDF

 

On Fri, Dec 7, 2012 at 1:20 PM, Steven Ayre <steveayre at gmail.com> wrote:

Try this parameter:

http://wiki.freeswitch.org/wiki/NDLB#NDLB-force-rport

 

or if that fails

http://wiki.freeswitch.org/wiki/NDLB#NDLB-connectile-dysfunction

 

On 7 December 2012 16:39, Sean Devoy <sdevoy at bizfocused.com> wrote:

HI All,

 

I am still banging my head against the wall here try to get a Polycom 335 to
register w/ FS.  I have checked all the SERVER and USER/AUTH fields like
1000 times and 900 variations.  I think my problem may be NAT related.  I
know on my CISCO 504G I had to enable several NAT features to work behind
our firewall.  I am totally new to Polycom, so some very basic help is
needed.

 

The server is remote but not behind a NAT there.  The phones are NAT'ed to
the internet.  In the sofia sip trace I see this over and over:

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

recv 552 bytes from udp/[71.127.152.57]:1026 at 16:26:07.358892:

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

   REGISTER sip:fs_bfis.bizfocused.com SIP/2.0

   Via: SIP/2.0/UDP 10.10.40.47:5060;branch=z9hG4bKbf81dbdc8E687A5

   From: "228 Sean" <sip:228 at fs_bfis.bizfocused.com
<mailto:sip%3A228 at fs_bfis.bizfocused.com> >;tag=3F42C046-B61A297

   To: <sip:228 at fs_bfis.bizfocused.com
<mailto:sip%3A228 at fs_bfis.bizfocused.com> >

   CSeq: 1 REGISTER

   Call-ID: 2f482c2-2599cc43-1fb1a78 at 10.10.40.47

   Contact: <sip:228 at 10.10.40.47:5060>;methods="INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"

   User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069

   Accept-Language: en

   Max-Forwards: 70

   Expires: 600

   Content-Length: 0

 

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

send 710 bytes to udp/[71.127.152.57]:5060 at 16:26:07.359067:

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

   SIP/2.0 401 Unauthorized

   Via: SIP/2.0/UDP
10.10.40.47:5060;branch=z9hG4bKbf81dbdc8E687A5;received=71.127.152.57

  From: "228 Sean" <sip:228 at fs_bfis.bizfocused.com
<mailto:sip%3A228 at fs_bfis.bizfocused.com> >;tag=3F42C046-B61A297

   To: <sip:228 at fs_bfis.bizfocused.com
<mailto:sip%3A228 at fs_bfis.bizfocused.com> >;tag=t232me1NSD02S

   Call-ID: 2f482c2-2599cc43-1fb1a78 at 10.10.40.47

   CSeq: 1 REGISTER

   User-Agent:
FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120
712T101002Z

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: precondition, path, replaces

   WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com",
nonce="b9583359-0163-4bf2-9818-788f64c34207", algorithm=MD5, qop="auth"

   Content-Length: 0

If I understand correctly, the server should be sending back this 401
message with the nonce so the phone can re-attempt the registration with an
encrypted password.  If NAT is failing, the phone is never seeing the 401 w/
the nonce.

 

So what do I do in the WEB config interface to enable NAT on this phone?

 

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

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

 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
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.





 

-- 
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.

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From francis.joanis at gmail.com  Fri Dec  7 21:54:05 2012
From: francis.joanis at gmail.com (Francis Joanis)
Date: Fri, 7 Dec 2012 13:54:05 -0500
Subject: [Freeswitch-users] RFC 4579 and FreeSWITCH
Message-ID: <CAPoB04xa1SeKRzi=ZnqUKRHLUk+6Paz7B2nDKktYZOYQ6ZieWw@mail.gmail.com>

Hi guys,

I started playing with FreeSWITCH to make conferences and I was curious
about the support for RFC 4579 (Call Control - Conferencing for User
Agents).

I was able to have FreeSWITCH return a ;isfocus parameter in the Contact
(from the 200 OK) and I tried to add a new participant to the conference
using a REFER (see RFC 4579 section 5.5), which generated the following log:

[ERR] sofia.c:7095 Cannot Blind Transfer 1 Legged calls

I think RFC 4579 is mentioned on the FS wiki as a "supported" RFC but is
this call flow currently supported?

Cheers and thanks for the awesome work in FS,
Francis
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From francis.joanis at gmail.com  Fri Dec  7 22:32:03 2012
From: francis.joanis at gmail.com (Francis Joanis)
Date: Fri, 7 Dec 2012 14:32:03 -0500
Subject: [Freeswitch-users] RFC 4579 and FreeSWITCH
In-Reply-To: <CAPoB04xa1SeKRzi=ZnqUKRHLUk+6Paz7B2nDKktYZOYQ6ZieWw@mail.gmail.com>
References: <CAPoB04xa1SeKRzi=ZnqUKRHLUk+6Paz7B2nDKktYZOYQ6ZieWw@mail.gmail.com>
Message-ID: <CAPoB04w66efnSrMs9822K+XuFH5Ju3kK-6Ev25HL76Axzj-tHw@mail.gmail.com>

Hi again,

On Fri, Dec 7, 2012 at 1:54 PM, Francis Joanis <francis.joanis at gmail.com>wrote:

> Hi guys,
>
> I started playing with FreeSWITCH to make conferences and I was curious
> about the support for RFC 4579 (Call Control - Conferencing for User
> Agents).
>
> I was able to have FreeSWITCH return a ;isfocus parameter in the Contact
> (from the 200 OK) and I tried to add a new participant to the conference
> using a REFER (see RFC 4579 section 5.5), which generated the following log:
>
> [ERR] sofia.c:7095 Cannot Blind Transfer 1 Legged calls
>
> I think RFC 4579 is mentioned on the FS wiki as a "supported" RFC but is
> this call flow currently supported?
>
> Cheers and thanks for the awesome work in FS,
> Francis
>

After digging through the code I realized my test is probably invalid since
I was reusing the same SIP dialog for the REFER as the one that already
existed for the existing call. I'll retest with a true out of dialog REFER
and post my results back...

Thanks,
Francis
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From msc at freeswitch.org  Sat Dec  8 00:56:16 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 Dec 2012 13:56:16 -0800
Subject: [Freeswitch-users] sip registration
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233A0D5@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2339A49@Mail-Kilo.squay.com>
	<CAFiqYu=cpMkTxdE99smQBM6NgitNQ5ZpePwfSYr_R23dpj7Hng@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339B39@Mail-Kilo.squay.com>
	<CAKzWOxUW21XZte0AoVDqTtC6vkJ4FmA3sbP5LgJVK5bj4+1Sbw@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339D42@Mail-Kilo.squay.com>
	<CAKzWOxVMWfR6vk8ODKkND1Tjxj-d5VFYn9U+dDoNyB_t2dE05w@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233A02B@Mail-Kilo.squay.com>
	<CAKzWOxVqn1tGWgHKFs+vCEDpXUg2rNyvpbw5-bBSPw846hg7Xw@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233A0D5@Mail-Kilo.squay.com>
Message-ID: <CAKzWOxVhphxwn8EhfwTfo2Ti5OgTNp2Q9hKsB_Y-+4p+P9Ns4A@mail.gmail.com>

Archana,

Did you see this page on the wiki?
http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Alphanumeric_to_numeric_user_mapping

It's been there for years. I promise you that the wiki has more information
than you are giving it credit for. Just remember that our wiki is a
microcosm of the Internet: it has what you're looking for but you need
Google to search it and you have to dig through all the results. :)

Try that and see if it does what you need.

As far as your question goes, there isn't a "script" that checks it.
Rather, the module that is doing the authentication will look in the XML
data for the user id and password. If mod_xml_curl is enable then FS will
go fetch the XML data from your server.

-MC

On Fri, Dec 7, 2012 at 1:21 AM, Archana Venugopan <a.venugopan at mundio.com>wrote:

>  Hi,****
>
> ** **
>
> I want to give some alphabets instead of number. I want to know which
> script checks this authentication name to corresponding DB table. Please
> let me know.****
>
> Thanks****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 06 December 2012 21:58
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
> ** **
>
> Use "100" for the authentication name as well.
> -MC****
>
> On Thu, Dec 6, 2012 at 9:25 AM, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
> Thanks. I thought it will look in mod_sofia code. In the below screen I
> register the ID ?100?. Now instead of ?100? in ?Authentication Name? I need
> to give some e-mail ID or name(Archana) which should validate in DB.****
>
> I tried giving a name in ?Authentication Name? but the phone was not
> registered. Am not sure this authentication name is being looked in which
> column in table too.****
>
> Please let me know if this will be picked from any sofia code or any C
> script? Once we register in the below screen which script validates the
> Settings in freeswitch?****
>
>  ****
>
> Sorry if am repeating the same question, but I could not get the exact
> code and am clueless.****
>
> *Global SIP Settings*****
>
> Top of Form****
>
> *Basic SIP Authentication Settings*****
>
>  ****
>
> Screen Name****
>
> Screen Name 2****
>
> Phone Number****
>
> Caller ID****
>
> Authentication Name****
>
> Password****
>
> Bottom of Form****
>
>  ****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 05 December 2012 20:34
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
>  ****
>
> When an event that requires a user lookup takes place then the system will
> look in the XML user directory unless it has been configured to look
> somewhere else. The other places to look are usually:
> mod_xml_curl
> One of the language like Lua, Perl, Python
>
> If it's xml_curl then FS will do a POST to your web server in hopes of
> receiving back the necessary XML for the given user. It would be up to you
> to have your web server handle the request, poll the database, then format
> and return the XML data. See this wiki page<http://wiki.freeswitch.org/wiki/Xml_curl#How_do_I_configure_mod_xml_curl.3F>for more info on xml curl.
>
> If it's a language then you'll have a "binding" in the conf file for the
> language that will handle the lookup. Again, your script will need to
> handle the communication with your database. See this wiki page<http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration>for more information.
>
> Hope this helps.
> -MC****
>
> On Wed, Dec 5, 2012 at 9:30 AM, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
> Thanks for the information. But sorry, how to access user_data API command.
> ****
>
>  ****
>
> Am not clear on the flow. Once we register domain and usernumber in sip
> what exactly happens? Which script picks up this domain and username and
> validates with our database?****
>
> Could you please provide me with an overview. ****
>
>  ****
>
> Many thanks****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 05 December 2012 17:13
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
>  ****
>
> If you're talking about the user configuration then yes, you could create
> an "email" parameter or variable and access it with the user_data API
> command.
> -MC****
>
> On Wed, Dec 5, 2012 at 1:35 AM, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
> In that case can I have 1 more column say e-mail and can this e-mail be
> checked in DB instead of checking reg_user(?100?)? Is that feasible?****
>
> Also which code should be changed any idea please?****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 04 December 2012 19:51
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
>  ****
>
> You can have a user 'ana' in the domain 'gmail.com'. Though using someone
> else's domain as local in your FS setup may not be a good idea.****
>
> You can't have a @ in the username itself (per the SIP standard, not
> limited to FreeSWITCH).****
>
>  ****
>
> On 4 December 2012 18:00, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
>  ****
>
> Currently we register authentication name as say ?100? in sip
> registration, this comes to freeswitch and it will check in our DB for 100
> and if its present then registrations would be successful. ****
>
>  ****
>
> freeswitch at internal> show registrations****
>
>
> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
> ****
>
> 100,fsfailover.uk01.com
> ,e4969067f9a8c098,sofia/internal/sip:100 at 192.168.2.234:5060
> ;transport=udp,1354638871,192.168.2.234,5060,udp,squay-laptop-1.squay.com*
> ***
>
>  ****
>
> I want to change this 100 to some e-mail address, so instead of 100 it
> will be something like ?ana at gmail.com?. Can we do this? While coming to
> freeswitch whether there would be any issues?****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
> ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Sat Dec  8 01:01:30 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 Dec 2012 14:01:30 -0800
Subject: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch
In-Reply-To: <1a1401cdd435$167f2940$437d7bc0$@bizfocused.com>
References: <186801cdd400$87a246a0$96e6d3e0$@bizfocused.com>
	<949E214D-7B59-4D8E-BE39-DC5A3145C6D7@endigotech.com>
	<1a1401cdd435$167f2940$437d7bc0$@bizfocused.com>
Message-ID: <CAKzWOxWzY=401Cuq2=hP5yDArBSypOcBMuemZkNyfnwAt8WHkA@mail.gmail.com>

Sean,

Is the Polycom behind NAT and FS on a public IP? If so I think you'll need
to modify the SIP profile. There's an "NDLB-force-rport" param. Check the
wiki, but I think there's several values ("false","true","safe") you can
try. IIRC "safe" is the one you'd try. This kinda sounds like the Polycom
not supporting rport bug that they've ignored for the past decade...

-MC

On Thu, Dec 6, 2012 at 8:41 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Thanks Brian.  I have looked up more info and that is correct.****
>
> However, I still get 401 Unauthorized.****
>
> ** **
>
> Any other ideas anyone?****
>
> ** **
>
> I am ready to return them and just by Cisco SPA504Gs.  I was hoping to add
> another choice for my customers but I am yet to come across any decent
> documentation from Polycom.  If I can?t even get them to log in, I am not
> hopeful about advanced features.****
>
> ** **
>
> Sean****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian Foster
> *Sent:* Thursday, December 06, 2012 5:38 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch**
> **
>
> ** **
>
> I think Address should be <extension>@<host name/ip>
>
> Sent from my iPhone****
>
>
> On Dec 6, 2012, at 5:25 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:***
> *
>
> I am new to Polycom phones.  We have always used CISCO 5xx Series.  I have
> been trying to use the WEB interface.  The example in the WIKI uses tftp
> through DHCP option 60.  MY home router/gateway?s DHCP does not support
> option 60.****
>
>  ****
>
> I am unable to get the Polycom 335 to register, it keeps getting
> ?Unauthorized?.  I have double checked by having my Cisco phone register
> with this information and it works fine.  I must be missing something.****
>
>  ****
>
> On the Polycom Web Interface I have the following values:****
>
> SIP TAB:****
>
> SERVERS:****
>
> Outbound Proxy is blank****
>
> SERVER 1:****
>
> Address: <my server?s FQDN >****
>
> Port: 5060****
>
> Transport: UDPOnly****
>
> Expires:3600****
>
> Register: 1****
>
> Retry Timeout: 0****
>
> Retry Max Count: 3****
>
> Line Seize Timeout: 30****
>
> SERVER 2: <same as server 1 for now except port 5062)****
>
> LINE TAB:****
>
> LINE 1:****
>
> Display Name: 228****
>
> Address: 228****
>
> Auth UserID:228****
>
> Auth Password: <actual password here>****
>
> Label: 228****
>
> Type: Private****
>
> 3rd Party Name: blank****
>
> Number of Line Keys: 1****
>
> Calls per line: 24****
>
> SERVER 1:****
>
> Address: <my server?s FQDN >****
>
> Port: 5060****
>
> Transport: UDPOnly****
>
> Expires:3600****
>
> Register: 1****
>
> Retry Timeout: 0****
>
> Retry Max Count: 3****
>
> Line Seize Timeout: 30****
>
> SERVER 2: <same as server 1 for now)****
>
> Everything else is the defaults.****
>
>  ****
>
> The resulting sofia trace messages:****
>
>    ------------------------------------------------------------------------
> ****
>
> recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.210138:****
>
>    ------------------------------------------------------------------------
> ****
>
>    REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0****
>
>    Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C****
>
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B****
>
>    To: <sip:228 at fs_bfis.bizfocused.com>****
>
>    CSeq: 1 REGISTER****
>
>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47****
>
>    Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL,
> OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"****
>
>    User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069****
>
>    Accept-Language: en****
>
>    Max-Forwards: 70****
>
>    Expires: 3600****
>
>    Content-Length: 0****
>
>    ****
>
>    ------------------------------------------------------------------------
> ****
>
> send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.211063:****
>
>    ------------------------------------------------------------------------
> ****
>
>    SIP/2.0 401 Unauthorized****
>
>    Via: SIP/2.0/UDP
> 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57****
>
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B****
>
>    To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre****
>
>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47****
>
>    CSeq: 1 REGISTER****
>
>    User-Agent:
> FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
> ****
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE****
>
>    Supported: precondition, path, replaces****
>
>    WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com",
> nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"***
> *
>
>    Content-Length: 0****
>
>    ****
>
>    ------------------------------------------------------------------------
> ****
>
> recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.712339:****
>
>    ------------------------------------------------------------------------
> ****
>
>    REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0****
>
>    Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C****
>
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B****
>
>    To: <sip:228 at fs_bfis.bizfocused.com>****
>
>    CSeq: 1 REGISTER****
>
>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47****
>
>    Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL,
> OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"****
>
>    User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069****
>
>    Accept-Language: en****
>
>    Max-Forwards: 70****
>
>    Expires: 3600****
>
>    Content-Length: 0****
>
>  ****
>
>   ------------------------------------------------------------------------
> ****
>
> send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.712479:****
>
>    ------------------------------------------------------------------------
> ****
>
>    SIP/2.0 401 Unauthorized****
>
>   Via: SIP/2.0/UDP
> 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57****
>
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B****
>
>   To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre****
>
>   Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47****
>
>    CSeq: 1 REGISTER****
>
>    User-Agent:
> FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
> ****
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE****
>
>    Supported: precondition, path, replaces****
>
>    WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com",
> nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"***
> *
>
>    Content-Length: 0****
>
>    ------------------------------------------------------------------------
> ****
>
>  ****
>
>  ****
>
> Any ideas?****
>
>  ****
>
>  ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Sat Dec  8 01:03:21 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 Dec 2012 14:03:21 -0800
Subject: [Freeswitch-users] sofia_reg.c
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233A2D2@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233A2D2@Mail-Kilo.squay.com>
Message-ID: <CAKzWOxUdWXAw332+DKH5d5kD78jNU9r4rTUoNxZe1QaGak1T7w@mail.gmail.com>

These come from the SIP headers for the dialog in question.
-MC

On Fri, Dec 7, 2012 at 10:04 AM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi,****
>
> ** **
>
> In sofia_reg.c I find variables like
> ?sip_auth_username?,?sip_auth_password?,etc. Can anyone please tell me from
> where it will get value for all these variables?****
>
> Is it from some query in lua script or is it from any sip URL?****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Sat Dec  8 01:09:10 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 Dec 2012 14:09:10 -0800
Subject: [Freeswitch-users] how to Ignore DTMF shorter than Fixed
	duration ?
In-Reply-To: <A9CAED77-5449-4C93-95DB-7C742F68CE06@edvina.net>
References: <CAHWtr1TbtvoxrxbCEY6y8r_rKgtsh2zHhXmeZYRA0+6P0dKGUQ@mail.gmail.com>
	<CAKzWOxXQSj9hPSoSmH9pQ19J=+eU5qM+8aNO51BoWzzN_gKj_w@mail.gmail.com>
	<CAHWtr1SfKpkXL7BeB0995sjN9k8yLwKEXRcxkXvZyvQfqqEjUg@mail.gmail.com>
	<CAKzWOxUoXCMFOb-P8ZxOVJzaK4=o5Kxt=dF+EnMHEt_cu2qQqQ@mail.gmail.com>
	<A9CAED77-5449-4C93-95DB-7C742F68CE06@edvina.net>
Message-ID: <CAKzWOxW5GScF3WA0fRJMyBBDS4m51W65-_35BLq-1x3MkXPZHQ@mail.gmail.com>

On Thu, Dec 6, 2012 at 11:08 PM, Olle E. Johansson <oej at edvina.net> wrote:

>
> 6 dec 2012 kl. 22:52 skrev Michael Collins <msc at freeswitch.org>:
>
> Bummer. I'm not aware of any way to ignore those. Anyone else have ideas?
>
>
> The problem with ignoring DTMF in RTP is that it will delay audio, always.
> Let's say you want to delete all DTMF shorter than 1200 ms.
> We receive a DTMF begin packet, then will have to wait to 1220 ms to see
> if the DTMF CONTINUE packets continue or if we actually get a DTMF END
> packet arriving in time. If it does, we'll delete the DTMF - but that
> will produce a gap in the audio we will have to fill. If it is longer than
> 1200 ms we will start playing it out on the other side of the call bridge.
>
> You seriously do not want to delay audio 1200 ms. Not even 100 ms.
>
> There no simple solution to this. We've had issues with GENBAND servers
> and their DSPs that send short DTMF in calls with
> female voices.
>
> In short. When we receive the start of the DTMF we have no idea about the
> duration. We can't make a decision until it's too late to make a
> decision... If you just focus on delaying DTMF, you will start playing DTMF
> on top of audio, cancelling out parts of the conversation.
>
> /O
>
>
OEJ,
+1
Many thanks for taking the time to explain that! That was an extremely well
written paragraph, easily understood by all.
-MC
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From bdfoster at endigotech.com  Sat Dec  8 01:09:17 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 7 Dec 2012 17:09:17 -0500
Subject: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch
In-Reply-To: <CAKzWOxWzY=401Cuq2=hP5yDArBSypOcBMuemZkNyfnwAt8WHkA@mail.gmail.com>
References: <186801cdd400$87a246a0$96e6d3e0$@bizfocused.com>
	<949E214D-7B59-4D8E-BE39-DC5A3145C6D7@endigotech.com>
	<1a1401cdd435$167f2940$437d7bc0$@bizfocused.com>
	<CAKzWOxWzY=401Cuq2=hP5yDArBSypOcBMuemZkNyfnwAt8WHkA@mail.gmail.com>
Message-ID: <1D7D4D6B-5E9C-4BB4-BA28-9CF93CEFA008@endigotech.com>

+1 we have that in our configs as well.

Sent from my iPhone

On Dec 7, 2012, at 5:01 PM, Michael Collins <msc at freeswitch.org> wrote:

> Sean,
> 
> Is the Polycom behind NAT and FS on a public IP? If so I think you'll need to modify the SIP profile. There's an "NDLB-force-rport" param. Check the wiki, but I think there's several values ("false","true","safe") you can try. IIRC "safe" is the one you'd try. This kinda sounds like the Polycom not supporting rport bug that they've ignored for the past decade...
> 
> -MC
> 
> On Thu, Dec 6, 2012 at 8:41 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>> Thanks Brian.  I have looked up more info and that is correct.
>> 
>> However, I still get 401 Unauthorized.
>> 
>>  
>> 
>> Any other ideas anyone?
>> 
>>  
>> 
>> I am ready to return them and just by Cisco SPA504Gs.  I was hoping to add another choice for my customers but I am yet to come across any decent documentation from Polycom.  If I can?t even get them to log in, I am not hopeful about advanced features.
>> 
>>  
>> 
>> Sean
>> 
>>  
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster
>> Sent: Thursday, December 06, 2012 5:38 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch
>> 
>>  
>> 
>> I think Address should be <extension>@<host name/ip>
>> 
>> Sent from my iPhone
>> 
>> 
>> On Dec 6, 2012, at 5:25 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:
>> 
>> I am new to Polycom phones.  We have always used CISCO 5xx Series.  I have been trying to use the WEB interface.  The example in the WIKI uses tftp through DHCP option 60.  MY home router/gateway?s DHCP does not support option 60.
>> 
>>  
>> 
>> I am unable to get the Polycom 335 to register, it keeps getting ?Unauthorized?.  I have double checked by having my Cisco phone register with this information and it works fine.  I must be missing something.
>> 
>>  
>> 
>> On the Polycom Web Interface I have the following values:
>> 
>> SIP TAB:
>> 
>> SERVERS:
>> 
>> Outbound Proxy is blank
>> 
>> SERVER 1:
>> 
>> Address: <my server?s FQDN >
>> 
>> Port: 5060
>> 
>> Transport: UDPOnly
>> 
>> Expires:3600
>> 
>> Register: 1
>> 
>> Retry Timeout: 0
>> 
>> Retry Max Count: 3
>> 
>> Line Seize Timeout: 30
>> 
>> SERVER 2: <same as server 1 for now except port 5062)
>> 
>> LINE TAB:
>> 
>> LINE 1:
>> 
>> Display Name: 228
>> 
>> Address: 228
>> 
>> Auth UserID:228
>> 
>> Auth Password: <actual password here>
>> 
>> Label: 228
>> 
>> Type: Private
>> 
>> 3rd Party Name: blank
>> 
>> Number of Line Keys: 1
>> 
>> Calls per line: 24
>> 
>> SERVER 1:
>> 
>> Address: <my server?s FQDN >
>> 
>> Port: 5060
>> 
>> Transport: UDPOnly
>> 
>> Expires:3600
>> 
>> Register: 1
>> 
>> Retry Timeout: 0
>> 
>> Retry Max Count: 3
>> 
>> Line Seize Timeout: 30
>> 
>> SERVER 2: <same as server 1 for now)
>> 
>> Everything else is the defaults.
>> 
>>  
>> 
>> The resulting sofia trace messages:
>> 
>>    ------------------------------------------------------------------------
>> 
>> recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.210138:
>> 
>>    ------------------------------------------------------------------------
>> 
>>    REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0
>> 
>>    Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C
>> 
>>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B
>> 
>>    To: <sip:228 at fs_bfis.bizfocused.com>
>> 
>>    CSeq: 1 REGISTER
>> 
>>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47
>> 
>>    Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
>> 
>>    User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069
>> 
>>    Accept-Language: en
>> 
>>    Max-Forwards: 70
>> 
>>    Expires: 3600
>> 
>>    Content-Length: 0
>> 
>>   
>> 
>>    ------------------------------------------------------------------------
>> 
>> send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.211063:
>> 
>>    ------------------------------------------------------------------------
>> 
>>    SIP/2.0 401 Unauthorized
>> 
>>    Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57
>> 
>>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B
>> 
>>    To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre
>> 
>>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47
>> 
>>    CSeq: 1 REGISTER
>> 
>>    User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
>> 
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> 
>>    Supported: precondition, path, replaces
>> 
>>    WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"
>> 
>>    Content-Length: 0
>> 
>>   
>> 
>>    ------------------------------------------------------------------------
>> 
>> recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.712339:
>> 
>>    ------------------------------------------------------------------------
>> 
>>    REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0
>> 
>>    Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C
>> 
>>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B
>> 
>>    To: <sip:228 at fs_bfis.bizfocused.com>
>> 
>>    CSeq: 1 REGISTER
>> 
>>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47
>> 
>>    Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
>> 
>>    User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069
>> 
>>    Accept-Language: en
>> 
>>    Max-Forwards: 70
>> 
>>    Expires: 3600
>> 
>>    Content-Length: 0
>> 
>>  
>> 
>>   ------------------------------------------------------------------------
>> 
>> send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.712479:
>> 
>>    ------------------------------------------------------------------------
>> 
>>    SIP/2.0 401 Unauthorized
>> 
>>   Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57
>> 
>>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B
>> 
>>   To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre
>> 
>>   Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47
>> 
>>    CSeq: 1 REGISTER
>> 
>>    User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
>> 
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> 
>>    Supported: precondition, path, replaces
>> 
>>    WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"
>> 
>>    Content-Length: 0
>> 
>>    ------------------------------------------------------------------------
>> 
>>  
>> 
>>  
>> 
>> Any ideas?
>> 
>>  
>> 
>>  
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 william.king at quentustech.com  Sat Dec  8 01:22:15 2012
From: william.king at quentustech.com (William King)
Date: Fri, 07 Dec 2012 14:22:15 -0800
Subject: [Freeswitch-users] pgsql as dsn
In-Reply-To: <E3AD9560-8E69-4588-B929-A2AD4B401C3D@carmickle.com>
References: <DF2B131E-2C82-4D7E-B4CE-085432BD3C0F@carmickle.com>
	<CAKbxfG_iwHwBpF7kcNL=W36KvawJ2zy-0VSmH3VnXobmn1Voxg@mail.gmail.com>
	<E3AD9560-8E69-4588-B929-A2AD4B401C3D@carmickle.com>
Message-ID: <50C26C17.60900@quentustech.com>

Debian packaging patched and pushed into master.

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 12/07/2012 11:57 AM, Frank Carmickle wrote:
> Thank you Tony.  I saw that the deb packages had depends of libpq so I thought it was built.  That fixed it.
> 
> --FC
> 
> 
> On Dec 6, 2012, at 4:09 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
>> try this from the build root
>>
>> ./configure --enable-core-pgsql-support --no-create --no-recursion
>>
>> then 
>>
>> make install
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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


From sdevoy at bizfocused.com  Sat Dec  8 01:33:51 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 7 Dec 2012 17:33:51 -0500
Subject: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch
In-Reply-To: <1D7D4D6B-5E9C-4BB4-BA28-9CF93CEFA008@endigotech.com>
References: <186801cdd400$87a246a0$96e6d3e0$@bizfocused.com>	<949E214D-7B59-4D8E-BE39-DC5A3145C6D7@endigotech.com>	<1a1401cdd435$167f2940$437d7bc0$@bizfocused.com>	<CAKzWOxWzY=401Cuq2=hP5yDArBSypOcBMuemZkNyfnwAt8WHkA@mail.gmail.com>
	<1D7D4D6B-5E9C-4BB4-BA28-9CF93CEFA008@endigotech.com>
Message-ID: <216801cdd4ca$ef0de540$cd29afc0$@bizfocused.com>

Thank you    thank you    thank you    thank you     thank you   thank you     thank you    thank you!

 

I had outbound calling working, but not inbound!!!

 

I added:

<variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>

<param name="NDLB-force-rport" value="true"/>

 

Now it is perfect.  Sorry to be so needy and thank you so much for all your work here.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster
Sent: Friday, December 07, 2012 5:09 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch

 

+1 we have that in our configs as well.

Sent from my iPhone


On Dec 7, 2012, at 5:01 PM, Michael Collins <msc at freeswitch.org> wrote:

Sean,

Is the Polycom behind NAT and FS on a public IP? If so I think you'll need to modify the SIP profile. There's an "NDLB-force-rport" param. Check the wiki, but I think there's several values ("false","true","safe") you can try. IIRC "safe" is the one you'd try. This kinda sounds like the Polycom not supporting rport bug that they've ignored for the past decade...

-MC

On Thu, Dec 6, 2012 at 8:41 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Thanks Brian.  I have looked up more info and that is correct.

However, I still get 401 Unauthorized.

 

Any other ideas anyone?

 

I am ready to return them and just by Cisco SPA504Gs.  I was hoping to add another choice for my customers but I am yet to come across any decent documentation from Polycom.  If I can?t even get them to log in, I am not hopeful about advanced features.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster
Sent: Thursday, December 06, 2012 5:38 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch

 

I think Address should be <extension>@<host name/ip>

Sent from my iPhone


On Dec 6, 2012, at 5:25 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

I am new to Polycom phones.  We have always used CISCO 5xx Series.  I have been trying to use the WEB interface.  The example in the WIKI uses tftp through DHCP option 60.  MY home router/gateway?s DHCP does not support option 60.

 

I am unable to get the Polycom 335 to register, it keeps getting ?Unauthorized?.  I have double checked by having my Cisco phone register with this information and it works fine.  I must be missing something.

 

On the Polycom Web Interface I have the following values:

SIP TAB:

SERVERS:

Outbound Proxy is blank

SERVER 1:

Address: <my server?s FQDN >

Port: 5060

Transport: UDPOnly

Expires:3600

Register: 1

Retry Timeout: 0

Retry Max Count: 3

Line Seize Timeout: 30

SERVER 2: <same as server 1 for now except port 5062)

LINE TAB:

LINE 1:

Display Name: 228

Address: 228

Auth UserID:228

Auth Password: <actual password here>

Label: 228

Type: Private

3rd Party Name: blank

Number of Line Keys: 1

Calls per line: 24

SERVER 1:

Address: <my server?s FQDN >

Port: 5060

Transport: UDPOnly

Expires:3600

Register: 1

Retry Timeout: 0

Retry Max Count: 3

Line Seize Timeout: 30

SERVER 2: <same as server 1 for now)

Everything else is the defaults.

 

The resulting sofia trace messages:

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

recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.210138:

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

   REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0

   Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C

   From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B

   To: <sip:228 at fs_bfis.bizfocused.com>

   CSeq: 1 REGISTER

   Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47

   Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"

   User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069

   Accept-Language: en

   Max-Forwards: 70

   Expires: 3600

   Content-Length: 0

   

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

send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.211063:

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

   SIP/2.0 401 Unauthorized

   Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57

   From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B

   To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre

   Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47

   CSeq: 1 REGISTER

   User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: precondition, path, replaces

   WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"

   Content-Length: 0

   

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

recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.712339:

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

   REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0

   Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C

   From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B

   To: <sip:228 at fs_bfis.bizfocused.com>

   CSeq: 1 REGISTER

   Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47

   Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"

   User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069

   Accept-Language: en

   Max-Forwards: 70

   Expires: 3600

   Content-Length: 0

 

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

send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.712479:

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

   SIP/2.0 401 Unauthorized

  Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57

   From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B

  To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre

  Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47

   CSeq: 1 REGISTER

   User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: precondition, path, replaces

   WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"

   Content-Length: 0

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

 

 

Any ideas?

 

 

_________________________________________________________________________
Professional FreeSWITCH Consulting 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

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




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org



_________________________________________________________________________
Professional FreeSWITCH Consulting 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

-------------- next part --------------
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From msc at freeswitch.org  Sat Dec  8 01:53:46 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 Dec 2012 14:53:46 -0800
Subject: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch
In-Reply-To: <216801cdd4ca$ef0de540$cd29afc0$@bizfocused.com>
References: <186801cdd400$87a246a0$96e6d3e0$@bizfocused.com>
	<949E214D-7B59-4D8E-BE39-DC5A3145C6D7@endigotech.com>
	<1a1401cdd435$167f2940$437d7bc0$@bizfocused.com>
	<CAKzWOxWzY=401Cuq2=hP5yDArBSypOcBMuemZkNyfnwAt8WHkA@mail.gmail.com>
	<1D7D4D6B-5E9C-4BB4-BA28-9CF93CEFA008@endigotech.com>
	<216801cdd4ca$ef0de540$cd29afc0$@bizfocused.com>
Message-ID: <CAKzWOxWbkaM7kgmEMmgr2TF1Gbkp7V0uU-4XXbJ6ikonEF59Wg@mail.gmail.com>

No problemo. Now you can help the next desperate Polycom user that comes
along. :)
Also, if this is a business then you can ask your boss to make a donation
to the project. We also have
wishlists<http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_So_where_do_I_find_wishlists_for_the_developers_of_this_awesome_software.3F>if
you're feeling especially thankful! :P

In all seriousness, though, please be sure to pass this hard-earned
knowledge on to others to save them the heartache you experienced.
-MC

On Fri, Dec 7, 2012 at 2:33 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Thank you    thank you    thank you    thank you     thank you   thank you
>     thank you    thank you!****
>
> ** **
>
> I had outbound calling working, but not inbound!!!****
>
> ** **
>
> I added:****
>
> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>**
> **
>
> <param name="NDLB-force-rport" value="true"/>****
>
> ** **
>
> Now it is perfect.  Sorry to be so needy and thank you so much for all
> your work here.****
>
> ** **
>
> Sean****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian Foster
> *Sent:* Friday, December 07, 2012 5:09 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch**
> **
>
> ** **
>
> +1 we have that in our configs as well.
>
> Sent from my iPhone****
>
>
> On Dec 7, 2012, at 5:01 PM, Michael Collins <msc at freeswitch.org> wrote:***
> *
>
> Sean,
>
> Is the Polycom behind NAT and FS on a public IP? If so I think you'll need
> to modify the SIP profile. There's an "NDLB-force-rport" param. Check the
> wiki, but I think there's several values ("false","true","safe") you can
> try. IIRC "safe" is the one you'd try. This kinda sounds like the Polycom
> not supporting rport bug that they've ignored for the past decade...
>
> -MC****
>
> On Thu, Dec 6, 2012 at 8:41 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:*
> ***
>
> Thanks Brian.  I have looked up more info and that is correct.****
>
> However, I still get 401 Unauthorized.****
>
>  ****
>
> Any other ideas anyone?****
>
>  ****
>
> I am ready to return them and just by Cisco SPA504Gs.  I was hoping to add
> another choice for my customers but I am yet to come across any decent
> documentation from Polycom.  If I can?t even get them to log in, I am not
> hopeful about advanced features.****
>
>  ****
>
> Sean****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian Foster
> *Sent:* Thursday, December 06, 2012 5:38 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] POLYCOM 335 registration w/ Freeswitch**
> **
>
>  ****
>
> I think Address should be <extension>@<host name/ip>
>
> Sent from my iPhone****
>
>
> On Dec 6, 2012, at 5:25 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:***
> *
>
> I am new to Polycom phones.  We have always used CISCO 5xx Series.  I have
> been trying to use the WEB interface.  The example in the WIKI uses tftp
> through DHCP option 60.  MY home router/gateway?s DHCP does not support
> option 60.****
>
>  ****
>
> I am unable to get the Polycom 335 to register, it keeps getting
> ?Unauthorized?.  I have double checked by having my Cisco phone register
> with this information and it works fine.  I must be missing something.****
>
>  ****
>
> On the Polycom Web Interface I have the following values:****
>
> SIP TAB:****
>
> SERVERS:****
>
> Outbound Proxy is blank****
>
> SERVER 1:****
>
> Address: <my server?s FQDN >****
>
> Port: 5060****
>
> Transport: UDPOnly****
>
> Expires:3600****
>
> Register: 1****
>
> Retry Timeout: 0****
>
> Retry Max Count: 3****
>
> Line Seize Timeout: 30****
>
> SERVER 2: <same as server 1 for now except port 5062)****
>
> LINE TAB:****
>
> LINE 1:****
>
> Display Name: 228****
>
> Address: 228****
>
> Auth UserID:228****
>
> Auth Password: <actual password here>****
>
> Label: 228****
>
> Type: Private****
>
> 3rd Party Name: blank****
>
> Number of Line Keys: 1****
>
> Calls per line: 24****
>
> SERVER 1:****
>
> Address: <my server?s FQDN >****
>
> Port: 5060****
>
> Transport: UDPOnly****
>
> Expires:3600****
>
> Register: 1****
>
> Retry Timeout: 0****
>
> Retry Max Count: 3****
>
> Line Seize Timeout: 30****
>
> SERVER 2: <same as server 1 for now)****
>
> Everything else is the defaults.****
>
>  ****
>
> The resulting sofia trace messages:****
>
>    ------------------------------------------------------------------------
> ****
>
> recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.210138:****
>
>    ------------------------------------------------------------------------
> ****
>
>    REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0****
>
>    Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C****
>
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B****
>
>    To: <sip:228 at fs_bfis.bizfocused.com>****
>
>    CSeq: 1 REGISTER****
>
>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47****
>
>    Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL,
> OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"****
>
>    User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069****
>
>    Accept-Language: en****
>
>    Max-Forwards: 70****
>
>    Expires: 3600****
>
>    Content-Length: 0****
>
>    ****
>
>    ------------------------------------------------------------------------
> ****
>
> send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.211063:****
>
>    ------------------------------------------------------------------------
> ****
>
>    SIP/2.0 401 Unauthorized****
>
>    Via: SIP/2.0/UDP
> 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57****
>
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B****
>
>    To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre****
>
>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47****
>
>    CSeq: 1 REGISTER****
>
>    User-Agent:
> FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
> ****
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE****
>
>    Supported: precondition, path, replaces****
>
>    WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com",
> nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"***
> *
>
>    Content-Length: 0****
>
>    ****
>
>    ------------------------------------------------------------------------
> ****
>
> recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.712339:****
>
>    ------------------------------------------------------------------------
> ****
>
>    REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0****
>
>    Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C****
>
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B****
>
>    To: <sip:228 at fs_bfis.bizfocused.com>****
>
>    CSeq: 1 REGISTER****
>
>    Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47****
>
>    Contact: <sip:228 at 10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL,
> OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"****
>
>    User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069****
>
>    Accept-Language: en****
>
>    Max-Forwards: 70****
>
>    Expires: 3600****
>
>    Content-Length: 0****
>
>  ****
>
>   ------------------------------------------------------------------------
> ****
>
> send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.712479:****
>
>    ------------------------------------------------------------------------
> ****
>
>    SIP/2.0 401 Unauthorized****
>
>   Via: SIP/2.0/UDP
> 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57****
>
>    From: "228" <sip:228 at fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B****
>
>   To: <sip:228 at fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre****
>
>   Call-ID: ca6990df-b25553e-44d1bb49 at 10.10.40.47****
>
>    CSeq: 1 REGISTER****
>
>    User-Agent:
> FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
> ****
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE****
>
>    Supported: precondition, path, replaces****
>
>    WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com",
> nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth"***
> *
>
>    Content-Length: 0****
>
>    ------------------------------------------------------------------------
> ****
>
>  ****
>
>  ****
>
> Any ideas?****
>
>  ****
>
>  ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
> ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20121207/15477921/attachment-0001.html 

From shaik.bawajan at gmail.com  Sat Dec  8 09:07:51 2012
From: shaik.bawajan at gmail.com (bawajan)
Date: Fri, 7 Dec 2012 22:07:51 -0800 (PST)
Subject: [Freeswitch-users] having problem using java with esl library
In-Reply-To: <ADF32A4A0F985A428C85B277B8E1C9D80F4924EE@NY1-EXMB-02.ip-soft.net>
References: <CAB3HJytM41noZgqhqsa492MC+nNEc4S7AvWmMi+8X4UcOdZA_w@mail.gmail.com>
	<ADF32A4A0F985A428C85B277B8E1C9D80F4924EE@NY1-EXMB-02.ip-soft.net>
Message-ID: <1354946871791-7585328.post@n2.nabble.com>

Thanks,

I will try this out.

Also i want to mentioned that, i saw one warning when restarting the
freeswitch ie., 

switch_event.c:632 Activate Eventing Engine.
2012-12-08 11:09:39.570194 [WARNING] switch_event.c:607 Create additional
event dispatch thread 0

But i have added the parameter initial-event-threads in switch.conf.xml and
still getting this warning.

So, kindly let me know where to configure it and how to resolve it.

Thanks in Advance



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/having-problem-using-java-with-esl-library-tp7585250p7585328.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From a.venugopan at mundio.com  Sat Dec  8 13:33:25 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Sat, 8 Dec 2012 10:33:25 +0000
Subject: [Freeswitch-users] sip registration
In-Reply-To: <CAKzWOxVhphxwn8EhfwTfo2Ti5OgTNp2Q9hKsB_Y-+4p+P9Ns4A@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF2339A49@Mail-Kilo.squay.com>
	<CAFiqYu=cpMkTxdE99smQBM6NgitNQ5ZpePwfSYr_R23dpj7Hng@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339B39@Mail-Kilo.squay.com>
	<CAKzWOxUW21XZte0AoVDqTtC6vkJ4FmA3sbP5LgJVK5bj4+1Sbw@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339D42@Mail-Kilo.squay.com>
	<CAKzWOxVMWfR6vk8ODKkND1Tjxj-d5VFYn9U+dDoNyB_t2dE05w@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233A02B@Mail-Kilo.squay.com>
	<CAKzWOxVqn1tGWgHKFs+vCEDpXUg2rNyvpbw5-bBSPw846hg7Xw@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233A0D5@Mail-Kilo.squay.com>
	<CAKzWOxVhphxwn8EhfwTfo2Ti5OgTNp2Q9hKsB_Y-+4p+P9Ns4A@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233A3B2@Mail-Kilo.squay.com>

Hi Michael,

Thanks. I have gone through that wiki already, but in our freeswitch we don't use XML to check the userid and password rather we use DB in some script(not sure where). Also mod_xml_curl has been disabled too.
But in none of the lua script I find that its verifying the password from the URL to DB too. SO am confused.

Sorry I know am missing something but not sure where.

Thanks again.

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 07 December 2012 21:56
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

Archana,

Did you see this page on the wiki?
http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Alphanumeric_to_numeric_user_mapping

It's been there for years. I promise you that the wiki has more information than you are giving it credit for. Just remember that our wiki is a microcosm of the Internet: it has what you're looking for but you need Google to search it and you have to dig through all the results. :)

Try that and see if it does what you need.

As far as your question goes, there isn't a "script" that checks it. Rather, the module that is doing the authentication will look in the XML data for the user id and password. If mod_xml_curl is enable then FS will go fetch the XML data from your server.

-MC
On Fri, Dec 7, 2012 at 1:21 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

I want to give some alphabets instead of number. I want to know which script checks this authentication name to corresponding DB table. Please let me know.
Thanks

Regards,
Archana

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: 06 December 2012 21:58
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

Use "100" for the authentication name as well.
-MC
On Thu, Dec 6, 2012 at 9:25 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
Thanks. I thought it will look in mod_sofia code. In the below screen I register the ID '100'. Now instead of '100' in "Authentication Name" I need to give some e-mail ID or name(Archana) which should validate in DB.
I tried giving a name in "Authentication Name" but the phone was not registered. Am not sure this authentication name is being looked in which column in table too.
Please let me know if this will be picked from any sofia code or any C script? Once we register in the below screen which script validates the Settings in freeswitch?

Sorry if am repeating the same question, but I could not get the exact code and am clueless.
Global SIP Settings
Top of Form
Basic SIP Authentication Settings



Screen Name

Screen Name 2

Phone Number

Caller ID

Authentication Name

Password

Bottom of Form


Regards,
Archana

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: 05 December 2012 20:34
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

When an event that requires a user lookup takes place then the system will look in the XML user directory unless it has been configured to look somewhere else. The other places to look are usually:
mod_xml_curl
One of the language like Lua, Perl, Python

If it's xml_curl then FS will do a POST to your web server in hopes of receiving back the necessary XML for the given user. It would be up to you to have your web server handle the request, poll the database, then format and return the XML data. See this wiki page<http://wiki.freeswitch.org/wiki/Xml_curl#How_do_I_configure_mod_xml_curl.3F> for more info on xml curl.

If it's a language then you'll have a "binding" in the conf file for the language that will handle the lookup. Again, your script will need to handle the communication with your database. See this wiki page<http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration> for more information.

Hope this helps.
-MC
On Wed, Dec 5, 2012 at 9:30 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
Thanks for the information. But sorry, how to access user_data API command.

Am not clear on the flow. Once we register domain and usernumber in sip what exactly happens? Which script picks up this domain and username and validates with our database?
Could you please provide me with an overview.

Many thanks

Regards,
Archana

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: 05 December 2012 17:13
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

If you're talking about the user configuration then yes, you could create an "email" parameter or variable and access it with the user_data API command.
-MC
On Wed, Dec 5, 2012 at 1:35 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
In that case can I have 1 more column say e-mail and can this e-mail be checked in DB instead of checking reg_user('100')? Is that feasible?
Also which code should be changed any idea please?

Regards,
Archana

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 Steven Ayre
Sent: 04 December 2012 19:51
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

You can have a user 'ana' in the domain 'gmail.com<http://gmail.com>'. Though using someone else's domain as local in your FS setup may not be a good idea.
You can't have a @ in the username itself (per the SIP standard, not limited to FreeSWITCH).

On 4 December 2012 18:00, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

Currently we register authentication name as say '100' in sip registration, this comes to freeswitch and it will check in our DB for 100 and if its present then registrations would be successful.

freeswitch at internal> show registrations
reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
100,fsfailover.uk01.com<http://fsfailover.uk01.com>,e4969067f9a8c098,sofia/internal/sip:100 at 192.168.2.234:5060;transport=udp,1354638871,192.168.2.234,5060,udp,squay-laptop-1.squay.com<http://squay-laptop-1.squay.com>

I want to change this 100 to some e-mail address, so instead of 100 it will be something like 'ana at gmail.com<mailto:ana at gmail.com>'. Can we do this? While coming to freeswitch whether there would be any issues?

Regards,
Archana


_________________________________________________________________________
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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

-------------- next part --------------
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From a.venugopan at mundio.com  Sat Dec  8 14:13:42 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Sat, 8 Dec 2012 11:13:42 +0000
Subject: [Freeswitch-users] sip registration
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233A3B2@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2339A49@Mail-Kilo.squay.com>
	<CAFiqYu=cpMkTxdE99smQBM6NgitNQ5ZpePwfSYr_R23dpj7Hng@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339B39@Mail-Kilo.squay.com>
	<CAKzWOxUW21XZte0AoVDqTtC6vkJ4FmA3sbP5LgJVK5bj4+1Sbw@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339D42@Mail-Kilo.squay.com>
	<CAKzWOxVMWfR6vk8ODKkND1Tjxj-d5VFYn9U+dDoNyB_t2dE05w@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233A02B@Mail-Kilo.squay.com>
	<CAKzWOxVqn1tGWgHKFs+vCEDpXUg2rNyvpbw5-bBSPw846hg7Xw@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233A0D5@Mail-Kilo.squay.com>
	<CAKzWOxVhphxwn8EhfwTfo2Ti5OgTNp2Q9hKsB_Y-+4p+P9Ns4A@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233A3B2@Mail-Kilo.squay.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233A3C3@Mail-Kilo.squay.com>

Hi,

I found this part of code in sofia_reg.c file. If I change authentication name alone to 'arch' its not getting registered. Is it something to do with below code?

    /* Optional check that auth name == SIP username */
        if ((regtype == REG_REGISTER) && sofia_test_pflag(profile, PFLAG_CHECKUSER)) {
                if (zstr(username) || zstr(to_user) || strcasecmp(to_user, username)) {
                        /* Names don't match, so fail */
                        if (profile->debug) {
                                switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "SIP username %s does not match auth username\n", switch_str_nil(to_user));
                        }
                        goto end;
                }
        }

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Archana Venugopan
Sent: 08 December 2012 10:33
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

Hi Michael,

Thanks. I have gone through that wiki already, but in our freeswitch we don't use XML to check the userid and password rather we use DB in some script(not sure where). Also mod_xml_curl has been disabled too.
But in none of the lua script I find that its verifying the password from the URL to DB too. SO am confused.

Sorry I know am missing something but not sure where.

Thanks again.

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 07 December 2012 21:56
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

Archana,

Did you see this page on the wiki?
http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Alphanumeric_to_numeric_user_mapping

It's been there for years. I promise you that the wiki has more information than you are giving it credit for. Just remember that our wiki is a microcosm of the Internet: it has what you're looking for but you need Google to search it and you have to dig through all the results. :)

Try that and see if it does what you need.

As far as your question goes, there isn't a "script" that checks it. Rather, the module that is doing the authentication will look in the XML data for the user id and password. If mod_xml_curl is enable then FS will go fetch the XML data from your server.

-MC
On Fri, Dec 7, 2012 at 1:21 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

I want to give some alphabets instead of number. I want to know which script checks this authentication name to corresponding DB table. Please let me know.
Thanks

Regards,
Archana

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: 06 December 2012 21:58
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

Use "100" for the authentication name as well.
-MC
On Thu, Dec 6, 2012 at 9:25 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
Thanks. I thought it will look in mod_sofia code. In the below screen I register the ID '100'. Now instead of '100' in "Authentication Name" I need to give some e-mail ID or name(Archana) which should validate in DB.
I tried giving a name in "Authentication Name" but the phone was not registered. Am not sure this authentication name is being looked in which column in table too.
Please let me know if this will be picked from any sofia code or any C script? Once we register in the below screen which script validates the Settings in freeswitch?

Sorry if am repeating the same question, but I could not get the exact code and am clueless.
Global SIP Settings
Top of Form
Basic SIP Authentication Settings



Screen Name

Screen Name 2

Phone Number

Caller ID

Authentication Name

Password

Bottom of Form


Regards,
Archana

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: 05 December 2012 20:34
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

When an event that requires a user lookup takes place then the system will look in the XML user directory unless it has been configured to look somewhere else. The other places to look are usually:
mod_xml_curl
One of the language like Lua, Perl, Python

If it's xml_curl then FS will do a POST to your web server in hopes of receiving back the necessary XML for the given user. It would be up to you to have your web server handle the request, poll the database, then format and return the XML data. See this wiki page<http://wiki.freeswitch.org/wiki/Xml_curl#How_do_I_configure_mod_xml_curl.3F> for more info on xml curl.

If it's a language then you'll have a "binding" in the conf file for the language that will handle the lookup. Again, your script will need to handle the communication with your database. See this wiki page<http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration> for more information.

Hope this helps.
-MC
On Wed, Dec 5, 2012 at 9:30 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
Thanks for the information. But sorry, how to access user_data API command.

Am not clear on the flow. Once we register domain and usernumber in sip what exactly happens? Which script picks up this domain and username and validates with our database?
Could you please provide me with an overview.

Many thanks

Regards,
Archana

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: 05 December 2012 17:13
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

If you're talking about the user configuration then yes, you could create an "email" parameter or variable and access it with the user_data API command.
-MC
On Wed, Dec 5, 2012 at 1:35 AM, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
In that case can I have 1 more column say e-mail and can this e-mail be checked in DB instead of checking reg_user('100')? Is that feasible?
Also which code should be changed any idea please?

Regards,
Archana

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 Steven Ayre
Sent: 04 December 2012 19:51
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip registration

You can have a user 'ana' in the domain 'gmail.com<http://gmail.com>'. Though using someone else's domain as local in your FS setup may not be a good idea.
You can't have a @ in the username itself (per the SIP standard, not limited to FreeSWITCH).

On 4 December 2012 18:00, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

Currently we register authentication name as say '100' in sip registration, this comes to freeswitch and it will check in our DB for 100 and if its present then registrations would be successful.

freeswitch at internal> show registrations
reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
100,fsfailover.uk01.com<http://fsfailover.uk01.com>,e4969067f9a8c098,sofia/internal/sip:100 at 192.168.2.234:5060;transport=udp,1354638871,192.168.2.234,5060,udp,squay-laptop-1.squay.com<http://squay-laptop-1.squay.com>

I want to change this 100 to some e-mail address, so instead of 100 it will be something like 'ana at gmail.com<mailto:ana at gmail.com>'. Can we do this? While coming to freeswitch whether there would be any issues?

Regards,
Archana


_________________________________________________________________________
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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From admin at blindi.net  Sat Dec  8 18:25:31 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sat, 8 Dec 2012 16:25:31 +0100 (CET)
Subject: [Freeswitch-users] Problem freeswitch and asterisk mediagateway
 bridge no direct connection
In-Reply-To: <CAFiqYunGtJ4qbbCyO-L-ppdpRfQDPgMK_3uAR2na61_GkjnyJA@mail.gmail.com>
References: <CACak_yxsejacKz6jAkdfdhdAV5tquAF4eBVR4a4yyC81kvuOvg@mail.gmail.com>
	<CAKzWOxXTDzOJ7Q0X4R+m06pG4R1w6kkkO7EA_7WzSbqO=crLZw@mail.gmail.com>
	<473CAD87-8625-4580-BEDC-08DE4475CE22@gmail.com>
	<CAFiqYunGtJ4qbbCyO-L-ppdpRfQDPgMK_3uAR2na61_GkjnyJA@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1212081544550.25422@1a-8140.antagus.de>

Hi,
i have setup a voicechat (social network  for handycap users).
Fs and asterisk works on the same host:
eth0 mainip fs
eth0:0 asteriskip
I connecting freeswitch (siprouter line managemnt), asterisk 
(mediagateway),
then i  connect from a sip provider
Fs answer the call, and forward to bridge to 
asterisk  fine. 
then i make a direct connection to fs: (no sipprovider)
via sipuri: "dorf at dorf.blindi.net"
fs answer the calls, but the bridge don.t works to asterisk.
the bridge hangup before asterisk answer the call from fs

the problem is only to connect fs from the world. 
Why works only fine from a sip provider?
must i edit the internal profile to bridge calls to another instance?

Can you help please? thanks.



i connection

---------------
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 covici at ccs.covici.com  Sat Dec  8 20:02:11 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 08 Dec 2012 12:02:11 -0500
Subject: [Freeswitch-users] Problem freeswitch and asterisk mediagateway
	bridge no direct connection
In-Reply-To: <Pine.LNX.4.64.1212081544550.25422@1a-8140.antagus.de>
References: <CACak_yxsejacKz6jAkdfdhdAV5tquAF4eBVR4a4yyC81kvuOvg@mail.gmail.com>
	<CAKzWOxXTDzOJ7Q0X4R+m06pG4R1w6kkkO7EA_7WzSbqO=crLZw@mail.gmail.com>
	<473CAD87-8625-4580-BEDC-08DE4475CE22@gmail.com>
	<CAFiqYunGtJ4qbbCyO-L-ppdpRfQDPgMK_3uAR2na61_GkjnyJA@mail.gmail.com>
	<Pine.LNX.4.64.1212081544550.25422@1a-8140.antagus.de>
Message-ID: <7523.1354986131@ccs.covici.com>

Why use asterisk at all?  What is it doing?

Thomas Hoellriegel <admin at blindi.net> wrote:

> Hi,
> i have setup a voicechat (social network  for handycap users).
> Fs and asterisk works on the same host:
> eth0 mainip fs
> eth0:0 asteriskip
> I connecting freeswitch (siprouter line managemnt), asterisk
> (mediagateway),
> then i  connect from a sip provider
> Fs answer the call, and forward to bridge to asterisk  fine. then i
> make a direct connection to fs: (no sipprovider)
> via sipuri: "dorf at dorf.blindi.net"
> fs answer the calls, but the bridge don.t works to asterisk.
> the bridge hangup before asterisk answer the call from fs
> 
> the problem is only to connect fs from the world. Why works only fine
> from a sip provider?
> must i edit the internal profile to bridge calls to another instance?
> 
> Can you help please? thanks.
> 
> 
> 
> i connection
> 
> ---------------
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 geddes.jeffrey at gmail.com  Sat Dec  8 20:32:45 2012
From: geddes.jeffrey at gmail.com (Jeff Geddes)
Date: Sat, 8 Dec 2012 13:32:45 -0400
Subject: [Freeswitch-users] xml_curl result not found
Message-ID: <CALAD2r74Cq8pDd5LRFZJvxAnfgZ6E=mhu0GsdJQkLew8WQsQhA@mail.gmail.com>

I'm using xml_curl to generate a dialplan and send back result status="not
found" if something throws an error.  I don't want it to fail back to the
in-memory dialplan if there is an error.  Is there a way to do this?

thanks
jeff
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From avi at avimarcus.net  Sat Dec  8 21:10:29 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 8 Dec 2012 20:10:29 +0200
Subject: [Freeswitch-users] xml_curl result not found
In-Reply-To: <CALAD2r74Cq8pDd5LRFZJvxAnfgZ6E=mhu0GsdJQkLew8WQsQhA@mail.gmail.com>
References: <CALAD2r74Cq8pDd5LRFZJvxAnfgZ6E=mhu0GsdJQkLew8WQsQhA@mail.gmail.com>
Message-ID: <CALNZuwkFdK6XvE4eQO_RBF70YQFbaRwqz7MEzs2biZaBTOdo0g@mail.gmail.com>

Why send a not found? Send a hangup with a failure reason, and that will be
the end of processing.

-Avi


On Sat, Dec 8, 2012 at 7:32 PM, Jeff Geddes <geddes.jeffrey at gmail.com>wrote:

> I'm using xml_curl to generate a dialplan and send back result status="not
> found" if something throws an error.  I don't want it to fail back to the
> in-memory dialplan if there is an error.  Is there a way to do this?
>
> thanks
> jeff
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 cal.leeming at simplicitymedialtd.co.uk  Sat Dec  8 23:48:33 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sat, 8 Dec 2012 20:48:33 +0000
Subject: [Freeswitch-users] xml_curl result not found
In-Reply-To: <CALAD2r74Cq8pDd5LRFZJvxAnfgZ6E=mhu0GsdJQkLew8WQsQhA@mail.gmail.com>
References: <CALAD2r74Cq8pDd5LRFZJvxAnfgZ6E=mhu0GsdJQkLew8WQsQhA@mail.gmail.com>
Message-ID: <CAHKQagHV6UbM4OReAYJ3Bg2gNzoJS=5UVp9Cfp51Ubn+tTAoSA@mail.gmail.com>

Exclude any dialplan sections from your config, and thus it won't have
anything to fall back onto.

However, if any other modules are configured to look for configs (??) then
you may need to use the approach Avi mentioned, i.e. sending back a hangup
with failure reason.

Cal

On Sat, Dec 8, 2012 at 5:32 PM, Jeff Geddes <geddes.jeffrey at gmail.com>wrote:

> I'm using xml_curl to generate a dialplan and send back result status="not
> found" if something throws an error.  I don't want it to fail back to the
> in-memory dialplan if there is an error.  Is there a way to do this?
>
> thanks
> jeff
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 cal.leeming at simplicitymedialtd.co.uk  Sun Dec  9 00:09:54 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sat, 8 Dec 2012 21:09:54 +0000
Subject: [Freeswitch-users] IVR menu delay - 1 second between events
In-Reply-To: <CAHKQagH-5HwgRpP77bVSLkOusVzMH6bU68Ox=jR7fvS7X-hkFg@mail.gmail.com>
References: <CAHKQagF3C76QiybFNrS7GCdiGqaTbtPpQzD4wFivtJ+EK1CirQ@mail.gmail.com>
	<CAPNTnnrVW4Vpj4LcaNyeQrcmYq99DDF26E3UXpFoJyf=-q11fg@mail.gmail.com>
	<CAHKQagG1dW+MUtLYgoJHN0mWa0xowKGo4-VFR4wV2nZqcMKWiw@mail.gmail.com>
	<CAHKQagH-5HwgRpP77bVSLkOusVzMH6bU68Ox=jR7fvS7X-hkFg@mail.gmail.com>
Message-ID: <CAHKQagFQBeB+ho+rrniqZiqS1aocWy6BNN16Lv_2xm-mFDr+Nw@mail.gmail.com>

I've raised a JIRA ticket for this;
http://jira.freeswitch.org/browse/FS-4924

Cal

On Fri, Dec 7, 2012 at 8:08 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Just trawled through the source and found this;
>
> src/switch_ivr_menu.c:
> if (status == SWITCH_STATUS_SUCCESS) {
>  status = switch_ivr_sleep(session, 1000, SWITCH_FALSE, NULL);
> }
>
> So, I changed 1000 to 10, the delay has now disappeared and it seems to
> work perfectly.
>
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From cal.leeming at simplicitymedialtd.co.uk  Sun Dec  9 01:18:37 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sat, 8 Dec 2012 22:18:37 +0000
Subject: [Freeswitch-users] recovery_profile_name - any explanation?
Message-ID: <CAHKQagH-b0waBO2jQS5B0_GBZcJvm8T5+_Uugk4QHJ_cK_=8mg@mail.gmail.com>

Hello,

I've just been trawling Google for an explanation of what exactly the
variable '*recovery_profile_name' means, but no luck.*
*
*
*It seems to relate to the following change from 4 months ago;*
https://github.com/freeSwitch/freeswitch/commit/2a8841ab666ec23ceb5688587bb14d16bf193b77

The commit message states 'change mod_sofia to use new core based recovery
engine'.

But, it is unclear when this variable should be relied upon, and under what
circumstances it will be used under.

Any clarifications would be good, I will of course put in time to update
the wiki too.

Thanks

Cal
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From bdfoster at endigotech.com  Sun Dec  9 02:05:47 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 8 Dec 2012 18:05:47 -0500
Subject: [Freeswitch-users] Hello hackers!
Message-ID: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>

Regarding a recent mailing list posting that included some of my IP addresses, most of you don't know that I do set up honeypots in hopes of catching some of the bad apples that try and hack into our phone systems. We have a centralized list of Bad IP's that end up getting sent to all of our other servers. Today, one of those servers was an IT guy that works for one of my clients. He has since been fired. If anyone is interested in the 180,000 IP's I've collected...sorry you can't have 'em.

-BDF

Sent from my iPhone

From admin at blindi.net  Sun Dec  9 02:31:34 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sun, 9 Dec 2012 00:31:34 +0100 (CET)
Subject: [Freeswitch-users] Problem freeswitch and asterisk mediagateway
 bridge no direct connection
In-Reply-To: <7523.1354986131@ccs.covici.com>
References: <CACak_yxsejacKz6jAkdfdhdAV5tquAF4eBVR4a4yyC81kvuOvg@mail.gmail.com>
	<CAKzWOxXTDzOJ7Q0X4R+m06pG4R1w6kkkO7EA_7WzSbqO=crLZw@mail.gmail.com>
	<473CAD87-8625-4580-BEDC-08DE4475CE22@gmail.com>
	<CAFiqYunGtJ4qbbCyO-L-ppdpRfQDPgMK_3uAR2na61_GkjnyJA@mail.gmail.com>
	<Pine.LNX.4.64.1212081544550.25422@1a-8140.antagus.de>
	<7523.1354986131@ccs.covici.com>
Message-ID: <Pine.LNX.4.64.1212090006140.28277@1a-8140.antagus.de>

Am 08.12.12 um 12:02 schrieb covici at ccs.covici.com:

> Why use asterisk at all?  What is it doing?

Fs make the maimmenu and Asterisk make the interface:
Fs have extensions to forward to asterisk on the same host.
Asterisk make the folowing functions:
a voice mail forum messageboard to leave messages for all users,
chat features; who is online, (onlinelist)
conferencemanagemnt: drop user from conference; mute user,
speak a system 
broadcastmessage to all current online users,
and more; voicemail, adressbook, blacklist distrobutionlist, whitelist, 
autoplay function in forums, calleridbased autologin, messagenotification, 
wakeupcalls,

Systemadministrationfeatures:
delete account, reset user passwords, block and
unblock acounts

I have programed a full features  chat system to help blind users 
via voice. (barrierfreee)
People call these chat for free, and enter digits. For
information.
Example:
press 1 for time and date, 2, for messageboards, 4 who ist online and so 
on.
Fs makes the maimenu, and asterisk make the rest easy.

This Chat is works in german language.
Then you enter the these instruction, press 2 you become a free 
voicemailbox to speak with some one else.

Fs and asterisk is very nice to program a voicechat platform.


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 geddes.jeffrey at gmail.com  Sun Dec  9 02:57:56 2012
From: geddes.jeffrey at gmail.com (Jeff Geddes)
Date: Sat, 8 Dec 2012 19:57:56 -0400
Subject: [Freeswitch-users] xml_curl result not found
Message-ID: <CALAD2r5J=sX=BwQTQnrU8vKwtvbbcfSRaiO-ZkHAp1GrY0h-3w@mail.gmail.com>

thanks guys

Avi,  your suggestion makes perfect sense, that's what I'll do

jeff

On Sat, Dec 8, 2012 at 6:19 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

> 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. Problem freeswitch and asterisk mediagateway bridge no direct
>       connection (Thomas Hoellriegel)
>    2. Re: Problem freeswitch and asterisk mediagateway  bridge no
>       direct connection (covici at ccs.covici.com)
>    3. xml_curl result not found (Jeff Geddes)
>    4. Re: xml_curl result not found (Avi Marcus)
>    5. Re: xml_curl result not found (Cal Leeming [Simplicity Media Ltd])
>    6. Re: IVR menu delay - 1 second between events
>       (Cal Leeming [Simplicity Media Ltd])
>    7. recovery_profile_name - any explanation?
>       (Cal Leeming [Simplicity Media Ltd])
>
>
> ---------- Forwarded message ----------
> From: Thomas Hoellriegel <admin at blindi.net>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Sat, 8 Dec 2012 16:25:31 +0100 (CET)
> Subject: [Freeswitch-users] Problem freeswitch and asterisk mediagateway
> bridge no direct connection
> Hi,
> i have setup a voicechat (social network  for handycap users).
> Fs and asterisk works on the same host:
> eth0 mainip fs
> eth0:0 asteriskip
> I connecting freeswitch (siprouter line managemnt), asterisk
> (mediagateway),
> then i  connect from a sip provider
> Fs answer the call, and forward to bridge to asterisk  fine. then i make a
> direct connection to fs: (no sipprovider)
> via sipuri: "dorf at dorf.blindi.net"
> fs answer the calls, but the bridge don.t works to asterisk.
> the bridge hangup before asterisk answer the call from fs
>
> the problem is only to connect fs from the world. Why works only fine from
> a sip provider?
> must i edit the internal profile to bridge calls to another instance?
>
> Can you help please? thanks.
>
>
>
> i connection
>
> ---------------
> 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>
>
>
> ---------- Forwarded message ----------
> From: covici at ccs.covici.com
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Sat, 08 Dec 2012 12:02:11 -0500
> Subject: Re: [Freeswitch-users] Problem freeswitch and asterisk
> mediagateway bridge no direct connection
> Why use asterisk at all?  What is it doing?
>
> Thomas Hoellriegel <admin at blindi.net> wrote:
>
> > Hi,
> > i have setup a voicechat (social network  for handycap users).
> > Fs and asterisk works on the same host:
> > eth0 mainip fs
> > eth0:0 asteriskip
> > I connecting freeswitch (siprouter line managemnt), asterisk
> > (mediagateway),
> > then i  connect from a sip provider
> > Fs answer the call, and forward to bridge to asterisk  fine. then i
> > make a direct connection to fs: (no sipprovider)
> > via sipuri: "dorf at dorf.blindi.net"
> > fs answer the calls, but the bridge don.t works to asterisk.
> > the bridge hangup before asterisk answer the call from fs
> >
> > the problem is only to connect fs from the world. Why works only fine
> > from a sip provider?
> > must i edit the internal profile to bridge calls to another instance?
> >
> > Can you help please? thanks.
> >
> >
> >
> > i connection
> >
> > ---------------
> > 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
>
>
>
> ---------- Forwarded message ----------
> From: Jeff Geddes <geddes.jeffrey at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Date: Sat, 8 Dec 2012 13:32:45 -0400
> Subject: [Freeswitch-users] xml_curl result not found
> I'm using xml_curl to generate a dialplan and send back result status="not
> found" if something throws an error.  I don't want it to fail back to the
> in-memory dialplan if there is an error.  Is there a way to do this?
>
> thanks
> jeff
>
>
> ---------- Forwarded message ----------
> From: Avi Marcus <avi at avimarcus.net>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Sat, 8 Dec 2012 20:10:29 +0200
> Subject: Re: [Freeswitch-users] xml_curl result not found
> Why send a not found? Send a hangup with a failure reason, and that will
> be the end of processing.
>
> -Avi
>
>
> On Sat, Dec 8, 2012 at 7:32 PM, Jeff Geddes <geddes.jeffrey at gmail.com>wrote:
>
>> I'm using xml_curl to generate a dialplan and send back result
>> status="not found" if something throws an error.  I don't want it to fail
>> back to the in-memory dialplan if there is an error.  Is there a way to do
>> this?
>>
>> thanks
>> jeff
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> ---------- Forwarded message ----------
> From: "Cal Leeming [Simplicity Media Ltd]" <
> cal.leeming at simplicitymedialtd.co.uk>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Sat, 8 Dec 2012 20:48:33 +0000
> Subject: Re: [Freeswitch-users] xml_curl result not found
> Exclude any dialplan sections from your config, and thus it won't have
> anything to fall back onto.
>
> However, if any other modules are configured to look for configs (??) then
> you may need to use the approach Avi mentioned, i.e. sending back a hangup
> with failure reason.
>
> Cal
>
> On Sat, Dec 8, 2012 at 5:32 PM, Jeff Geddes <geddes.jeffrey at gmail.com>wrote:
>
>> I'm using xml_curl to generate a dialplan and send back result
>> status="not found" if something throws an error.  I don't want it to fail
>> back to the in-memory dialplan if there is an error.  Is there a way to do
>> this?
>>
>> thanks
>> jeff
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> ---------- Forwarded message ----------
> From: "Cal Leeming [Simplicity Media Ltd]" <
> cal.leeming at simplicitymedialtd.co.uk>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Sat, 8 Dec 2012 21:09:54 +0000
> Subject: Re: [Freeswitch-users] IVR menu delay - 1 second between events
> I've raised a JIRA ticket for this;
> http://jira.freeswitch.org/browse/FS-4924
>
> Cal
>
> On Fri, Dec 7, 2012 at 8:08 PM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> Just trawled through the source and found this;
>>
>> src/switch_ivr_menu.c:
>> if (status == SWITCH_STATUS_SUCCESS) {
>>  status = switch_ivr_sleep(session, 1000, SWITCH_FALSE, NULL);
>> }
>>
>> So, I changed 1000 to 10, the delay has now disappeared and it seems to
>> work perfectly.
>>
>
>
>
> ---------- Forwarded message ----------
> From: "Cal Leeming [Simplicity Media Ltd]" <
> cal.leeming at simplicitymedialtd.co.uk>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Sat, 8 Dec 2012 22:18:37 +0000
> Subject: [Freeswitch-users] recovery_profile_name - any explanation?
> Hello,
>
> I've just been trawling Google for an explanation of what exactly the
> variable '*recovery_profile_name' means, but no luck.*
> *
> *
> *It seems to relate to the following change from 4 months ago;*
>
> https://github.com/freeSwitch/freeswitch/commit/2a8841ab666ec23ceb5688587bb14d16bf193b77
>
> The commit message states 'change mod_sofia to use new core based
> recovery engine'.
>
> But, it is unclear when this variable should be relied upon, and under
> what circumstances it will be used under.
>
> Any clarifications would be good, I will of course put in time to update
> the wiki too.
>
> Thanks
>
> Cal
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec  9 03:20:00 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 8 Dec 2012 19:20:00 -0500
Subject: [Freeswitch-users] Problem freeswitch and asterisk mediagateway
	bridge no direct connection
In-Reply-To: <Pine.LNX.4.64.1212090006140.28277@1a-8140.antagus.de>
References: <CACak_yxsejacKz6jAkdfdhdAV5tquAF4eBVR4a4yyC81kvuOvg@mail.gmail.com>
	<CAKzWOxXTDzOJ7Q0X4R+m06pG4R1w6kkkO7EA_7WzSbqO=crLZw@mail.gmail.com>
	<473CAD87-8625-4580-BEDC-08DE4475CE22@gmail.com>
	<CAFiqYunGtJ4qbbCyO-L-ppdpRfQDPgMK_3uAR2na61_GkjnyJA@mail.gmail.com>
	<Pine.LNX.4.64.1212081544550.25422@1a-8140.antagus.de>
	<7523.1354986131@ccs.covici.com>
	<Pine.LNX.4.64.1212090006140.28277@1a-8140.antagus.de>
Message-ID: <A08B3A8A-9487-47B8-B95C-B62DDDB464F6@endigotech.com>

This can be done in Freeswitch, and probably easier than doing so in Asterisk, but I'm sure there are other factors involved so that topic now comes to a rest.

As far as your problem at hand, please submit a PCAP and a better description of your issue, maybe we can take a look for you to see what's the matter. 

-BDF

Sent from my iPhone

On Dec 8, 2012, at 6:31 PM, Thomas Hoellriegel <admin at blindi.net> wrote:

> Am 08.12.12 um 12:02 schrieb covici at ccs.covici.com:
> 
>> Why use asterisk at all?  What is it doing?
> 
> Fs make the maimmenu and Asterisk make the interface:
> Fs have extensions to forward to asterisk on the same host.
> Asterisk make the folowing functions:
> a voice mail forum messageboard to leave messages for all users,
> chat features; who is online, (onlinelist)
> conferencemanagemnt: drop user from conference; mute user,
> speak a system broadcastmessage to all current online users,
> and more; voicemail, adressbook, blacklist distrobutionlist, whitelist, autoplay function in forums, calleridbased autologin, messagenotification, wakeupcalls,
> 
> Systemadministrationfeatures:
> delete account, reset user passwords, block and
> unblock acounts
> 
> I have programed a full features  chat system to help blind users via voice. (barrierfreee)
> People call these chat for free, and enter digits. For
> information.
> Example:
> press 1 for time and date, 2, for messageboards, 4 who ist online and so on.
> Fs makes the maimenu, and asterisk make the rest easy.
> 
> This Chat is works in german language.
> Then you enter the these instruction, press 2 you become a free voicemailbox to speak with some one else.
> 
> Fs and asterisk is very nice to program a voicechat platform.
> 
> 
> 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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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


From cal.leeming at simplicitymedialtd.co.uk  Sun Dec  9 03:28:30 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 9 Dec 2012 00:28:30 +0000
Subject: [Freeswitch-users] Hello hackers!
In-Reply-To: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
Message-ID: <CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>

Hi Brian,

I had contemplated replying off-list, but was interested to hear other
peoples thoughts on this too.

First - could you elaborate further on the 'bad apple' that you found,
exactly what justifies an attempt to 'hack into our phone systems', and why
this person in your story has been fired because of it?

Second, in reference to the 180k IPs.. There are other companies out there
that share abusive IP information from a variety of sources. Why do they
share? Because it's nice to share. If the FreeSWITCH developers took the
same attitude as your post here, then you wouldn't have FreeSWITCH.

Third, why are you telling us this on a public mailing list? If the
honeypots are designed to catch people unwittingly, then this post does the
exact opposite. This leads me to think that a more probable story is that
you actually don't have any honey pots (or the story is slightly
exaggerated), and when you realised you gave out potentially damaging
information, you panic'd and tried to discourage by asserting this email.
If this is the case, then you are taking the lay approach of security
through obscurity.

Fourth, if someone is wanting to break into your phone system, they
probably don't care about losing their job.. and if they do, then this post
will just give them more reason to be careful about hiding themselves.

I apologise in advance if this reply is inappropriate in anyway.

Cal

On Sat, Dec 8, 2012 at 11:05 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> Regarding a recent mailing list posting that included some of my IP
> addresses, most of you don't know that I do set up honeypots in hopes of
> catching some of the bad apples that try and hack into our phone systems.
> We have a centralized list of Bad IP's that end up getting sent to all of
> our other servers. Today, one of those servers was an IT guy that works for
> one of my clients. He has since been fired. If anyone is interested in the
> 180,000 IP's I've collected...sorry you can't have 'em.
>
> -BDF
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec  9 04:29:05 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 8 Dec 2012 20:29:05 -0500
Subject: [Freeswitch-users] Hello hackers!
In-Reply-To: <CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
Message-ID: <CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>

The 'bad apple' in which I was referring to was using the same IP as a client of ours. He was trying to DOS the honeypot from an IP I posted on the mailing list when doing some testing for someone. I have no idea if he read the post on the mailing list or not. It's not really of my concern.

If you got on the wiki and searched for fail2ban, you would be setting up your server to jail the same IP's we are under the same circumstances. The only difference is we log who gets caught by fail2ban and distribute the list internally.

We do not release this information per company policy. We also do not gather this information from other sources. We only use the information we gather through the processes we put in to place.

My comment on the 180K IP's was mostly sarcastic, however. It probably wasn't appropriate and I do apologize for that. 

I'm not exactly up to date on the legalities of releasing that type of information so we rather not release it. It's nothing against the freeswitch community or the open source community. We just don't like getting in trouble.

If we did spend the resources into making sure everything was legal on the information regarding the 180K IP's, we would certainly release these free of charge. It's not something I would be interested in making money from.

As far as telling this story on a public mailing list, it won't stop anyone from trying to hack into anyone's server. It does frustrate me that I have to do any of this stuff at all, but there's always going to be someone out there trying to screw it up for the rest of us. These servers are also set up for testing, which is why I use them when trying to help people on the mailing list. There is really nothing you can do to these machines to 'screw them up'. They are VPS's. There are no accounts tied to them. We can change those IP's in a heartbeat. There's really no risk. Besides, hackers can't read ;)

The biggest thing you should take away from this post is that I'm pissed off that I have to go through all of this. Even though it makes our lives easier in the long run, it's still an expense we could live without. 

Believe it or not the whole reason why I started doing honeypots is that about 8 months ago I DID release IP's that I shouldn't have, by accident. Since then I have added more resources to help curve the attacks on other servers we have contracts on.



-BDF
Sent from my iPhone

On Dec 8, 2012, at 7:28 PM, "Cal Leeming [Simplicity Media Ltd]" <cal.leeming at simplicitymedialtd.co.uk> wrote:

> Hi Brian,
> 
> I had contemplated replying off-list, but was interested to hear other peoples thoughts on this too.
> 
> First - could you elaborate further on the 'bad apple' that you found, exactly what justifies an attempt to 'hack into our phone systems', and why this person in your story has been fired because of it?
> 
> Second, in reference to the 180k IPs.. There are other companies out there that share abusive IP information from a variety of sources. Why do they share? Because it's nice to share. If the FreeSWITCH developers took the same attitude as your post here, then you wouldn't have FreeSWITCH.
> 
> Third, why are you telling us this on a public mailing list? If the honeypots are designed to catch people unwittingly, then this post does the exact opposite. This leads me to think that a more probable story is that you actually don't have any honey pots (or the story is slightly exaggerated), and when you realised you gave out potentially damaging information, you panic'd and tried to discourage by asserting this email. If this is the case, then you are taking the lay approach of security through obscurity.
> 
> Fourth, if someone is wanting to break into your phone system, they probably don't care about losing their job.. and if they do, then this post will just give them more reason to be careful about hiding themselves.
> 
> I apologise in advance if this reply is inappropriate in anyway.
> 
> Cal
> 
> On Sat, Dec 8, 2012 at 11:05 PM, Brian Foster <bdfoster at endigotech.com> wrote:
>> Regarding a recent mailing list posting that included some of my IP addresses, most of you don't know that I do set up honeypots in hopes of catching some of the bad apples that try and hack into our phone systems. We have a centralized list of Bad IP's that end up getting sent to all of our other servers. Today, one of those servers was an IT guy that works for one of my clients. He has since been fired. If anyone is interested in the 180,000 IP's I've collected...sorry you can't have 'em.
>> 
>> -BDF
>> 
>> 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 jmesquita at freeswitch.org  Sun Dec  9 07:22:27 2012
From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Sun, 9 Dec 2012 02:22:27 -0200
Subject: [Freeswitch-users] Compiling FSComm
In-Reply-To: <1354878605.5967.44.camel@vmmarces.vm.marces.com>
References: <1354878605.5967.44.camel@vmmarces.vm.marces.com>
Message-ID: <21EB7A95-BDA4-48FE-A944-DFB5662BBB14@freeswitch.org>

Antonio, I am the dev of FSComm and I am glad to hear there is still interest on it. I will try to fix the problem tomorrow and if not Monday as I am out of the country. 

Please help us out and file a Jira for it?

Sent from my iPhone

On Dec 7, 2012, at 9:10 AM, Antonio Silva <asilva at wirelessmundi.com> wrote:

> Hi,
> 
> i'm trying to compile fscomm but i have the following errors:
> 
> "
> freeswitch-git/fscomm# qmake
> freeswitch-git/fscomm# make
> /usr/bin/uic-qt4 mainwindow.ui -o ui_mainwindow.h
> /usr/bin/uic-qt4 preferences/prefdialog.ui -o ui_prefdialog.h
> /usr/bin/uic-qt4 preferences/accountdialog.ui -o ui_accountdialog.h
> /usr/bin/uic-qt4 widgets/codecwidget.ui -o ui_codecwidget.h
> /usr/bin/uic-qt4 debugtools/consolewindow.ui -o ui_consolewindow.h
> /usr/bin/uic-qt4 debugtools/statedebugdialog.ui -o ui_statedebugdialog.h
> Warning: name layoutWidget is already used
> Warning: name layoutWidget is already used
> g++ -c -pipe -O2 -Wall -W -D_REENTRANT -DQT_NO_DEBUG -DQT_XML_LIB -DQT_GUI_LIB -DQT_CORE_LIB -DQT_SHARED -I/usr/share/qt4/mkspecs/linux-g++ -I. -I/usr/include/qt4/QtCore -I/usr/include/qt4/QtGui -I/usr/include/qt4/QtXml -I/usr/include/qt4 -I../src/include -I../libs/apr/include -I../libs/libteletone/src -I. -I. -o main.o main.cpp
> In file included from mainwindow.h:38,
>                  from main.cpp:32:
> ../src/include/switch.h:110:18: error: stfu.h: No such file or directory
> In file included from ../src/include/switch.h:121,
>                  from mainwindow.h:38,
>                  from main.cpp:32:
> ../src/include/switch_core.h:752: error: expected constructor, destructor, or type conversion before ?*? token
> In file included from ../src/include/switch_loadable_module.h:46,
>                  from ../src/include/switch.h:122,
>                  from mainwindow.h:38,
>                  from main.cpp:32:
> ../src/include/switch_module_interfaces.h:121: error: expected initializer before ?*? token
> ../src/include/switch_module_interfaces.h:162: error: ?switch_io_get_jb_t? does not name a type
> In file included from ../src/include/switch.h:134,
>                  from mainwindow.h:38,
>                  from main.cpp:32:
> ../src/include/switch_rtp.h:243: error: expected constructor, destructor, or type conversion before ?*? token
> ./fshost.h:43: warning: ?void eventHandlerCallback(switch_event_t*)? declared ?static? but never defined
> ./fshost.h:44: warning: ?switch_status_t loggerHandler(const switch_log_node_t*, switch_log_level_t)? declared ?static? but never defined
> make: *** [main.o] Error 1
> "
> 
> watching this error: 
> "../src/include/switch.h:110:18: error: stfu.h: No such file or directory"
> 
> i manually change switch.h to fix the problem with the include by adding the following:
> 
> "
> diff --git a/src/include/switch.h b/src/include/switch.h
> index c7ea7b0..2847112 100644
> --- a/src/include/switch.h
> +++ b/src/include/switch.h
> @@ -107,7 +107,8 @@
> #include <errno.h>
> 
> #ifndef WIN32
> -#include "stfu.h"
> +/* #include "stfu.h" */
> +#include "../../../libs/stfu/stfu.h"
> #else
> #include "../../../libs/stfu/stfu.h"
> #endif
> 
> "
> 
> I could compile fscomm, but now i can't start it... i have the following error:
> "
> Initializing core... 
> Failed to initialize FreeSWITCH's core: Cannot Open log directory or XML Root!
> Everything OK, Entering runtime loop ... 
> Segmentation fault
> "
> i had try the fix in the wiki: "chmod  644 ~/.fscomm/conf/freeswitch.xml", and even "chmod -R 777 ~/.fscomm",  but no luck...
> 
> Can you help me to go further...?
> 
> I'm trying it to install on a debian squeeze, i installed qt4-dev-tools. Is it possible to install in debian squeeze our i should just  give up... and try another distro?
> The freeswitch-git is the lasted head. 
> 
> 
> Thanks,
> Ant?nio
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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


From cal.leeming at simplicitymedialtd.co.uk  Sun Dec  9 07:46:57 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 9 Dec 2012 04:46:57 +0000
Subject: [Freeswitch-users] Hello hackers!
In-Reply-To: <CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
Message-ID: <CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>

Thank you for the detailed response.

On Sun, Dec 9, 2012 at 1:29 AM, Brian Foster <bdfoster at endigotech.com>wrote:

> The 'bad apple' in which I was referring to was using the same IP as a
> client of ours. He was trying to DOS the honeypot from an IP I posted on
> the mailing list when doing some testing for someone. I have no idea if he
> read the post on the mailing list or not. It's not really of my concern.
>

You know what happens when someone attacks one of our clients?

We track them down, introduce them to the CTO/CEO of the company they
attacked, and give them an opportunity to prove themselves. I have been
involved in this process on several occasions now where the outcome has
been extremely positive. I'm not saying this works all the time, but
sometimes people don't need punishment, they need guidance.


>
> If you got on the wiki and searched for fail2ban, you would be setting up
> your server to jail the same IP's we are under the same circumstances. The
> only difference is we log who gets caught by fail2ban and distribute the
> list internally.
>

> We do not release this information per company policy. We also do not
> gather this information from other sources. We only use the information we
> gather through the processes we put in to place.
>
> My comment on the 180K IP's was mostly sarcastic, however. It probably
> wasn't appropriate and I do apologize for that.
>
> I'm not exactly up to date on the legalities of releasing that type of
> information so we rather not release it. It's nothing against the
> freeswitch community or the open source community. We just don't like
> getting in trouble.
>
> If we did spend the resources into making sure everything was legal on the
> information regarding the 180K IP's, we would certainly release these free
> of charge. It's not something I would be interested in making money from.
>

The concept is no different to email blacklist databases (e.g. XBL), and
there would be no legalities stopping you from releasing this information
into the public domain - only internal red tape and policies. I can say
this with at least some authority on the subject (although I'm by no means
an expert).

Right now we live in a society where often companies can't/won't share
information for one reason or another (I hear the 'company policy' story a
lot), but yet feel it's okay to use years of time/development in open
source for free. I mean no direct disrespect, I just personally find this
quite irritating.


> As far as telling this story on a public mailing list, it won't stop
> anyone from trying to hack into anyone's server. It does frustrate me that
> I have to do any of this stuff at all, but there's always going to be
> someone out there trying to screw it up for the rest of us. These servers
> are also set up for testing, which is why I use them when trying to help
> people on the mailing list. There is really nothing you can do to these
> machines to 'screw them up'. They are VPS's. There are no accounts tied to
> them. We can change those IP's in a heartbeat. There's really no risk.
> Besides, hackers can't read ;)
>

> The biggest thing you should take away from this post is that I'm pissed
> off that I have to go through all of this. Even though it makes our lives
> easier in the long run, it's still an expense we could live without.
>

Giving away IPs shouldn't amount to any concern in the first place though,
hence the previous comment about security through obscurity..

It all comes down to stacking... there is no one big solution, just lots of
small solutions (assuming you don't believe those AF/WAF sales guys selling
god damn snake oil)

Production services really shouldn't be live without at least being behind
some form of DPI/SPI appliance (L7 deep/stateful packet inspection).

I will agree with you that cost can be a contributing factor.. but hey, I
don't like paying tax.. still gotta pay it!


>
> Believe it or not the whole reason why I started doing honeypots is that
> about 8 months ago I DID release IP's that I shouldn't have, by accident.
> Since then I have added more resources to help curve the attacks on other
> servers we have contracts on.
>

>
>
> -BDF
> Sent from my iPhone
>
> On Dec 8, 2012, at 7:28 PM, "Cal Leeming [Simplicity Media Ltd]" <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
> Hi Brian,
>
> I had contemplated replying off-list, but was interested to hear other
> peoples thoughts on this too.
>
> First - could you elaborate further on the 'bad apple' that you found,
> exactly what justifies an attempt to 'hack into our phone systems', and why
> this person in your story has been fired because of it?
>
> Second, in reference to the 180k IPs.. There are other companies out there
> that share abusive IP information from a variety of sources. Why do they
> share? Because it's nice to share. If the FreeSWITCH developers took the
> same attitude as your post here, then you wouldn't have FreeSWITCH.
>
> Third, why are you telling us this on a public mailing list? If the
> honeypots are designed to catch people unwittingly, then this post does the
> exact opposite. This leads me to think that a more probable story is that
> you actually don't have any honey pots (or the story is slightly
> exaggerated), and when you realised you gave out potentially damaging
> information, you panic'd and tried to discourage by asserting this email.
> If this is the case, then you are taking the lay approach of security
> through obscurity.
>
> Fourth, if someone is wanting to break into your phone system, they
> probably don't care about losing their job.. and if they do, then this post
> will just give them more reason to be careful about hiding themselves.
>
> I apologise in advance if this reply is inappropriate in anyway.
>
> Cal
>
> On Sat, Dec 8, 2012 at 11:05 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> Regarding a recent mailing list posting that included some of my IP
>> addresses, most of you don't know that I do set up honeypots in hopes of
>> catching some of the bad apples that try and hack into our phone systems.
>> We have a centralized list of Bad IP's that end up getting sent to all of
>> our other servers. Today, one of those servers was an IT guy that works for
>> one of my clients. He has since been fired. If anyone is interested in the
>> 180,000 IP's I've collected...sorry you can't have 'em.
>>
>> -BDF
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec  9 09:49:27 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 9 Dec 2012 01:49:27 -0500
Subject: [Freeswitch-users] Hello hackers!
In-Reply-To: <CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
	<CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
Message-ID: <CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>

I appreciate your comments and concerns, and I respect the fact that we can
talk about this stuff without things getting ugly. My comments are below.

-BDF


On Sat, Dec 8, 2012 at 11:46 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Thank you for the detailed response.
>
> On Sun, Dec 9, 2012 at 1:29 AM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> The 'bad apple' in which I was referring to was using the same IP as a
>> client of ours. He was trying to DOS the honeypot from an IP I posted on
>> the mailing list when doing some testing for someone. I have no idea if he
>> read the post on the mailing list or not. It's not really of my concern.
>>
>
> You know what happens when someone attacks one of our clients?
>
> We track them down, introduce them to the CTO/CEO of the company they
> attacked, and give them an opportunity to prove themselves. I have been
> involved in this process on several occasions now where the outcome has
> been extremely positive. I'm not saying this works all the time, but
> sometimes people don't need punishment, they need guidance.
>

This is not really a practice of ours because there are so many people out
there that contribute to the problem. It's too time consuming to educate
those people. I really wish I could do that, but it's just not feasible. We
do not actively pursue these abusive IP's nor do we DDOS them or fight back
in any other way other than fighting back some of the noise through
blocking those activities on machines we support.


>
>> If you got on the wiki and searched for fail2ban, you would be setting up
>> your server to jail the same IP's we are under the same circumstances. The
>> only difference is we log who gets caught by fail2ban and distribute the
>> list internally.
>>
>
>> We do not release this information per company policy. We also do not
>> gather this information from other sources. We only use the information we
>> gather through the processes we put in to place.
>>
>> My comment on the 180K IP's was mostly sarcastic, however. It probably
>> wasn't appropriate and I do apologize for that.
>>
>> I'm not exactly up to date on the legalities of releasing that type of
>> information so we rather not release it. It's nothing against the
>> freeswitch community or the open source community. We just don't like
>> getting in trouble.
>>
>> If we did spend the resources into making sure everything was legal on
>> the information regarding the 180K IP's, we would certainly release these
>> free of charge. It's not something I would be interested in making money
>> from.
>>
>
> The concept is no different to email blacklist databases (e.g. XBL), and
> there would be no legalities stopping you from releasing this information
> into the public domain - only internal red tape and policies. I can say
> this with at least some authority on the subject (although I'm by no means
> an expert).
>

I'm open to this idea, but I would have to consult attorneys to do this. We
operate very cautiously as in we do not operate in grey areas. If this is a
potential grey area we will certainly take the extra precautions in order
to prevent legal issues.

This isn't some big company that has endless amounts of resources. We're a
small business, just like those we support. We also have to consider how
this effects our clients as well, and we're not about to take the risk of
operating in a grey area. I can't afford mistakes like that especially
since we're still a brand new company in the grand scheme of things.


> Right now we live in a society where often companies can't/won't share
> information for one reason or another (I hear the 'company policy' story a
> lot), but yet feel it's okay to use years of time/development in open
> source for free. I mean no direct disrespect, I just personally find this
> quite irritating.
>

First of all I'd like to introduce you to the CEO of Endigo Computer LLC.
He's 23 years old, and is currently in school pursuing his Bachelors in
Computer Science and eventually his Masters. He also works part time at a
real estate firm. Hi :)

We are not foreign to the open source world. I founded this company back in
2010 to help promote the use of open source software. We seek clients who
are tired of proprietary, non-moldable software for an open-source based
solution that works for them. Most of our technical staff are hired from
within the open source community and contribute to various open source
projects both on and off company time. I'm extremely happy to find people
who live and breathe the open source philosophy.


>
>> As far as telling this story on a public mailing list, it won't stop
>> anyone from trying to hack into anyone's server. It does frustrate me that
>> I have to do any of this stuff at all, but there's always going to be
>> someone out there trying to screw it up for the rest of us. These servers
>> are also set up for testing, which is why I use them when trying to help
>> people on the mailing list. There is really nothing you can do to these
>> machines to 'screw them up'. They are VPS's. There are no accounts tied to
>> them. We can change those IP's in a heartbeat. There's really no risk.
>> Besides, hackers can't read ;)
>>
>
>> The biggest thing you should take away from this post is that I'm pissed
>> off that I have to go through all of this. Even though it makes our lives
>> easier in the long run, it's still an expense we could live without.
>>
>
> Giving away IPs shouldn't amount to any concern in the first place though,
> hence the previous comment about security through obscurity..
>

> It all comes down to stacking... there is no one big solution, just lots
> of small solutions (assuming you don't believe those AF/WAF sales guys
> selling god damn snake oil)
>
> Production services really shouldn't be live without at least being behind
> some form of DPI/SPI appliance (L7 deep/stateful packet inspection).
>
> I will agree with you that cost can be a contributing factor.. but hey, I
> don't like paying tax.. still gotta pay it!
>

This strategy of using honeypots is certainly not the only tactic we use to
fight attacks like this. It's just one piece of the puzzle. I can assure
you there are many other processes in place to actively and passively
protect the machines we support.


>
>
>>
>> Believe it or not the whole reason why I started doing honeypots is that
>> about 8 months ago I DID release IP's that I shouldn't have, by accident.
>> Since then I have added more resources to help curve the attacks on other
>> servers we have contracts on.
>>
>
>>
>>
>> -BDF
>> Sent from my iPhone
>>
>> On Dec 8, 2012, at 7:28 PM, "Cal Leeming [Simplicity Media Ltd]" <
>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>
>> Hi Brian,
>>
>> I had contemplated replying off-list, but was interested to hear other
>> peoples thoughts on this too.
>>
>> First - could you elaborate further on the 'bad apple' that you found,
>> exactly what justifies an attempt to 'hack into our phone systems', and why
>> this person in your story has been fired because of it?
>>
>> Second, in reference to the 180k IPs.. There are other companies out
>> there that share abusive IP information from a variety of sources. Why do
>> they share? Because it's nice to share. If the FreeSWITCH developers took
>> the same attitude as your post here, then you wouldn't have FreeSWITCH.
>>
>> Third, why are you telling us this on a public mailing list? If the
>> honeypots are designed to catch people unwittingly, then this post does the
>> exact opposite. This leads me to think that a more probable story is that
>> you actually don't have any honey pots (or the story is slightly
>> exaggerated), and when you realised you gave out potentially damaging
>> information, you panic'd and tried to discourage by asserting this email.
>> If this is the case, then you are taking the lay approach of security
>> through obscurity.
>>
>> Fourth, if someone is wanting to break into your phone system, they
>> probably don't care about losing their job.. and if they do, then this post
>> will just give them more reason to be careful about hiding themselves.
>>
>> I apologise in advance if this reply is inappropriate in anyway.
>>
>> Cal
>>
>> On Sat, Dec 8, 2012 at 11:05 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>>
>>> Regarding a recent mailing list posting that included some of my IP
>>> addresses, most of you don't know that I do set up honeypots in hopes of
>>> catching some of the bad apples that try and hack into our phone systems.
>>> We have a centralized list of Bad IP's that end up getting sent to all of
>>> our other servers. Today, one of those servers was an IT guy that works for
>>> one of my clients. He has since been fired. If anyone is interested in the
>>> 180,000 IP's I've collected...sorry you can't have 'em.
>>>
>>> -BDF
>>>
>>> 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
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.
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From admin at blindi.net  Sun Dec  9 11:25:37 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sun, 9 Dec 2012 09:25:37 +0100 (CET)
Subject: [Freeswitch-users] easy way to crash freeswitch
In-Reply-To: <CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
	<CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
	<CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1212090907450.32638@1a-8140.antagus.de>

Hi guys, i found a easy way to crash fs.

first, you must record a testsoundfile 5 secs, for example: 
/usr/local/freeswitch/sounds/test.alaw
Write the following luascript in:
/usr/local/freeswitch/script/simpy-crash.lua
with:

-- simply script to crash freeswitch
session:sleep(100);
session:answer()
session:sleep(200);
  session:setInputCallback("onInput", "");
function onInput(s, type, obj)
   freeswitch.consoleLog("info", "Callback with type " .. type .. "\n");
--  freeswitch.consoleLog("info", "DTMF Digit: " .. obj.digit .. "\n");

   if ( obj['digit'] == '1' ) then

session:streamFile("/usr/local/freeswitch/sounds/test.alaw")
read_msg_menu()
  return("false");

end

   if ( obj['digit'] == '2' ) then

-- if (obj.digit   == "2")  then 
--  session:streamFile(msg_file_play)
session:streamFile("/usr/local/freeswitch/sounds/test.alaw")
read_msg_menu()
  return("false");

end

   if ( obj['digit'] == '3' ) then

-- if (obj.digit   == "3")  then 
--  session:streamFile(msg_file_play)
session:streamFile("/usr/local/freeswitch/sounds/test.alaw")
read_msg_menu()
  return("false");

end


end

function read_msg_menu( )
  session:setInputCallback("onInput");
session:streamFile("/usr/local/freeswitch/sounds/test.alaw")
end


  read_msg_menu()

-- end script


edit:
/usr/local/freeswitch/dialplan/default/crash.xml

<!-- crash test lua  -->
   <!--    -->

<extension name="crashtest" >
<condition field="destination_number" expression="^104$">
<action application="lua" data="simply-crash.lua"/>
<!-- <action application="start_dtmf" />    --> 
</condition>
</extension>

reload the dialplan:
fs_cli -x "reloadxml"

Dial extension 104, and heare your voice.
you must break the voice: press 90 times 1.
Fs is ready to crash.
Fs crash with segfault.
This is a good bug for sysadmins.-).









From avi at avimarcus.net  Sun Dec  9 11:51:29 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 9 Dec 2012 10:51:29 +0200
Subject: [Freeswitch-users] easy way to crash freeswitch
In-Reply-To: <Pine.LNX.4.64.1212090907450.32638@1a-8140.antagus.de>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
	<CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
	<CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
	<Pine.LNX.4.64.1212090907450.32638@1a-8140.antagus.de>
Message-ID: <CALNZuwnN=z_ODN8CU9JX5_wwE3xGBMW0WhU6LkThxW+pBEJY1w@mail.gmail.com>

If you WANT to crash FreeSWITCH to test your recovery, monitoring, etc you
can run "fsctl crash".
This sounds like a bug.

-Avi


On Sun, Dec 9, 2012 at 10:25 AM, Thomas Hoellriegel <admin at blindi.net>wrote:

> Hi guys, i found a easy way to crash fs.
>
> first, you must record a testsoundfile 5 secs, for example:
> /usr/local/freeswitch/sounds/test.alaw
> Write the following luascript in:
> /usr/local/freeswitch/script/simpy-crash.lua
> with:
>
> -- simply script to crash freeswitch
> session:sleep(100);
> session:answer()
> session:sleep(200);
>   session:setInputCallback("onInput", "");
> function onInput(s, type, obj)
>    freeswitch.consoleLog("info", "Callback with type " .. type .. "\n");
> --  freeswitch.consoleLog("info", "DTMF Digit: " .. obj.digit .. "\n");
>
>    if ( obj['digit'] == '1' ) then
>
> session:streamFile("/usr/local/freeswitch/sounds/test.alaw")
> read_msg_menu()
>   return("false");
>
> end
>
>    if ( obj['digit'] == '2' ) then
>
> -- if (obj.digit   == "2")  then
> --  session:streamFile(msg_file_play)
> session:streamFile("/usr/local/freeswitch/sounds/test.alaw")
> read_msg_menu()
>   return("false");
>
> end
>
>    if ( obj['digit'] == '3' ) then
>
> -- if (obj.digit   == "3")  then
> --  session:streamFile(msg_file_play)
> session:streamFile("/usr/local/freeswitch/sounds/test.alaw")
> read_msg_menu()
>   return("false");
>
> end
>
>
> end
>
> function read_msg_menu( )
>   session:setInputCallback("onInput");
> session:streamFile("/usr/local/freeswitch/sounds/test.alaw")
> end
>
>
>   read_msg_menu()
>
> -- end script
>
>
> edit:
> /usr/local/freeswitch/dialplan/default/crash.xml
>
> <!-- crash test lua  -->
>    <!--    -->
>
> <extension name="crashtest" >
> <condition field="destination_number" expression="^104$">
> <action application="lua" data="simply-crash.lua"/>
> <!-- <action application="start_dtmf" />    -->
> </condition>
> </extension>
>
> reload the dialplan:
> fs_cli -x "reloadxml"
>
> Dial extension 104, and heare your voice.
> you must break the voice: press 90 times 1.
> Fs is ready to crash.
> Fs crash with segfault.
> This is a good bug for sysadmins.-).
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 gabe at gundy.org  Sun Dec  9 11:59:45 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sun, 9 Dec 2012 01:59:45 -0700
Subject: [Freeswitch-users] easy way to crash freeswitch
In-Reply-To: <Pine.LNX.4.64.1212090907450.32638@1a-8140.antagus.de>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
	<CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
	<CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
	<Pine.LNX.4.64.1212090907450.32638@1a-8140.antagus.de>
Message-ID: <CAMwZ-tKu1QJwGqpZRNM_CHxZaRr8sjWbgY_V0rEsON125AbRdQ@mail.gmail.com>

On Sun, Dec 9, 2012 at 1:25 AM, Thomas Hoellriegel <admin at blindi.net> wrote:
> Fs is ready to crash.
> Fs crash with segfault.
> This is a good bug for sysadmins.-).

Strange, I searched http://jira.freeswitch.org for simpy-crash.lua and
couldn't find it.


From steveayre at gmail.com  Sun Dec  9 13:13:47 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 9 Dec 2012 10:13:47 +0000
Subject: [Freeswitch-users] easy way to crash freeswitch
In-Reply-To: <Pine.LNX.4.64.1212090907450.32638@1a-8140.antagus.de>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
	<CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
	<CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
	<Pine.LNX.4.64.1212090907450.32638@1a-8140.antagus.de>
Message-ID: <CAFiqYukEqT9VWtFHhKtZieQUieLjDkEixoqCuk+f5OesEdF0pQ@mail.gmail.com>

Anything that crashes FS is *always* a bug, and should be fixed.

Please test it on git head of the master branch to confirm if it still
applies there, or whether it is a bug that has already been resolved.

Bugs can be reporterd at http://jira.freeswitch.org/

There is never a 'good' bug for sysadmins... 'fsctl crash' will
intentionally crash FS with much less effort if you ever need to test crash
handling.




On 9 December 2012 08:25, Thomas Hoellriegel <admin at blindi.net> wrote:

> Hi guys, i found a easy way to crash fs.
>
> first, you must record a testsoundfile 5 secs, for example:
> /usr/local/freeswitch/sounds/test.alaw
> Write the following luascript in:
> /usr/local/freeswitch/script/simpy-crash.lua
> with:
>
> -- simply script to crash freeswitch
> session:sleep(100);
> session:answer()
> session:sleep(200);
>   session:setInputCallback("onInput", "");
> function onInput(s, type, obj)
>    freeswitch.consoleLog("info", "Callback with type " .. type .. "\n");
> --  freeswitch.consoleLog("info", "DTMF Digit: " .. obj.digit .. "\n");
>
>    if ( obj['digit'] == '1' ) then
>
> session:streamFile("/usr/local/freeswitch/sounds/test.alaw")
> read_msg_menu()
>   return("false");
>
> end
>
>    if ( obj['digit'] == '2' ) then
>
> -- if (obj.digit   == "2")  then
> --  session:streamFile(msg_file_play)
> session:streamFile("/usr/local/freeswitch/sounds/test.alaw")
> read_msg_menu()
>   return("false");
>
> end
>
>    if ( obj['digit'] == '3' ) then
>
> -- if (obj.digit   == "3")  then
> --  session:streamFile(msg_file_play)
> session:streamFile("/usr/local/freeswitch/sounds/test.alaw")
> read_msg_menu()
>   return("false");
>
> end
>
>
> end
>
> function read_msg_menu( )
>   session:setInputCallback("onInput");
> session:streamFile("/usr/local/freeswitch/sounds/test.alaw")
> end
>
>
>   read_msg_menu()
>
> -- end script
>
>
> edit:
> /usr/local/freeswitch/dialplan/default/crash.xml
>
> <!-- crash test lua  -->
>    <!--    -->
>
> <extension name="crashtest" >
> <condition field="destination_number" expression="^104$">
> <action application="lua" data="simply-crash.lua"/>
> <!-- <action application="start_dtmf" />    -->
> </condition>
> </extension>
>
> reload the dialplan:
> fs_cli -x "reloadxml"
>
> Dial extension 104, and heare your voice.
> you must break the voice: press 90 times 1.
> Fs is ready to crash.
> Fs crash with segfault.
> This is a good bug for sysadmins.-).
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bote_radio at botecomm.com  Sun Dec  9 13:15:08 2012
From: bote_radio at botecomm.com (Bote Man)
Date: Sun, 9 Dec 2012 05:15:08 -0500
Subject: [Freeswitch-users] Avaya CS1000M integration guide
In-Reply-To: <031701cdc783$b0eae640$12c0b2c0$@bizfocused.com>
References: <031701cdc783$b0eae640$12c0b2c0$@bizfocused.com>
Message-ID: <005301cdd5f6$11bc5a00$35350e00$@com>

I know this is somewhat off-topic for this list, but I'm quickly running
out of options.

 

I need pointers to what Avaya considers an "integration guide" that
tells them what to enter into their CS1k M pbx to a plain SIP trunk to
talk to FreeSWITCH. Their pbx is brand new so I assume it is running the
latest 7.5 or thereabouts.

 

I'm hitting the bureaucratic roadblock with their tech support on this
point so I need to send them document that will move us off dead-center.

 

Thanks.

 

 

John Boteler

Bote Communications

Fort Lauderdale, FL

 

 

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From 8f27e956 at gmail.com  Sun Dec  9 13:46:24 2012
From: 8f27e956 at gmail.com (S. Scott)
Date: Sun, 9 Dec 2012 05:46:24 -0500
Subject: [Freeswitch-users] Avaya CS1000M integration guide
In-Reply-To: <005301cdd5f6$11bc5a00$35350e00$@com>
References: <031701cdc783$b0eae640$12c0b2c0$@bizfocused.com>
	<005301cdd5f6$11bc5a00$35350e00$@com>
Message-ID: <3855001073712662482@unknownmsgid>

Have you googled 'avaya sip trunking'.  Many, many returns, including
avaya's own doc's.

https://devconnect.avaya.com/public/download/interop/SES-CUCM-SIPtrk.pdf

Cheers,

?????
iThing:  Big thumbs & little keys. Please excuse typo, spelling and grammar
errors ? Last night I played a blank CD at full blast. The Mime next door
went nuts.


On 2012-12-09, at 5:20, Bote Man <bote_radio at botecomm.com> wrote:

I know this is somewhat off-topic for this list, but I'm quickly running
out of options.



I need pointers to what Avaya considers an "integration guide" that tells
them what to enter into their CS1k M pbx to a plain SIP trunk to talk to
FreeSWITCH. Their pbx is brand new so I assume it is running the latest 7.5
or thereabouts.



I'm hitting the bureaucratic roadblock with their tech support on this
point so I need to send them document that will move us off dead-center.



Thanks.





John Boteler

Bote Communications

Fort Lauderdale, FL





_________________________________________________________________________
Professional FreeSWITCH Consulting 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 cal.leeming at simplicitymedialtd.co.uk  Sun Dec  9 18:02:16 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 9 Dec 2012 15:02:16 +0000
Subject: [Freeswitch-users] Hello hackers!
In-Reply-To: <CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
	<CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
	<CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
Message-ID: <CAHKQagGOREuTuSyYhMi2zWg-ndOrqhtvKZre7EEc936xuv8SwQ@mail.gmail.com>

Sounds like I might have slightly misinterpreted your initial email and
jumped too fast to a conclusion, so my apologies for this!

I agree that it's rare to have a discussion about such topics without
things getting hot, and hearing someone else's thoughts on the subject has
been quite interesting.

Cal

On Sun, Dec 9, 2012 at 6:49 AM, Brian Foster <bdfoster at endigotech.com>wrote:

> I appreciate your comments and concerns, and I respect the fact that we
> can talk about this stuff without things getting ugly. My comments are
> below.
>
> -BDF
>
>
> On Sat, Dec 8, 2012 at 11:46 PM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> Thank you for the detailed response.
>>
>> On Sun, Dec 9, 2012 at 1:29 AM, Brian Foster <bdfoster at endigotech.com>wrote:
>>
>>> The 'bad apple' in which I was referring to was using the same IP as a
>>> client of ours. He was trying to DOS the honeypot from an IP I posted on
>>> the mailing list when doing some testing for someone. I have no idea if he
>>> read the post on the mailing list or not. It's not really of my concern.
>>>
>>
>> You know what happens when someone attacks one of our clients?
>>
>> We track them down, introduce them to the CTO/CEO of the company they
>> attacked, and give them an opportunity to prove themselves. I have been
>> involved in this process on several occasions now where the outcome has
>> been extremely positive. I'm not saying this works all the time, but
>> sometimes people don't need punishment, they need guidance.
>>
>
> This is not really a practice of ours because there are so many people out
> there that contribute to the problem. It's too time consuming to educate
> those people. I really wish I could do that, but it's just not feasible. We
> do not actively pursue these abusive IP's nor do we DDOS them or fight back
> in any other way other than fighting back some of the noise through
> blocking those activities on machines we support.
>
>
>>
>>> If you got on the wiki and searched for fail2ban, you would be setting
>>> up your server to jail the same IP's we are under the same circumstances.
>>> The only difference is we log who gets caught by fail2ban and distribute
>>> the list internally.
>>>
>>
>>> We do not release this information per company policy. We also do not
>>> gather this information from other sources. We only use the information we
>>> gather through the processes we put in to place.
>>>
>>> My comment on the 180K IP's was mostly sarcastic, however. It probably
>>> wasn't appropriate and I do apologize for that.
>>>
>>> I'm not exactly up to date on the legalities of releasing that type of
>>> information so we rather not release it. It's nothing against the
>>> freeswitch community or the open source community. We just don't like
>>> getting in trouble.
>>>
>>> If we did spend the resources into making sure everything was legal on
>>> the information regarding the 180K IP's, we would certainly release these
>>> free of charge. It's not something I would be interested in making money
>>> from.
>>>
>>
>> The concept is no different to email blacklist databases (e.g. XBL), and
>> there would be no legalities stopping you from releasing this information
>> into the public domain - only internal red tape and policies. I can say
>> this with at least some authority on the subject (although I'm by no means
>> an expert).
>>
>
> I'm open to this idea, but I would have to consult attorneys to do this.
> We operate very cautiously as in we do not operate in grey areas. If this
> is a potential grey area we will certainly take the extra precautions in
> order to prevent legal issues.
>
> This isn't some big company that has endless amounts of resources. We're a
> small business, just like those we support. We also have to consider how
> this effects our clients as well, and we're not about to take the risk of
> operating in a grey area. I can't afford mistakes like that especially
> since we're still a brand new company in the grand scheme of things.
>
>
>> Right now we live in a society where often companies can't/won't share
>> information for one reason or another (I hear the 'company policy' story a
>> lot), but yet feel it's okay to use years of time/development in open
>> source for free. I mean no direct disrespect, I just personally find this
>> quite irritating.
>>
>
> First of all I'd like to introduce you to the CEO of Endigo Computer LLC.
> He's 23 years old, and is currently in school pursuing his Bachelors in
> Computer Science and eventually his Masters. He also works part time at a
> real estate firm. Hi :)
>
> We are not foreign to the open source world. I founded this company back
> in 2010 to help promote the use of open source software. We seek clients
> who are tired of proprietary, non-moldable software for an open-source
> based solution that works for them. Most of our technical staff are hired
> from within the open source community and contribute to various open source
> projects both on and off company time. I'm extremely happy to find people
> who live and breathe the open source philosophy.
>
>
>>
>>> As far as telling this story on a public mailing list, it won't stop
>>> anyone from trying to hack into anyone's server. It does frustrate me that
>>> I have to do any of this stuff at all, but there's always going to be
>>> someone out there trying to screw it up for the rest of us. These servers
>>> are also set up for testing, which is why I use them when trying to help
>>> people on the mailing list. There is really nothing you can do to these
>>> machines to 'screw them up'. They are VPS's. There are no accounts tied to
>>> them. We can change those IP's in a heartbeat. There's really no risk.
>>> Besides, hackers can't read ;)
>>>
>>
>>> The biggest thing you should take away from this post is that I'm pissed
>>> off that I have to go through all of this. Even though it makes our lives
>>> easier in the long run, it's still an expense we could live without.
>>>
>>
>> Giving away IPs shouldn't amount to any concern in the first place
>> though, hence the previous comment about security through obscurity..
>>
>
>> It all comes down to stacking... there is no one big solution, just lots
>> of small solutions (assuming you don't believe those AF/WAF sales guys
>> selling god damn snake oil)
>>
>> Production services really shouldn't be live without at least being
>> behind some form of DPI/SPI appliance (L7 deep/stateful packet inspection).
>>
>> I will agree with you that cost can be a contributing factor.. but hey, I
>> don't like paying tax.. still gotta pay it!
>>
>
> This strategy of using honeypots is certainly not the only tactic we use
> to fight attacks like this. It's just one piece of the puzzle. I can assure
> you there are many other processes in place to actively and passively
> protect the machines we support.
>
>
>>
>>
>>>
>>> Believe it or not the whole reason why I started doing honeypots is that
>>> about 8 months ago I DID release IP's that I shouldn't have, by accident.
>>> Since then I have added more resources to help curve the attacks on other
>>> servers we have contracts on.
>>>
>>
>>>
>>>
>>> -BDF
>>> Sent from my iPhone
>>>
>>> On Dec 8, 2012, at 7:28 PM, "Cal Leeming [Simplicity Media Ltd]" <
>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>
>>> Hi Brian,
>>>
>>> I had contemplated replying off-list, but was interested to hear other
>>> peoples thoughts on this too.
>>>
>>> First - could you elaborate further on the 'bad apple' that you found,
>>> exactly what justifies an attempt to 'hack into our phone systems', and why
>>> this person in your story has been fired because of it?
>>>
>>> Second, in reference to the 180k IPs.. There are other companies out
>>> there that share abusive IP information from a variety of sources. Why do
>>> they share? Because it's nice to share. If the FreeSWITCH developers took
>>> the same attitude as your post here, then you wouldn't have FreeSWITCH.
>>>
>>> Third, why are you telling us this on a public mailing list? If the
>>> honeypots are designed to catch people unwittingly, then this post does the
>>> exact opposite. This leads me to think that a more probable story is that
>>> you actually don't have any honey pots (or the story is slightly
>>> exaggerated), and when you realised you gave out potentially damaging
>>> information, you panic'd and tried to discourage by asserting this email.
>>> If this is the case, then you are taking the lay approach of security
>>> through obscurity.
>>>
>>> Fourth, if someone is wanting to break into your phone system, they
>>> probably don't care about losing their job.. and if they do, then this post
>>> will just give them more reason to be careful about hiding themselves.
>>>
>>> I apologise in advance if this reply is inappropriate in anyway.
>>>
>>> Cal
>>>
>>> On Sat, Dec 8, 2012 at 11:05 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>>>
>>>> Regarding a recent mailing list posting that included some of my IP
>>>> addresses, most of you don't know that I do set up honeypots in hopes of
>>>> catching some of the bad apples that try and hack into our phone systems.
>>>> We have a centralized list of Bad IP's that end up getting sent to all of
>>>> our other servers. Today, one of those servers was an IT guy that works for
>>>> one of my clients. He has since been fired. If anyone is interested in the
>>>> 180,000 IP's I've collected...sorry you can't have 'em.
>>>>
>>>> -BDF
>>>>
>>>> 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
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> --
> 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 admin at blindi.net  Sun Dec  9 19:14:10 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sun, 9 Dec 2012 17:14:10 +0100 (CET)
Subject: [Freeswitch-users] easy way to crash freeswitch
In-Reply-To: <CAMwZ-tKu1QJwGqpZRNM_CHxZaRr8sjWbgY_V0rEsON125AbRdQ@mail.gmail.com>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
	<CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
	<CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
	<Pine.LNX.4.64.1212090907450.32638@1a-8140.antagus.de>
	<CAMwZ-tKu1QJwGqpZRNM_CHxZaRr8sjWbgY_V0rEsON125AbRdQ@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1212091706130.2416@1a-8140.antagus.de>

Am 09.12.12 um 01:59 schrieb Gabriel Gunderson:

> Strange, I searched http://jira.freeswitch.org for simpy-crash.lua and
> couldn't find it.

I have the actual git version.
But i can.t create a jira.
I.m a blind user, my browser is lynx.
I don.t no how can i upload with lynx?
I can  paste only 25 x 80 screenblocks with my brailledisyplay.
This is a handycap for me.
I can post bugs descriptions only in this list, to hope somebody can help 
me.
or to reproduce a bug.

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 steveayre at gmail.com  Sun Dec  9 23:03:29 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 9 Dec 2012 20:03:29 +0000
Subject: [Freeswitch-users] easy way to crash freeswitch
In-Reply-To: <Pine.LNX.4.64.1212091706130.2416@1a-8140.antagus.de>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
	<CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
	<CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
	<Pine.LNX.4.64.1212090907450.32638@1a-8140.antagus.de>
	<CAMwZ-tKu1QJwGqpZRNM_CHxZaRr8sjWbgY_V0rEsON125AbRdQ@mail.gmail.com>
	<Pine.LNX.4.64.1212091706130.2416@1a-8140.antagus.de>
Message-ID: <CAFiqYuk1xEshM886shyTMLpR6YKcT3TF0gW2vRCVJSKu-45spg@mail.gmail.com>

Are you able to use gdb to collect a backtrace from the coredump file, and
share it here along with the git revision?

-Steve



On 9 December 2012 16:14, Thomas Hoellriegel <admin at blindi.net> wrote:

> Am 09.12.12 um 01:59 schrieb Gabriel Gunderson:
>
>
>  Strange, I searched http://jira.freeswitch.org for simpy-crash.lua and
>> couldn't find it.
>>
>
> I have the actual git version.
> But i can.t create a jira.
> I.m a blind user, my browser is lynx.
> I don.t no how can i upload with lynx?
> I can  paste only 25 x 80 screenblocks with my brailledisyplay.
> This is a handycap for me.
> I can post bugs descriptions only in this list, to hope somebody can help
> me.
> or to reproduce a bug.
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Dec 10 00:50:50 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 9 Dec 2012 15:50:50 -0600
Subject: [Freeswitch-users] recovery_profile_name - any explanation?
In-Reply-To: <CAHKQagH-b0waBO2jQS5B0_GBZcJvm8T5+_Uugk4QHJ_cK_=8mg@mail.gmail.com>
References: <CAHKQagH-b0waBO2jQS5B0_GBZcJvm8T5+_Uugk4QHJ_cK_=8mg@mail.gmail.com>
Message-ID: <CAKbxfG800DH1egeHjBBziEh2eN9FXK4u69t_v0xWNTKx+r6Qgg@mail.gmail.com>

Its used internally so the core recovery engine knows which profile name
goes with the call.
The profile name to the core is just an arbitrary sub category of the call
where to mod_sofia it means the sip profile name.

You personally don't really have much use for it.


On Sat, Dec 8, 2012 at 4:18 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Hello,
>
> I've just been trawling Google for an explanation of what exactly the
> variable '*recovery_profile_name' means, but no luck.*
> *
> *
> *It seems to relate to the following change from 4 months ago;*
>
> https://github.com/freeSwitch/freeswitch/commit/2a8841ab666ec23ceb5688587bb14d16bf193b77
>
> The commit message states 'change mod_sofia to use new core based
> recovery engine'.
>
> But, it is unclear when this variable should be relied upon, and under
> what circumstances it will be used under.
>
> Any clarifications would be good, I will of course put in time to update
> the wiki too.
>
> Thanks
>
> Cal
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Anthony Minessale II

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
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From cal.leeming at simplicitymedialtd.co.uk  Mon Dec 10 01:57:17 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 9 Dec 2012 22:57:17 +0000
Subject: [Freeswitch-users] IVR menu delay - 1 second between events
In-Reply-To: <CAHKQagFQBeB+ho+rrniqZiqS1aocWy6BNN16Lv_2xm-mFDr+Nw@mail.gmail.com>
References: <CAHKQagF3C76QiybFNrS7GCdiGqaTbtPpQzD4wFivtJ+EK1CirQ@mail.gmail.com>
	<CAPNTnnrVW4Vpj4LcaNyeQrcmYq99DDF26E3UXpFoJyf=-q11fg@mail.gmail.com>
	<CAHKQagG1dW+MUtLYgoJHN0mWa0xowKGo4-VFR4wV2nZqcMKWiw@mail.gmail.com>
	<CAHKQagH-5HwgRpP77bVSLkOusVzMH6bU68Ox=jR7fvS7X-hkFg@mail.gmail.com>
	<CAHKQagFQBeB+ho+rrniqZiqS1aocWy6BNN16Lv_2xm-mFDr+Nw@mail.gmail.com>
Message-ID: <CAHKQagF=jW5ry3=6d8e3i0rwXMZUsRo5g5v_VHLQapDscQ7obg@mail.gmail.com>

This has now been patched in the core (sleep has been completely removed),
see JIRA ticket for details.

Thanks Ant!

Cal

On Sat, Dec 8, 2012 at 9:09 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> I've raised a JIRA ticket for this;
> http://jira.freeswitch.org/browse/FS-4924
>
> Cal
>
> On Fri, Dec 7, 2012 at 8:08 PM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> Just trawled through the source and found this;
>>
>> src/switch_ivr_menu.c:
>> if (status == SWITCH_STATUS_SUCCESS) {
>>  status = switch_ivr_sleep(session, 1000, SWITCH_FALSE, NULL);
>> }
>>
>> So, I changed 1000 to 10, the delay has now disappeared and it seems to
>> work perfectly.
>>
>
>
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From cal.leeming at simplicitymedialtd.co.uk  Mon Dec 10 02:02:13 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 9 Dec 2012 23:02:13 +0000
Subject: [Freeswitch-users] recovery_profile_name - any explanation?
In-Reply-To: <CAKbxfG800DH1egeHjBBziEh2eN9FXK4u69t_v0xWNTKx+r6Qgg@mail.gmail.com>
References: <CAHKQagH-b0waBO2jQS5B0_GBZcJvm8T5+_Uugk4QHJ_cK_=8mg@mail.gmail.com>
	<CAKbxfG800DH1egeHjBBziEh2eN9FXK4u69t_v0xWNTKx+r6Qgg@mail.gmail.com>
Message-ID: <CAHKQagE+Tvf-2qDsexgQQJf8dy8F9dpdSdDQ=Dxbpbz92t346Q@mail.gmail.com>

Documentation updated;
http://wiki.freeswitch.org/wiki/Variable_recovery_profile_name

Thanks!

Cal

On Sun, Dec 9, 2012 at 9:50 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Its used internally so the core recovery engine knows which profile name
> goes with the call.
> The profile name to the core is just an arbitrary sub category of the call
> where to mod_sofia it means the sip profile name.
>
> You personally don't really have much use for it.
>
>
> On Sat, Dec 8, 2012 at 4:18 PM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> Hello,
>>
>> I've just been trawling Google for an explanation of what exactly the
>> variable '*recovery_profile_name' means, but no luck.*
>> *
>> *
>> *It seems to relate to the following change from 4 months ago;*
>>
>> https://github.com/freeSwitch/freeswitch/commit/2a8841ab666ec23ceb5688587bb14d16bf193b77
>>
>> The commit message states 'change mod_sofia to use new core based
>> recovery engine'.
>>
>> But, it is unclear when this variable should be relied upon, and under
>> what circumstances it will be used under.
>>
>> Any clarifications would be good, I will of course put in time to update
>> the wiki too.
>>
>> Thanks
>>
>> Cal
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Dec 10 02:10:00 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sun, 9 Dec 2012 23:10:00 +0000
Subject: [Freeswitch-users] easy way to crash freeswitch
In-Reply-To: <CAFiqYuk1xEshM886shyTMLpR6YKcT3TF0gW2vRCVJSKu-45spg@mail.gmail.com>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
	<CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
	<CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
	<Pine.LNX.4.64.1212090907450.32638@1a-8140.antagus.de>
	<CAMwZ-tKu1QJwGqpZRNM_CHxZaRr8sjWbgY_V0rEsON125AbRdQ@mail.gmail.com>
	<Pine.LNX.4.64.1212091706130.2416@1a-8140.antagus.de>
	<CAFiqYuk1xEshM886shyTMLpR6YKcT3TF0gW2vRCVJSKu-45spg@mail.gmail.com>
Message-ID: <CAMbW9oK7kUuR2Aa-zY67-rVSK+AB=3O-iyMhGvG_D7orRepESQ@mail.gmail.com>

Obligatory:



On 9 December 2012 20:03, Steven Ayre <steveayre at gmail.com> wrote:

> Are you able to use gdb to collect a backtrace from the coredump file, and
> share it here along with the git revision?
>
> -Steve
>
>
>
> On 9 December 2012 16:14, Thomas Hoellriegel <admin at blindi.net> wrote:
>
>> Am 09.12.12 um 01:59 schrieb Gabriel Gunderson:
>>
>>
>>  Strange, I searched http://jira.freeswitch.org for simpy-crash.lua and
>>> couldn't find it.
>>>
>>
>> I have the actual git version.
>> But i can.t create a jira.
>> I.m a blind user, my browser is lynx.
>> I don.t no how can i upload with lynx?
>> I can  paste only 25 x 80 screenblocks with my brailledisyplay.
>> This is a handycap for me.
>> I can post bugs descriptions only in this list, to hope somebody can help
>> me.
>> or to reproduce a bug.
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
*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  Mon Dec 10 02:27:30 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 9 Dec 2012 15:27:30 -0800
Subject: [Freeswitch-users] Hello hackers!
In-Reply-To: <CAHKQagGOREuTuSyYhMi2zWg-ndOrqhtvKZre7EEc936xuv8SwQ@mail.gmail.com>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
	<CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
	<CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
	<CAHKQagGOREuTuSyYhMi2zWg-ndOrqhtvKZre7EEc936xuv8SwQ@mail.gmail.com>
Message-ID: <CAMWiMy_=h7ncK6s2F2DMtsAgHUPbZTNBeQkNf=Ks2c_-vC7cSg@mail.gmail.com>

S/hackers/script kidiots/g

On Sunday, December 9, 2012, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:
> Sounds like I might have slightly misinterpreted your initial email and
jumped too fast to a conclusion, so my apologies for this!
> I agree that it's rare to have a discussion about such topics without
things getting hot, and hearing someone else's thoughts on the subject has
been quite interesting.
> Cal
>
> On Sun, Dec 9, 2012 at 6:49 AM, Brian Foster <bdfoster at endigotech.com>
wrote:
>
> I appreciate your comments and concerns, and I respect the fact that we
can talk about this stuff without things getting ugly. My comments are
below.
> -BDF
>
>
> On Sat, Dec 8, 2012 at 11:46 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:
>
> Thank you for the detailed response.
>
> On Sun, Dec 9, 2012 at 1:29 AM, Brian Foster <bdfoster at endigotech.com>
wrote:
>
> The 'bad apple' in which I was referring to was using the same IP as a
client of ours. He was trying to DOS the honeypot from an IP I posted on
the mailing list when doing some testing for someone. I have no idea if he
read the post on the mailing list or not. It's not really of my concern.
>
> You know what happens when someone attacks one of our clients?
> We track them down, introduce them to the CTO/CEO of the company they
attacked, and give them an opportunity to prove themselves. I have been
involved in this process on several occasions now where the outcome has
been extremely positive. I'm not saying this works all the time, but
sometimes people don't need punishment, they need guidance.
>
> This is not really a practice of ours because there are so many people
out there that contribute to the problem. It's too time consuming to
educate those people. I really wish I could do that, but it's just not
feasible. We do not actively pursue these abusive IP's nor do we DDOS them
or fight back in any other way other than fighting back some of the noise
through blocking those activities on machines we support.
>
>
> If you got on the wiki and searched for fail2ban, you would be setting up
your server to jail the same IP's we are under the same circumstances. The
only difference is we log who gets caught by fail2ban and distribute the
list internally.
>
> We do not release this information per company policy. We also do not
gather this information from other sources. We only use the information we
gather through the processes we put in to place.
> My comment on the 180K IP's was mostly sarcastic, however. It probably
wasn't appropriate and I do apologize for that.
> I'm not exactly up to date on the legalities of releasing that type of
information so we rather not release it. It's nothing against the
freeswitch community or the open source community. We just don't like
getting in trouble.
> If we did spend the resources into making sure everything was legal on
the information regarding the 180K IP's, we would certainly release these
free of charge. It's not something I would be interested in making money
from.
>
> The concept is no different to email blacklist databases (e.g. XBL), and
there would be no legalities stopping you from releasing this information
into the public domain - only internal red tape and policies. I can say
this with at least some authority on the subject (although I'm by no means
an expert).
>
> I'm open to this idea, but I would have to consult attorneys to do this.
We operate very cautiously as in we do not operate in grey areas. If this
is a potential grey area we will certainly take the extra precautions in
order to prevent legal issues.
> This isn't some big company that has endless amounts of resources. We're
a small business, just like those we support. We also have to consider how
this effects our clients as well, and we
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From cal.leeming at simplicitymedialtd.co.uk  Mon Dec 10 03:08:34 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Mon, 10 Dec 2012 00:08:34 +0000
Subject: [Freeswitch-users] Hello hackers!
In-Reply-To: <CAMWiMy_=h7ncK6s2F2DMtsAgHUPbZTNBeQkNf=Ks2c_-vC7cSg@mail.gmail.com>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
	<CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
	<CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
	<CAHKQagGOREuTuSyYhMi2zWg-ndOrqhtvKZre7EEc936xuv8SwQ@mail.gmail.com>
	<CAMWiMy_=h7ncK6s2F2DMtsAgHUPbZTNBeQkNf=Ks2c_-vC7cSg@mail.gmail.com>
Message-ID: <CAHKQagFekD9_5Kug=hY0mt4Ywvk+7g4HOHjnh-r-nFMJUEbHkQ@mail.gmail.com>

That's an entirely different argument, one that has been debated and argued
far too many times - and is even more off-topic than the original
conversation ;)

Cal

On Sun, Dec 9, 2012 at 11:27 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> S/hackers/script kidiots/g
>
>
> On Sunday, December 9, 2012, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
> > Sounds like I might have slightly misinterpreted your initial email and
> jumped too fast to a conclusion, so my apologies for this!
> > I agree that it's rare to have a discussion about such topics without
> things getting hot, and hearing someone else's thoughts on the subject has
> been quite interesting.
> > Cal
> >
> > On Sun, Dec 9, 2012 at 6:49 AM, Brian Foster <bdfoster at endigotech.com>
> wrote:
> >
> > I appreciate your comments and concerns, and I respect the fact that we
> can talk about this stuff without things getting ugly. My comments are
> below.
> > -BDF
> >
> >
> > On Sat, Dec 8, 2012 at 11:46 PM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
> >
> > Thank you for the detailed response.
> >
> > On Sun, Dec 9, 2012 at 1:29 AM, Brian Foster <bdfoster at endigotech.com>
> wrote:
> >
> > The 'bad apple' in which I was referring to was using the same IP as a
> client of ours. He was trying to DOS the honeypot from an IP I posted on
> the mailing list when doing some testing for someone. I have no idea if he
> read the post on the mailing list or not. It's not really of my concern.
> >
> > You know what happens when someone attacks one of our clients?
> > We track them down, introduce them to the CTO/CEO of the company they
> attacked, and give them an opportunity to prove themselves. I have been
> involved in this process on several occasions now where the outcome has
> been extremely positive. I'm not saying this works all the time, but
> sometimes people don't need punishment, they need guidance.
> >
> > This is not really a practice of ours because there are so many people
> out there that contribute to the problem. It's too time consuming to
> educate those people. I really wish I could do that, but it's just not
> feasible. We do not actively pursue these abusive IP's nor do we DDOS them
> or fight back in any other way other than fighting back some of the noise
> through blocking those activities on machines we support.
> >
> >
> > If you got on the wiki and searched for fail2ban, you would be setting
> up your server to jail the same IP's we are under the same circumstances.
> The only difference is we log who gets caught by fail2ban and distribute
> the list internally.
> >
> > We do not release this information per company policy. We also do not
> gather this information from other sources. We only use the information we
> gather through the processes we put in to place.
> > My comment on the 180K IP's was mostly sarcastic, however. It probably
> wasn't appropriate and I do apologize for that.
> > I'm not exactly up to date on the legalities of releasing that type of
> information so we rather not release it. It's nothing against the
> freeswitch community or the open source community. We just don't like
> getting in trouble.
> > If we did spend the resources into making sure everything was legal on
> the information regarding the 180K IP's, we would certainly release these
> free of charge. It's not something I would be interested in making money
> from.
> >
> > The concept is no different to email blacklist databases (e.g. XBL), and
> there would be no legalities stopping you from releasing this information
> into the public domain - only internal red tape and policies. I can say
> this with at least some authority on the subject (although I'm by no means
> an expert).
> >
> > I'm open to this idea, but I would have to consult attorneys to do this.
> We operate very cautiously as in we do not operate in grey areas. If this
> is a potential grey area we will certainly take the extra precautions in
> order to prevent legal issues.
> > This isn't some big company that has endless amounts of resources. We're
> a small business, just like those we support. We also have to consider how
> this effects our clients as well, and we
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 krice at freeswitch.org  Mon Dec 10 05:36:58 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 09 Dec 2012 20:36:58 -0600
Subject: [Freeswitch-users] easy way to crash freeswitch
In-Reply-To: <Pine.LNX.4.64.1212091706130.2416@1a-8140.antagus.de>
Message-ID: <CCEAA6EA.61895%krice@freeswitch.org>

Hey Thomas,
Thanks for being the perfect Object lesson here...

As a blind person, Thomas is one of the few users that has an excuse to not
open a jira. Yes for the vision impaired Jira is a pain to use. I'm working
on some ways around this. However, this is the perfect time to address how
to handle this.

1) there are tools in tree such as ${FS_SOURCE}/support-d/fscore_pb  which
is a small script that will gather a backtrace from the core file resulting
from a segfault and pastebin it to the FreeSwitch Pastebin. (please note you
may have to configure your system to allow FreeSWITCH to actuall drop a core
file.)

2) Log files. (the stuff from /usr/local/freeswitch/log/freeswitch.log is
perfect here)

3) description of the problem. With an example of how to on demand tickle
the bug if known.

4) Email the list let us know you need help.

Now that being said, This is not an invitation for everyone to start
emailing the list for help on posting a Jira, 99% of the FS user base are
sighted users. That means you can open a jira yourself. Like any telephony
software we have a fairly good sized blind user community. We know who they
are, and for the most part they are fairly well involved.

I wont call them out individually, but they know who they are, and quite
frankly all of them are super nice guys.

Thomas, don't take the "well you didn't open a jira" thing personally, its
just we deal with able bodied people that are just way too lazy to follow
the procedure that helps us the most.

K


On 12/9/12 10:14 AM, "Thomas Hoellriegel" <admin at blindi.net> wrote:

> Am 09.12.12 um 01:59 schrieb Gabriel Gunderson:
> 
>> Strange, I searched http://jira.freeswitch.org for simpy-crash.lua and
>> couldn't find it.
> 
> I have the actual git version.
> But i can.t create a jira.
> I.m a blind user, my browser is lynx.
> I don.t no how can i upload with lynx?
> I can  paste only 25 x 80 screenblocks with my brailledisyplay.
> This is a handycap for me.
> I can post bugs descriptions only in this list, to hope somebody can help
> me.
> or to reproduce a bug.
> 
> 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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From moises.silva at gmail.com  Mon Dec 10 07:48:24 2012
From: moises.silva at gmail.com (Moises Silva)
Date: Sun, 9 Dec 2012 23:48:24 -0500
Subject: [Freeswitch-users] FreeTDM ftmod pri tap error
In-Reply-To: <CAEymo133NELNUgy1jwUbtEQwH_gznq6M2v=cQNaJ9Lmj9woH0w@mail.gmail.com>
References: <CAEymo11XeECiWjUpckjinxRiY0AO7HcAqmK6HTjpdPto6CwpiQ@mail.gmail.com>
	<CAEymo12bugji+6UbV05FshCLAVBnJbOXpUj4gH1GyA0S_qa=Wg@mail.gmail.com>
	<CAEymo133NELNUgy1jwUbtEQwH_gznq6M2v=cQNaJ9Lmj9woH0w@mail.gmail.com>
Message-ID: <CAA4nhyCQZmkGmmXUGRd+mZjre6MkyiPnDqmgEZaDGHWiZ+GQuQ@mail.gmail.com>

It's being used in production in several sites. Having said that, some
features such as NFAS are not supported.

*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>





On Tue, Nov 27, 2012 at 11:52 AM, Gopalakrishnan N <
gopalakrishnan.an at gmail.com> wrote:

> Has somebody used pritap module?
>
>
> On Mon, Nov 26, 2012 at 7:13 PM, Gopalakrishnan N <
> gopalakrishnan.an at gmail.com> wrote:
>
>> Even I tried the moy tap1.4 complete patch with libpri, when i try to
>> execute with this ./configure --with-libpri --with-pritap
>> --prefix=/usr/local/freeswitch/ in make I am getting error like,
>>
>> make: *** [ftmod_libpri_la-ftmod_libpri.lo] Error 1
>>
>> Regards.
>>
>>
>> On Mon, Nov 26, 2012 at 6:21 PM, Gopalakrishnan N <
>> gopalakrishnan.an at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> Am trying to install ftmod pri tap by following this link
>>> http://wiki.freeswitch.org/wiki/FreeTDM#Tapping and enabled
>>> --with-pritap while doing ./configure, and while doing make I get error.
>>>
>>> Can someone help me on this. Do I need to update any patch for tap?
>>>
>>> My installation log is attached here.
>>>
>>> Regards.
>>> Gopal.
>>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 gabe at gundy.org  Mon Dec 10 09:41:43 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sun, 9 Dec 2012 23:41:43 -0700
Subject: [Freeswitch-users] easy way to crash freeswitch
In-Reply-To: <Pine.LNX.4.64.1212091706130.2416@1a-8140.antagus.de>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
	<CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
	<CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
	<Pine.LNX.4.64.1212090907450.32638@1a-8140.antagus.de>
	<CAMwZ-tKu1QJwGqpZRNM_CHxZaRr8sjWbgY_V0rEsON125AbRdQ@mail.gmail.com>
	<Pine.LNX.4.64.1212091706130.2416@1a-8140.antagus.de>
Message-ID: <CAMwZ-tJHBKfHsrHGwp_45Zf8jy5HA-vhSg9Pj0qCMMHtnDPUmA@mail.gmail.com>

On Sun, Dec 9, 2012 at 9:14 AM, Thomas Hoellriegel <admin at blindi.net> wrote:
> I can post bugs descriptions only in this list, to hope somebody can help
> me. or to reproduce a bug.

Heh, I didn't know. :) I will help file the bug for you. Give me a few
days for work to slow down. I'll follow the steps you outline and make
sure I can reproduce it. When I can, I'll file and follow up on the
Jira.

Thanks for bringing it to our attention and being a good sport about a
gentle ribbing.

Best,
Gabe


From Sirish.MasurMohan at oa.com.au  Mon Dec 10 10:40:24 2012
From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan)
Date: Mon, 10 Dec 2012 18:40:24 +1100
Subject: [Freeswitch-users] FreeSWITCH+Sangoma with dialup modems
Message-ID: <965759A53E43FE439E43565A7715E5F058F1BE0EE8@oa-exchange1.oa.com.au>

Hello All,

I am a newbie to FreeSWITCH, and have been experimenting with it. I am currently facing a problem, which, to be honest, I am not sure if is a FreeSWITCH misconfiguration problem, or if it is totally unrelated!

My setup is as follows:
Serial port based device(Caller)->Dlink dialup modem(DFM562E)->PBX->Sangoma A102 E1 Link(and FreeSWITCH)->PBX->Dlink dialup modem(DFM562E)->Server(Receiver)

Problem:
When the caller makes the call (using the modem with a lower baud rate 1200), Receiver modem receives the RING and answers the call (using the ATA command). But for some reason, both the ends timeout waiting for modem CONNECT.

Freeswitch logs show:
freeswitch at clock2> 2012-12-10 16:53:30.975599 [INFO] ftmod_sangoma_isdn_stack_rcv.c:96 [s1c30][1:31] Received SETUP (suId:1 suInstId:0 spInstId:5)
2012-12-10 16:53:30.975599 [INFO] ftmod_sangoma_isdn_stack_hndl.c:153 [s1c30][1:31] Incoming call: Called No:[997870] Calling No:[988869]
2012-12-10 16:53:30.975599 [NOTICE] switch_channel.c:953 New Channel FreeTDM/1:30/997870 [0f80e853-5e9b-4c33-af82-b960b261af19]
2012-12-10 16:53:30.975599 [INFO] ftmod_sangoma_isdn_stack_out.c:175 [s1c30][1:31] Sending PROCEED (suId:1 suInstId:5 spInstId:5 dchan:1 ces:0)
2012-12-10 16:53:30.975599 [INFO] mod_dialplan_xml.c:498 Processing  <988869>->997870 in context default
2012-12-10 16:53:30.975599 [NOTICE] switch_channel.c:953 New Channel FreeTDM/1:1/988713 [b7eed81e-0404-4a79-b9db-17fa59add3bd]
2012-12-10 16:53:30.975599 [INFO] ftmod_sangoma_isdn_stack_out.c:62 [s1c1][1:1] Outgoing call: Called No:[988713] Calling No:[988869]
2012-12-10 16:53:30.975599 [INFO] ftmod_sangoma_isdn_stack_out.c:79 [s1c1][1:1] Sending SETUP (suId:1 suInstId:6 spInstId:0 dchan:1 ces:0)
2012-12-10 16:53:31.195550 [INFO] ftmod_sangoma_isdn_stack_rcv.c:194 [s1c1][1:1] Received PROCEED (suId:1 suInstId:6 spInstId:6 ces:0)
2012-12-10 16:53:32.695541 [INFO] ftmod_sangoma_isdn_stack_rcv.c:194 [s1c1][1:1] Received ALERT (suId:1 suInstId:6 spInstId:6 ces:0)
2012-12-10 16:53:32.695541 [NOTICE] mod_freetdm.c:2716 Ring-Ready FreeTDM/1:1/988713!
2012-12-10 16:53:32.695541 [INFO] ftmod_sangoma_isdn_stack_out.c:233 [s1c30][1:31] Sending ALERT (suId:1 suInstId:5 spInstId:5 dchan:1 ces:0)
2012-12-10 16:53:32.695541 [NOTICE] switch_ivr_originate.c:521 Ring Ready FreeTDM/1:30/997870!
2012-12-10 16:53:32.695541 [NOTICE] switch_ivr_originate.c:521 Ring-Ready FreeTDM/1:30/997870!
2012-12-10 16:53:34.635576 [INFO] ftmod_sangoma_isdn_stack_rcv.c:144 [s1c1][1:1] Received CONNECT/CONNECT ACK (suId:1 suInstId:6 spInstId:6 ces:0)
2012-12-10 16:53:34.635576 [NOTICE] mod_freetdm.c:2686 Channel [FreeTDM/1:1/988713] has been answered
2012-12-10 16:53:34.635576 [INFO] ftmod_sangoma_isdn_stack_out.c:269 [s1c30][1:31] Sending CONNECT (suId:1 suInstId:5 spInstId:5 dchan:1 ces:0)
2012-12-10 16:53:34.635576 [NOTICE] switch_ivr_originate.c:3376 Channel [FreeTDM/1:30/997870] has been answered
2012-12-10 16:53:34.735754 [INFO] ftmod_sangoma_isdn_stack_rcv.c:144 [s1c30][1:31] Received CONNECT/CONNECT ACK (suId:1 suInstId:5 spInstId:5 ces:0)


And my dial plan is:
    <extension name="997870">
      <condition field="destination_number" expression="997870">
        <action application="set" data="api_hangup_hook=lua hook-test.lua"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="TEST_VAR=${lua(pool-test.lua)}"/>
        <action application="set" data="MY_VAR=${lua(helloworld.lua arg1 arg2)}"/>
        <action application="bridge" data="freetdm/wp1/a/988713"/>
        <!--<action application="answer"/>
        <action application="sleep" data="10000"/>
        <action application="hangup"/>-->
      </condition>
    </extension>

Could you please help?

With regards,
Sirish

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From andrew at cassidywebservices.co.uk  Mon Dec 10 11:54:54 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Mon, 10 Dec 2012 08:54:54 +0000
Subject: [Freeswitch-users] easy way to crash freeswitch
In-Reply-To: <CAMwZ-tJHBKfHsrHGwp_45Zf8jy5HA-vhSg9Pj0qCMMHtnDPUmA@mail.gmail.com>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
	<CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
	<CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
	<Pine.LNX.4.64.1212090907450.32638@1a-8140.antagus.de>
	<CAMwZ-tKu1QJwGqpZRNM_CHxZaRr8sjWbgY_V0rEsON125AbRdQ@mail.gmail.com>
	<Pine.LNX.4.64.1212091706130.2416@1a-8140.antagus.de>
	<CAMwZ-tJHBKfHsrHGwp_45Zf8jy5HA-vhSg9Pj0qCMMHtnDPUmA@mail.gmail.com>
Message-ID: <CAMbW9o+E9Zoi9c_KgD1UFMpo76f34rdtpAR6-fzhccYEG-NayQ@mail.gmail.com>

That'll teach me to read the whole thread wont it!

My wife is actually a rehab worker for people with visual impairments, she
works for a charity that helps people in our local area in the UK.

On 10 December 2012 06:41, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sun, Dec 9, 2012 at 9:14 AM, Thomas Hoellriegel <admin at blindi.net>
> wrote:
> > I can post bugs descriptions only in this list, to hope somebody can help
> > me. or to reproduce a bug.
>
> Heh, I didn't know. :) I will help file the bug for you. Give me a few
> days for work to slow down. I'll follow the steps you outline and make
> sure I can reproduce it. When I can, I'll file and follow up on the
> Jira.
>
> Thanks for bringing it to our attention and being a good sport about a
> gentle ribbing.
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 miha at softnet.si  Mon Dec 10 12:21:33 2012
From: miha at softnet.si (Miha)
Date: Mon, 10 Dec 2012 10:21:33 +0100
Subject: [Freeswitch-users] media problem afet 302
Message-ID: <50C5A99D.1030305@softnet.si>

Hi,

Scenario:

fs ----> SBC---> innovaphone pbx

calls goes throught and media is ok (g711u). When on other side 
(innovaphone pbx) someone do a 302 redirect innovaphone sends a sdb 
request in which prepose g711u, FS trys with g711u and send back not 
accabtable and media does not work.

In attachment I am sending jpg of wireshark.

 From left to right: FS, SBC, SBC MEDIA (on different ip than sbc).

How to deal with this?

Thanks!
Miha
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From avi at avimarcus.net  Mon Dec 10 12:52:47 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 10 Dec 2012 11:52:47 +0200
Subject: [Freeswitch-users] media problem afet 302
In-Reply-To: <50C5A99D.1030305@softnet.si>
References: <50C5A99D.1030305@softnet.si>
Message-ID: <CALNZuwmM4Tphy+B1MvKwmBD_csszGrBF_FknJQ-H7pmoDmykyA@mail.gmail.com>

Uploading the actual wirrshark file (don't attach) will likely help someone
diagnose this easier.

-Avi

On Dec 10, 2012 11:24 AM, "Miha" <miha at softnet.si> wrote:

Hi,

Scenario:

fs ----> SBC---> innovaphone pbx

calls goes throught and media is ok (g711u). When on other side
(innovaphone pbx) someone do a 302 redirect innovaphone sends a sdb request
in which prepose g711u, FS trys with g711u and send back not accabtable and
media does not work.

In attachment I am sending jpg of wireshark.

>From left to right: FS, SBC, SBC MEDIA (on different ip than sbc).

How to deal with this?

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From foxb at abv.bg  Mon Dec 10 13:12:22 2012
From: foxb at abv.bg (Hristo Benev)
Date: Mon, 10 Dec 2012 12:12:22 +0200 (EET)
Subject: [Freeswitch-users] freeswitch repo
Message-ID: <874722339.33234.1355134342790.JavaMail.apache@nm52.abv.bg>

When updating I have following error:

Package freeswitch-application-httapi-1.2.5.1-1.el6.i386.rpm is not signed


Is that normal?

Thanks


From lists at kavun.ch  Mon Dec 10 16:29:45 2012
From: lists at kavun.ch (Emrah)
Date: Mon, 10 Dec 2012 08:29:45 -0500
Subject: [Freeswitch-users] Improve generated tones quality
Message-ID: <65F91A2E-F875-4070-8910-D61DC5349BD1@kavun.ch>

Hi list,

I am experiencing a recurring problem with generated tones across different machines and FS versions. 

The start or the end of the tone is often distorted. For some interesting reasons, this does not happen with all equipments and not all codecs.
Typical scenario in PCMU/A:
If someone calls me on an inbound DID, I pick up the call and than transfer the call to an internal extension, the original caller will hear MOH followed by a ringback that starts with a sort of "fade in".
Something similar happens with the voicemail where the end of the beep tone is faded out.

Even though this hasn't much to do with Early Media, I tried playing with bridge_early_media to no avail. It added a 5 seconds latency at the start of my call and did not solve my issue.

Your take on this would be greatly appreciated. Is there a way to fine tune PCMU / PCMA for better performance? Can we initiate a little buffering on system generated tones?

Cheers,
Emrah
Ps: It sounds weird, but I believe this also happens if the played tone is an actual sound file.

From dgarcia at anew.com.ve  Mon Dec 10 16:57:05 2012
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Mon, 10 Dec 2012 09:27:05 -0430
Subject: [Freeswitch-users] Avaya CS1000M integration guide
In-Reply-To: <005301cdd5f6$11bc5a00$35350e00$@com>
References: <031701cdc783$b0eae640$12c0b2c0$@bizfocused.com>
	<005301cdd5f6$11bc5a00$35350e00$@com>
Message-ID: <50C5EA31.1030408@anew.com.ve>

Hi, Bote,

First, good luck with the CS1000. CS1000 could be brand new but realese 
7.5 is not new, when Avaya bought Nortel, Avaya will not continue CS1000 
development, just support.

My advice, use all debugging tools that you can find, CS1000 sometimes 
can be tricky (and if the pbx team support is not "skilled") you could 
get some headaches.

Second, try to get info about CS1000 architecture. CS1000 use "media 
card" that make the conversion from TDM<=>SIP, how many cards, and IP 
services will uses these cards, etc. Your FS installation could process 
hundreds of calls but the CS1000 have to provide enough cards to support 
all call legs. Apparently, CS1000 does not do trunking optimization, so 
one call could consume several media cards resources.




On 12/9/2012 5:45 AM, Bote Man wrote:
>
> I know this is somewhat off-topic for this list, but I'm quickly 
> running out of options.
>
> I need pointers to what Avaya considers an "integration guide" that 
> tells them what to enter into their CS1k M pbx to a plain SIP trunk to 
> talk to FreeSWITCH. Their pbx is brand new so I assume it is running 
> the latest 7.5 or thereabouts.
>
> I'm hitting the bureaucratic roadblock with their tech support on this 
> point so I need to send them document that will move us off dead-center.
>
> Thanks.
>
> John Boteler
>
> Bote Communications
>
> Fort Lauderdale, FL
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2221 / Virus Database: 2634/5438 - Release Date: 12/05/12
>


-- 
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 vbvbrj at gmail.com  Mon Dec 10 16:59:06 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Mon, 10 Dec 2012 15:59:06 +0200
Subject: [Freeswitch-users] Natting question.
Message-ID: <50C5EAAA.6040100@gmail.com>

Hello.

I have a FS behind firewall on which ports 5080,5060 udp are mapped to 
the FS server. A client connected directly to Internet, say using GPRS, 
authorise to FS and calling a number audio is heared. When this client 
is connected via WiFi to some router connected directly to Internet, 
client authorises, calls but there is no audio.

So I'm stump on where to look for issue: on FS server side with firewall 
or on client side behind router?

Thank you.
-- 
Mimiko desu.


From miha at softnet.si  Mon Dec 10 17:13:04 2012
From: miha at softnet.si (Miha)
Date: Mon, 10 Dec 2012 15:13:04 +0100
Subject: [Freeswitch-users] media problem afet 302
In-Reply-To: <CALNZuwmM4Tphy+B1MvKwmBD_csszGrBF_FknJQ-H7pmoDmykyA@mail.gmail.com>
References: <50C5A99D.1030305@softnet.si>
	<CALNZuwmM4Tphy+B1MvKwmBD_csszGrBF_FknJQ-H7pmoDmykyA@mail.gmail.com>
Message-ID: <50C5EDF0.5070805@softnet.si>

HI,

here is a sip trace from FS.

http://pastebin.freeswitch.org/20305

xxx.xxx.xxx.xxx is FS, yyy.yyy.yyy.yyy is SBC, aaa.aaa.aaa.aaa SBC media :)

thanks for help!

Miha

Dne 12/10/2012 10:52 AM, pis(e Avi Marcus:
>
> Uploading the actual wirrshark file (don't attach) will likely help 
> someone diagnose this easier.
>
> -Avi
>
>> On Dec 10, 2012 11:24 AM, "Miha" <miha at softnet.si 
>> <mailto:miha at softnet.si>> wrote:
>>
>> Hi,
>>
>> Scenario:
>>
>> fs ----> SBC---> innovaphone pbx
>>
>> calls goes throught and media is ok (g711u). When on other side 
>> (innovaphone pbx) someone do a 302 redirect innovaphone sends a sdb 
>> request in which prepose g711u, FS trys with g711u and send back not 
>> accabtable and media does not work.
>>
>> In attachment I am sending jpg of wireshark.
>>
>> >From left to right: FS, SBC, SBC MEDIA (on different ip than sbc).
>>
>> How to deal with this?
>>
>> Thanks!
>> Miha
>>
>> _________________________________________________________________________
>> 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
> 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 garbytrash at gmail.com  Mon Dec 10 17:58:21 2012
From: garbytrash at gmail.com (Zenny)
Date: Mon, 10 Dec 2012 15:58:21 +0100
Subject: [Freeswitch-users] Natting question.
In-Reply-To: <50C5EAAA.6040100@gmail.com>
References: <50C5EAAA.6040100@gmail.com>
Message-ID: <CACuV5sAJHmkLGtYZ0w+B127Lfz8Gi5YgK-yku5H6VoHiFK9DuQ@mail.gmail.com>

It seems like it is associated with either the portforwarding of the
wifi router or NAT prolem, I guess. The following link could be of
help, imho:
http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT

/zenny

On 12/10/12, Mimiko <vbvbrj at gmail.com> wrote:
> Hello.
>
> I have a FS behind firewall on which ports 5080,5060 udp are mapped to
> the FS server. A client connected directly to Internet, say using GPRS,
> authorise to FS and calling a number audio is heared. When this client
> is connected via WiFi to some router connected directly to Internet,
> client authorises, calls but there is no audio.
>
> So I'm stump on where to look for issue: on FS server side with firewall
> or on client side behind router?
>
> Thank you.
> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>


From marcdecorny at gmail.com  Mon Dec 10 18:19:24 2012
From: marcdecorny at gmail.com (Marc de Corny)
Date: Mon, 10 Dec 2012 15:19:24 +0000
Subject: [Freeswitch-users] Cannot Kill stale channels
Message-ID: <CA+myZ6dBpJoZWfrA-Wc+jiAxtTP-sd7NvgDFFaxK2d8=2eTAdg@mail.gmail.com>

Hi

I have calls that come into a queue and then get pulled out by a lua script
in background and transfered to a destination. all works fine except that
it looks like the sessions a not clearing from FS when all the parties
clear the calls.

When I do show channels, I get this for example:
         uuid direction created created_epoch name state cid_name cid_num
ip_addr dest presence_id presence_data callstate callee_name callee_num
callee_direction call_uuid hostname sent_callee_name sent_callee_num
5e8215e7-b036-4cce-9b96-e57db92f13ee outbound 03/12/2012 09:32 1.35E+09
sofia/external/02031950164 CS_HANGUP Outbound Call 2031950164 135.196.144.32
agent_to_queue_paymentsense_queue_service  ACTIVE  SEND
5e8215e7-b036-4cce-9b96-e57db92f13ee freeswitch2 4.4209E+11 2089623100

But when I try to get kill or query it, I get a message that the call is
not anywhere in the FS.

/usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_kill
5e8215e7-b036-4cce-9b96-e57db92f13ee"
-ERR No Such Channel!

/usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_exists
5e8215e7-b036-4cce-9b96-e57db92f13ee"
false

Does anyone have an idea how I can kill that call. If I restart the FS,
they clear, but the problem is that they are hitting my limit on number of
simultaneous calls.

Any help much appreciated.

Thansk
marc
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From krice at freeswitch.org  Mon Dec 10 18:25:22 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 10 Dec 2012 09:25:22 -0600
Subject: [Freeswitch-users] freeswitch repo
In-Reply-To: <874722339.33234.1355134342790.JavaMail.apache@nm52.abv.bg>
Message-ID: <CCEB5B02.618C5%krice@freeswitch.org>

Its being worked ok


On 12/10/12 4:12 AM, "Hristo Benev" <foxb at abv.bg> wrote:

> When updating I have following error:
> 
> Package freeswitch-application-httapi-1.2.5.1-1.el6.i386.rpm is not signed
> 
> 
> Is that normal?
> 
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From bpriddy at bryantschools.org  Mon Dec 10 18:39:16 2012
From: bpriddy at bryantschools.org (Blake Priddy)
Date: Mon, 10 Dec 2012 09:39:16 -0600
Subject: [Freeswitch-users] 401
Message-ID: <CADK-Bjtuu+dEXQRxOxDzkHSdwO-7JuLg=ciuCUD6S4iUq5hE_w@mail.gmail.com>

So I updated to the latest git head and now several phones NOT ALL,
including mine :( are giving me

2012-12-10 09:36:04.584347 [DEBUG] sofia_reg.c:1498 Send challenge for [
1206 at pbx.bryantschools.org]
2012-12-10 09:36:04.584347 [WARNING] sofia_reg.c:1502 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [1206 at pbx.bryantschools.org]
from ip 10.10.24.20
send 679 bytes to udp/[10.10.24.20]:3072 at 15:36:04.585091:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 10.10.24.20:3072;branch=z9hG4bK-n757mrbdlqdw;rport=3072
   From: "Blake Priddy" <sip:1206 at pbx.bryantschools.org>;tag=dbou3jy9kf
   To: "Blake Priddy" <sip:1206 at pbx.bryantschools.org>;tag=v2Kpa56439DgK
   Call-ID: 4e000000e08c-cm5eg6nr96al
   CSeq: 7373 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.3.9b+git~20121206T182231Z~a7fafb2039
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   WWW-Authenticate: Digest realm="pbx.bryantschools.org",
nonce="4f0b07b6-42df-11e2-ab4a-4bb963400da6", algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 884 bytes from udp/[10.10.24.20]:3072 at 15:36:04.680840:
   ------------------------------------------------------------------------
   REGISTER sip:pbx.bryantschools.org SIP/2.0
   Via: SIP/2.0/UDP 10.10.24.20:3072;branch=z9hG4bK-cpukgis6pexk;rport
   From: "Blake Priddy" <sip:1206 at pbx.bryantschools.org>;tag=dbou3jy9kf
   To: "Blake Priddy" <sip:1206 at pbx.bryantschools.org>
   Call-ID: 4e000000e08c-cm5eg6nr96al
   CSeq: 7374 REGISTER
   Max-Forwards: 70
   Contact: <sip:1206 at 10.10.24.20:3072
>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:cb7105d7-ce2c-4835-8300-000413411CA5>"
   User-Agent: snom870/8.7.3.10
   Allow-Events: dialog
   X-Real-IP: 10.10.24.20
   Supported: path, gruu
   WWW-Contact: <http://10.10.24.20:80>
   WWW-Contact: <https://10.10.24.20:443>
   Authorization: Digest username="1206",realm="pbx.bryantschools.org
",nonce="4f0b07b6-42df-11e2-ab4a-4bb963400da6",uri="sip:
pbx.bryantschools.org
",qop=auth,nc=00000001,cnonce="4afc06b9",response="8c9a1fcedd906467bd0afa2b98b50829",algorithm=MD5
   Expires: 3600
   Content-Length: 0


So........ Yea ha

-- 

*Blakelund Priddy*
Network Systems Engineer
Bryant Public School District
Bryant, Arkansas 72022
http://www.bryantschools.org
p 501-653-5038
f 501-847-5656
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From peter.olsson at visionutveckling.se  Mon Dec 10 19:39:27 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 10 Dec 2012 16:39:27 +0000
Subject: [Freeswitch-users] 401
In-Reply-To: <CADK-Bjtuu+dEXQRxOxDzkHSdwO-7JuLg=ciuCUD6S4iUq5hE_w@mail.gmail.com>
References: <CADK-Bjtuu+dEXQRxOxDzkHSdwO-7JuLg=ciuCUD6S4iUq5hE_w@mail.gmail.com>
Message-ID: <A9BD8A0C-C0A1-4102-9600-CE2A0F57846E@visionutveckling.se>

You need to give more information. Is something not working? As far as I can see this is just normal authentication stuff - 401 is sent so the client sends the credentials back to the server.

/Peter

10 dec 2012 kl. 16:46 skrev "Blake Priddy" <bpriddy at bryantschools.org<mailto:bpriddy at bryantschools.org>>:

So I updated to the latest git head and now several phones NOT ALL, including mine :( are giving me

2012-12-10 09:36:04.584347 [DEBUG] sofia_reg.c:1498 Send challenge for [1206 at pbx.bryantschools.org<mailto:1206 at pbx.bryantschools.org>]
2012-12-10 09:36:04.584347 [WARNING] sofia_reg.c:1502 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1206 at pbx.bryantschools.org<mailto:1206 at pbx.bryantschools.org>] from ip 10.10.24.20
send 679 bytes to udp/[10.10.24.20]:3072 at 15:36:04.585091:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 10.10.24.20:3072;branch=z9hG4bK-n757mrbdlqdw;rport=3072
   From: "Blake Priddy" <sip:1206 at pbx.bryantschools.org<mailto:sip%3A1206 at pbx.bryantschools.org>>;tag=dbou3jy9kf
   To: "Blake Priddy" <sip:1206 at pbx.bryantschools.org<mailto:sip%3A1206 at pbx.bryantschools.org>>;tag=v2Kpa56439DgK
   Call-ID: 4e000000e08c-cm5eg6nr96al
   CSeq: 7373 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.3.9b+git~20121206T182231Z~a7fafb2039
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   WWW-Authenticate: Digest realm="pbx.bryantschools.org<http://pbx.bryantschools.org>", nonce="4f0b07b6-42df-11e2-ab4a-4bb963400da6", algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 884 bytes from udp/[10.10.24.20]:3072 at 15:36:04.680840:
   ------------------------------------------------------------------------
   REGISTER sip:pbx.bryantschools.org<http://pbx.bryantschools.org> SIP/2.0
   Via: SIP/2.0/UDP 10.10.24.20:3072;branch=z9hG4bK-cpukgis6pexk;rport
   From: "Blake Priddy" <sip:1206 at pbx.bryantschools.org<mailto:sip%3A1206 at pbx.bryantschools.org>>;tag=dbou3jy9kf
   To: "Blake Priddy" <sip:1206 at pbx.bryantschools.org<mailto:sip%3A1206 at pbx.bryantschools.org>>
   Call-ID: 4e000000e08c-cm5eg6nr96al
   CSeq: 7374 REGISTER
   Max-Forwards: 70
   Contact: <sip:1206 at 10.10.24.20:3072<http://sip:1206 at 10.10.24.20:3072>>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:cb7105d7-ce2c-4835-8300-000413411CA5>"
   User-Agent: snom870/8.7.3.10<http://8.7.3.10>
   Allow-Events: dialog
   X-Real-IP: 10.10.24.20
   Supported: path, gruu
   WWW-Contact: <http://10.10.24.20:80>
   WWW-Contact: <https://10.10.24.20:443>
   Authorization: Digest username="1206",realm="pbx.bryantschools.org<http://pbx.bryantschools.org>",nonce="4f0b07b6-42df-11e2-ab4a-4bb963400da6",uri="sip:pbx.bryantschools.org<http://pbx.bryantschools.org>",qop=auth,nc=00000001,cnonce="4afc06b9",response="8c9a1fcedd906467bd0afa2b98b50829",algorithm=MD5
   Expires: 3600
   Content-Length: 0


So........ Yea ha

--
[http://dl.dropbox.com/u/6313391/BryantSchoolDist_Seal_final_color%20%281%291.jpg]
Blakelund Priddy
Network Systems Engineer
Bryant Public School District
Bryant, Arkansas 72022
http://www.bryantschools.org<http://www.bryantschools.org/>
p 501-653-5038
f 501-847-5656

!DSPAM:50c5ffa732761244698123!
_________________________________________________________________________
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>
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From peter.olsson at visionutveckling.se  Mon Dec 10 19:40:31 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 10 Dec 2012 16:40:31 +0000
Subject: [Freeswitch-users] Cannot Kill stale channels
In-Reply-To: <CA+myZ6dBpJoZWfrA-Wc+jiAxtTP-sd7NvgDFFaxK2d8=2eTAdg@mail.gmail.com>
References: <CA+myZ6dBpJoZWfrA-Wc+jiAxtTP-sd7NvgDFFaxK2d8=2eTAdg@mail.gmail.com>
Message-ID: <4424748D-98BF-4BDA-A616-F6809467C3D0@visionutveckling.se>

Are you exiting the lua script? Usually when this happens it means you have not
released all references to the call session object.

/Peter

10 dec 2012 kl. 16:28 skrev "Marc de Corny" <marcdecorny at gmail.com<mailto:marcdecorny at gmail.com>>:

Hi

I have calls that come into a queue and then get pulled out by a lua script in background and transfered to a destination. all works fine except that it looks like the sessions a not clearing from FS when all the parties clear the calls.

When I do show channels, I get this for example:
uuid    direction       created created_epoch   name    state   cid_name        cid_num ip_addr dest    presence_id     presence_data   callstate       callee_name     callee_num      callee_direction        call_uuid       hostname        sent_callee_name        sent_callee_num
5e8215e7-b036-4cce-9b96-e57db92f13ee    outbound        03/12/2012 09:32        1.35E+09        sofia/external/02031950164      CS_HANGUP       Outbound Call   2031950164      135.196.144.32  agent_to_queue_paymentsense_queue_service                       ACTIVE                  SEND    5e8215e7-b036-4cce-9b96-e57db92f13ee    freeswitch2     4.4209E+11      2089623100

But when I try to get kill or query it, I get a message that the call is not anywhere in the FS.

/usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_kill 5e8215e7-b036-4cce-9b96-e57db92f13ee"
-ERR No Such Channel!

/usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_exists 5e8215e7-b036-4cce-9b96-e57db92f13ee"
false

Does anyone have an idea how I can kill that call. If I restart the FS, they clear, but the problem is that they are hitting my limit on number of simultaneous calls.

Any help much appreciated.

Thansk
marc






!DSPAM:50c5fb8c32767955115405!
_________________________________________________________________________
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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
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From sdevoy at bizfocused.com  Mon Dec 10 19:57:45 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 10 Dec 2012 11:57:45 -0500
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
Message-ID: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>

HI,

 

I have a bridge statement used dozens and dozens of times a day:

EXECUTE sofia/external_noauth/anonymous at 64.136.174.30
bridge(user/300 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/301 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/203 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/302 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/303 at fs_mbri2.bizfocused.com)

 

Very occasionally, recently extensions (at least 300 and 302) have returned
[CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

All of these extensions are at the same remote location using the same wan
links/routers, etc.

 

What does that mean?  

 

Pastebin: http://pastebin.freeswitch.org/pastebin.php?dl=20306

 

Also, the 8 second delay appears to be ignored (I think, I have never asked
the customer or been on site to test).  Any ideas?

 

Thanks,

Sean

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From sdevoy at bizfocused.com  Mon Dec 10 20:08:29 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 10 Dec 2012 12:08:29 -0500
Subject: [Freeswitch-users] Wish Lists - 'tis the season
Message-ID: <31fb01cdd6f8$fa356850$eea038f0$@bizfocused.com>

Hey Guys,

 

Update those wish lists.  Anthony's starts at like $495!  I want to show my
love, but I am a poor VOIP provider and not spending $495 on my wife this
season!  I hooked up BF and MC but they could use a few more suggestions in
the $25 to $50 range.

 

Anybody ever setup a contribution app so users could leave a contribution
TOWARD the purchase of one of the expensive items that you guys deserve.
Then others could see that and add to it?  I could hook that up easy enough.
Keeping track of what was purchased outright on the other list might be too
confusing, but separate lists should be a breeze.

 

Just a thought.

Sean

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From msc at freeswitch.org  Mon Dec 10 20:26:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Dec 2012 09:26:27 -0800
Subject: [Freeswitch-users] sip registration
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233A3B2@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF2339A49@Mail-Kilo.squay.com>
	<CAFiqYu=cpMkTxdE99smQBM6NgitNQ5ZpePwfSYr_R23dpj7Hng@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339B39@Mail-Kilo.squay.com>
	<CAKzWOxUW21XZte0AoVDqTtC6vkJ4FmA3sbP5LgJVK5bj4+1Sbw@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF2339D42@Mail-Kilo.squay.com>
	<CAKzWOxVMWfR6vk8ODKkND1Tjxj-d5VFYn9U+dDoNyB_t2dE05w@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233A02B@Mail-Kilo.squay.com>
	<CAKzWOxVqn1tGWgHKFs+vCEDpXUg2rNyvpbw5-bBSPw846hg7Xw@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233A0D5@Mail-Kilo.squay.com>
	<CAKzWOxVhphxwn8EhfwTfo2Ti5OgTNp2Q9hKsB_Y-+4p+P9Ns4A@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233A3B2@Mail-Kilo.squay.com>
Message-ID: <CAKzWOxVao=9ibwjKTXnAS4yPeQ5HyHsEA2XPvgSZ7JZDbAQGuA@mail.gmail.com>

Well, that is a problem. If you don't know where the DB check is done then
we certainly won't know. I would consult the person(s) who configured your
system in the first place. The other alternative is to seek paid,
professional assistance. You can do so by emailing an inquiry to
consulting at freeswitch.org.

I apologize but I'm pretty sure that the community has given you all the
assistance they possibly can, at least for this specific issue. Of course,
that doesn't mean you can't ask us other questions. :)

Thanks,
MC

On Sat, Dec 8, 2012 at 2:33 AM, Archana Venugopan <a.venugopan at mundio.com>wrote:

>  Hi Michael,****
>
> ** **
>
> Thanks. I have gone through that wiki already, but in our freeswitch we
> don?t use XML to check the userid and password rather we use DB in some
> script(not sure where). Also mod_xml_curl has been disabled too.****
>
> But in none of the lua script I find that its verifying the password from
> the URL to DB too. SO am confused.****
>
> ** **
>
> Sorry I know am missing something but not sure where.****
>
> ** **
>
> Thanks again.****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 07 December 2012 21:56
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
>  ** **
>
> Archana,
>
> Did you see this page on the wiki?
>
> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Alphanumeric_to_numeric_user_mapping
>
> It's been there for years. I promise you that the wiki has more
> information than you are giving it credit for. Just remember that our wiki
> is a microcosm of the Internet: it has what you're looking for but you need
> Google to search it and you have to dig through all the results. :)
>
> Try that and see if it does what you need.
>
> As far as your question goes, there isn't a "script" that checks it.
> Rather, the module that is doing the authentication will look in the XML
> data for the user id and password. If mod_xml_curl is enable then FS will
> go fetch the XML data from your server.
>
> -MC****
>
> On Fri, Dec 7, 2012 at 1:21 AM, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
>  ****
>
> I want to give some alphabets instead of number. I want to know which
> script checks this authentication name to corresponding DB table. Please
> let me know.****
>
> Thanks****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 06 December 2012 21:58
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
>  ****
>
> Use "100" for the authentication name as well.
> -MC****
>
> On Thu, Dec 6, 2012 at 9:25 AM, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
> Thanks. I thought it will look in mod_sofia code. In the below screen I
> register the ID ?100?. Now instead of ?100? in ?Authentication Name? I need
> to give some e-mail ID or name(Archana) which should validate in DB.****
>
> I tried giving a name in ?Authentication Name? but the phone was not
> registered. Am not sure this authentication name is being looked in which
> column in table too.****
>
> Please let me know if this will be picked from any sofia code or any C
> script? Once we register in the below screen which script validates the
> Settings in freeswitch?****
>
>  ****
>
> Sorry if am repeating the same question, but I could not get the exact
> code and am clueless.****
>
> *Global SIP Settings*****
>
> Top of Form****
>
> *Basic SIP Authentication Settings*****
>
>  ****
>
> Screen Name****
>
> Screen Name 2****
>
> Phone Number****
>
> Caller ID****
>
> Authentication Name****
>
> Password****
>
> Bottom of Form****
>
>  ****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 05 December 2012 20:34
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
>  ****
>
> When an event that requires a user lookup takes place then the system will
> look in the XML user directory unless it has been configured to look
> somewhere else. The other places to look are usually:
> mod_xml_curl
> One of the language like Lua, Perl, Python
>
> If it's xml_curl then FS will do a POST to your web server in hopes of
> receiving back the necessary XML for the given user. It would be up to you
> to have your web server handle the request, poll the database, then format
> and return the XML data. See this wiki page<http://wiki.freeswitch.org/wiki/Xml_curl#How_do_I_configure_mod_xml_curl.3F>for more info on xml curl.
>
> If it's a language then you'll have a "binding" in the conf file for the
> language that will handle the lookup. Again, your script will need to
> handle the communication with your database. See this wiki page<http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration>for more information.
>
> Hope this helps.
> -MC****
>
> On Wed, Dec 5, 2012 at 9:30 AM, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
> Thanks for the information. But sorry, how to access user_data API command.
> ****
>
>  ****
>
> Am not clear on the flow. Once we register domain and usernumber in sip
> what exactly happens? Which script picks up this domain and username and
> validates with our database?****
>
> Could you please provide me with an overview. ****
>
>  ****
>
> Many thanks****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 05 December 2012 17:13
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
>  ****
>
> If you're talking about the user configuration then yes, you could create
> an "email" parameter or variable and access it with the user_data API
> command.
> -MC****
>
> On Wed, Dec 5, 2012 at 1:35 AM, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
> In that case can I have 1 more column say e-mail and can this e-mail be
> checked in DB instead of checking reg_user(?100?)? Is that feasible?****
>
> Also which code should be changed any idea please?****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 04 December 2012 19:51
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
>  ****
>
> You can have a user 'ana' in the domain 'gmail.com'. Though using someone
> else's domain as local in your FS setup may not be a good idea.****
>
> You can't have a @ in the username itself (per the SIP standard, not
> limited to FreeSWITCH).****
>
>  ****
>
> On 4 December 2012 18:00, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
>  ****
>
> Currently we register authentication name as say ?100? in sip
> registration, this comes to freeswitch and it will check in our DB for 100
> and if its present then registrations would be successful. ****
>
>  ****
>
> freeswitch at internal> show registrations****
>
>
> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
> ****
>
> 100,fsfailover.uk01.com
> ,e4969067f9a8c098,sofia/internal/sip:100 at 192.168.2.234:5060
> ;transport=udp,1354638871,192.168.2.234,5060,udp,squay-laptop-1.squay.com*
> ***
>
>  ****
>
> I want to change this 100 to some e-mail address, so instead of 100 it
> will be something like ?ana at gmail.com?. Can we do this? While coming to
> freeswitch whether there would be any issues?****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
> ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From peter.olsson at visionutveckling.se  Mon Dec 10 20:30:47 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 10 Dec 2012 17:30:47 +0000
Subject: [Freeswitch-users] git IPv6 down?
Message-ID: <1FFF97C269757C458224B7C895F35F151E3575@cantor.std.visionutv.se>

Hi all,

Ever since the DoS attack a couple of weeks ago, I can't access git over IPv6 anymore. Are there still issues that need to be resolved, or is it just me?

It fallbacks to IPv4, so it works just fine, I just wanted to let you know.

/Peter


From anthony.minessale at gmail.com  Mon Dec 10 20:33:16 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 Dec 2012 11:33:16 -0600
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
Message-ID: <CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>

The SERVICE_NOT_IMPLEMENTED is the q.850 translation of the 406 you
received over SIP.
If you turn on the siptrace with "sofia global siptrace on" you may get
more details.

Also leg_delay_start does not do much good in enterprise originate as each
url in the list separated by :_: is an entire originate string so you are
effectively doing 5 concurrent calls to originate and only supplying one
leg.  I don't believe that feature works with only one leg supplied.



On Mon, Dec 10, 2012 at 10:57 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> HI,****
>
> ** **
>
> I have a bridge statement used dozens and dozens of times a day:****
>
> EXECUTE sofia/external_noauth/anonymous at 64.136.174.30 bridge(user/
> 300 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> 301 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> 203 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> 302 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> 303 at fs_mbri2.bizfocused.com)****
>
> ** **
>
> Very occasionally, recently extensions (at least 300 and 302) have
> returned [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]****
>
> ** **
>
> All of these extensions are at the same remote location using the same wan
> links/routers, etc.****
>
> ** **
>
> What does that mean?  ****
>
> ** **
>
> Pastebin: http://pastebin.freeswitch.org/pastebin.php?dl=20306****
>
> ** **
>
> Also, the 8 second delay appears to be ignored (I think, I have never
> asked the customer or been on site to test).  Any 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
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From msc at freeswitch.org  Mon Dec 10 20:40:13 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Dec 2012 09:40:13 -0800
Subject: [Freeswitch-users] IVR menu delay - 1 second between events
In-Reply-To: <CAHKQagF=jW5ry3=6d8e3i0rwXMZUsRo5g5v_VHLQapDscQ7obg@mail.gmail.com>
References: <CAHKQagF3C76QiybFNrS7GCdiGqaTbtPpQzD4wFivtJ+EK1CirQ@mail.gmail.com>
	<CAPNTnnrVW4Vpj4LcaNyeQrcmYq99DDF26E3UXpFoJyf=-q11fg@mail.gmail.com>
	<CAHKQagG1dW+MUtLYgoJHN0mWa0xowKGo4-VFR4wV2nZqcMKWiw@mail.gmail.com>
	<CAHKQagH-5HwgRpP77bVSLkOusVzMH6bU68Ox=jR7fvS7X-hkFg@mail.gmail.com>
	<CAHKQagFQBeB+ho+rrniqZiqS1aocWy6BNN16Lv_2xm-mFDr+Nw@mail.gmail.com>
	<CAHKQagF=jW5ry3=6d8e3i0rwXMZUsRo5g5v_VHLQapDscQ7obg@mail.gmail.com>
Message-ID: <CAKzWOxX15YgszgD_siyHsMHRz5Bj80dsFLYegLpuZvObf1QHAw@mail.gmail.com>

Another satisfied customer... and proof that opening a Jira not only makes
Tony's life easier, it actually WORKS! Thanks for opening the Jira and
thanks for reporting that your issue has been resolved.
-MC

On Sun, Dec 9, 2012 at 2:57 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> This has now been patched in the core (sleep has been completely removed),
> see JIRA ticket for details.
>
> Thanks Ant!
>
> Cal
>
>
> On Sat, Dec 8, 2012 at 9:09 PM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> I've raised a JIRA ticket for this;
>> http://jira.freeswitch.org/browse/FS-4924
>>
>> Cal
>>
>> On Fri, Dec 7, 2012 at 8:08 PM, Cal Leeming [Simplicity Media Ltd] <
>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>
>>> Just trawled through the source and found this;
>>>
>>> src/switch_ivr_menu.c:
>>> if (status == SWITCH_STATUS_SUCCESS) {
>>>  status = switch_ivr_sleep(session, 1000, SWITCH_FALSE, NULL);
>>> }
>>>
>>> So, I changed 1000 to 10, the delay has now disappeared and it seems to
>>> work perfectly.
>>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From vbvbrj at gmail.com  Mon Dec 10 20:46:18 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Mon, 10 Dec 2012 19:46:18 +0200
Subject: [Freeswitch-users] Natting question.
In-Reply-To: <CACuV5sAJHmkLGtYZ0w+B127Lfz8Gi5YgK-yku5H6VoHiFK9DuQ@mail.gmail.com>
References: <50C5EAAA.6040100@gmail.com>
	<CACuV5sAJHmkLGtYZ0w+B127Lfz8Gi5YgK-yku5H6VoHiFK9DuQ@mail.gmail.com>
Message-ID: <50C61FEA.1000109@gmail.com>

On 10.12.2012 16:58, Zenny wrote:
> It seems like it is associated with either the portforwarding of the
> wifi router or NAT prolem, I guess. The following link could be of
> help, imho:
> http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT
>

I've read all pages on wiki related to nat and still have trouble 
finding the cause.

-- 
Mimiko desu.


From steveayre at gmail.com  Mon Dec 10 20:49:22 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 10 Dec 2012 17:49:22 +0000
Subject: [Freeswitch-users] 401
In-Reply-To: <CADK-Bjtuu+dEXQRxOxDzkHSdwO-7JuLg=ciuCUD6S4iUq5hE_w@mail.gmail.com>
References: <CADK-Bjtuu+dEXQRxOxDzkHSdwO-7JuLg=ciuCUD6S4iUq5hE_w@mail.gmail.com>
Message-ID: <CAFiqYum-4HCc=cddz1DHi+z=tozwK0nU3=M1jQekZjixwKrqwQ@mail.gmail.com>

Please provide more information. Is your authentication failing?

401 Unauthorized is 100% normal behaviour for a SIP call to authenticate.

The call flow is:
> REGISTER
< 401 Unauthorized (with WWW-Authenticate header)
> REGISTER (with Authorization header)
< 200 OK
Ditto for INVITE.

The 401 is required to give the WWW-Authenticate challenge which is
required to generate the Authorization header. Thus for a authenticated
call the 401 'error' is normal and expected.

If authentication is failing after the upgrade or you were not requiring
authentication before the upgrade, perhaps during the upgrade your
configuration files were modified or the location has moved?

-Steve



On 10 December 2012 15:39, Blake Priddy <bpriddy at bryantschools.org> wrote:

> So I updated to the latest git head and now several phones NOT ALL,
> including mine :( are giving me
>
> 2012-12-10 09:36:04.584347 [DEBUG] sofia_reg.c:1498 Send challenge for [
> 1206 at pbx.bryantschools.org]
> 2012-12-10 09:36:04.584347 [WARNING] sofia_reg.c:1502 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [1206 at pbx.bryantschools.org]
> from ip 10.10.24.20
> send 679 bytes to udp/[10.10.24.20]:3072 at 15:36:04.585091:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP 10.10.24.20:3072
> ;branch=z9hG4bK-n757mrbdlqdw;rport=3072
>    From: "Blake Priddy" <sip:1206 at pbx.bryantschools.org>;tag=dbou3jy9kf
>    To: "Blake Priddy" <sip:1206 at pbx.bryantschools.org>;tag=v2Kpa56439DgK
>    Call-ID: 4e000000e08c-cm5eg6nr96al
>    CSeq: 7373 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.3.9b+git~20121206T182231Z~a7fafb2039
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    WWW-Authenticate: Digest realm="pbx.bryantschools.org",
> nonce="4f0b07b6-42df-11e2-ab4a-4bb963400da6", algorithm=MD5, qop="auth"
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 884 bytes from udp/[10.10.24.20]:3072 at 15:36:04.680840:
>    ------------------------------------------------------------------------
>    REGISTER sip:pbx.bryantschools.org SIP/2.0
>    Via: SIP/2.0/UDP 10.10.24.20:3072;branch=z9hG4bK-cpukgis6pexk;rport
>    From: "Blake Priddy" <sip:1206 at pbx.bryantschools.org>;tag=dbou3jy9kf
>    To: "Blake Priddy" <sip:1206 at pbx.bryantschools.org>
>    Call-ID: 4e000000e08c-cm5eg6nr96al
>    CSeq: 7374 REGISTER
>     Max-Forwards: 70
>    Contact: <sip:1206 at 10.10.24.20:3072
> >;reg-id=1;q=1.0;+sip.instance="<urn:uuid:cb7105d7-ce2c-4835-8300-000413411CA5>"
>    User-Agent: snom870/8.7.3.10
>    Allow-Events: dialog
>    X-Real-IP: 10.10.24.20
>    Supported: path, gruu
>    WWW-Contact: <http://10.10.24.20:80>
>    WWW-Contact: <https://10.10.24.20:443>
>    Authorization: Digest username="1206",realm="pbx.bryantschools.org
> ",nonce="4f0b07b6-42df-11e2-ab4a-4bb963400da6",uri="sip:
> pbx.bryantschools.org
> ",qop=auth,nc=00000001,cnonce="4afc06b9",response="8c9a1fcedd906467bd0afa2b98b50829",algorithm=MD5
>    Expires: 3600
>    Content-Length: 0
>
>
> So........ Yea ha
>
> --
>
> *Blakelund Priddy*
>  Network Systems Engineer
> Bryant Public School District
> Bryant, Arkansas 72022
> http://www.bryantschools.org
> p 501-653-5038
> f 501-847-5656
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 steveayre at gmail.com  Mon Dec 10 21:00:28 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 10 Dec 2012 18:00:28 +0000
Subject: [Freeswitch-users] Natting question.
In-Reply-To: <50C5EAAA.6040100@gmail.com>
References: <50C5EAAA.6040100@gmail.com>
Message-ID: <CAFiqYumipKg+-D0-c3NqXrUdJ0BMioZyDuueq67gf9213C8_Tg@mail.gmail.com>

So to clarify:

Client -> Internet -> NAT -> FS
works

Client -> NAT -> Internet -> NAT -> FS
does not?

Since the FS NAT side appears to be working (from the 1st case), I'm
guessing it's probably either a problem at the client side (with client or
NAT router) or perhaps the double-NAT is causing an issue. Personally I
haven't run FS behind NAT, so it's not a situation I've had to handle.

Do a packet trace of a call at both the client and FS ends. The FS end is
easy - "sofia global siptrace on if you have low volumes, or a tool like
tcpdump/ngrep if you have higher volumes. FS will actually log the SDP
information in at debug-level console logs and in XML CDRs. At the client
side try using Wireshark on the client PC.

Compare the SDP information of the call, to check the IP and ports seen at
either end. Check whether they look reasonable. FS will need to see the
public IP of the client, and the client see the public IP of FS for audio
to work in both directions. The FS/client may not need to know their public
IP on the internal side of the router as many routers have SIP ALG that
will rewrite the internal IP to an external one, but if the client has
already done so that often causes more problems than it solves.

FS can handle only knowing the internal IP of the client if they're able to
send RTP to FS - it will see the IP/port the audio is coming from and send
audio back to that address. That does mean audio won't work on such a call
until it has received RTP from the client though.

-Steve







On 10 December 2012 13:59, Mimiko <vbvbrj at gmail.com> wrote:

> Hello.
>
> I have a FS behind firewall on which ports 5080,5060 udp are mapped to
> the FS server. A client connected directly to Internet, say using GPRS,
> authorise to FS and calling a number audio is heared. When this client
> is connected via WiFi to some router connected directly to Internet,
> client authorises, calls but there is no audio.
>
> So I'm stump on where to look for issue: on FS server side with firewall
> or on client side behind router?
>
> Thank you.
> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 vbvbrj at gmail.com  Mon Dec 10 21:22:12 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Mon, 10 Dec 2012 20:22:12 +0200
Subject: [Freeswitch-users] Natting question.
In-Reply-To: <CAFiqYumipKg+-D0-c3NqXrUdJ0BMioZyDuueq67gf9213C8_Tg@mail.gmail.com>
References: <50C5EAAA.6040100@gmail.com>
	<CAFiqYumipKg+-D0-c3NqXrUdJ0BMioZyDuueq67gf9213C8_Tg@mail.gmail.com>
Message-ID: <50C62854.60102@gmail.com>

On 10.12.2012 20:00, Steven Ayre wrote:
> So to clarify:
>
> Client -> Internet -> NAT -> FS
> works
>
> Client -> NAT -> Internet -> NAT -> FS
> does not?
>
> Since the FS NAT side appears to be working (from the 1st case), I'm
> guessing it's probably either a problem at the client side (with client
> or NAT router) or perhaps the double-NAT is causing an issue. Personally
> I haven't run FS behind NAT, so it's not a situation I've had to handle.

Thank you for a comprehensive response.

What I've seen now. Audio actually comes up, but it is taking a while. 
Sometimes is ~5sec, sometimes more, sometimes never. Also, sip commands 
does not take action. The other party have to hung up, for example, in 
order for call to end.

-- 
Mimiko desu.


From msc at freeswitch.org  Mon Dec 10 21:42:58 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Dec 2012 10:42:58 -0800
Subject: [Freeswitch-users] 401
In-Reply-To: <CAFiqYum-4HCc=cddz1DHi+z=tozwK0nU3=M1jQekZjixwKrqwQ@mail.gmail.com>
References: <CADK-Bjtuu+dEXQRxOxDzkHSdwO-7JuLg=ciuCUD6S4iUq5hE_w@mail.gmail.com>
	<CAFiqYum-4HCc=cddz1DHi+z=tozwK0nU3=M1jQekZjixwKrqwQ@mail.gmail.com>
Message-ID: <CAKzWOxWSzXWwncfJWf_F+Ehpy+jVptEyxbFYbyc-KU1Qm6u1Fw@mail.gmail.com>

FYI,

Darren Schreiber did a nice presentation on SIP 101 that covers this. Look
at June 20, 2012 here:
http://wiki.freeswitch.org/wiki/Weekly_Conference_Call

There is a PDF to go along with the audio recording. I highly recommend any
SIP n00bs check it out.

On Mon, Dec 10, 2012 at 9:49 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Please provide more information. Is your authentication failing?
>
> 401 Unauthorized is 100% normal behaviour for a SIP call to authenticate.
>
> The call flow is:
> > REGISTER
> < 401 Unauthorized (with WWW-Authenticate header)
> > REGISTER (with Authorization header)
> < 200 OK
> Ditto for INVITE.
>
> The 401 is required to give the WWW-Authenticate challenge which is
> required to generate the Authorization header. Thus for a authenticated
> call the 401 'error' is normal and expected.
>
> If authentication is failing after the upgrade or you were not requiring
> authentication before the upgrade, perhaps during the upgrade your
> configuration files were modified or the location has moved?
>
> -Steve
>
>
>
> On 10 December 2012 15:39, Blake Priddy <bpriddy at bryantschools.org> wrote:
>
>> So I updated to the latest git head and now several phones NOT ALL,
>> including mine :( are giving me
>>
>> 2012-12-10 09:36:04.584347 [DEBUG] sofia_reg.c:1498 Send challenge for [
>> 1206 at pbx.bryantschools.org]
>> 2012-12-10 09:36:04.584347 [WARNING] sofia_reg.c:1502 SIP auth challenge
>> (REGISTER) on sofia profile 'internal' for [1206 at pbx.bryantschools.org]
>> from ip 10.10.24.20
>> send 679 bytes to udp/[10.10.24.20]:3072 at 15:36:04.585091:
>>
>>  ------------------------------------------------------------------------
>>    SIP/2.0 401 Unauthorized
>>    Via: SIP/2.0/UDP 10.10.24.20:3072
>> ;branch=z9hG4bK-n757mrbdlqdw;rport=3072
>>    From: "Blake Priddy" <sip:1206 at pbx.bryantschools.org>;tag=dbou3jy9kf
>>    To: "Blake Priddy" <sip:1206 at pbx.bryantschools.org>;tag=v2Kpa56439DgK
>>    Call-ID: 4e000000e08c-cm5eg6nr96al
>>    CSeq: 7373 REGISTER
>>    User-Agent: FreeSWITCH-mod_sofia/1.3.9b+git~20121206T182231Z~a7fafb2039
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, precondition, path, replaces
>>    WWW-Authenticate: Digest realm="pbx.bryantschools.org",
>> nonce="4f0b07b6-42df-11e2-ab4a-4bb963400da6", algorithm=MD5, qop="auth"
>>    Content-Length: 0
>>
>>
>>  ------------------------------------------------------------------------
>> recv 884 bytes from udp/[10.10.24.20]:3072 at 15:36:04.680840:
>>
>>  ------------------------------------------------------------------------
>>    REGISTER sip:pbx.bryantschools.org SIP/2.0
>>    Via: SIP/2.0/UDP 10.10.24.20:3072;branch=z9hG4bK-cpukgis6pexk;rport
>>    From: "Blake Priddy" <sip:1206 at pbx.bryantschools.org>;tag=dbou3jy9kf
>>    To: "Blake Priddy" <sip:1206 at pbx.bryantschools.org>
>>    Call-ID: 4e000000e08c-cm5eg6nr96al
>>    CSeq: 7374 REGISTER
>>     Max-Forwards: 70
>>    Contact: <sip:1206 at 10.10.24.20:3072
>> >;reg-id=1;q=1.0;+sip.instance="<urn:uuid:cb7105d7-ce2c-4835-8300-000413411CA5>"
>>    User-Agent: snom870/8.7.3.10
>>    Allow-Events: dialog
>>    X-Real-IP: 10.10.24.20
>>    Supported: path, gruu
>>    WWW-Contact: <http://10.10.24.20:80>
>>    WWW-Contact: <https://10.10.24.20:443>
>>    Authorization: Digest username="1206",realm="pbx.bryantschools.org
>> ",nonce="4f0b07b6-42df-11e2-ab4a-4bb963400da6",uri="sip:
>> pbx.bryantschools.org
>> ",qop=auth,nc=00000001,cnonce="4afc06b9",response="8c9a1fcedd906467bd0afa2b98b50829",algorithm=MD5
>>    Expires: 3600
>>    Content-Length: 0
>>
>>
>> So........ Yea ha
>>
>> --
>>
>> *Blakelund Priddy*
>>  Network Systems Engineer
>> Bryant Public School District
>> Bryant, Arkansas 72022
>> http://www.bryantschools.org
>> p 501-653-5038
>> f 501-847-5656
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Mon Dec 10 21:47:29 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Dec 2012 10:47:29 -0800
Subject: [Freeswitch-users] Improve generated tones quality
In-Reply-To: <65F91A2E-F875-4070-8910-D61DC5349BD1@kavun.ch>
References: <65F91A2E-F875-4070-8910-D61DC5349BD1@kavun.ch>
Message-ID: <CAKzWOxU+xEnr66sJaWNXEb0KQ19u1wK7jsbsZ6WhP7UEi6--9Q@mail.gmail.com>

Please file this as a feature request in Jira.
-MC

On Mon, Dec 10, 2012 at 5:29 AM, Emrah <lists at kavun.ch> wrote:

> Hi list,
>
> I am experiencing a recurring problem with generated tones across
> different machines and FS versions.
>
> The start or the end of the tone is often distorted. For some interesting
> reasons, this does not happen with all equipments and not all codecs.
> Typical scenario in PCMU/A:
> If someone calls me on an inbound DID, I pick up the call and than
> transfer the call to an internal extension, the original caller will hear
> MOH followed by a ringback that starts with a sort of "fade in".
> Something similar happens with the voicemail where the end of the beep
> tone is faded out.
>
> Even though this hasn't much to do with Early Media, I tried playing with
> bridge_early_media to no avail. It added a 5 seconds latency at the start
> of my call and did not solve my issue.
>
> Your take on this would be greatly appreciated. Is there a way to fine
> tune PCMU / PCMA for better performance? Can we initiate a little buffering
> on system generated tones?
>
> Cheers,
> Emrah
> Ps: It sounds weird, but I believe this also happens if the played tone is
> an actual sound file.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From anthony.minessale at gmail.com  Mon Dec 10 22:16:08 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 Dec 2012 13:16:08 -0600
Subject: [Freeswitch-users] easy way to crash freeswitch
In-Reply-To: <CAMbW9o+E9Zoi9c_KgD1UFMpo76f34rdtpAR6-fzhccYEG-NayQ@mail.gmail.com>
References: <026150BF-2E8A-44F2-ACF6-99E6A0186044@endigotech.com>
	<CAHKQagGcO62Ku+p3k9eh7m=EqB4JNShGd3cr9Dna=ai+vjrKLA@mail.gmail.com>
	<CB1CE14E-D1B7-4889-A964-47A38B24A1FA@endigotech.com>
	<CAHKQagHD5wncPKJtzRwS=YA7V=DfB5MvzWvhoinFSCSZksQJvg@mail.gmail.com>
	<CAKV-jFy5skhPsL1r1WVf2WaU+Roq8TPhPqkrsTy_wizDVRYEXw@mail.gmail.com>
	<Pine.LNX.4.64.1212090907450.32638@1a-8140.antagus.de>
	<CAMwZ-tKu1QJwGqpZRNM_CHxZaRr8sjWbgY_V0rEsON125AbRdQ@mail.gmail.com>
	<Pine.LNX.4.64.1212091706130.2416@1a-8140.antagus.de>
	<CAMwZ-tJHBKfHsrHGwp_45Zf8jy5HA-vhSg9Pj0qCMMHtnDPUmA@mail.gmail.com>
	<CAMbW9o+E9Zoi9c_KgD1UFMpo76f34rdtpAR6-fzhccYEG-NayQ@mail.gmail.com>
Message-ID: <CAKbxfG_q7Pd4VoUs7qL85f8BS1gh5Tegy0Vou0Cxohjm21Xd6A@mail.gmail.com>

This was a recursion error caused by calling stream file from an input
callback thus streaming a file in the middle of the top level stream file
who in turn collected input and did the same so on and so forth.  So every
time you type 1 you added a bunch of calls to the stack recursively until
you used up all the stack memory and it aborted.

There is a patch in tree now that only allows you to nest 25 times before
it gets mad but really you should avoid doing this at all if you can help
it.



On Mon, Dec 10, 2012 at 2:54 AM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> That'll teach me to read the whole thread wont it!
>
> My wife is actually a rehab worker for people with visual impairments, she
> works for a charity that helps people in our local area in the UK.
>
>
> On 10 December 2012 06:41, Gabriel Gunderson <gabe at gundy.org> wrote:
>
>> On Sun, Dec 9, 2012 at 9:14 AM, Thomas Hoellriegel <admin at blindi.net>
>> wrote:
>> > I can post bugs descriptions only in this list, to hope somebody can
>> help
>> > me. or to reproduce a bug.
>>
>> Heh, I didn't know. :) I will help file the bug for you. Give me a few
>> days for work to slow down. I'll follow the steps you outline and make
>> sure I can reproduce it. When I can, I'll file and follow up on the
>> Jira.
>>
>> Thanks for bringing it to our attention and being a good sport about a
>> gentle ribbing.
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
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From msc at freeswitch.org  Mon Dec 10 22:33:28 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Dec 2012 11:33:28 -0800
Subject: [Freeswitch-users] Wish Lists - 'tis the season
In-Reply-To: <31fb01cdd6f8$fa356850$eea038f0$@bizfocused.com>
References: <31fb01cdd6f8$fa356850$eea038f0$@bizfocused.com>
Message-ID: <CAKzWOxV3yi3_tXBtZK8UCMv98UUSFasztJBjOqSGhVz9vPhu5w@mail.gmail.com>

Many thanks, sir!

We'll look into the possibility of doing a donation thing. The other thing
we can do is maybe have each person put a $25 Amazon gift card on their
wishlist and then people can buy as many or as few as they wish.

Thanks!
-MC

On Mon, Dec 10, 2012 at 9:08 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Hey Guys,****
>
> ** **
>
> Update those wish lists.  Anthony?s starts at like $495!  I want to show
> my love, but I am a poor VOIP provider and not spending $495 on my wife
> this season!  I hooked up BF and MC but they could use a few more
> suggestions in the $25 to $50 range.****
>
> ** **
>
> Anybody ever setup a contribution app so users could leave a contribution
> TOWARD the purchase of one of the expensive items that you guys deserve.
> Then others could see that and add to it?  I could hook that up easy
> enough.  Keeping track of what was purchased outright on the other list
> might be too confusing, but separate lists should be a breeze.****
>
> ** **
>
> Just a thought.****
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From sdevoy at bizfocused.com  Mon Dec 10 23:00:00 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 10 Dec 2012 15:00:00 -0500
Subject: [Freeswitch-users] Natting question.
In-Reply-To: <50C5EAAA.6040100@gmail.com>
References: <50C5EAAA.6040100@gmail.com>
Message-ID: <33fa01cdd710$f0168a30$d0439e90$@bizfocused.com>

Hi Mimiko,

I have had hours and hours of problems with NAT, so I can at least tell you
what helped me get past them.

All of my problems have been in the configuration     PHONE > NAT > Internet
> FS  .

CISCO SPECIFIC ISSUES:
I learned through a long tough week with a new customer that CISCO 3xx and
5xx Series have a web setup section that is critical when behind NAT.  When
logged in as an ADMIN, on the SIP Tab, these NAT Support Parameters should
be set to yes:
Handle VIA received,  Insert VIA received,  Substitute VIA Addr,  Handle VIA
rport, Insert VIA rport, Send Resp To Src Port
With those all set I have had pretty good luck with CISCO.

GENERIC TO ALL PHONES:
Brian and Michael have just worked me through a similar Registration issue
with Polycom phones, but I am sure this applies to many others too.  The key
to understanding that it was a NAT issue was that the phones CSeq (in the
Sophia global sipstrace on) did not change from one attempt to the next.  It
was always "CSeq: 1 REGISTER" and the second one would be "2 REGISTER" with
the auth stuff attached.  That shows it did not receive the response.  In
this case I added a parameter and a variable to the USER section of the
"directory" on the server.
          <variable name="sip-force-contact"
value="NDLB-connectile-dysfunction"/>
         <param name="NDLB-force-rport" value="true"/>

I am not sure if that fits your case, but maybe it will help others in the
future.

Sean

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mimiko
Sent: Monday, December 10, 2012 8:59 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Natting question.

Hello.

I have a FS behind firewall on which ports 5080,5060 udp are mapped to the
FS server. A client connected directly to Internet, say using GPRS,
authorise to FS and calling a number audio is heared. When this client is
connected via WiFi to some router connected directly to Internet, client
authorises, calls but there is no audio.

So I'm stump on where to look for issue: on FS server side with firewall or
on client side behind router?

Thank you.
--
Mimiko desu.

_________________________________________________________________________
Professional FreeSWITCH Consulting 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




From moises.silva at gmail.com  Mon Dec 10 23:00:14 2012
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 10 Dec 2012 15:00:14 -0500
Subject: [Freeswitch-users] FreeSWITCH+Sangoma with dialup modems
In-Reply-To: <965759A53E43FE439E43565A7715E5F058F1BE0EE8@oa-exchange1.oa.com.au>
References: <965759A53E43FE439E43565A7715E5F058F1BE0EE8@oa-exchange1.oa.com.au>
Message-ID: <CAA4nhyD-uMSecPn=4P24=VeNY4h8Mt=sd_BA7T5JSPoA4TfYgQ@mail.gmail.com>

On Mon, Dec 10, 2012 at 2:40 AM, Sirish Masur Mohan <
Sirish.MasurMohan at oa.com.au> wrote:

> My setup is as follows:
>
> **
>
> Serial port based device(Caller)->Dlink dialup
> modem(DFM562E)->PBX->Sangoma A102 E1 Link(and FreeSWITCH)->PBX->Dlink
> dialup modem(DFM562E)->Server(Receiver)****
>
> ** **
>
> Problem:****
>
> When the caller makes the call (using the modem with a lower baud rate
> 1200), Receiver modem receives the RING and answers the call (using the ATA
> command). But for some reason, both the ends timeout waiting for modem
> CONNECT.
>

Are you sure there is no echo cancellers in the middle (make sure you have
an A102 without echo canceller or that you disable it in wanpipe). You can
always make sure you disable the ec in the dialplan too (<action
application="disable_ec">)


*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 marketing at cluecon.com  Mon Dec 10 23:29:01 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 10 Dec 2012 12:29:01 -0800
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxWG_5d5_jBVB-2H-VShLOmU8UB=BWqkhW4h4yywsAfKZg@mail.gmail.com>

Greetings!

We are glad to report that the FreeSWITCH team has tagged version 1.2.5.3.
You can download the tarball
here<http://files.freeswitch.org/freeswitch-1.2.5.3.tar.bz2>.
Anyone using 1.2.5.x should update as soon as possible. We appreciate all
those who have helped us with testing and tracking down some sneaky and
pernicious little bugs.

On last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_05>we spent
some time talking about the XML parser and some of its
pre-processor directives. We discussed specifically how you can use the
"exec" command to execute a shell script in the middle of XML processing.
We also discussed a few tricks on how to look at the source code when you
need to learn about some FreeSWITCH functionality that otherwise is not
documented. This week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_12>subject is
still pending, so stay tuned!

One other item I'd like to mention is that we've had several reports of
FreeSWITCH success stories. We will be providing more information about
those in upcoming stories on our Web site. We've got people using
FreeSWITCH in various situations as well as software developers who've
added support for FreeSWITCH to their offerings. The FreeSWITCH ecosystem
continues to grow and flourish! Thank you all for being a part of it.

Take care and have a great week!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From bpriddy at bryantschools.org  Mon Dec 10 23:35:04 2012
From: bpriddy at bryantschools.org (Blake Priddy)
Date: Mon, 10 Dec 2012 14:35:04 -0600
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKzWOxWG_5d5_jBVB-2H-VShLOmU8UB=BWqkhW4h4yywsAfKZg@mail.gmail.com>
References: <CAKzWOxWG_5d5_jBVB-2H-VShLOmU8UB=BWqkhW4h4yywsAfKZg@mail.gmail.com>
Message-ID: <CADK-Bjv7sCR0KZAyhXYmQ_Dywn2QkODY0QYKqo2G8p8GTbG8vw@mail.gmail.com>

FREESWITCH FTW!!!! :)


On Mon, Dec 10, 2012 at 2:29 PM, Michael Collins <marketing at cluecon.com>wrote:

> Greetings!
>
> We are glad to report that the FreeSWITCH team has tagged version 1.2.5.3.
> You can download the tarball here<http://files.freeswitch.org/freeswitch-1.2.5.3.tar.bz2>.
> Anyone using 1.2.5.x should update as soon as possible. We appreciate all
> those who have helped us with testing and tracking down some sneaky and
> pernicious little bugs.
>
> On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_05>we spent some time talking about the XML parser and some of its
> pre-processor directives. We discussed specifically how you can use the
> "exec" command to execute a shell script in the middle of XML processing.
> We also discussed a few tricks on how to look at the source code when you
> need to learn about some FreeSWITCH functionality that otherwise is not
> documented. This week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_12>subject is still pending, so stay tuned!
>
> One other item I'd like to mention is that we've had several reports of
> FreeSWITCH success stories. We will be providing more information about
> those in upcoming stories on our Web site. We've got people using
> FreeSWITCH in various situations as well as software developers who've
> added support for FreeSWITCH to their offerings. The FreeSWITCH ecosystem
> continues to grow and flourish! Thank you all for being a part of it.
>
> Take care and have a great week!
>
> --
> Michael S Collins
> ClueCon Team
> http://www.cluecon.com
> 877-7-4ACLUE
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 

*Blakelund Priddy*
Network Systems Engineer
Bryant Public School District
Bryant, Arkansas 72022
http://www.bryantschools.org
p 501-653-5038
f 501-847-5656
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From marcdecorny at gmail.com  Mon Dec 10 23:42:25 2012
From: marcdecorny at gmail.com (Marc de Corny)
Date: Mon, 10 Dec 2012 20:42:25 +0000
Subject: [Freeswitch-users] Cannot Kill stale channels
In-Reply-To: <4424748D-98BF-4BDA-A616-F6809467C3D0@visionutveckling.se>
References: <CA+myZ6dBpJoZWfrA-Wc+jiAxtTP-sd7NvgDFFaxK2d8=2eTAdg@mail.gmail.com>
	<4424748D-98BF-4BDA-A616-F6809467C3D0@visionutveckling.se>
Message-ID: <CA+myZ6f2bfAZm5K2t7Cjki1nrMsdw3ayM1OkWEhg1o4Y0tM3Wg@mail.gmail.com>

Thanks for your response Peter, that sounds very likely.

My diaplan sends everything into the queue and then my background lua
script empties the queue every 10 seconds and tries to connect the call.
Ideally my lua scripts "forgets" about the connected call so that it does
not get stuck and can take another call out of the queue 10 seconds later
independantly of the previous call being hung up.


Am i going about this the wrong way? the only reason I do this is that the
mod_fifo as it is does not give me enough flexibility.

My script basically calls out from the FS and if connected successfully (
new_session), perform a
new_session:execute("transfer", "agent_to_queue_paymentsense_queue_service
XML default");
and connect via the dialplan with :<action application="fifo" data="queue77
out nowait"/>

if I use a bridge instead of the transfer, the scripts sleeps until the
call is hung up, with transfer, it can go 10 seconds later and take another
call out of the queue.

so is there a command I need to add to the dialplan like
hangup_after_bridge on the outbound call?
<action application="set" data="hangup_after_bridge=true"/>

any ideas?

thanks
marc



On Mon, Dec 10, 2012 at 4:40 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Are you exiting the lua script? Usually when this happens it means you
> have not
> released all references to the call session object.
>
> /Peter
>
> 10 dec 2012 kl. 16:28 skrev "Marc de Corny" <marcdecorny at gmail.com<mailto:
> marcdecorny at gmail.com>>:
>
> Hi
>
> I have calls that come into a queue and then get pulled out by a lua
> script in background and transfered to a destination. all works fine except
> that it looks like the sessions a not clearing from FS when all the parties
> clear the calls.
>
> When I do show channels, I get this for example:
> uuid    direction       created created_epoch   name    state   cid_name
>      cid_num ip_addr dest    presence_id     presence_data   callstate
>   callee_name     callee_num      callee_direction        call_uuid
> hostname        sent_callee_name        sent_callee_num
> 5e8215e7-b036-4cce-9b96-e57db92f13ee    outbound        03/12/2012 09:32
>      1.35E+09        sofia/external/02031950164      CS_HANGUP
> Outbound Call   2031950164      135.196.144.32
>  agent_to_queue_paymentsense_queue_service                       ACTIVE
>              SEND    5e8215e7-b036-4cce-9b96-e57db92f13ee    freeswitch2
>   4.4209E+11      2089623100
>
> But when I try to get kill or query it, I get a message that the call is
> not anywhere in the FS.
>
> /usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_kill
> 5e8215e7-b036-4cce-9b96-e57db92f13ee"
> -ERR No Such Channel!
>
> /usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_exists
> 5e8215e7-b036-4cce-9b96-e57db92f13ee"
> false
>
> Does anyone have an idea how I can kill that call. If I restart the FS,
> they clear, but the problem is that they are hitting my limit on number of
> simultaneous calls.
>
> Any help much appreciated.
>
> Thansk
> marc
>
>
>
>
>
>
> !DSPAM:50c5fb8c32767955115405!
> _________________________________________________________________________
> 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://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org
>
>
> !DSPAM:50c5fb8c32767955115405!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 miha at softnet.si  Mon Dec 10 23:49:18 2012
From: miha at softnet.si (Miha)
Date: Mon, 10 Dec 2012 21:49:18 +0100
Subject: [Freeswitch-users] media problem afet 302
In-Reply-To: <50C5EDF0.5070805@softnet.si>
References: <50C5A99D.1030305@softnet.si>
	<CALNZuwmM4Tphy+B1MvKwmBD_csszGrBF_FknJQ-H7pmoDmykyA@mail.gmail.com>
	<50C5EDF0.5070805@softnet.si>
Message-ID: <web-88303730@softnet.si>

Hi,

can some pls help me:)

Is this maybe a bug and I should post to jira?

BR,
Miha

On Mon, 10 Dec 2012 15:13:04 +0100
 Miha <miha at softnet.si> wrote:
> HI,
> 
> here is a sip trace from FS.
> 
> http://pastebin.freeswitch.org/20305
> 
> xxx.xxx.xxx.xxx is FS, yyy.yyy.yyy.yyy is SBC,
> aaa.aaa.aaa.aaa SBC media :)
> 
> thanks for help!
> 
> Miha
> 
> Dne 12/10/2012 10:52 AM, pis(e Avi Marcus:
> >
> > Uploading the actual wirrshark file (don't attach) will
> likely help someone diagnose this easier.
> >
> > -Avi
> >
> >> On Dec 10, 2012 11:24 AM, "Miha" <miha at softnet.si
> <mailto:miha at softnet.si>> wrote:
> >>
> >> Hi,
> >>
> >> Scenario:
> >>
> >> fs ----> SBC---> innovaphone pbx
> >>
> >> calls goes throught and media is ok (g711u). When on
> other side (innovaphone pbx) someone do a 302 redirect
> innovaphone sends a sdb request in which prepose g711u,
> FS trys with g711u and send back not accabtable and media
> does not work.
> >>
> >> In attachment I am sending jpg of wireshark.
> >>
> >> >From left to right: FS, SBC, SBC MEDIA (on different
> ip than sbc).
> >>
> >> How to deal with this?
> >>
> >> Thanks!
> >> Miha
> >>
> >>
>
_________________________________________________________________________
> >> 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
> >>
> >> 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
> >
> > FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> > 
> >
> > 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
> 



From msc at freeswitch.org  Mon Dec 10 23:55:29 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Dec 2012 12:55:29 -0800
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <CADK-Bjv7sCR0KZAyhXYmQ_Dywn2QkODY0QYKqo2G8p8GTbG8vw@mail.gmail.com>
References: <CAKzWOxWG_5d5_jBVB-2H-VShLOmU8UB=BWqkhW4h4yywsAfKZg@mail.gmail.com>
	<CADK-Bjv7sCR0KZAyhXYmQ_Dywn2QkODY0QYKqo2G8p8GTbG8vw@mail.gmail.com>
Message-ID: <CAKzWOxVnK_=XuK5qg4qGMrYD40iD4zRbNyy0C8YcgYRsWJ19aQ@mail.gmail.com>

On Mon, Dec 10, 2012 at 12:35 PM, Blake Priddy <bpriddy at bryantschools.org>wrote:

> FREESWITCH FTW!!!! :)
>
>
So sayeth one of our success stories. :)
-MC
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From msc at freeswitch.org  Tue Dec 11 00:10:50 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Dec 2012 13:10:50 -0800
Subject: [Freeswitch-users] media problem afet 302
In-Reply-To: <web-88303730@softnet.si>
References: <50C5A99D.1030305@softnet.si>
	<CALNZuwmM4Tphy+B1MvKwmBD_csszGrBF_FknJQ-H7pmoDmykyA@mail.gmail.com>
	<50C5EDF0.5070805@softnet.si> <web-88303730@softnet.si>
Message-ID: <CAKzWOxW-cZZcHvkUmXGWnZZCPr8C8Tzq=_pjFAeO5A80YHRfuQ@mail.gmail.com>

I highly doubt this is a bug in FreeSWITCH. It looks like you're having
codec mismatch problems. I can see a call starting with just PCMU and also
a call starting with just PCMA enabled. See lines 160 and 280, respectively.

I doubt anyone can help without a full console debug log of all the calls
involved. We also would need to see an explanation of what each call leg is
supposed to be doing. Lastly it would be helpful to know where the pcap was
taken from, i.e. from FS or the SBC so that we know the perspective of each
point in the chart. I know you're trying to obfuscate away IP addresses,
however in doing so you've also made it really hard for us to help you
because we don't always know what is what.

-MC

On Mon, Dec 10, 2012 at 12:49 PM, Miha <miha at softnet.si> wrote:

> Hi,
>
> can some pls help me:)
>
> Is this maybe a bug and I should post to jira?
>
> BR,
> Miha
>
> On Mon, 10 Dec 2012 15:13:04 +0100
>  Miha <miha at softnet.si> wrote:
> > HI,
> >
> > here is a sip trace from FS.
> >
> > http://pastebin.freeswitch.org/20305
> >
> > xxx.xxx.xxx.xxx is FS, yyy.yyy.yyy.yyy is SBC,
> > aaa.aaa.aaa.aaa SBC media :)
> >
> > thanks for help!
> >
> > Miha
> >
> > Dne 12/10/2012 10:52 AM, pis(e Avi Marcus:
> > >
> > > Uploading the actual wirrshark file (don't attach) will
> > likely help someone diagnose this easier.
> > >
> > > -Avi
> > >
> > >> On Dec 10, 2012 11:24 AM, "Miha" <miha at softnet.si
> > <mailto:miha at softnet.si>> wrote:
> > >>
> > >> Hi,
> > >>
> > >> Scenario:
> > >>
> > >> fs ----> SBC---> innovaphone pbx
> > >>
> > >> calls goes throught and media is ok (g711u). When on
> > other side (innovaphone pbx) someone do a 302 redirect
> > innovaphone sends a sdb request in which prepose g711u,
> > FS trys with g711u and send back not accabtable and media
> > does not work.
> > >>
> > >> In attachment I am sending jpg of wireshark.
> > >>
> > >> >From left to right: FS, SBC, SBC MEDIA (on different
> > ip than sbc).
> > >>
> > >> How to deal with this?
> > >>
> > >> Thanks!
> > >> Miha
> > >>
> > >>
> >
> _________________________________________________________________________
> > >> 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
> > >>
> > >> 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
> > >
> > > FreeSWITCH-powered IP PBX: The CudaTel Communication
> > Server
> > > 
> > >
> > > 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Tue Dec 11 00:17:57 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Dec 2012 13:17:57 -0800
Subject: [Freeswitch-users] Cannot Kill stale channels
In-Reply-To: <CA+myZ6f2bfAZm5K2t7Cjki1nrMsdw3ayM1OkWEhg1o4Y0tM3Wg@mail.gmail.com>
References: <CA+myZ6dBpJoZWfrA-Wc+jiAxtTP-sd7NvgDFFaxK2d8=2eTAdg@mail.gmail.com>
	<4424748D-98BF-4BDA-A616-F6809467C3D0@visionutveckling.se>
	<CA+myZ6f2bfAZm5K2t7Cjki1nrMsdw3ayM1OkWEhg1o4Y0tM3Wg@mail.gmail.com>
Message-ID: <CAKzWOxU5RqNWnGFpAEkst7K+TjjQ3w645ahtt0gBvqrPRz_TAg@mail.gmail.com>

On Mon, Dec 10, 2012 at 12:42 PM, Marc de Corny <marcdecorny at gmail.com>wrote:

> Thanks for your response Peter, that sounds very likely.
>
> My diaplan sends everything into the queue and then my background lua
> script empties the queue every 10 seconds and tries to connect the call.
> Ideally my lua scripts "forgets" about the connected call so that it does
> not get stuck and can take another call out of the queue 10 seconds later
> independantly of the previous call being hung up.
>
>
> Am i going about this the wrong way? the only reason I do this is that the
> mod_fifo as it is does not give me enough flexibility.
>
> My script basically calls out from the FS and if connected successfully (
> new_session), perform a
>
What does this mean? You have something like this?
<action application="lua" data="my_script.lua"/>


> new_session:execute("transfer", "agent_to_queue_paymentsense_queue_service
> XML default");
> and connect via the dialplan with :<action application="fifo"
> data="queue77 out nowait"/>
>
> if I use a bridge instead of the transfer, the scripts sleeps until the
> call is hung up, with transfer, it can go 10 seconds later and take another
> call out of the queue.
>
> so is there a command I need to add to the dialplan like
> hangup_after_bridge on the outbound call?
> <action application="set" data="hangup_after_bridge=true"/>
>
> any ideas?
>
What, exactly does your Lua script do? Do you have an explicit exit clause
in there anywhere?
-MC


>
> thanks
> marc
>
>
>
> On Mon, Dec 10, 2012 at 4:40 PM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
>> Are you exiting the lua script? Usually when this happens it means you
>> have not
>> released all references to the call session object.
>>
>> /Peter
>>
>> 10 dec 2012 kl. 16:28 skrev "Marc de Corny" <marcdecorny at gmail.com
>> <mailto:marcdecorny at gmail.com>>:
>>
>> Hi
>>
>> I have calls that come into a queue and then get pulled out by a lua
>> script in background and transfered to a destination. all works fine except
>> that it looks like the sessions a not clearing from FS when all the parties
>> clear the calls.
>>
>> When I do show channels, I get this for example:
>> uuid    direction       created created_epoch   name    state   cid_name
>>        cid_num ip_addr dest    presence_id     presence_data   callstate
>>     callee_name     callee_num      callee_direction        call_uuid
>> hostname        sent_callee_name        sent_callee_num
>> 5e8215e7-b036-4cce-9b96-e57db92f13ee    outbound        03/12/2012 09:32
>>        1.35E+09        sofia/external/02031950164      CS_HANGUP
>> Outbound Call   2031950164      135.196.144.32
>>  agent_to_queue_paymentsense_queue_service                       ACTIVE
>>              SEND    5e8215e7-b036-4cce-9b96-e57db92f13ee    freeswitch2
>>   4.4209E+11      2089623100
>>
>> But when I try to get kill or query it, I get a message that the call is
>> not anywhere in the FS.
>>
>> /usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_kill
>> 5e8215e7-b036-4cce-9b96-e57db92f13ee"
>> -ERR No Such Channel!
>>
>> /usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_exists
>> 5e8215e7-b036-4cce-9b96-e57db92f13ee"
>> false
>>
>> Does anyone have an idea how I can kill that call. If I restart the FS,
>> they clear, but the problem is that they are hitting my limit on number of
>> simultaneous calls.
>>
>> Any help much appreciated.
>>
>> Thansk
>> marc
>>
>>
>>
>>
>>
>>
>> !DSPAM:50c5fb8c32767955115405!
>> _________________________________________________________________________
>> 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://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org
>>
>>
>> !DSPAM:50c5fb8c32767955115405!
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From steveayre at gmail.com  Tue Dec 11 00:31:00 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 10 Dec 2012 21:31:00 +0000
Subject: [Freeswitch-users] Natting question.
In-Reply-To: <50C62854.60102@gmail.com>
References: <50C5EAAA.6040100@gmail.com>
	<CAFiqYumipKg+-D0-c3NqXrUdJ0BMioZyDuueq67gf9213C8_Tg@mail.gmail.com>
	<50C62854.60102@gmail.com>
Message-ID: <CAFiqYuma7BJSt4_WQnNAcQ8B6sWi13HkLVNXAu-phXgcpPm2bA@mail.gmail.com>

Do you see any entries in the FS logs relating to rtp auto-adjust at about
the time the audio starts working?

If so that could indicate FS is getting an incorrect (internal) IP in the
SDP from the client, and audio is starting when the client starts sending
to FS after a few seconds.

The other party have to hung up, for example, in order for call to end.


This also indicates a NAT issue. The BYE hangup gets sent to the IP:port in
the Contact header, not the address the INVITE came from. If the contact
header is incorrect the BYE goes to the wrong place and so the client never
sees it.

If you can't correct that on the client end (STUN, set manually, SIP ALG
etc) then try
http://wiki.freeswitch.org/wiki/NDLB#NDLB-connectile-dysfunction and/or
http://wiki.freeswitch.org/wiki/NDLB#NDLB-force-rport which will tell FS to
break from the SIP standard to support buggy clients.

-Steve




On 10 December 2012 18:22, Mimiko <vbvbrj at gmail.com> wrote:

> On 10.12.2012 20:00, Steven Ayre wrote:
> > So to clarify:
> >
> > Client -> Internet -> NAT -> FS
> > works
> >
> > Client -> NAT -> Internet -> NAT -> FS
> > does not?
> >
> > Since the FS NAT side appears to be working (from the 1st case), I'm
> > guessing it's probably either a problem at the client side (with client
> > or NAT router) or perhaps the double-NAT is causing an issue. Personally
> > I haven't run FS behind NAT, so it's not a situation I've had to handle.
>
> Thank you for a comprehensive response.
>
> What I've seen now. Audio actually comes up, but it is taking a while.
> Sometimes is ~5sec, sometimes more, sometimes never. Also, sip commands
> does not take action. The other party have to hung up, for example, in
> order for call to end.
>
> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 curriegrad2004 at gmail.com  Tue Dec 11 01:42:10 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 10 Dec 2012 14:42:10 -0800
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKzWOxVnK_=XuK5qg4qGMrYD40iD4zRbNyy0C8YcgYRsWJ19aQ@mail.gmail.com>
References: <CAKzWOxWG_5d5_jBVB-2H-VShLOmU8UB=BWqkhW4h4yywsAfKZg@mail.gmail.com>
	<CADK-Bjv7sCR0KZAyhXYmQ_Dywn2QkODY0QYKqo2G8p8GTbG8vw@mail.gmail.com>
	<CAKzWOxVnK_=XuK5qg4qGMrYD40iD4zRbNyy0C8YcgYRsWJ19aQ@mail.gmail.com>
Message-ID: <CAMWiMy9vtMYZ8hP6p15R9Nh=4u6BfJYMKufaLaskWrLyCuY=WA@mail.gmail.com>

At least FreeSWITCH is easier to understand than Asterisk for me ;)

On Mon, Dec 10, 2012 at 12:55 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Mon, Dec 10, 2012 at 12:35 PM, Blake Priddy <bpriddy at bryantschools.org>
> wrote:
>>
>> FREESWITCH FTW!!!! :)
>>
>
> So sayeth one of our success stories. :)
> -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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From krice at freeswitch.org  Tue Dec 11 02:23:50 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 10 Dec 2012 17:23:50 -0600
Subject: [Freeswitch-users] Wanted Perl Programmer to help with a little
 Project for the Visually Impaired FreeSWITCH users.
Message-ID: <CCEBCB26.61D69%krice@freeswitch.org>

As many of you know we have a pretty large number of FreeSWITCH users that
are visually impaired (from significant vision loss to right out blind).

Most of these users use screen readers to help them, and in some cases they
use other technologies like Brail output devices. However, these things only
go so far.

One of the issues they have is with Jira, and its lack of screen reader
?friendlyness?... I think I have a solution for this... If you are a
reasonably good perl programmer and want to help me out on a small project
drop me an email offlist. I have an idea that should make it much easier for
this group of users.

K

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From anthony.minessale at gmail.com  Tue Dec 11 03:25:40 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 Dec 2012 18:25:40 -0600
Subject: [Freeswitch-users] Wish Lists - 'tis the season
In-Reply-To: <CAKzWOxV3yi3_tXBtZK8UCMv98UUSFasztJBjOqSGhVz9vPhu5w@mail.gmail.com>
References: <31fb01cdd6f8$fa356850$eea038f0$@bizfocused.com>
	<CAKzWOxV3yi3_tXBtZK8UCMv98UUSFasztJBjOqSGhVz9vPhu5w@mail.gmail.com>
Message-ID: <CAKbxfG--PB1qs5NnjFS2HnPE8ULtF1hcq67-X9wTQ8NkX-_Pfw@mail.gmail.com>

I just asked my family to help update mine.  I'm usually too busy coding to
stop and keep up w/ Amazon =p



On Mon, Dec 10, 2012 at 1:33 PM, Michael Collins <msc at freeswitch.org> wrote:

> Many thanks, sir!
>
> We'll look into the possibility of doing a donation thing. The other thing
> we can do is maybe have each person put a $25 Amazon gift card on their
> wishlist and then people can buy as many or as few as they wish.
>
> Thanks!
> -MC
>
> On Mon, Dec 10, 2012 at 9:08 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>
>> Hey Guys,****
>>
>> ** **
>>
>> Update those wish lists.  Anthony?s starts at like $495!  I want to show
>> my love, but I am a poor VOIP provider and not spending $495 on my wife
>> this season!  I hooked up BF and MC but they could use a few more
>> suggestions in the $25 to $50 range.****
>>
>> ** **
>>
>> Anybody ever setup a contribution app so users could leave a contribution
>> TOWARD the purchase of one of the expensive items that you guys deserve.
>> Then others could see that and add to it?  I could hook that up easy
>> enough.  Keeping track of what was purchased outright on the other list
>> might be too confusing, but separate lists should be a breeze.****
>>
>> ** **
>>
>> Just a thought.****
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Anthony Minessale II

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
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From bdfoster at endigotech.com  Tue Dec 11 04:15:38 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 10 Dec 2012 20:15:38 -0500
Subject: [Freeswitch-users] Wish Lists - 'tis the season
In-Reply-To: <31fb01cdd6f8$fa356850$eea038f0$@bizfocused.com>
References: <31fb01cdd6f8$fa356850$eea038f0$@bizfocused.com>
Message-ID: <A535DEBB-85AD-4C99-BA20-694038CCC7BB@endigotech.com>

Wait a minute, I have a wish list?

Sent from my iPhone

On Dec 10, 2012, at 12:08 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> Hey Guys,
>  
> Update those wish lists.  Anthony?s starts at like $495!  I want to show my love, but I am a poor VOIP provider and not spending $495 on my wife this season!  I hooked up BF and MC but they could use a few more suggestions in the $25 to $50 range.
>  
> Anybody ever setup a contribution app so users could leave a contribution TOWARD the purchase of one of the expensive items that you guys deserve.  Then others could see that and add to it?  I could hook that up easy enough.  Keeping track of what was purchased outright on the other list might be too confusing, but separate lists should be a breeze.
>  
> Just a thought.
> Sean
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From gabe at gundy.org  Tue Dec 11 05:18:43 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 10 Dec 2012 19:18:43 -0700
Subject: [Freeswitch-users] Wish Lists - 'tis the season
In-Reply-To: <A535DEBB-85AD-4C99-BA20-694038CCC7BB@endigotech.com>
References: <31fb01cdd6f8$fa356850$eea038f0$@bizfocused.com>
	<A535DEBB-85AD-4C99-BA20-694038CCC7BB@endigotech.com>
Message-ID: <CAMwZ-tLM_yN8oeMRvqrAWTVMCs-XhkVe3QYK2=j8n=RFOLNi5A@mail.gmail.com>

You wish.

Best,
Gabe

(disclaimer about spelling errors, grammar, punctuation etc. --sent from my
mobile phone)
On Dec 10, 2012 6:18 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> Wait a minute, I have a wish list?
>
> Sent from my iPhone
>
> On Dec 10, 2012, at 12:08 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:
>
> Hey Guys,****
>
> ** **
>
> Update those wish lists.  Anthony?s starts at like $495!  I want to show
> my love, but I am a poor VOIP provider and not spending $495 on my wife
> this season!  I hooked up BF and MC but they could use a few more
> suggestions in the $25 to $50 range.****
>
> ** **
>
> Anybody ever setup a contribution app so users could leave a contribution
> TOWARD the purchase of one of the expensive items that you guys deserve.
> Then others could see that and add to it?  I could hook that up easy
> enough.  Keeping track of what was purchased outright on the other list
> might be too confusing, but separate lists should be a breeze.****
>
> ** **
>
> Just a thought.****
>
> Sean****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Tue Dec 11 06:07:12 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 10 Dec 2012 22:07:12 -0500
Subject: [Freeswitch-users] Wish Lists - 'tis the season
In-Reply-To: <CAMwZ-tLM_yN8oeMRvqrAWTVMCs-XhkVe3QYK2=j8n=RFOLNi5A@mail.gmail.com>
References: <31fb01cdd6f8$fa356850$eea038f0$@bizfocused.com>
	<A535DEBB-85AD-4C99-BA20-694038CCC7BB@endigotech.com>
	<CAMwZ-tLM_yN8oeMRvqrAWTVMCs-XhkVe3QYK2=j8n=RFOLNi5A@mail.gmail.com>
Message-ID: <3F97033D-ADF1-4965-85FB-5F2AD53E3A82@endigotech.com>

I do now lol

Sent from my iPhone

On Dec 10, 2012, at 9:18 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> You wish.
> 
> Best,
> Gabe
> 
> (disclaimer about spelling errors, grammar, punctuation etc. --sent from my mobile phone)
> 
> On Dec 10, 2012 6:18 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>> Wait a minute, I have a wish list?
>> 
>> Sent from my iPhone
>> 
>> On Dec 10, 2012, at 12:08 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:
>> 
>>> Hey Guys,
>>> 
>>>  
>>> 
>>> Update those wish lists.  Anthony?s starts at like $495!  I want to show my love, but I am a poor VOIP provider and not spending $495 on my wife this season!  I hooked up BF and MC but they could use a few more suggestions in the $25 to $50 range.
>>> 
>>>  
>>> 
>>> Anybody ever setup a contribution app so users could leave a contribution TOWARD the purchase of one of the expensive items that you guys deserve.  Then others could see that and add to it?  I could hook that up easy enough.  Keeping track of what was purchased outright on the other list might be too confusing, but separate lists should be a breeze.
>>> 
>>>  
>>> 
>>> Just a thought.
>>> 
>>> 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
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From Sirish.MasurMohan at oa.com.au  Tue Dec 11 05:24:01 2012
From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan)
Date: Tue, 11 Dec 2012 13:24:01 +1100
Subject: [Freeswitch-users] FreeSWITCH+Sangoma with dialup modems
In-Reply-To: <CAA4nhyD-uMSecPn=4P24=VeNY4h8Mt=sd_BA7T5JSPoA4TfYgQ@mail.gmail.com>
References: <965759A53E43FE439E43565A7715E5F058F1BE0EE8@oa-exchange1.oa.com.au>
	<CAA4nhyD-uMSecPn=4P24=VeNY4h8Mt=sd_BA7T5JSPoA4TfYgQ@mail.gmail.com>
Message-ID: <965759A53E43FE439E43565A7715E5F058F1BE0F26@oa-exchange1.oa.com.au>

Hi Moises,

Thanks for the reply. I have been able to resolve this issue ? looks like I had to explicitly choose the modem carrier (1200) using the +MS command.
However, curious to know a bit more on the echo canceller that you are referring to ? we are using the Sangoma card with echo canceller, and I haven?t explicitly turned it off!

With regards,
Sirish

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises Silva
Sent: Tuesday, 11 December 2012 7:00 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSWITCH+Sangoma with dialup modems

On Mon, Dec 10, 2012 at 2:40 AM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au<mailto:Sirish.MasurMohan at oa.com.au>> wrote:
My setup is as follows:
Serial port based device(Caller)->Dlink dialup modem(DFM562E)->PBX->Sangoma A102 E1 Link(and FreeSWITCH)->PBX->Dlink dialup modem(DFM562E)->Server(Receiver)

Problem:
When the caller makes the call (using the modem with a lower baud rate 1200), Receiver modem receives the RING and answers the call (using the ATA command). But for some reason, both the ends timeout waiting for modem CONNECT.

Are you sure there is no echo cancellers in the middle (make sure you have an A102 without echo canceller or that you disable it in wanpipe). You can always make sure you disable the ec in the dialplan too (<action application="disable_ec">)



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 | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>


 <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>
 <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>
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From moises.silva at gmail.com  Tue Dec 11 08:22:19 2012
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 11 Dec 2012 00:22:19 -0500
Subject: [Freeswitch-users] FreeSWITCH+Sangoma with dialup modems
In-Reply-To: <965759A53E43FE439E43565A7715E5F058F1BE0F26@oa-exchange1.oa.com.au>
References: <965759A53E43FE439E43565A7715E5F058F1BE0EE8@oa-exchange1.oa.com.au>
	<CAA4nhyD-uMSecPn=4P24=VeNY4h8Mt=sd_BA7T5JSPoA4TfYgQ@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F1BE0F26@oa-exchange1.oa.com.au>
Message-ID: <CAA4nhyCHF9zOYP5ABvidGLouR1_go9FA89v3G0T_+1F745odYA@mail.gmail.com>

On Mon, Dec 10, 2012 at 9:24 PM, Sirish Masur Mohan <
Sirish.MasurMohan at oa.com.au> wrote:

> Hi Moises,****
>
> ** **
>
> Thanks for the reply. I have been able to resolve this issue ? looks like
> I had to explicitly choose the modem carrier (1200) using the +MS command.
> ****
>
> However, curious to know a bit more on the echo canceller that you are
> referring to ? we are using the Sangoma card with echo canceller, and I
> haven?t explicitly turned it off!****
>
> **
>
Echo cancellation is not a good idea in data calls. Mind you, depending on
the type of data call there is variable tolerance. Also, some data calls
(ie fax) start by sending a well-known tone that Sangoma echo canceller
recognizes and then disables itself (see CED tone here:
http://telecom.tbi.net/fax-call.htm).

If it's working for you, good :) ... if you start having issues with some
calls succeeding some not, now you know another thing to try (turn off the
ec).


*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 Sirish.MasurMohan at oa.com.au  Tue Dec 11 09:16:23 2012
From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan)
Date: Tue, 11 Dec 2012 17:16:23 +1100
Subject: [Freeswitch-users] FreeSWITCH+Sangoma with dialup modems
In-Reply-To: <CAA4nhyCHF9zOYP5ABvidGLouR1_go9FA89v3G0T_+1F745odYA@mail.gmail.com>
References: <965759A53E43FE439E43565A7715E5F058F1BE0EE8@oa-exchange1.oa.com.au>
	<CAA4nhyD-uMSecPn=4P24=VeNY4h8Mt=sd_BA7T5JSPoA4TfYgQ@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F1BE0F26@oa-exchange1.oa.com.au>
	<CAA4nhyCHF9zOYP5ABvidGLouR1_go9FA89v3G0T_+1F745odYA@mail.gmail.com>
Message-ID: <965759A53E43FE439E43565A7715E5F058F1BE0F48@oa-exchange1.oa.com.au>

Thanks Moises, noted!

With regards,
Sirish

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises Silva
Sent: Tuesday, 11 December 2012 4:22 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSWITCH+Sangoma with dialup modems

On Mon, Dec 10, 2012 at 9:24 PM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au<mailto:Sirish.MasurMohan at oa.com.au>> wrote:
Hi Moises,

Thanks for the reply. I have been able to resolve this issue ? looks like I had to explicitly choose the modem carrier (1200) using the +MS command.
However, curious to know a bit more on the echo canceller that you are referring to ? we are using the Sangoma card with echo canceller, and I haven?t explicitly turned it off!
Echo cancellation is not a good idea in data calls. Mind you, depending on the type of data call there is variable tolerance. Also, some data calls (ie fax) start by sending a well-known tone that Sangoma echo canceller recognizes and then disables itself (see CED tone here: http://telecom.tbi.net/fax-call.htm).

If it's working for you, good :) ... if you start having issues with some calls succeeding some not, now you know another thing to try (turn off the ec).



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 | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>


 <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>
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From miha at softnet.si  Tue Dec 11 10:26:05 2012
From: miha at softnet.si (Miha)
Date: Tue, 11 Dec 2012 08:26:05 +0100
Subject: [Freeswitch-users] media problem afet 302
In-Reply-To: <CAKzWOxW-cZZcHvkUmXGWnZZCPr8C8Tzq=_pjFAeO5A80YHRfuQ@mail.gmail.com>
References: <50C5A99D.1030305@softnet.si>
	<CALNZuwmM4Tphy+B1MvKwmBD_csszGrBF_FknJQ-H7pmoDmykyA@mail.gmail.com>
	<50C5EDF0.5070805@softnet.si> <web-88303730@softnet.si>
	<CAKzWOxW-cZZcHvkUmXGWnZZCPr8C8Tzq=_pjFAeO5A80YHRfuQ@mail.gmail.com>
Message-ID: <50C6E00D.7090603@softnet.si>

Hi Michael,

thanks for your reply.

http://pastebin.freeswitch.com/20318

In pastebin I post full debug log from FS.

xxx.xxx.xxx.xxx is FS, yyy.yyy.yyy.yyy is SBC, aaa.aaa.aaa.aaa is SBC media.

 From phone (linksys) is called number 074901690, number is matched in 
dialplan, naptr look up is made and  on FS is checked if number is on 
FS, if not than number with 386+ nrn code (enum prefix) is send to sbc.
Sbc send call to a PBX where call is pick up (IVR, call is set up with 
g711u). On other side (innovaphone pbx) ivr tells you which number to 
use so that you will be redirected to different group of people.

I dial 1, which do in pbx 302 redirect and send phone call to outside 
(mobile carrier). When this is made pbx send new sdp invite to FS, to 
change codect to g711a, which FS refuse and send not acceptable.

Michael I also send you on private trace which was mad od FS and on SBC. 
I can send it also to someone, but I will prefer not to post it to user 
group as recently we are dealing with a loot of sip attacks.

BR,
Miha

Dne 12/10/2012 10:10 PM, pis(e Michael Collins:
> I highly doubt this is a bug in FreeSWITCH. It looks like you're 
> having codec mismatch problems. I can see a call starting with just 
> PCMU and also a call starting with just PCMA enabled. See lines 160 
> and 280, respectively.
>
> I doubt anyone can help without a full console debug log of all the 
> calls involved. We also would need to see an explanation of what each 
> call leg is supposed to be doing. Lastly it would be helpful to know 
> where the pcap was taken from, i.e. from FS or the SBC so that we know 
> the perspective of each point in the chart. I know you're trying to 
> obfuscate away IP addresses, however in doing so you've also made it 
> really hard for us to help you because we don't always know what is what.
>
> -MC
>
> On Mon, Dec 10, 2012 at 12:49 PM, Miha <miha at softnet.si 
> <mailto:miha at softnet.si>> wrote:
>
>     Hi,
>
>     can some pls help me:)
>
>     Is this maybe a bug and I should post to jira?
>
>     BR,
>     Miha
>
>     On Mon, 10 Dec 2012 15:13:04 +0100
>      Miha <miha at softnet.si <mailto:miha at softnet.si>> wrote:
>     > HI,
>     >
>     > here is a sip trace from FS.
>     >
>     > http://pastebin.freeswitch.org/20305
>     >
>     > xxx.xxx.xxx.xxx is FS, yyy.yyy.yyy.yyy is SBC,
>     > aaa.aaa.aaa.aaa SBC media :)
>     >
>     > thanks for help!
>     >
>     > Miha
>     >
>     > Dne 12/10/2012 10:52 AM, pis(e Avi Marcus:
>     > >
>     > > Uploading the actual wirrshark file (don't attach) will
>     > likely help someone diagnose this easier.
>     > >
>     > > -Avi
>     > >
>     > >> On Dec 10, 2012 11:24 AM, "Miha" <miha at softnet.si
>     <mailto:miha at softnet.si>
>     > <mailto:miha at softnet.si <mailto:miha at softnet.si>>> wrote:
>     > >>
>     > >> Hi,
>     > >>
>     > >> Scenario:
>     > >>
>     > >> fs ----> SBC---> innovaphone pbx
>     > >>
>     > >> calls goes throught and media is ok (g711u). When on
>     > other side (innovaphone pbx) someone do a 302 redirect
>     > innovaphone sends a sdb request in which prepose g711u,
>     > FS trys with g711u and send back not accabtable and media
>     > does not work.
>     > >>
>     > >> In attachment I am sending jpg of wireshark.
>     > >>
>     > >> >From left to right: FS, SBC, SBC MEDIA (on different
>     > ip than sbc).
>     > >>
>     > >> How to deal with this?
>     > >>
>     > >> Thanks!
>     > >> Miha
>     > >>
>     > >>
>     >
>     _________________________________________________________________________
>     > >> 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
>     > >>
>     > >> FreeSWITCH-powered IP PBX: The CudaTel Communication
>     > Server
>     > >> 
>     > >>
>     > >> 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>
>     > <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>
>     > > 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
>     > >
>     > > 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
>
>     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
>
>
>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 sdevoy at bizfocused.com  Tue Dec 11 12:02:18 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 11 Dec 2012 04:02:18 -0500
Subject: [Freeswitch-users] Wish Lists - 'tis the season
In-Reply-To: <A535DEBB-85AD-4C99-BA20-694038CCC7BB@endigotech.com>
References: <31fb01cdd6f8$fa356850$eea038f0$@bizfocused.com>
	<A535DEBB-85AD-4C99-BA20-694038CCC7BB@endigotech.com>
Message-ID: <371f01cdd77e$3972d310$ac587930$@bizfocused.com>

Oh CRAP!  I sent it to Brian West.  You Brian?s all look alike to me.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster
Sent: Monday, December 10, 2012 8:16 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Wish Lists - 'tis the season

 

Wait a minute, I have a wish list?

Sent from my iPhone


On Dec 10, 2012, at 12:08 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

Hey Guys,

 

Update those wish lists.  Anthony?s starts at like $495!  I want to show my love, but I am a poor VOIP provider and not spending $495 on my wife this season!  I hooked up BF and MC but they could use a few more suggestions in the $25 to $50 range.

 

Anybody ever setup a contribution app so users could leave a contribution TOWARD the purchase of one of the expensive items that you guys deserve.  Then others could see that and add to it?  I could hook that up easy enough.  Keeping track of what was purchased outright on the other list might be too confusing, but separate lists should be a breeze.

 

Just a thought.

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

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

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From asilva at wirelessmundi.com  Tue Dec 11 13:46:01 2012
From: asilva at wirelessmundi.com (Antonio)
Date: Tue, 11 Dec 2012 11:46:01 +0100
Subject: [Freeswitch-users] Compiling FSComm
In-Reply-To: <21EB7A95-BDA4-48FE-A944-DFB5662BBB14@freeswitch.org>
References: <1354878605.5967.44.camel@vmmarces.vm.marces.com>
	<21EB7A95-BDA4-48FE-A944-DFB5662BBB14@freeswitch.org>
Message-ID: <1355222761.7628.21.camel@marces.madrid.commsmundi.com>

Hi Jo?o,

I just open you the jira, http://jira.freeswitch.org/browse/FSCOMM-11


Thanks,
Ant?nio




On Sun, 2012-12-09 at 02:22 -0200, Jo?o Mesquita wrote: 

> Antonio, I am the dev of FSComm and I am glad to hear there is still interest on it. I will try to fix the problem tomorrow and if not Monday as I am out of the country. 
> 
> Please help us out and file a Jira for it?
> 
> Sent from my iPhone
> 
> On Dec 7, 2012, at 9:10 AM, Antonio Silva <asilva at wirelessmundi.com> wrote:
> 
> > Hi,
> > 
> > i'm trying to compile fscomm but i have the following errors:
> > 
> > "
> > freeswitch-git/fscomm# qmake
> > freeswitch-git/fscomm# make
> > /usr/bin/uic-qt4 mainwindow.ui -o ui_mainwindow.h
> > /usr/bin/uic-qt4 preferences/prefdialog.ui -o ui_prefdialog.h
> > /usr/bin/uic-qt4 preferences/accountdialog.ui -o ui_accountdialog.h
> > /usr/bin/uic-qt4 widgets/codecwidget.ui -o ui_codecwidget.h
> > /usr/bin/uic-qt4 debugtools/consolewindow.ui -o ui_consolewindow.h
> > /usr/bin/uic-qt4 debugtools/statedebugdialog.ui -o ui_statedebugdialog.h
> > Warning: name layoutWidget is already used
> > Warning: name layoutWidget is already used
> > g++ -c -pipe -O2 -Wall -W -D_REENTRANT -DQT_NO_DEBUG -DQT_XML_LIB -DQT_GUI_LIB -DQT_CORE_LIB -DQT_SHARED -I/usr/share/qt4/mkspecs/linux-g++ -I. -I/usr/include/qt4/QtCore -I/usr/include/qt4/QtGui -I/usr/include/qt4/QtXml -I/usr/include/qt4 -I../src/include -I../libs/apr/include -I../libs/libteletone/src -I. -I. -o main.o main.cpp
> > In file included from mainwindow.h:38,
> >                  from main.cpp:32:
> > ../src/include/switch.h:110:18: error: stfu.h: No such file or directory
> > In file included from ../src/include/switch.h:121,
> >                  from mainwindow.h:38,
> >                  from main.cpp:32:
> > ../src/include/switch_core.h:752: error: expected constructor, destructor, or type conversion before ?*? token
> > In file included from ../src/include/switch_loadable_module.h:46,
> >                  from ../src/include/switch.h:122,
> >                  from mainwindow.h:38,
> >                  from main.cpp:32:
> > ../src/include/switch_module_interfaces.h:121: error: expected initializer before ?*? token
> > ../src/include/switch_module_interfaces.h:162: error: ?switch_io_get_jb_t? does not name a type
> > In file included from ../src/include/switch.h:134,
> >                  from mainwindow.h:38,
> >                  from main.cpp:32:
> > ../src/include/switch_rtp.h:243: error: expected constructor, destructor, or type conversion before ?*? token
> > ./fshost.h:43: warning: ?void eventHandlerCallback(switch_event_t*)? declared ?static? but never defined
> > ./fshost.h:44: warning: ?switch_status_t loggerHandler(const switch_log_node_t*, switch_log_level_t)? declared ?static? but never defined
> > make: *** [main.o] Error 1
> > "
> > 
> > watching this error: 
> > "../src/include/switch.h:110:18: error: stfu.h: No such file or directory"
> > 
> > i manually change switch.h to fix the problem with the include by adding the following:
> > 
> > "
> > diff --git a/src/include/switch.h b/src/include/switch.h
> > index c7ea7b0..2847112 100644
> > --- a/src/include/switch.h
> > +++ b/src/include/switch.h
> > @@ -107,7 +107,8 @@
> > #include <errno.h>
> > 
> > #ifndef WIN32
> > -#include "stfu.h"
> > +/* #include "stfu.h" */
> > +#include "../../../libs/stfu/stfu.h"
> > #else
> > #include "../../../libs/stfu/stfu.h"
> > #endif
> > 
> > "
> > 
> > I could compile fscomm, but now i can't start it... i have the following error:
> > "
> > Initializing core... 
> > Failed to initialize FreeSWITCH's core: Cannot Open log directory or XML Root!
> > Everything OK, Entering runtime loop ... 
> > Segmentation fault
> > "
> > i had try the fix in the wiki: "chmod  644 ~/.fscomm/conf/freeswitch.xml", and even "chmod -R 777 ~/.fscomm",  but no luck...
> > 
> > Can you help me to go further...?
> > 
> > I'm trying it to install on a debian squeeze, i installed qt4-dev-tools. Is it possible to install in debian squeeze our i should just  give up... and try another distro?
> > The freeswitch-git is the lasted head. 
> > 
> > 
> > Thanks,
> > Ant?nio
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 

Un cordial saludo / Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com
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From acrow at integrafin.co.uk  Tue Dec 11 20:19:48 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Tue, 11 Dec 2012 17:19:48 +0000
Subject: [Freeswitch-users] Dialplan for FS modems - odd behaviour or am I
	doing it wrong?
Message-ID: <50C76B34.6080007@integrafin.co.uk>

Hi,

Please see the below log snippet. I am accepting faxes from an ISDN 
gateway on sequential numbers and attempting to route them into distinct 
freeswitch modems by taking the last two digits of the called number thusly:

    <condition field="destination_number" 
expression="^12([1-2][0-9]|3[0-6])$">
         <!--<action application="set" data="ignore_early_media=true"/>-->
         <action application="set" data="sip_ignore_reinvites=true"/>
         <action application="set" 
data="absolute_codec_string='PCMA,PCMU'"/>
         <action application="set" data="jitterbuffer_msec=600:600:60"/>
         <action application="set" data="sip_jitter_buffer_plc=false"/>
         <action application="bridge" data="modem/$1/a"/>
    </condition>

And then have Hylafax email them on depending on which modem they 
arrived on (via FaxDispatch).

However as you can see from the below log snippets, it seems that FS is 
quite happy to change the modem device from what is actually delivered 
via the dialplan. Consequently a number of my faxes are now being routed 
to the wrong people.

Is my bridge dest wrong? Should it be something like modem/$1/0 instead 
of "a"?

Thanks

Alex

Dialplan: sofia/internal/anonymous at anonymous.invalid Action 
set(sip_ignore_reinvites=true)
Dialplan: sofia/internal/anonymous at anonymous.invalid Action 
set(absolute_codec_string='PCMA,PCMU')
Dialplan: sofia/internal/anonymous at anonymous.invalid Action 
set(jitterbuffer_msec=600:600:60)
Dialplan: sofia/internal/anonymous at anonymous.invalid Action 
set(sip_jitter_buffer_plc=false)
Dialplan: sofia/internal/anonymous at anonymous.invalid Action 
bridge(modem/13/a)
2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:167 
(sofia/internal/anonymous at anonymous.invalid) State Change CS_ROUTING -> 
CS_EXECUTE
2012-12-11 14:43:18.873181 [DEBUG] switch_core_session.c:1283 Send 
signal sofia/internal/anonymous at anonymous.invalid [BREAK]
2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:470 
(sofia/internal/anonymous at anonymous.invalid) State ROUTING going to sleep
2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:415 
(sofia/internal/anonymous at anonymous.invalid) Running State Change CS_EXECUTE
2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:477 
(sofia/internal/anonymous at anonymous.invalid) State EXECUTE
2012-12-11 14:43:18.873181 [DEBUG] mod_sofia.c:242 
sofia/internal/anonymous at anonymous.invalid SOFIA EXECUTE
2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:209 
sofia/internal/anonymous at anonymous.invalid Standard EXECUTE
EXECUTE sofia/internal/anonymous at anonymous.invalid set(call_direction=local)
2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344 
sofia/internal/anonymous at anonymous.invalid SET [call_direction]=[local]
EXECUTE sofia/internal/anonymous at anonymous.invalid set(open=true)
2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344 
sofia/internal/anonymous at anonymous.invalid SET [open]=[true]
EXECUTE sofia/internal/anonymous at anonymous.invalid 
hash(insert/192.168.20.245-spymap/anonymous/4e5c4699-aa22-4998-b7fa-b50a52c940fb)
EXECUTE sofia/internal/anonymous at anonymous.invalid 
hash(insert/192.168.20.245-last_dial/anonymous/1213)
EXECUTE sofia/internal/anonymous at anonymous.invalid 
hash(insert/192.168.20.245-last_dial/global/4e5c4699-aa22-4998-b7fa-b50a52c940fb)
EXECUTE sofia/internal/anonymous at anonymous.invalid set(RFC2822_DATE=Tue, 
11 Dec 2012 14:43:18 +0000)
2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344 
sofia/internal/anonymous at anonymous.invalid SET [RFC2822_DATE]=[Tue, 11 
Dec 2012 14:43:18 +0000]
EXECUTE sofia/internal/anonymous at anonymous.invalid 
set(sip_ignore_reinvites=true)
2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344 
sofia/internal/anonymous at anonymous.invalid SET [sip_ignore_reinvites]=[true]
EXECUTE sofia/internal/anonymous at anonymous.invalid 
set(absolute_codec_string='PCMA,PCMU')
2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344 
sofia/internal/anonymous at anonymous.invalid SET 
[absolute_codec_string]=['PCMA,PCMU']
EXECUTE sofia/internal/anonymous at anonymous.invalid 
set(jitterbuffer_msec=600:600:60)
2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344 
sofia/internal/anonymous at anonymous.invalid SET 
[jitterbuffer_msec]=[600:600:60]
EXECUTE sofia/internal/anonymous at anonymous.invalid 
set(sip_jitter_buffer_plc=false)
2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344 
sofia/internal/anonymous at anonymous.invalid SET 
[sip_jitter_buffer_plc]=[false]
EXECUTE sofia/internal/anonymous at anonymous.invalid bridge(modem/13/a)
2012-12-11 14:43:18.873181 [DEBUG] switch_channel.c:1089 
sofia/internal/anonymous at anonymous.invalid EXPORTING[export_vars] 
[domain_name]=[192.168.20.245] to event
2012-12-11 14:43:18.873181 [DEBUG] switch_ivr_originate.c:2022 Parsing 
global variables
2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:1417 Modem 
/dev/FS/FS10 [ONHOOK] - Changing state to ACQUIRED
2012-12-11 14:43:18.873181 [NOTICE] switch_channel.c:968 New Channel 
modem/10/a [a202aef1-1628-45e8-b720-9f5368aa4565]
2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:769 modem/10/a 
setup codec L16/8000/20
2012-12-11 14:43:18.873181 [DEBUG] switch_channel.c:1135 EXPORT 
(export_vars) [rtp_autoflush_during_bridge]=[false]
2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:896 (modem/10/a) 
State Change CS_NEW -> CS_INIT
2012-12-11 14:43:18.873181 [DEBUG] switch_core_session.c:1283 Send 
signal modem/10/a [BREAK]
2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:581 modem/10/a 
CHANNEL KILL
2012-12-11 14:43:18.893118 [DEBUG] switch_core_state_machine.c:415 
(modem/10/a) Running State Change CS_INIT
2012-12-11 14:43:18.893118 [DEBUG] switch_core_state_machine.c:454 
(modem/10/a) State INIT
2012-12-11 14:43:18.893118 [DEBUG] mod_spandsp_modem.c:461 Modem 
/dev/FS/FS10 [ACQUIRED] - Changing state to RINGING
2012-12-11 14:43:18.893118 [DEBUG] mod_spandsp_modem.c:1131 Modem 
/dev/FS/FS10 [RINGING] - RNG 1


And another one:


EXECUTE sofia/internal/anonymous at anonymous.invalid bridge(modem/23/a)
2012-12-11 13:42:20.813107 [DEBUG] switch_channel.c:1089 
sofia/internal/anonymous at anonymous.invalid EXPORTING[export_vars] 
[domain_name]=[192.168.20.245] to event
2012-12-11 13:42:20.813107 [DEBUG] switch_ivr_originate.c:2022 Parsing 
global variables
2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:1417 Modem 
/dev/FS/FS26 [ONHOOK] - Changing state to ACQUIRED
2012-12-11 13:42:20.813107 [NOTICE] switch_channel.c:968 New Channel 
modem/26/a [32872ae9-6f78-45b0-af56-4b18c9de7160]
2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:769 modem/26/a 
setup codec L16/8000/20
2012-12-11 13:42:20.813107 [DEBUG] switch_channel.c:1135 EXPORT 
(export_vars) [rtp_autoflush_during_bridge]=[false]
2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:896 (modem/26/a) 
State Change CS_NEW -> CS_INIT
2012-12-11 13:42:20.813107 [DEBUG] switch_core_session.c:1283 Send 
signal modem/26/a [BREAK]
2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:581 modem/26/a 
CHANNEL KILL
2012-12-11 13:42:20.813107 [DEBUG] switch_core_state_machine.c:415 
(modem/26/a) Running State Change CS_INIT
2012-12-11 13:42:20.813107 [DEBUG] switch_core_state_machine.c:454 
(modem/26/a) State INIT
2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:461 Modem 
/dev/FS/FS26 [ACQUIRED] - Changing state to RINGING
2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:1131 Modem 
/dev/FS/FS26 [RINGING] - RNG 1
2012-12-11 13:42:20.833137 [DEBUG] mod_spandsp_modem.c:1038 Modem 
/dev/FS/FS26 [RINGING] - Answering
2012-12-11 13:42:20.833137 [DEBUG] mod_spandsp_modem.c:1040 Modem 
/dev/FS/FS26 [RINGING] - Changing state to ANSWERED


From msc at freeswitch.org  Tue Dec 11 20:23:14 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Dec 2012 09:23:14 -0800
Subject: [Freeswitch-users] media problem afet 302
In-Reply-To: <50C6E00D.7090603@softnet.si>
References: <50C5A99D.1030305@softnet.si>
	<CALNZuwmM4Tphy+B1MvKwmBD_csszGrBF_FknJQ-H7pmoDmykyA@mail.gmail.com>
	<50C5EDF0.5070805@softnet.si> <web-88303730@softnet.si>
	<CAKzWOxW-cZZcHvkUmXGWnZZCPr8C8Tzq=_pjFAeO5A80YHRfuQ@mail.gmail.com>
	<50C6E00D.7090603@softnet.si>
Message-ID: <CAKzWOxUN2GBp=VXB+Oy+fJiWRBXkkgLG78r_s8HszbgCtUkujg@mail.gmail.com>

Miha,

Can you turn off sofia debugging. That's just too much noise and it doesn't
help you. Create a new pastebin without the the sofia debug. Leave the sip
trace on. ("sofia global siptrace on")

I'm probably not the person who is most qualified to help you figure this
out but I will definitely take a look.

-MC

On Mon, Dec 10, 2012 at 11:26 PM, Miha <miha at softnet.si> wrote:

>  Hi Michael,
>
> thanks for your reply.
>
> http://pastebin.freeswitch.com/20318
>
> In pastebin I post full debug log from FS.
>
> xxx.xxx.xxx.xxx is FS, yyy.yyy.yyy.yyy is SBC, aaa.aaa.aaa.aaa is SBC
> media.
>
> From phone (linksys) is called number 074901690, number is matched in
> dialplan, naptr look up is made and  on FS is checked if number is on FS,
> if not than number with 386+ nrn code (enum prefix) is send to sbc.
> Sbc send call to a PBX where call is pick up (IVR, call is set up with
> g711u). On other side (innovaphone pbx) ivr tells you which number to use
> so that you will be redirected to different group of people.
>
> I dial 1, which do in pbx 302 redirect and send phone call to outside
> (mobile carrier). When this is made pbx send new sdp invite to FS, to
> change codect to g711a, which FS refuse and send not acceptable.
>
> Michael I also send you on private trace which was mad od FS and on SBC. I
> can send it also to someone, but I will prefer not to post it to user group
> as recently we are dealing with a loot of sip attacks.
>
> BR,
>
> Miha
>
> Dne 12/10/2012 10:10 PM, pi?e Michael Collins:
>
> I highly doubt this is a bug in FreeSWITCH. It looks like you're having
> codec mismatch problems. I can see a call starting with just PCMU and also
> a call starting with just PCMA enabled. See lines 160 and 280, respectively.
>
> I doubt anyone can help without a full console debug log of all the calls
> involved. We also would need to see an explanation of what each call leg is
> supposed to be doing. Lastly it would be helpful to know where the pcap was
> taken from, i.e. from FS or the SBC so that we know the perspective of each
> point in the chart. I know you're trying to obfuscate away IP addresses,
> however in doing so you've also made it really hard for us to help you
> because we don't always know what is what.
>
> -MC
>
> On Mon, Dec 10, 2012 at 12:49 PM, Miha <miha at softnet.si> wrote:
>
>> Hi,
>>
>> can some pls help me:)
>>
>> Is this maybe a bug and I should post to jira?
>>
>> BR,
>> Miha
>>
>> On Mon, 10 Dec 2012 15:13:04 +0100
>>  Miha <miha at softnet.si> wrote:
>> > HI,
>> >
>> > here is a sip trace from FS.
>> >
>> > http://pastebin.freeswitch.org/20305
>> >
>> > xxx.xxx.xxx.xxx is FS, yyy.yyy.yyy.yyy is SBC,
>> > aaa.aaa.aaa.aaa SBC media :)
>> >
>> > thanks for help!
>> >
>> > Miha
>> >
>>  > Dne 12/10/2012 10:52 AM, pis(e Avi Marcus:
>> > >
>> > > Uploading the actual wirrshark file (don't attach) will
>> > likely help someone diagnose this easier.
>> > >
>> > > -Avi
>> > >
>> > >> On Dec 10, 2012 11:24 AM, "Miha" <miha at softnet.si
>>  > <mailto:miha at softnet.si>> wrote:
>> > >>
>> > >> Hi,
>> > >>
>> > >> Scenario:
>> > >>
>> > >> fs ----> SBC---> innovaphone pbx
>> > >>
>> > >> calls goes throught and media is ok (g711u). When on
>> > other side (innovaphone pbx) someone do a 302 redirect
>> > innovaphone sends a sdb request in which prepose g711u,
>> > FS trys with g711u and send back not accabtable and media
>> > does not work.
>> > >>
>> > >> In attachment I am sending jpg of wireshark.
>> > >>
>> > >> >From left to right: FS, SBC, SBC MEDIA (on different
>> > ip than sbc).
>> > >>
>> > >> How to deal with this?
>> > >>
>> > >> Thanks!
>> > >> Miha
>> > >>
>> > >>
>> >
>> _________________________________________________________________________
>> > >> 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
>> > >>
>> > >> 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
>> > >
>> > > FreeSWITCH-powered IP PBX: The CudaTel Communication
>> > Server
>> > > 
>> > >
>> > > 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Tue Dec 11 20:24:36 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Dec 2012 09:24:36 -0800
Subject: [Freeswitch-users] Wish Lists - 'tis the season
In-Reply-To: <371f01cdd77e$3972d310$ac587930$@bizfocused.com>
References: <31fb01cdd6f8$fa356850$eea038f0$@bizfocused.com>
	<A535DEBB-85AD-4C99-BA20-694038CCC7BB@endigotech.com>
	<371f01cdd77e$3972d310$ac587930$@bizfocused.com>
Message-ID: <CAKzWOxVnqQ3DP2SdRpC=cmY-d4NpQFqa6KYmixBMVqowt6j0dg@mail.gmail.com>

On Tue, Dec 11, 2012 at 1:02 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Oh CRAP!  I sent it to Brian West.  You Brian?s all look alike to me.
>

You sent it to that slacker West?!? Hahaha
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From miha at softnet.si  Tue Dec 11 21:33:19 2012
From: miha at softnet.si (Miha)
Date: Tue, 11 Dec 2012 19:33:19 +0100
Subject: [Freeswitch-users] media problem afet 302
In-Reply-To: <CAKzWOxUN2GBp=VXB+Oy+fJiWRBXkkgLG78r_s8HszbgCtUkujg@mail.gmail.com>
References: <50C5A99D.1030305@softnet.si>
	<CALNZuwmM4Tphy+B1MvKwmBD_csszGrBF_FknJQ-H7pmoDmykyA@mail.gmail.com>
	<50C5EDF0.5070805@softnet.si> <web-88303730@softnet.si>
	<CAKzWOxW-cZZcHvkUmXGWnZZCPr8C8Tzq=_pjFAeO5A80YHRfuQ@mail.gmail.com>
	<50C6E00D.7090603@softnet.si>
	<CAKzWOxUN2GBp=VXB+Oy+fJiWRBXkkgLG78r_s8HszbgCtUkujg@mail.gmail.com>
Message-ID: <web-88323817@softnet.si>

Hi Michael,

thank you very much for your help.

I manage to figure it out.

First I did <action application="export"
data="nolocal:sip_renegotiate_codec_on_reinvite=true" />

and this did not work. Then I did:

<action application="bridge"
data="{sip_renegotiate_codec_on_reinvite=true}sofia/export/......"
/>

and this happens to work. After 302 is made, PBX sends to
FS new sdp invite on g711a and fs change codec. I do not
get error codec any more.

As I read on wiki this nolocal or if I put variable in to
the bridge {} is should be the same but I guess in this
case is not the same.

http://wiki.freeswitch.org/wiki/Codec_negotiation#Early_Negotiation_parameters

Thanks Mihael again!!!

BR,
Miha

On Tue, 11 Dec 2012 09:23:14 -0800
 Michael Collins <msc at freeswitch.org> wrote:
> Miha,
> 
> Can you turn off sofia debugging. That's just too much
> noise and it doesn't
> help you. Create a new pastebin without the the sofia
> debug. Leave the sip
> trace on. ("sofia global siptrace on")
> 
> I'm probably not the person who is most qualified to help
> you figure this
> out but I will definitely take a look.
> 
> -MC
> 
> On Mon, Dec 10, 2012 at 11:26 PM, Miha <miha at softnet.si>
> wrote:
> 
> >  Hi Michael,
> >
> > thanks for your reply.
> >
> > http://pastebin.freeswitch.com/20318
> >
> > In pastebin I post full debug log from FS.
> >
> > xxx.xxx.xxx.xxx is FS, yyy.yyy.yyy.yyy is SBC,
> aaa.aaa.aaa.aaa is SBC
> > media.
> >
> > From phone (linksys) is called number 074901690, number
> is matched in
> > dialplan, naptr look up is made and  on FS is checked
> if number is on FS,
> > if not than number with 386+ nrn code (enum prefix) is
> send to sbc.
> > Sbc send call to a PBX where call is pick up (IVR, call
> is set up with
> > g711u). On other side (innovaphone pbx) ivr tells you
> which number to use
> > so that you will be redirected to different group of
> people.
> >
> > I dial 1, which do in pbx 302 redirect and send phone
> call to outside
> > (mobile carrier). When this is made pbx send new sdp
> invite to FS, to
> > change codect to g711a, which FS refuse and send not
> acceptable.
> >
> > Michael I also send you on private trace which was mad
> od FS and on SBC. I
> > can send it also to someone, but I will prefer not to
> post it to user group
> > as recently we are dealing with a loot of sip attacks.
> >
> > BR,
> >
> > Miha
> >
> > Dne 12/10/2012 10:10 PM, pi?e Michael Collins:
> >
> > I highly doubt this is a bug in FreeSWITCH. It looks
> like you're having
> > codec mismatch problems. I can see a call starting with
> just PCMU and also
> > a call starting with just PCMA enabled. See lines 160
> and 280, respectively.
> >
> > I doubt anyone can help without a full console debug
> log of all the calls
> > involved. We also would need to see an explanation of
> what each call leg is
> > supposed to be doing. Lastly it would be helpful to
> know where the pcap was
> > taken from, i.e. from FS or the SBC so that we know the
> perspective of each
> > point in the chart. I know you're trying to obfuscate
> away IP addresses,
> > however in doing so you've also made it really hard for
> us to help you
> > because we don't always know what is what.
> >
> > -MC
> >
> > On Mon, Dec 10, 2012 at 12:49 PM, Miha
> <miha at softnet.si> wrote:
> >
> >> Hi,
> >>
> >> can some pls help me:)
> >>
> >> Is this maybe a bug and I should post to jira?
> >>
> >> BR,
> >> Miha
> >>
> >> On Mon, 10 Dec 2012 15:13:04 +0100
> >>  Miha <miha at softnet.si> wrote:
> >> > HI,
> >> >
> >> > here is a sip trace from FS.
> >> >
> >> > http://pastebin.freeswitch.org/20305
> >> >
> >> > xxx.xxx.xxx.xxx is FS, yyy.yyy.yyy.yyy is SBC,
> >> > aaa.aaa.aaa.aaa SBC media :)
> >> >
> >> > thanks for help!
> >> >
> >> > Miha
> >> >
> >>  > Dne 12/10/2012 10:52 AM, pis(e Avi Marcus:
> >> > >
> >> > > Uploading the actual wirrshark file (don't attach)
> will
> >> > likely help someone diagnose this easier.
> >> > >
> >> > > -Avi
> >> > >
> >> > >> On Dec 10, 2012 11:24 AM, "Miha" <miha at softnet.si
> >>  > <mailto:miha at softnet.si>> wrote:
> >> > >>
> >> > >> Hi,
> >> > >>
> >> > >> Scenario:
> >> > >>
> >> > >> fs ----> SBC---> innovaphone pbx
> >> > >>
> >> > >> calls goes throught and media is ok (g711u). When
> on
> >> > other side (innovaphone pbx) someone do a 302
> redirect
> >> > innovaphone sends a sdb request in which prepose
> g711u,
> >> > FS trys with g711u and send back not accabtable and
> media
> >> > does not work.
> >> > >>
> >> > >> In attachment I am sending jpg of wireshark.
> >> > >>
> >> > >> >From left to right: FS, SBC, SBC MEDIA (on
> different
> >> > ip than sbc).
> >> > >>
> >> > >> How to deal with this?
> >> > >>
> >> > >> Thanks!
> >> > >> Miha
> >> > >>
> >> > >>
> >> >
> >>
>
_________________________________________________________________________
> >> > >> 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
> >> > >>
> >> > >> 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
> >> > >
> >> > > FreeSWITCH-powered IP PBX: The CudaTel
> Communication
> >> > Server
> >> > > 
> >> > >
> >> > > 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
> >>
> >> FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> >> 
> >>
> >> 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
> >>
> >
> >
> >
> > --
> > Michael S Collins
> > Twitter: @mercutioviz
> > http://www.FreeSWITCH.org
> > http://www.ClueCon.com
> > http://www.OSTAG.org
> >
> >
> >
> >
> >
>
_________________________________________________________________________
> > Professional FreeSWITCH Consulting
>
Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
> >
> > FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> >
> > Official FreeSWITCH
>
Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
> >
> > 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
> >
> > FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> > 
> >
> > 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
> >
> >
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org


From anthony.minessale at gmail.com  Tue Dec 11 21:37:49 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 11 Dec 2012 12:37:49 -0600
Subject: [Freeswitch-users] Dialplan for FS modems - odd behaviour or am
 I doing it wrong?
In-Reply-To: <50C76B34.6080007@integrafin.co.uk>
References: <50C76B34.6080007@integrafin.co.uk>
Message-ID: <CAKbxfG-3tqMdeoLuwwxjtNRM17itT9m_4rBwhzAhfXRUgPyOPw@mail.gmail.com>

hard to tell its not a complete log, is $1 not the same as what you
captured with the parens?



On Tue, Dec 11, 2012 at 11:19 AM, Alex Crow <acrow at integrafin.co.uk> wrote:

> Hi,
>
> Please see the below log snippet. I am accepting faxes from an ISDN
> gateway on sequential numbers and attempting to route them into distinct
> freeswitch modems by taking the last two digits of the called number
> thusly:
>
>     <condition field="destination_number"
> expression="^12([1-2][0-9]|3[0-6])$">
>          <!--<action application="set" data="ignore_early_media=true"/>-->
>          <action application="set" data="sip_ignore_reinvites=true"/>
>          <action application="set"
> data="absolute_codec_string='PCMA,PCMU'"/>
>          <action application="set" data="jitterbuffer_msec=600:600:60"/>
>          <action application="set" data="sip_jitter_buffer_plc=false"/>
>          <action application="bridge" data="modem/$1/a"/>
>     </condition>
>
> And then have Hylafax email them on depending on which modem they
> arrived on (via FaxDispatch).
>
> However as you can see from the below log snippets, it seems that FS is
> quite happy to change the modem device from what is actually delivered
> via the dialplan. Consequently a number of my faxes are now being routed
> to the wrong people.
>
> Is my bridge dest wrong? Should it be something like modem/$1/0 instead
> of "a"?
>
> Thanks
>
> Alex
>
> Dialplan: sofia/internal/anonymous at anonymous.invalid Action
> set(sip_ignore_reinvites=true)
> Dialplan: sofia/internal/anonymous at anonymous.invalid Action
> set(absolute_codec_string='PCMA,PCMU')
> Dialplan: sofia/internal/anonymous at anonymous.invalid Action
> set(jitterbuffer_msec=600:600:60)
> Dialplan: sofia/internal/anonymous at anonymous.invalid Action
> set(sip_jitter_buffer_plc=false)
> Dialplan: sofia/internal/anonymous at anonymous.invalid Action
> bridge(modem/13/a)
> 2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:167
> (sofia/internal/anonymous at anonymous.invalid) State Change CS_ROUTING ->
> CS_EXECUTE
> 2012-12-11 14:43:18.873181 [DEBUG] switch_core_session.c:1283 Send
> signal sofia/internal/anonymous at anonymous.invalid [BREAK]
> 2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:470
> (sofia/internal/anonymous at anonymous.invalid) State ROUTING going to sleep
> 2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/anonymous at anonymous.invalid) Running State Change
> CS_EXECUTE
> 2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:477
> (sofia/internal/anonymous at anonymous.invalid) State EXECUTE
> 2012-12-11 14:43:18.873181 [DEBUG] mod_sofia.c:242
> sofia/internal/anonymous at anonymous.invalid SOFIA EXECUTE
> 2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:209
> sofia/internal/anonymous at anonymous.invalid Standard EXECUTE
> EXECUTE sofia/internal/anonymous at anonymous.invalidset(call_direction=local)
> 2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344
> sofia/internal/anonymous at anonymous.invalid SET [call_direction]=[local]
> EXECUTE sofia/internal/anonymous at anonymous.invalid set(open=true)
> 2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344
> sofia/internal/anonymous at anonymous.invalid SET [open]=[true]
> EXECUTE sofia/internal/anonymous at anonymous.invalid
>
> hash(insert/192.168.20.245-spymap/anonymous/4e5c4699-aa22-4998-b7fa-b50a52c940fb)
> EXECUTE sofia/internal/anonymous at anonymous.invalid
> hash(insert/192.168.20.245-last_dial/anonymous/1213)
> EXECUTE sofia/internal/anonymous at anonymous.invalid
>
> hash(insert/192.168.20.245-last_dial/global/4e5c4699-aa22-4998-b7fa-b50a52c940fb)
> EXECUTE sofia/internal/anonymous at anonymous.invalid set(RFC2822_DATE=Tue,
> 11 Dec 2012 14:43:18 +0000)
> 2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344
> sofia/internal/anonymous at anonymous.invalid SET [RFC2822_DATE]=[Tue, 11
> Dec 2012 14:43:18 +0000]
> EXECUTE sofia/internal/anonymous at anonymous.invalid
> set(sip_ignore_reinvites=true)
> 2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344
> sofia/internal/anonymous at anonymous.invalid SET
> [sip_ignore_reinvites]=[true]
> EXECUTE sofia/internal/anonymous at anonymous.invalid
> set(absolute_codec_string='PCMA,PCMU')
> 2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344
> sofia/internal/anonymous at anonymous.invalid SET
> [absolute_codec_string]=['PCMA,PCMU']
> EXECUTE sofia/internal/anonymous at anonymous.invalid
> set(jitterbuffer_msec=600:600:60)
> 2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344
> sofia/internal/anonymous at anonymous.invalid SET
> [jitterbuffer_msec]=[600:600:60]
> EXECUTE sofia/internal/anonymous at anonymous.invalid
> set(sip_jitter_buffer_plc=false)
> 2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344
> sofia/internal/anonymous at anonymous.invalid SET
> [sip_jitter_buffer_plc]=[false]
> EXECUTE sofia/internal/anonymous at anonymous.invalid bridge(modem/13/a)
> 2012-12-11 14:43:18.873181 [DEBUG] switch_channel.c:1089
> sofia/internal/anonymous at anonymous.invalid EXPORTING[export_vars]
> [domain_name]=[192.168.20.245] to event
> 2012-12-11 14:43:18.873181 [DEBUG] switch_ivr_originate.c:2022 Parsing
> global variables
> 2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:1417 Modem
> /dev/FS/FS10 [ONHOOK] - Changing state to ACQUIRED
> 2012-12-11 14:43:18.873181 [NOTICE] switch_channel.c:968 New Channel
> modem/10/a [a202aef1-1628-45e8-b720-9f5368aa4565]
> 2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:769 modem/10/a
> setup codec L16/8000/20
> 2012-12-11 14:43:18.873181 [DEBUG] switch_channel.c:1135 EXPORT
> (export_vars) [rtp_autoflush_during_bridge]=[false]
> 2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:896 (modem/10/a)
> State Change CS_NEW -> CS_INIT
> 2012-12-11 14:43:18.873181 [DEBUG] switch_core_session.c:1283 Send
> signal modem/10/a [BREAK]
> 2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:581 modem/10/a
> CHANNEL KILL
> 2012-12-11 14:43:18.893118 [DEBUG] switch_core_state_machine.c:415
> (modem/10/a) Running State Change CS_INIT
> 2012-12-11 14:43:18.893118 [DEBUG] switch_core_state_machine.c:454
> (modem/10/a) State INIT
> 2012-12-11 14:43:18.893118 [DEBUG] mod_spandsp_modem.c:461 Modem
> /dev/FS/FS10 [ACQUIRED] - Changing state to RINGING
> 2012-12-11 14:43:18.893118 [DEBUG] mod_spandsp_modem.c:1131 Modem
> /dev/FS/FS10 [RINGING] - RNG 1
>
>
> And another one:
>
>
> EXECUTE sofia/internal/anonymous at anonymous.invalid bridge(modem/23/a)
> 2012-12-11 13:42:20.813107 [DEBUG] switch_channel.c:1089
> sofia/internal/anonymous at anonymous.invalid EXPORTING[export_vars]
> [domain_name]=[192.168.20.245] to event
> 2012-12-11 13:42:20.813107 [DEBUG] switch_ivr_originate.c:2022 Parsing
> global variables
> 2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:1417 Modem
> /dev/FS/FS26 [ONHOOK] - Changing state to ACQUIRED
> 2012-12-11 13:42:20.813107 [NOTICE] switch_channel.c:968 New Channel
> modem/26/a [32872ae9-6f78-45b0-af56-4b18c9de7160]
> 2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:769 modem/26/a
> setup codec L16/8000/20
> 2012-12-11 13:42:20.813107 [DEBUG] switch_channel.c:1135 EXPORT
> (export_vars) [rtp_autoflush_during_bridge]=[false]
> 2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:896 (modem/26/a)
> State Change CS_NEW -> CS_INIT
> 2012-12-11 13:42:20.813107 [DEBUG] switch_core_session.c:1283 Send
> signal modem/26/a [BREAK]
> 2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:581 modem/26/a
> CHANNEL KILL
> 2012-12-11 13:42:20.813107 [DEBUG] switch_core_state_machine.c:415
> (modem/26/a) Running State Change CS_INIT
> 2012-12-11 13:42:20.813107 [DEBUG] switch_core_state_machine.c:454
> (modem/26/a) State INIT
> 2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:461 Modem
> /dev/FS/FS26 [ACQUIRED] - Changing state to RINGING
> 2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:1131 Modem
> /dev/FS/FS26 [RINGING] - RNG 1
> 2012-12-11 13:42:20.833137 [DEBUG] mod_spandsp_modem.c:1038 Modem
> /dev/FS/FS26 [RINGING] - Answering
> 2012-12-11 13:42:20.833137 [DEBUG] mod_spandsp_modem.c:1040 Modem
> /dev/FS/FS26 [RINGING] - Changing state to ANSWERED
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Anthony Minessale II

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
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From fs-list at communicatefreely.net  Tue Dec 11 21:42:05 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Tue, 11 Dec 2012 13:42:05 -0500
Subject: [Freeswitch-users] git IPv6 down?
In-Reply-To: <1FFF97C269757C458224B7C895F35F151E3575@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F151E3575@cantor.std.visionutv.se>
Message-ID: <50C77E7D.1000600@communicatefreely.net>

Likewise - I get "connection refused" at the IP address 2606:d900:0:24:1024:ff:fe00:1234



Peter Olsson wrote:
> Hi all,
> 
> Ever since the DoS attack a couple of weeks ago, I can't access git over IPv6 anymore. Are there still issues that need to be resolved, or is it just me?
> 
> It fallbacks to IPv4, so it works just fine, I just wanted to let you know.
> 
> /Peter
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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



From kbdfck at gmail.com  Tue Dec 11 21:55:09 2012
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 11 Dec 2012 22:55:09 +0400
Subject: [Freeswitch-users] Mod_opal & PTLIB
In-Reply-To: <006001cdbf60$1c408b10$54c1a130$@com>
References: <006001cdbf60$1c408b10$54c1a130$@com>
Message-ID: <CA+GmxbZ79-_mfHJWJFd69VhkgRStdyBOOKvNECkmLcSOLOcCCg@mail.gmail.com>

There is some error in configure script maybe
It creates /lib/ptbuildopts.h (!) instead of ./include/ptbuildopts.h, so
defines don't get their values and build fails.

I'm now trying to get these ptlib / opal to work in order to build mod_opal

2012/11/10 Phil Quesinberry <philq at qsystemsengineering.com>

> **
>
> I?m trying to compile mod_opal here but I can?t get it to work.  I?ve
> tried the buildopal.sh script, the manual install, sacrificial offerings,
> etc. to no avail.  I was getting an error about PTLib being too old, so I
> grabbed PTLib 2.10.7 source here:
>
> *http://www.linuxfromscratch.org/blfs/view/svn/general/ptlib.html*<http://www.linuxfromscratch.org/blfs/view/svn/general/ptlib.html>
>
> Trying to compile it, I get a ?No operating system selected? error.  I?ve
> pasted in the make attempt below, Distro is CentOS 5.8.
>
> It looks like a lot of other folks have had/are having problems with
> mod_opal and ptlib, but I?m not seeing much in the way of solutions.  Has
> anyone else gotten past this problem?  Any pointers?  (no pun intended)
>
> [root at Tyrion ptlib]# make
>
> Setting default PTLIBDIR to /root/ptlib
>
> make[1]: Entering directory `/root/ptlib'
>
> make[2]: Entering directory `/root/ptlib'
>
> make[3]: Entering directory `/root/ptlib'
>
> make[3]: Leaving directory `/root/ptlib'
>
> make[2]: Leaving directory `/root/ptlib'
>
> set -e; if test -d /root/ptlib/src ; then make -C /root/ptlib/src
> optshared; fi;  if test -d /root/ptlib/plugins ; then make
> -C/root/ptlib/plugins optshared; fi;
>
> make[2]: Entering directory `/root/ptlib/src'
>
> make[3]: Entering directory `/root/ptlib/src'
>
> [CC] ptclib/psasl.cxx
>
> In file included from /root/ptlib/include/ptlib/object.h:44,
>
>                  from /root/ptlib/include/ptlib.h:47,
>
>                  from ptclib/psasl.cxx:35:
>
> /root/ptlib/include/ptlib/unix/ptlib/platform.h:555:2: error: #error No
> operating system selected.
>
> /root/ptlib/include/ptlib/mutex.h:109: error: ?PThreadIdentifier? does not
> name a type
>
> /root/ptlib/include/ptlib/thread.h:292: error: ?PThreadIdentifier? does
> not name a type
>
> /root/ptlib/include/ptlib/thread.h:293: error: ?PThreadIdentifier? does
> not name a type
>
> /root/ptlib/include/ptlib/thread.h:403: error: ?PThreadIdentifier? does
> not name a type
>
> /root/ptlib/include/ptlib/syncthrd.h:326: error: ?PThreadIdentifier? was
> not declared in this scope
>
> /root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 1 is
> invalid
>
> /root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 3 is
> invalid
>
> /root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 4 is
> invalid
>
> /root/ptlib/include/ptclib/cypher.h:382: error: ?PUInt32l? does not name a
> type
>
> make[3]: *** [/root/ptlib/lib_linux_x86_64/obj/psasl.o] Error 1
>
> make[3]: Leaving directory `/root/ptlib/src'
>
> make[2]: *** [optshared] Error 2
>
> make[2]: Leaving directory `/root/ptlib/src'
>
> make[1]: *** [optshared] Error 2
>
> make[1]: Leaving directory `/root/ptlib'
>
> make: *** [default] Error 2
>
> Many thanks,
>
> *******Phil Quesinberry*
>
> Q Systems Engineering, Inc.
>
> Electronic Controls and Embedded Systems Development
>
> (410) 969-8002
>
> *****http://www.qsystemsengineering.com*<http://www.qsystemsengineering.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 tnelson at rockbochs.com  Tue Dec 11 22:26:32 2012
From: tnelson at rockbochs.com (Tim Nelson)
Date: Tue, 11 Dec 2012 13:26:32 -0600 (CST)
Subject: [Freeswitch-users] Dialplan for FS modems - odd behaviour or am
 I	doing it wrong?
In-Reply-To: <50C76B34.6080007@integrafin.co.uk>
Message-ID: <7654262.147478.1355253992585.JavaMail.root@rockbochs.com>

----- Original Message -----
> 
> <snip>
> 
> And then have Hylafax email them on depending on which modem they
> arrived on (via FaxDispatch).
> 
> However as you can see from the below log snippets, it seems that FS
> is
> quite happy to change the modem device from what is actually
> delivered
> via the dialplan. Consequently a number of my faxes are now being
> routed
> to the wrong people.
> 

Slightly off topic to this list... but with Hylafax you could certainly also route based on the DID, not the modem. Check out the examples at hylafax.sourceforge.net relating to FaxDispatch.

--Tim


From msc at freeswitch.org  Tue Dec 11 22:32:08 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Dec 2012 11:32:08 -0800
Subject: [Freeswitch-users] Mod_opal & PTLIB
In-Reply-To: <CA+GmxbZ79-_mfHJWJFd69VhkgRStdyBOOKvNECkmLcSOLOcCCg@mail.gmail.com>
References: <006001cdbf60$1c408b10$54c1a130$@com>
	<CA+GmxbZ79-_mfHJWJFd69VhkgRStdyBOOKvNECkmLcSOLOcCCg@mail.gmail.com>
Message-ID: <CAKzWOxVh9SrS7-AMwh2_B9+dh=uhG+c6yS6XH9_YeAJAV_01eA@mail.gmail.com>

Can you modify the script, verify that it works for you, then open a jira
as a patch and attach your diff file? That would make our lives much
easier...

Thanks,
MC

On Tue, Dec 11, 2012 at 10:55 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> There is some error in configure script maybe
> It creates /lib/ptbuildopts.h (!) instead of ./include/ptbuildopts.h, so
> defines don't get their values and build fails.
>
> I'm now trying to get these ptlib / opal to work in order to build mod_opal
>
> 2012/11/10 Phil Quesinberry <philq at qsystemsengineering.com>
>
>> **
>>
>> I?m trying to compile mod_opal here but I can?t get it to work.  I?ve
>> tried the buildopal.sh script, the manual install, sacrificial
>> offerings, etc. to no avail.  I was getting an error about PTLib being
>> too old, so I grabbed PTLib 2.10.7 source here:
>>
>> *http://www.linuxfromscratch.org/blfs/view/svn/general/ptlib.html*<http://www.linuxfromscratch.org/blfs/view/svn/general/ptlib.html>
>>
>> Trying to compile it, I get a ?No operating system selected? error.  I?ve
>> pasted in the make attempt below, Distro is CentOS 5.8.
>>
>> It looks like a lot of other folks have had/are having problems with
>> mod_opal and ptlib, but I?m not seeing much in the way of solutions.  Has
>> anyone else gotten past this problem?  Any pointers?  (no pun intended)
>>
>> [root at Tyrion ptlib]# make
>>
>> Setting default PTLIBDIR to /root/ptlib
>>
>> make[1]: Entering directory `/root/ptlib'
>>
>> make[2]: Entering directory `/root/ptlib'
>>
>> make[3]: Entering directory `/root/ptlib'
>>
>> make[3]: Leaving directory `/root/ptlib'
>>
>> make[2]: Leaving directory `/root/ptlib'
>>
>> set -e; if test -d /root/ptlib/src ; then make -C /root/ptlib/src
>> optshared; fi;  if test -d /root/ptlib/plugins ; then make
>> -C/root/ptlib/plugins optshared; fi;
>>
>> make[2]: Entering directory `/root/ptlib/src'
>>
>> make[3]: Entering directory `/root/ptlib/src'
>>
>> [CC] ptclib/psasl.cxx
>>
>> In file included from /root/ptlib/include/ptlib/object.h:44,
>>
>>                  from /root/ptlib/include/ptlib.h:47,
>>
>>                  from ptclib/psasl.cxx:35:
>>
>> /root/ptlib/include/ptlib/unix/ptlib/platform.h:555:2: error: #error No
>> operating system selected.
>>
>> /root/ptlib/include/ptlib/mutex.h:109: error: ?PThreadIdentifier? does
>> not name a type
>>
>> /root/ptlib/include/ptlib/thread.h:292: error: ?PThreadIdentifier? does
>> not name a type
>>
>> /root/ptlib/include/ptlib/thread.h:293: error: ?PThreadIdentifier? does
>> not name a type
>>
>> /root/ptlib/include/ptlib/thread.h:403: error: ?PThreadIdentifier? does
>> not name a type
>>
>> /root/ptlib/include/ptlib/syncthrd.h:326: error: ?PThreadIdentifier? was
>> not declared in this scope
>>
>> /root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 1 is
>> invalid
>>
>> /root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 3 is
>> invalid
>>
>> /root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 4 is
>> invalid
>>
>> /root/ptlib/include/ptclib/cypher.h:382: error: ?PUInt32l? does not name
>> a type
>>
>> make[3]: *** [/root/ptlib/lib_linux_x86_64/obj/psasl.o] Error 1
>>
>> make[3]: Leaving directory `/root/ptlib/src'
>>
>> make[2]: *** [optshared] Error 2
>>
>> make[2]: Leaving directory `/root/ptlib/src'
>>
>> make[1]: *** [optshared] Error 2
>>
>> make[1]: Leaving directory `/root/ptlib'
>>
>> make: *** [default] Error 2
>>
>> Many thanks,
>>
>> *******Phil Quesinberry*
>>
>> Q Systems Engineering, Inc.
>>
>> Electronic Controls and Embedded Systems Development
>>
>> (410) 969-8002
>>
>> *****http://www.qsystemsengineering.com*<http://www.qsystemsengineering.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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From acrow at integrafin.co.uk  Tue Dec 11 22:41:56 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Tue, 11 Dec 2012 19:41:56 +0000
Subject: [Freeswitch-users] Dialplan for FS modems - odd behaviour or am
 I doing it wrong?
In-Reply-To: <CAKbxfG-3tqMdeoLuwwxjtNRM17itT9m_4rBwhzAhfXRUgPyOPw@mail.gmail.com>
References: <50C76B34.6080007@integrafin.co.uk>
	<CAKbxfG-3tqMdeoLuwwxjtNRM17itT9m_4rBwhzAhfXRUgPyOPw@mail.gmail.com>
Message-ID: <50C78C84.8080309@integrafin.co.uk>

Anthony,

Thanks for replying so fast.

It should be enough log I think - you can see that the dialplan first 
goes to execute:

Dialplan: sofia/internal/anonymous at anonymous.invalid Action
bridge(modem/13/a)

*but* for some reasons it ends up being answered by /dev/FS/FS10 (I just 
patched one line in the source to make it create the links in a subdir 
so the permissions could be fixed for running as non-root user).

2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:1417 Modem
/dev/FS/FS10 [ONHOOK] - Changing state to ACQUIRED

And then for some bizarre reason FS continues as if we'd bridged to 
modem/10/whatever:

2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:769 modem/10/a
setup codec L16/8000/20

These are all definitely executed from the same extension in the 
dialplan. I've seen about 7 instances out of about 120 inbound faxes 
over the last few days. I've reverted to t38_gatewaying into t38modem 
endpoints for now as it seems to obey normal DP rules.

PS.

Perhaps related to this issue (think the code is in mod_spandsp) is that 
I've also found that the version of FS I compiled from GIT stable (? - 
not head at least - a bit confused about the new structure) a couple of 
days ago won't let me use t38_passthru with t38modem any more - even 
though it works reasonably well with and SPA3102. I am using the same 
binaries of t38modem I used with an older release which worked 
perfectly, same debian server, same kernel. The only thing I can tell 
you about the previous version that worked is that I built .debs from 
git on April 9 this year - I don't have the git downloaded tree any more 
to give you the revision (unless you can tell me another way of finding 
it without having the source tree). Luckily using proxy_media provides a 
fallback here, but I'd prefer to use the preferred method.

Also I've tried to use the FS modems for outbound faxing but I have a 
lot of problems on sessions with ECM where the destination sees 
increasingly more missing frames and tries to retrain the sender to a 
lower speed, and the lost frames get worse and worse. Researching on 
Hylafax mailing lists seems to suggest this means the sending modem does 
not do flow control properly and so the sending Hylafax does not know to 
stop pushing data at the modem. Consequently as the receiver asks to 
drop the speed the problem gets worse - by the time we get to 4800bps 
not even a single frame makes it through! The only inbound problem is 
what I describe above.

Without ECM an outbound fax completes but on the RX side there are large 
chunks of the page missing even though Hylafax on the sending side 
reports send OK. I guess that's why you should never disable ECM as 
Brian said.

Happy to open Jira for these last two but there is hardly anything in 
the FS log that looked different, just ended up with 100% hangups in the 
first place and sporadic faxing out in the second case.

Best regards

Alex


On 11/12/12 18:37, Anthony Minessale wrote:
> hard to tell its not a complete log, is $1 not the same as what you 
> captured with the parens?
>
>
>
> On Tue, Dec 11, 2012 at 11:19 AM, Alex Crow <acrow at integrafin.co.uk 
> <mailto:acrow at integrafin.co.uk>> wrote:
>
>     Hi,
>
>     Please see the below log snippet. I am accepting faxes from an ISDN
>     gateway on sequential numbers and attempting to route them into
>     distinct
>     freeswitch modems by taking the last two digits of the called
>     number thusly:
>
>         <condition field="destination_number"
>     expression="^12([1-2][0-9]|3[0-6])$">
>              <!--<action application="set"
>     data="ignore_early_media=true"/>-->
>              <action application="set" data="sip_ignore_reinvites=true"/>
>              <action application="set"
>     data="absolute_codec_string='PCMA,PCMU'"/>
>              <action application="set"
>     data="jitterbuffer_msec=600:600:60"/>
>              <action application="set"
>     data="sip_jitter_buffer_plc=false"/>
>              <action application="bridge" data="modem/$1/a"/>
>         </condition>
>
>     And then have Hylafax email them on depending on which modem they
>     arrived on (via FaxDispatch).
>
>     However as you can see from the below log snippets, it seems that
>     FS is
>     quite happy to change the modem device from what is actually delivered
>     via the dialplan. Consequently a number of my faxes are now being
>     routed
>     to the wrong people.
>
>     Is my bridge dest wrong? Should it be something like modem/$1/0
>     instead
>     of "a"?
>
>     Thanks
>
>     Alex
>
>     Dialplan: sofia/internal/anonymous at anonymous.invalid Action
>     set(sip_ignore_reinvites=true)
>     Dialplan: sofia/internal/anonymous at anonymous.invalid Action
>     set(absolute_codec_string='PCMA,PCMU')
>     Dialplan: sofia/internal/anonymous at anonymous.invalid Action
>     set(jitterbuffer_msec=600:600:60)
>     Dialplan: sofia/internal/anonymous at anonymous.invalid Action
>     set(sip_jitter_buffer_plc=false)
>     Dialplan: sofia/internal/anonymous at anonymous.invalid Action
>     bridge(modem/13/a)
>     2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:167
>     (sofia/internal/anonymous at anonymous.invalid) State Change
>     CS_ROUTING ->
>     CS_EXECUTE
>     2012-12-11 14:43:18.873181 [DEBUG] switch_core_session.c:1283 Send
>     signal sofia/internal/anonymous at anonymous.invalid [BREAK]
>     2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:470
>     (sofia/internal/anonymous at anonymous.invalid) State ROUTING going
>     to sleep
>     2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:415
>     (sofia/internal/anonymous at anonymous.invalid) Running State Change
>     CS_EXECUTE
>     2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:477
>     (sofia/internal/anonymous at anonymous.invalid) State EXECUTE
>     2012-12-11 14:43:18.873181 [DEBUG] mod_sofia.c:242
>     sofia/internal/anonymous at anonymous.invalid SOFIA EXECUTE
>     2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:209
>     sofia/internal/anonymous at anonymous.invalid Standard EXECUTE
>     EXECUTE sofia/internal/anonymous at anonymous.invalid
>     set(call_direction=local)
>     2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344
>     sofia/internal/anonymous at anonymous.invalid SET
>     [call_direction]=[local]
>     EXECUTE sofia/internal/anonymous at anonymous.invalid set(open=true)
>     2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344
>     sofia/internal/anonymous at anonymous.invalid SET [open]=[true]
>     EXECUTE sofia/internal/anonymous at anonymous.invalid
>     hash(insert/192.168.20.245-spymap/anonymous/4e5c4699-aa22-4998-b7fa-b50a52c940fb)
>     EXECUTE sofia/internal/anonymous at anonymous.invalid
>     hash(insert/192.168.20.245-last_dial/anonymous/1213)
>     EXECUTE sofia/internal/anonymous at anonymous.invalid
>     hash(insert/192.168.20.245-last_dial/global/4e5c4699-aa22-4998-b7fa-b50a52c940fb)
>     EXECUTE sofia/internal/anonymous at anonymous.invalid
>     set(RFC2822_DATE=Tue,
>     11 Dec 2012 14:43:18 +0000)
>     2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344
>     sofia/internal/anonymous at anonymous.invalid SET [RFC2822_DATE]=[Tue, 11
>     Dec 2012 14:43:18 +0000]
>     EXECUTE sofia/internal/anonymous at anonymous.invalid
>     set(sip_ignore_reinvites=true)
>     2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344
>     sofia/internal/anonymous at anonymous.invalid SET
>     [sip_ignore_reinvites]=[true]
>     EXECUTE sofia/internal/anonymous at anonymous.invalid
>     set(absolute_codec_string='PCMA,PCMU')
>     2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344
>     sofia/internal/anonymous at anonymous.invalid SET
>     [absolute_codec_string]=['PCMA,PCMU']
>     EXECUTE sofia/internal/anonymous at anonymous.invalid
>     set(jitterbuffer_msec=600:600:60)
>     2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344
>     sofia/internal/anonymous at anonymous.invalid SET
>     [jitterbuffer_msec]=[600:600:60]
>     EXECUTE sofia/internal/anonymous at anonymous.invalid
>     set(sip_jitter_buffer_plc=false)
>     2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344
>     sofia/internal/anonymous at anonymous.invalid SET
>     [sip_jitter_buffer_plc]=[false]
>     EXECUTE sofia/internal/anonymous at anonymous.invalid bridge(modem/13/a)
>     2012-12-11 14:43:18.873181 [DEBUG] switch_channel.c:1089
>     sofia/internal/anonymous at anonymous.invalid EXPORTING[export_vars]
>     [domain_name]=[192.168.20.245] to event
>     2012-12-11 14:43:18.873181 [DEBUG] switch_ivr_originate.c:2022 Parsing
>     global variables
>     2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:1417 Modem
>     /dev/FS/FS10 [ONHOOK] - Changing state to ACQUIRED
>     2012-12-11 14:43:18.873181 [NOTICE] switch_channel.c:968 New Channel
>     modem/10/a [a202aef1-1628-45e8-b720-9f5368aa4565]
>     2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:769 modem/10/a
>     setup codec L16/8000/20
>     2012-12-11 14:43:18.873181 [DEBUG] switch_channel.c:1135 EXPORT
>     (export_vars) [rtp_autoflush_during_bridge]=[false]
>     2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:896
>     (modem/10/a)
>     State Change CS_NEW -> CS_INIT
>     2012-12-11 14:43:18.873181 [DEBUG] switch_core_session.c:1283 Send
>     signal modem/10/a [BREAK]
>     2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:581 modem/10/a
>     CHANNEL KILL
>     2012-12-11 14:43:18.893118 [DEBUG] switch_core_state_machine.c:415
>     (modem/10/a) Running State Change CS_INIT
>     2012-12-11 14:43:18.893118 [DEBUG] switch_core_state_machine.c:454
>     (modem/10/a) State INIT
>     2012-12-11 14:43:18.893118 [DEBUG] mod_spandsp_modem.c:461 Modem
>     /dev/FS/FS10 [ACQUIRED] - Changing state to RINGING
>     2012-12-11 14:43:18.893118 [DEBUG] mod_spandsp_modem.c:1131 Modem
>     /dev/FS/FS10 [RINGING] - RNG 1
>
>
>     And another one:
>
>
>     EXECUTE sofia/internal/anonymous at anonymous.invalid bridge(modem/23/a)
>     2012-12-11 13:42:20.813107 [DEBUG] switch_channel.c:1089
>     sofia/internal/anonymous at anonymous.invalid EXPORTING[export_vars]
>     [domain_name]=[192.168.20.245] to event
>     2012-12-11 13:42:20.813107 [DEBUG] switch_ivr_originate.c:2022 Parsing
>     global variables
>     2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:1417 Modem
>     /dev/FS/FS26 [ONHOOK] - Changing state to ACQUIRED
>     2012-12-11 13:42:20.813107 [NOTICE] switch_channel.c:968 New Channel
>     modem/26/a [32872ae9-6f78-45b0-af56-4b18c9de7160]
>     2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:769 modem/26/a
>     setup codec L16/8000/20
>     2012-12-11 13:42:20.813107 [DEBUG] switch_channel.c:1135 EXPORT
>     (export_vars) [rtp_autoflush_during_bridge]=[false]
>     2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:896
>     (modem/26/a)
>     State Change CS_NEW -> CS_INIT
>     2012-12-11 13:42:20.813107 [DEBUG] switch_core_session.c:1283 Send
>     signal modem/26/a [BREAK]
>     2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:581 modem/26/a
>     CHANNEL KILL
>     2012-12-11 13:42:20.813107 [DEBUG] switch_core_state_machine.c:415
>     (modem/26/a) Running State Change CS_INIT
>     2012-12-11 13:42:20.813107 [DEBUG] switch_core_state_machine.c:454
>     (modem/26/a) State INIT
>     2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:461 Modem
>     /dev/FS/FS26 [ACQUIRED] - Changing state to RINGING
>     2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:1131 Modem
>     /dev/FS/FS26 [RINGING] - RNG 1
>     2012-12-11 13:42:20.833137 [DEBUG] mod_spandsp_modem.c:1038 Modem
>     /dev/FS/FS26 [RINGING] - Answering
>     2012-12-11 13:42:20.833137 [DEBUG] mod_spandsp_modem.c:1040 Modem
>     /dev/FS/FS26 [RINGING] - Changing state to ANSWERED
>
>     _________________________________________________________________________
>     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
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> -- 
> This message has been scanned for viruses and
> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is
> believed to be clean.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 acrow at integrafin.co.uk  Tue Dec 11 22:43:05 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Tue, 11 Dec 2012 19:43:05 +0000
Subject: [Freeswitch-users] Dialplan for FS modems - odd behaviour or am
 I	doing it wrong?
In-Reply-To: <7654262.147478.1355253992585.JavaMail.root@rockbochs.com>
References: <7654262.147478.1355253992585.JavaMail.root@rockbochs.com>
Message-ID: <50C78CC9.5080901@integrafin.co.uk>


> Slightly off topic to this list... but with Hylafax you could certainly also route based on the DID, not the modem. Check out the examples at hylafax.sourceforge.net relating to FaxDispatch.
>
> --Tim
>
>

Tim,

I thought about that but I don't actually see the DID logged on calls 
from the Freeswitch modems.

Cheers

Alex


From anthony.minessale at gmail.com  Tue Dec 11 23:00:46 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 11 Dec 2012 14:00:46 -0600
Subject: [Freeswitch-users] Dialplan for FS modems - odd behaviour or am
 I doing it wrong?
In-Reply-To: <50C78CC9.5080901@integrafin.co.uk>
References: <7654262.147478.1355253992585.JavaMail.root@rockbochs.com>
	<50C78CC9.5080901@integrafin.co.uk>
Message-ID: <CAKbxfG-v1JLnCGvvNCs_H5CFDk+=gq7+QzD1aiyCvQZm22QGKg@mail.gmail.com>

Looking again, I'm not sure what the modem/10/a is supposed to be

There is a such thing as modem/a/<number> but

if you are looking to bridge to slot 23 try

modem/23/${destination_number}

if you are getting a different slot maybe you simlinks are messed up, you
need to make sure they are recreated whenever you restart so maybe that
patch you are using has an issue.



BTW I would not have asked to see the real log if I didn't mean it.
I generally dislike pre-chosen log snippets, you have no idea what I may be
looking for.







On Tue, Dec 11, 2012 at 1:43 PM, Alex Crow <acrow at integrafin.co.uk> wrote:

>
> > Slightly off topic to this list... but with Hylafax you could certainly
> also route based on the DID, not the modem. Check out the examples at
> hylafax.sourceforge.net relating to FaxDispatch.
> >
> > --Tim
> >
> >
>
> Tim,
>
> I thought about that but I don't actually see the DID logged on calls
> from the Freeswitch modems.
>
> Cheers
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
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From acrow at integrafin.co.uk  Tue Dec 11 23:16:43 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Tue, 11 Dec 2012 20:16:43 +0000
Subject: [Freeswitch-users] Dialplan for FS modems - odd behaviour or am
 I doing it wrong?
In-Reply-To: <CAKbxfG-v1JLnCGvvNCs_H5CFDk+=gq7+QzD1aiyCvQZm22QGKg@mail.gmail.com>
References: <7654262.147478.1355253992585.JavaMail.root@rockbochs.com>
	<50C78CC9.5080901@integrafin.co.uk>
	<CAKbxfG-v1JLnCGvvNCs_H5CFDk+=gq7+QzD1aiyCvQZm22QGKg@mail.gmail.com>
Message-ID: <50C794AB.4060904@integrafin.co.uk>

On 11/12/12 20:00, Anthony Minessale wrote:
> Looking again, I'm not sure what the modem/10/a is supposed to be
>
> There is a such thing as modem/a/<number> but
>
> if you are looking to bridge to slot 23 try
>
> modem/23/${destination_number}
>
> if you are getting a different slot maybe you simlinks are messed up, 
> you need to make sure they are recreated whenever you restart so maybe 
> that patch you are using has an issue.
>
>
>

Hi,

I will try to get full logs to you when I can, possibly tomorrow as I 
have a stinking cold today. However I can tell you my symlinks are fine 
- and 90% of the time the inbound fax gets to the right modem. Just that 
every so often FS seems to send it to the wrong one.

On calls that are coming *in* to a modem device, I don't see how there 
even can be a destination number. It's just a modem surely - the 
destination is simply a device on which Hylafax is listening. I can 
completely understand how when you call out through the virtual modem 
you need to supply a number, but on inbound it doesn't make any sense to 
me, we're not routing into a SIP endpoint but a virtual device.

Looking at the source code told me to use modem/slot/<something>, but 
for inbound calls I don't know what <something> should be.

If I have got something topsy-turvy please enlighten me, I'm quite happy 
to be corrected!

Cheers

Alex


From anthony.minessale at gmail.com  Wed Dec 12 00:36:22 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 11 Dec 2012 15:36:22 -0600
Subject: [Freeswitch-users] Dialplan for FS modems - odd behaviour or am
 I doing it wrong?
In-Reply-To: <50C794AB.4060904@integrafin.co.uk>
References: <7654262.147478.1355253992585.JavaMail.root@rockbochs.com>
	<50C78CC9.5080901@integrafin.co.uk>
	<CAKbxfG-v1JLnCGvvNCs_H5CFDk+=gq7+QzD1aiyCvQZm22QGKg@mail.gmail.com>
	<50C794AB.4060904@integrafin.co.uk>
Message-ID: <CAKbxfG8MqT96cC4L9GYexUwyCOH7VxJ-HfP8R-+miK6mnyt=1Q@mail.gmail.com>

The <something> is a number saved in the variable modem_digits on the
channel to do whatever you want with in the cdr etc....Its not passed to
the the modem device.

This is an obvious case for a JIRA.... I need to come up with a way to
explain the order of magnitude difference there is trying to help someone
over the mailing list vs JIRA when you have to start providing patches etc.

If your would take this patch and run it to get more details and open a
JIRA and post the results I would appreciate it.





On Tue, Dec 11, 2012 at 2:16 PM, Alex Crow <acrow at integrafin.co.uk> wrote:

> On 11/12/12 20:00, Anthony Minessale wrote:
> > Looking again, I'm not sure what the modem/10/a is supposed to be
> >
> > There is a such thing as modem/a/<number> but
> >
> > if you are looking to bridge to slot 23 try
> >
> > modem/23/${destination_number}
> >
> > if you are getting a different slot maybe you simlinks are messed up,
> > you need to make sure they are recreated whenever you restart so maybe
> > that patch you are using has an issue.
> >
> >
> >
>
> Hi,
>
> I will try to get full logs to you when I can, possibly tomorrow as I
> have a stinking cold today. However I can tell you my symlinks are fine
> - and 90% of the time the inbound fax gets to the right modem. Just that
> every so often FS seems to send it to the wrong one.
>
> On calls that are coming *in* to a modem device, I don't see how there
> even can be a destination number. It's just a modem surely - the
> destination is simply a device on which Hylafax is listening. I can
> completely understand how when you call out through the virtual modem
> you need to supply a number, but on inbound it doesn't make any sense to
> me, we're not routing into a SIP endpoint but a virtual device.
>
> Looking at the source code told me to use modem/slot/<something>, but
> for inbound calls I don't know what <something> should be.
>
> If I have got something topsy-turvy please enlighten me, I'm quite happy
> to be corrected!
>
> Cheers
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
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From lists at kavun.ch  Wed Dec 12 00:39:20 2012
From: lists at kavun.ch (Emrah)
Date: Tue, 11 Dec 2012 16:39:20 -0500
Subject: [Freeswitch-users] Wanted Perl Programmer to help with a little
	Project for the Visually Impaired FreeSWITCH users.
In-Reply-To: <CCEBCB26.61D69%krice@freeswitch.org>
References: <CCEBCB26.61D69%krice@freeswitch.org>
Message-ID: <B08D4477-20BB-4F49-B18C-BCC1372D2BFE@kavun.ch>

People like you make all the difference in the world. Congrats for the initiative.
I myself cannot use Jira on Mac OS X with VoiceOver and would love an alternative. I have quite a few bugs queued up for the day I can file them.

Good luck and thanks!
Ps: if Jira is skinable, maybe it's just a question of using a less dynamic skin?
On Dec 10, 2012, at 6:23 PM, Ken Rice <krice at freeswitch.org> wrote:

> As many of you know we have a pretty large number of FreeSWITCH users that are visually impaired (from significant vision loss to right out blind).
> 
> Most of these users use screen readers to help them, and in some cases they use other technologies like Brail output devices. However, these things only go so far.
> 
> One of the issues they have is with Jira, and its lack of screen reader ?friendlyness?... I think I have a solution for this... If you are a reasonably good perl programmer and want to help me out on a small project drop me an email offlist. I have an idea that should make it much easier for this group of users.
> 
> K
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From acrow at integrafin.co.uk  Wed Dec 12 00:41:21 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Tue, 11 Dec 2012 21:41:21 +0000
Subject: [Freeswitch-users] Dialplan for FS modems - odd behaviour or am
 I doing it wrong?
In-Reply-To: <50C794AB.4060904@integrafin.co.uk>
References: <7654262.147478.1355253992585.JavaMail.root@rockbochs.com>
	<50C78CC9.5080901@integrafin.co.uk>
	<CAKbxfG-v1JLnCGvvNCs_H5CFDk+=gq7+QzD1aiyCvQZm22QGKg@mail.gmail.com>
	<50C794AB.4060904@integrafin.co.uk>
Message-ID: <50C7A881.5010400@integrafin.co.uk>

 >if you are getting a different slot maybe you simlinks are messed up, 
you need to make sure they are recreated whenever you restart so maybe 
that patch you are using has an issue.

Anthony,

Looking at the above again gave me a thought. In my 
/etc/init.d/freeswitch script I'd added a line to create /dev/FS and 
chown it to the freeswitch user. However perhaps it is significant that 
I didn't delete the previous /dev/FS/* links before so the old ones were 
left in place.

I have added a line to delete all of /dev/FS before creating the folder 
again and starting freeswitch. Will give feedback if it helps.

Will wikify if it proves successful and send you logs for the other two 
problems.

Thanks

Alex


From lists at kavun.ch  Wed Dec 12 01:02:09 2012
From: lists at kavun.ch (Emrah)
Date: Tue, 11 Dec 2012 17:02:09 -0500
Subject: [Freeswitch-users] ILBC takes precedence regardless of the codec
	prefs
Message-ID: <D37F8F9F-301A-4B29-9147-7C5E3E54C9DF@kavun.ch>

Hi all,

I am using FreeSWITCH Version 1.2.5.3+git~20121206T050429Z~91eef34d5c (git 91eef34 2012-12-06 05:04:29Z).
When mod_ilbc is loaded, ILBC is used in any session where the client offers it, regardless of my codec preferences. This gets even stranger when you look at the SIP trace? FS doesn't offer ILBC. This is brand new, I wanted to enable ILBC just in case, and found out that I prefer it to G729.

Here's some output of a trace:
Client:
v=0.
o=- 3564251257 3564251257 IN IP4 10.0.0.102.
s=pjmedia.
c=IN IP4 10.0.0.102.
t=0 0.
a=X-nat:0.
m=audio 4000 RTP/AVP 103 102 104 109 3 0 8 9 101.
a=rtcp:4001 IN IP4 10.0.1.132.
a=rtpmap:103 speex/16000.
a=rtpmap:102 speex/8000.
a=rtpmap:104 speex/32000.
a=rtpmap:109 iLBC/8000.
a=fmtp:109 mode=30.
a=rtpmap:3 GSM/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:9 G722/8000.
a=sendrecv.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

FS:
v=0.
o=FreeSWITCH 1355242302 1355242303 IN IP4 1.2.3.4.
s=FreeSWITCH.
c=IN IP4 1.2.3.4.
t=0 0.
m=audio 20042 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.


Your input is greatly appreciated, as always.

All the best,
Emrah

From andrew at cassidywebservices.co.uk  Wed Dec 12 01:17:53 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Tue, 11 Dec 2012 22:17:53 +0000
Subject: [Freeswitch-users] Wanted Perl Programmer to help with a little
 Project for the Visually Impaired FreeSWITCH users.
In-Reply-To: <B08D4477-20BB-4F49-B18C-BCC1372D2BFE@kavun.ch>
References: <CCEBCB26.61D69%krice@freeswitch.org>
	<B08D4477-20BB-4F49-B18C-BCC1372D2BFE@kavun.ch>
Message-ID: <CAMbW9oKpD5F3U+y_9r44VzRBNuYBWMBQJUJA5MgVtHUTSz=-ZQ@mail.gmail.com>

As I mentioned in a previous thread, my wife works for a local blindness
charity, and she'll be following this too.

On 11 December 2012 21:39, Emrah <lists at kavun.ch> wrote:

> People like you make all the difference in the world. Congrats for the
> initiative.
> I myself cannot use Jira on Mac OS X with VoiceOver and would love an
> alternative. I have quite a few bugs queued up for the day I can file them.
>
> Good luck and thanks!
> Ps: if Jira is skinable, maybe it's just a question of using a less
> dynamic skin?
> On Dec 10, 2012, at 6:23 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> > As many of you know we have a pretty large number of FreeSWITCH users
> that are visually impaired (from significant vision loss to right out
> blind).
> >
> > Most of these users use screen readers to help them, and in some cases
> they use other technologies like Brail output devices. However, these
> things only go so far.
> >
> > One of the issues they have is with Jira, and its lack of screen reader
> ?friendlyness?... I think I have a solution for this... If you are a
> reasonably good perl programmer and want to help me out on a small project
> drop me an email offlist. I have an idea that should make it much easier
> for this group of users.
> >
> > K
> >
> > --
> > Ken
> > http://www.FreeSWITCH.org
> > http://www.ClueCon.com
> > http://www.OSTAG.org
> > irc.freenode.net #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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
*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 lists at kavun.ch  Wed Dec 12 03:17:07 2012
From: lists at kavun.ch (Emrah)
Date: Tue, 11 Dec 2012 19:17:07 -0500
Subject: [Freeswitch-users] mod_perl - Can't call method "serialize" on an
	undefined value
Message-ID: <ADD8CA27-CB8D-4D61-BE4F-F03DFB354746@kavun.ch>

Hi there,

I am trying to list my env variables and it fails with "Can't call method "serialize" on an undefined value".
Something like $env->getHeader('uuid'); fails as well.
Everything else seems to be working fine. Mod_perl is loaded.

Google pointed me out to this post: http://lists.freeswitch.org/pipermail/freeswitch-users/2008-November/036404.html. No solution there? 

Any idea why this is happening? 

Thanks

From francis.joanis at gmail.com  Wed Dec 12 03:26:14 2012
From: francis.joanis at gmail.com (Francis Joanis)
Date: Tue, 11 Dec 2012 19:26:14 -0500
Subject: [Freeswitch-users] RFC 4579 and FreeSWITCH
In-Reply-To: <CAPoB04w66efnSrMs9822K+XuFH5Ju3kK-6Ev25HL76Axzj-tHw@mail.gmail.com>
References: <CAPoB04xa1SeKRzi=ZnqUKRHLUk+6Paz7B2nDKktYZOYQ6ZieWw@mail.gmail.com>
	<CAPoB04w66efnSrMs9822K+XuFH5Ju3kK-6Ev25HL76Axzj-tHw@mail.gmail.com>
Message-ID: <CAPoB04zF0b5_ff7-SQ4QBe7heahDPEdpo3DEgU6+0njrg6f_3g@mail.gmail.com>

Hi,

On Fri, Dec 7, 2012 at 2:32 PM, Francis Joanis <francis.joanis at gmail.com>wrote:

> Hi again,
>
> On Fri, Dec 7, 2012 at 1:54 PM, Francis Joanis <francis.joanis at gmail.com>wrote:
>
>> Hi guys,
>>
>> I started playing with FreeSWITCH to make conferences and I was curious
>> about the support for RFC 4579 (Call Control - Conferencing for User
>> Agents).
>>
>> I was able to have FreeSWITCH return a ;isfocus parameter in the Contact
>> (from the 200 OK) and I tried to add a new participant to the conference
>> using a REFER (see RFC 4579 section 5.5), which generated the following log:
>>
>> [ERR] sofia.c:7095 Cannot Blind Transfer 1 Legged calls
>>
>> I think RFC 4579 is mentioned on the FS wiki as a "supported" RFC but is
>> this call flow currently supported?
>>
>> Cheers and thanks for the awesome work in FS,
>> Francis
>>
>
> After digging through the code I realized my test is probably invalid
> since I was reusing the same SIP dialog for the REFER as the one that
> already existed for the existing call. I'll retest with a true out of
> dialog REFER and post my results back...
>
> Thanks,
> Francis
>

That was it: it works with an out of dialog REFER. I also had to add
"rfc-4579" to the conference-flags.

Francis
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From brian at freeswitch.org  Wed Dec 12 06:46:53 2012
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Dec 2012 21:46:53 -0600
Subject: [Freeswitch-users] Join the G+ Group!
Message-ID: <BFBBA7B3-26DE-41B3-BF86-76550AEE6BBF@freeswitch.org>

https://plus.google.com/u/0/communities/110013892794022037793?cfem=1




https://www.facebook.com/groups/2344193362/?fref=ts



Come one come all!


--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531

Twitter: @FreeSWITCH_Wire
http://freeswitchcookbook.com
http://freeswitchbook.com

T: +1.213.286.0410  |  F: +1.213.286.0401  |  M: +1.918.424.WEST
iNUM: +883 5100 1286 0410

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From joelrosenfield at yahoo.com  Wed Dec 12 06:04:46 2012
From: joelrosenfield at yahoo.com (Joel Rosenfield)
Date: Tue, 11 Dec 2012 19:04:46 -0800 (PST)
Subject: [Freeswitch-users] WebRTC Call
Message-ID: <1355281486.93598.YahooMailNeo@web162202.mail.bf1.yahoo.com>

In the default FreeSWITCH?configuration, I used the webrtc2sip proxy to connect a Chrome 23 client to user 1001. ?The SIP signaling is fine for REGISTER and INVITE, however?FreeSWITCH?returns 488 Not Acceptable Here.
I saw a post on the?discuss-doubango?Google Groups list from July 20 that said:

"The problem is not with the crypto its with the a=crypto being inside the AVP vs SAVP (denoting secure)
see?http://jira.freeswitch. org/browse/FS-636

I am willing to lift this restriction or at least make it configurable since the alternative is you must send a double sized sdp with AVP for non secure stuff and SAVP for the secure."

Is there something?that I need to configure?on FreeSWITCH to remove this restriction? ?Or, what should be different in the SDP offer from webrtc2sip so that?FreeSWITCH?will accept it?
Below is the output from the?FreeSWITCH?version command, the SDP offer from webrtc2sip, and?FreeSWITCH?error message.

Thanks,
- Joel

version

FreeSWITCH Version 1.3.8b+git~20121205T191750Z~ 924c524197 (git 924c524 2012-12-05 19:17:50Z)
. . . .?
? ?----------------------------- ------------------------------ -------------
recv 1931 bytes from udp/[10.159.25.56]:10060 at 22:42:51.414083:
? ?----------------------------- ------------------------------ -------------
? ?INVITE sip:9195@ . . .?
? ? . . .?
? ?User-Agent: webrtc2sip Media Server 2.0

? ?P-Preferred-Identity: <sip:webrtc2sip>

? ?v=0
? ?o=doubango 1983 678901 IN IP4 10.159.25.56
? ?s=-
? ?c=IN IP4 10.159.25.56
? ?t=0 0
? ?m=audio 60326 RTP/AVP 0 8 101
? ?c=IN IP4 10.159.25.56
? ?a=ptime:20
? ?a=silenceSupp:off - - - -
? ?a=rtpmap:0 PCMU/8000/1
? ?a=rtpmap:8 PCMA/8000/1
? ?a=rtpmap:101 telephone-event/8000/1
? ?a=fmtp:101 0-16
? ?a=tcap:1 RTP/SAVP
? ?a=pcfg:1 t=1
? ?a=sendrecv
? ?a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline: CG0ol5gUQjNxOXyMhXSIV2RltFltbx 99IkjHmXsT
? ?a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:aiZ33ixxaTy5dX5iy72n/ OAORs8lIezYKhuzti6G
? ?a=rtcp-mux
? ?a=ssrc:2783445458 cname:ldjWoB60jbyQlR6e
? ?a=ssrc:2783445458 mslabel:6994f7d1-6ce9-4fbd- acfd-84e5131ca2e2
? ?a=ssrc:2783445458 label:Doubango
? ?a=ice-ufrag:kWutfVwWe7bWeB4
? ?a=ice-pwd: ziAr5WnKhzu1KsZsMinJw
? ?a=mid:audio
? ?a=candidate:35wALsfKp 1 udp 2130706431 10.159.25.56 60326 typ host
? ?a=candidate:35wALsfKp 2 udp 2130706430 10.159.25.56 60327 typ host
? ?a=candidate:srflx35wA 2 udp 1694498814 107.21.197.144 60327 typ srflx
? ?a=candidate:srflx35wA 1 udp 1694498815 107.21.197.144 60326 typ srflx
? ?----------------------------- ------------------------------ -------------
send 400 bytes to udp/[10.159.25.56]:10060 at 22:42:51.414473:
? ?----------------------------- ------------------------------ -------------
? ?SIP/2.0 100 Trying
? ?. . .?
? ?----------------------------- ------------------------------ -------------
2012-12-11 22:42:51.734425 [INFO] mod_dialplan_xml.c:498 Processing 1001 <1001>->9195 in context default
-> 2012-12-11 22:42:51.894429 [ERR] sofia_glue.c:4922 a=crypto in RTP/AVP, refer to rfc3711
2012-12-11 22:42:51.894429 [NOTICE] switch_channel.c:3484 Hangup sofia/internal/1001 at . . .?
?[CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
send 924 bytes to udp/[10.159.25.56]:10060 at 22:42:52.054746:
? ?----------------------------- ------------------------------ -------------
? ?SIP/2.0 488 Not Acceptable Here
? ?Via: SIP/2.0/UDP 10.159.25.56:10060;branch= z9hG4bK1356314285436;rport= 10060

NOTE: ?I hacked the webrtc2sip to send "a=tcap:1 RTP/SAVP" rather than "a=tcap:1 RTP/SAVPF", but as you can see, no luck; same result either way.
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From kbdfck at gmail.com  Wed Dec 12 10:39:20 2012
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 12 Dec 2012 11:39:20 +0400
Subject: [Freeswitch-users] Mod_opal & PTLIB
In-Reply-To: <CAKzWOxVh9SrS7-AMwh2_B9+dh=uhG+c6yS6XH9_YeAJAV_01eA@mail.gmail.com>
References: <006001cdbf60$1c408b10$54c1a130$@com>
	<CA+GmxbZ79-_mfHJWJFd69VhkgRStdyBOOKvNECkmLcSOLOcCCg@mail.gmail.com>
	<CAKzWOxVh9SrS7-AMwh2_B9+dh=uhG+c6yS6XH9_YeAJAV_01eA@mail.gmail.com>
Message-ID: <CA+GmxbaiPs=npkVEzPudsS7ZvTpUCnfWJKM-rXfff1-Xnif_Pg@mail.gmail.com>

I have not so much experience with autoconf, but I'll try later, after I
verify that module is built correctly and works.

There is also some strange thing with top-level Makefile in latest versions
of both ptlib and opal.  For some reason it re-runs configure on make and
seems to do this incorrectly, without actually launching configure with
previously set configure args.
Maybe this effect is present only on my system, I don't know.

2012/12/11 Michael Collins <msc at freeswitch.org>

> Can you modify the script, verify that it works for you, then open a jira
> as a patch and attach your diff file? That would make our lives much
> easier...
>
> Thanks,
> MC
>
>
> On Tue, Dec 11, 2012 at 10:55 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>
>> There is some error in configure script maybe
>> It creates /lib/ptbuildopts.h (!) instead of ./include/ptbuildopts.h, so
>> defines don't get their values and build fails.
>>
>> I'm now trying to get these ptlib / opal to work in order to build
>> mod_opal
>>
>> 2012/11/10 Phil Quesinberry <philq at qsystemsengineering.com>
>>
>>> **
>>>
>>> I?m trying to compile mod_opal here but I can?t get it to work.  I?ve
>>> tried the buildopal.sh script, the manual install, sacrificial
>>> offerings, etc. to no avail.  I was getting an error about PTLib being
>>> too old, so I grabbed PTLib 2.10.7 source here:
>>>
>>> *http://www.linuxfromscratch.org/blfs/view/svn/general/ptlib.html*<http://www.linuxfromscratch.org/blfs/view/svn/general/ptlib.html>
>>>
>>> Trying to compile it, I get a ?No operating system selected? error.  I?ve
>>> pasted in the make attempt below, Distro is CentOS 5.8.
>>>
>>> It looks like a lot of other folks have had/are having problems with
>>> mod_opal and ptlib, but I?m not seeing much in the way of solutions.  Has
>>> anyone else gotten past this problem?  Any pointers?  (no pun intended)
>>>
>>> [root at Tyrion ptlib]# make
>>>
>>> Setting default PTLIBDIR to /root/ptlib
>>>
>>> make[1]: Entering directory `/root/ptlib'
>>>
>>> make[2]: Entering directory `/root/ptlib'
>>>
>>> make[3]: Entering directory `/root/ptlib'
>>>
>>> make[3]: Leaving directory `/root/ptlib'
>>>
>>> make[2]: Leaving directory `/root/ptlib'
>>>
>>> set -e; if test -d /root/ptlib/src ; then make -C /root/ptlib/src
>>> optshared; fi;  if test -d /root/ptlib/plugins ; then make
>>> -C/root/ptlib/plugins optshared; fi;
>>>
>>> make[2]: Entering directory `/root/ptlib/src'
>>>
>>> make[3]: Entering directory `/root/ptlib/src'
>>>
>>> [CC] ptclib/psasl.cxx
>>>
>>> In file included from /root/ptlib/include/ptlib/object.h:44,
>>>
>>>                  from /root/ptlib/include/ptlib.h:47,
>>>
>>>                  from ptclib/psasl.cxx:35:
>>>
>>> /root/ptlib/include/ptlib/unix/ptlib/platform.h:555:2: error: #error No
>>> operating system selected.
>>>
>>> /root/ptlib/include/ptlib/mutex.h:109: error: ?PThreadIdentifier? does
>>> not name a type
>>>
>>> /root/ptlib/include/ptlib/thread.h:292: error: ?PThreadIdentifier? does
>>> not name a type
>>>
>>> /root/ptlib/include/ptlib/thread.h:293: error: ?PThreadIdentifier? does
>>> not name a type
>>>
>>> /root/ptlib/include/ptlib/thread.h:403: error: ?PThreadIdentifier? does
>>> not name a type
>>>
>>> /root/ptlib/include/ptlib/syncthrd.h:326: error: ?PThreadIdentifier? was
>>> not declared in this scope
>>>
>>> /root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 1 is
>>> invalid
>>>
>>> /root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 3 is
>>> invalid
>>>
>>> /root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 4 is
>>> invalid
>>>
>>> /root/ptlib/include/ptclib/cypher.h:382: error: ?PUInt32l? does not name
>>> a type
>>>
>>> make[3]: *** [/root/ptlib/lib_linux_x86_64/obj/psasl.o] Error 1
>>>
>>> make[3]: Leaving directory `/root/ptlib/src'
>>>
>>> make[2]: *** [optshared] Error 2
>>>
>>> make[2]: Leaving directory `/root/ptlib/src'
>>>
>>> make[1]: *** [optshared] Error 2
>>>
>>> make[1]: Leaving directory `/root/ptlib'
>>>
>>> make: *** [default] Error 2
>>>
>>> Many thanks,
>>>
>>> *******Phil Quesinberry*
>>>
>>> Q Systems Engineering, Inc.
>>>
>>> Electronic Controls and Embedded Systems Development
>>>
>>> (410) 969-8002
>>>
>>> *****http://www.qsystemsengineering.com*<http://www.qsystemsengineering.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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From virbhati at gmail.com  Wed Dec 12 10:44:41 2012
From: virbhati at gmail.com (virendra bhati)
Date: Wed, 12 Dec 2012 13:14:41 +0530
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 78, Issue 60
In-Reply-To: <mailman.3250.1354872124.576.freeswitch-users@lists.freeswitch.org>
References: <mailman.3250.1354872124.576.freeswitch-users@lists.freeswitch.org>
Message-ID: <CANNhuhe06V3PjusRY4FH4Cjsd71=1XHDYFgNod4+M7mRK7sG2w@mail.gmail.com>

Hi Team,

Is I am the first who is facing this issue with nibblebill ?
1. Nibblebill negative balance (virendra bhati)


On Fri, Dec 7, 2012 at 2:52 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

> 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. Nibblebill negative balance (virendra bhati)
>    2. Re: sip registration (Archana Venugopan)
>
>
> ---------- Forwarded message ----------
> From: virendra bhati <virbhati at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Date: Fri, 7 Dec 2012 14:22:42 +0530
> Subject: [Freeswitch-users] Nibblebill negative balance
> Hi team,
>
> I have configure nibblebill with my freeswitch and it's working. But I am
> facing an issue with billing. Balance goes to -ve and after that calls also
> throw as well....
>
> Is that configuration issue or bug in nibblebill ?
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-9250078532
> Asterisk Developer
> E-mail-: virbhati at gmail.com
> Skype id:- virbhati2
> New Delhi(India)
> [image: View my profile on LinkedIn]<http://in.linkedin.com/pub/virendra-bhati/6/a30/755>
>
>
>
> ---------- Forwarded message ----------
> From: Archana Venugopan <a.venugopan at mundio.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Fri, 7 Dec 2012 09:21:30 +0000
> Subject: Re: [Freeswitch-users] sip registration
>
> Hi,****
>
> ** **
>
> I want to give some alphabets instead of number. I want to know which
> script checks this authentication name to corresponding DB table. Please
> let me know.****
>
> Thanks****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 06 December 2012 21:58
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
> ** **
>
> Use "100" for the authentication name as well.
> -MC****
>
> On Thu, Dec 6, 2012 at 9:25 AM, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
> Thanks. I thought it will look in mod_sofia code. In the below screen I
> register the ID ?100?. Now instead of ?100? in ?Authentication Name? I need
> to give some e-mail ID or name(Archana) which should validate in DB.****
>
> I tried giving a name in ?Authentication Name? but the phone was not
> registered. Am not sure this authentication name is being looked in which
> column in table too.****
>
> Please let me know if this will be picked from any sofia code or any C
> script? Once we register in the below screen which script validates the
> Settings in freeswitch?****
>
>  ****
>
> Sorry if am repeating the same question, but I could not get the exact
> code and am clueless.****
>
> *Global SIP Settings*****
>
> Top of Form****
>
> *Basic SIP Authentication Settings*****
>
>  ****
>
> Screen Name****
>
> Screen Name 2****
>
> Phone Number****
>
> Caller ID****
>
> Authentication Name****
>
> Password****
>
> Bottom of Form****
>
>  ****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 05 December 2012 20:34
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
>  ****
>
> When an event that requires a user lookup takes place then the system will
> look in the XML user directory unless it has been configured to look
> somewhere else. The other places to look are usually:
> mod_xml_curl
> One of the language like Lua, Perl, Python
>
> If it's xml_curl then FS will do a POST to your web server in hopes of
> receiving back the necessary XML for the given user. It would be up to you
> to have your web server handle the request, poll the database, then format
> and return the XML data. See this wiki page<http://wiki.freeswitch.org/wiki/Xml_curl#How_do_I_configure_mod_xml_curl.3F>for more info on xml curl.
>
> If it's a language then you'll have a "binding" in the conf file for the
> language that will handle the lookup. Again, your script will need to
> handle the communication with your database. See this wiki page<http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration>for more information.
>
> Hope this helps.
> -MC****
>
> On Wed, Dec 5, 2012 at 9:30 AM, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
> Thanks for the information. But sorry, how to access user_data API command.
> ****
>
>  ****
>
> Am not clear on the flow. Once we register domain and usernumber in sip
> what exactly happens? Which script picks up this domain and username and
> validates with our database?****
>
> Could you please provide me with an overview. ****
>
>  ****
>
> Many thanks****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* 05 December 2012 17:13
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
>  ****
>
> If you're talking about the user configuration then yes, you could create
> an "email" parameter or variable and access it with the user_data API
> command.
> -MC****
>
> On Wed, Dec 5, 2012 at 1:35 AM, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
> In that case can I have 1 more column say e-mail and can this e-mail be
> checked in DB instead of checking reg_user(?100?)? Is that feasible?****
>
> Also which code should be changed any idea please?****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 04 December 2012 19:51
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] sip registration****
>
>  ****
>
> You can have a user 'ana' in the domain 'gmail.com'. Though using someone
> else's domain as local in your FS setup may not be a good idea.****
>
> You can't have a @ in the username itself (per the SIP standard, not
> limited to FreeSWITCH).****
>
>  ****
>
> On 4 December 2012 18:00, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
>  ****
>
> Currently we register authentication name as say ?100? in sip
> registration, this comes to freeswitch and it will check in our DB for 100
> and if its present then registrations would be successful. ****
>
>  ****
>
> freeswitch at internal> show registrations****
>
>
> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
> ****
>
> 100,fsfailover.uk01.com,e4969067f9a8c098,sofia/internal/sip:100@
> 192.168.2.234:5060;transport=udp,1354638871,192.168.2.234,5060,udp,
> squay-laptop-1.squay.com****
>
>  ****
>
> I want to change this 100 to some e-mail address, so instead of 100 it
> will be something like ?ana at gmail.com?. Can we do this? While coming to
> freeswitch whether there would be any issues?****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
> ****
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

Thanks and regards

 Virendra Bhati
+91-9250078532
Asterisk Developer
E-mail-: virbhati at gmail.com
Skype id:- virbhati2
New Delhi(India)
[image: View my profile on
LinkedIn]<http://in.linkedin.com/pub/virendra-bhati/6/a30/755>
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From shaheryarkh at gmail.com  Wed Dec 12 11:31:34 2012
From: shaheryarkh at gmail.com (Muhammad Shahzad)
Date: Wed, 12 Dec 2012 09:31:34 +0100
Subject: [Freeswitch-users] WebRTC Call
In-Reply-To: <1355281486.93598.YahooMailNeo@web162202.mail.bf1.yahoo.com>
References: <1355281486.93598.YahooMailNeo@web162202.mail.bf1.yahoo.com>
Message-ID: <CAFZQphzGO+JEyUzAm_zaXLorchxW4ybXy4Uockp+zySDK1PATQ@mail.gmail.com>

If you are using newer (doubango based) version of WebRTC2SIP gateway then
you don't need to do anything, it will take care of both signalling and
media. However you might need media breaker module for transcoding. Though
FS supports both ISAC and VP8 codecs (for audio and video respectively) but
i am not sure if they are compatible / working with Chrome, since chrome
has lot of development going around and they make so many changes on daily
basis, resulting in every new version pretty much incompatible with older
ones and so on.

Please make sure you follow this wiki,

http://code.google.com/p/webrtc2sip/wiki/Building_Source_v2_0

Thank you.


On Wed, Dec 12, 2012 at 4:04 AM, Joel Rosenfield
<joelrosenfield at yahoo.com>wrote:

> In the default FreeSWITCH configuration, I used the webrtc2sip proxy to
> connect a Chrome 23 client to user 1001.  The SIP signaling is fine for
> REGISTER and INVITE, however FreeSWITCH returns 488 Not Acceptable Here.
> I saw a post on the discuss-doubango<https://groups.google.com/forum/#!forum/doubango> Google
> Groups list from July 20 that said:
>
> "The problem is not with the crypto its with the a=crypto being inside
> the AVP vs SAVP (denoting secure)
> see http://jira.freeswitch. org/browse/FS-636<http://jira.freeswitch.org/browse/FS-636>
>
> I am willing to lift this restriction or at least make it configurable
> since the alternative is you must send a double sized sdp with AVP for non
> secure stuff and SAVP for the secure."
>
> Is there something that I need to configure on FreeSWITCH to remove this
> restriction?  Or, what should be different in the SDP offer from webrtc2sip
> so that FreeSWITCH will accept it?
> Below is the output from the FreeSWITCH version command, the SDP offer
> from webrtc2sip, and FreeSWITCH error message.
>
> Thanks,
> - Joel
>
> version
>
> FreeSWITCH Version 1.3.8b+git~20121205T191750Z~ 924c524197 (git 924c524
> 2012-12-05 19:17:50Z)
> . . . .
>    ----------------------------- ------------------------------
> -------------
> recv 1931 bytes from udp/[10.159.25.56]:10060 at 22:42:51.414083:
>    ----------------------------- ------------------------------
> -------------
>    INVITE sip:9195@ . . .
>     . . .
>    User-Agent: webrtc2sip Media Server 2.0
>    P-Preferred-Identity: <sip:webrtc2sip>
>
>    v=0
>    o=doubango 1983 678901 IN IP4 10.159.25.56
>    s=-
>    c=IN IP4 10.159.25.56
>    t=0 0
>    m=audio 60326 RTP/AVP 0 8 101
>    c=IN IP4 10.159.25.56
>    a=ptime:20
>    a=silenceSupp:off - - - -
>    a=rtpmap:0 PCMU/8000/1
>    a=rtpmap:8 PCMA/8000/1
>    a=rtpmap:101 telephone-event/8000/1
>    a=fmtp:101 0-16
>    a=tcap:1 RTP/SAVP
>    a=pcfg:1 t=1
>    a=sendrecv
>    a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:
> CG0ol5gUQjNxOXyMhXSIV2RltFltbx 99IkjHmXsT
>    a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:aiZ33ixxaTy5dX5iy72n/
> OAORs8lIezYKhuzti6G
>    a=rtcp-mux
>    a=ssrc:2783445458 cname:ldjWoB60jbyQlR6e
>    a=ssrc:2783445458 mslabel:6994f7d1-6ce9-4fbd- acfd-84e5131ca2e2
>    a=ssrc:2783445458 label:Doubango
>    a=ice-ufrag:kWutfVwWe7bWeB4
>    a=ice-pwd: ziAr5WnKhzu1KsZsMinJw
>    a=mid:audio
>    a=candidate:35wALsfKp 1 udp 2130706431 10.159.25.56 60326 typ host
>    a=candidate:35wALsfKp 2 udp 2130706430 10.159.25.56 60327 typ host
>    a=candidate:srflx35wA 2 udp 1694498814 107.21.197.144 60327 typ srflx
>    a=candidate:srflx35wA 1 udp 1694498815 107.21.197.144 60326 typ srflx
>    ----------------------------- ------------------------------
> -------------
> send 400 bytes to udp/[10.159.25.56]:10060 at 22:42:51.414473:
>    ----------------------------- ------------------------------
> -------------
>    SIP/2.0 100 Trying
>    . . .
>    ----------------------------- ------------------------------
> -------------
> 2012-12-11 22:42:51.734425 [INFO] mod_dialplan_xml.c:498 Processing 1001
> <1001>->9195 in context default
> -> 2012-12-11 22:42:51.894429 [ERR] sofia_glue.c:4922 a=crypto in RTP/AVP,
> refer to rfc3711
> 2012-12-11 22:42:51.894429 [NOTICE] switch_channel.c:3484 Hangup
> sofia/internal/1001 at . . .
>  [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
> send 924 bytes to udp/[10.159.25.56]:10060 at 22:42:52.054746:
>    ----------------------------- ------------------------------
> -------------
>    SIP/2.0 488 Not Acceptable Here
>    Via: SIP/2.0/UDP 10.159.25.56:10060;branch=
> z9hG4bK1356314285436;rport= 10060
>
> NOTE:  I hacked the webrtc2sip to send "a=tcap:1 RTP/SAVP" rather than
> "a=tcap:1 RTP/SAVPF", but as you can see, no luck; same result either way.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From miha at softnet.si  Wed Dec 12 13:43:24 2012
From: miha at softnet.si (Miha)
Date: Wed, 12 Dec 2012 11:43:24 +0100
Subject: [Freeswitch-users] gateway tcp
Message-ID: <50C85FCC.5030405@softnet.si>

Hi,

I configured gw like this:

<include>
         <gateway name="lync_telprom">
                 <param name="username" value="none"/>
                 <param name="password" value="none"/>
                 <param name="proxy" value="xxx.xxx.xxx.xxx;transport=tcp"/>
                 <param name="register-transport" value="tcp"/>
                 <param name="register" value="false"/>
                 <param name="caller-id-in-from" value="true"/>
         </gateway>
</include>

I noticed that FS sends to trunk trafic via tcp but suddenly it sends 
one part of media via upd. Is this normal?

I am using 1.0.6 version of fs, should upgrade to latest stable verion 
of Fs solve this?

br,
Miha
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From miha at softnet.si  Wed Dec 12 13:54:01 2012
From: miha at softnet.si (Miha)
Date: Wed, 12 Dec 2012 11:54:01 +0100
Subject: [Freeswitch-users] gateway tcp
In-Reply-To: <50C85FCC.5030405@softnet.si>
References: <50C85FCC.5030405@softnet.si>
Message-ID: <50C86249.2040908@softnet.si>

Please ignore this as I was misguided by same lync administrator...

sorry for posting this.

Br,
Miha

Dne 12/12/2012 11:43 AM, pis(e Miha:
> Hi,
>
> I configured gw like this:
>
> <include>
>         <gateway name="lync_telprom">
>                 <param name="username" value="none"/>
>                 <param name="password" value="none"/>
>                 <param name="proxy" 
> value="xxx.xxx.xxx.xxx;transport=tcp"/>
>                 <param name="register-transport" value="tcp"/>
>                 <param name="register" value="false"/>
>                 <param name="caller-id-in-from" value="true"/>
>         </gateway>
> </include>
>
> I noticed that FS sends to trunk trafic via tcp but suddenly it sends 
> one part of media via upd. Is this normal?
>
> I am using 1.0.6 version of fs, should upgrade to latest stable verion 
> of Fs solve this?
>
> br,
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 12 14:15:51 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 12 Dec 2012 06:15:51 -0500
Subject: [Freeswitch-users] Nibblebill negative balance
In-Reply-To: <CANNhuhfyXQTao-4OriaS3Av_MC2=nm+GOLvfwJQAeJ76jRmVNg@mail.gmail.com>
References: <CANNhuhfyXQTao-4OriaS3Av_MC2=nm+GOLvfwJQAeJ76jRmVNg@mail.gmail.com>
Message-ID: <F5794F00-C367-40A6-8FAC-D5DD8130B2EF@endigotech.com>

If you're talking about calls being interrupted due to a negative balance, yes that's a feature that can be configured. 

http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_Call_When_the_Balance_Is_Depleted



Sent from my iPhone

On Dec 7, 2012, at 3:52 AM, virendra bhati <virbhati at gmail.com> wrote:

> Hi team,
> 
> I have configure nibblebill with my freeswitch and it's working. But I am facing an issue with billing. Balance goes to -ve and after that calls also throw as well....
> 
> Is that configuration issue or bug in nibblebill ? 
> 
> -- 
> 
> Thanks and regards
> 
>  Virendra Bhati
> +91-9250078532
> Asterisk Developer
> E-mail-: virbhati at gmail.com
> Skype id:- virbhati2
> New Delhi(India)
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 andrew at cassidywebservices.co.uk  Wed Dec 12 14:44:52 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 12 Dec 2012 11:44:52 +0000
Subject: [Freeswitch-users] Nibblebill negative balance
In-Reply-To: <F5794F00-C367-40A6-8FAC-D5DD8130B2EF@endigotech.com>
References: <CANNhuhfyXQTao-4OriaS3Av_MC2=nm+GOLvfwJQAeJ76jRmVNg@mail.gmail.com>
	<F5794F00-C367-40A6-8FAC-D5DD8130B2EF@endigotech.com>
Message-ID: <CAMbW9oJKi==P+RT3OQ+nB02=EVJk_D4DuhpJ1PMh679+=wUwrg@mail.gmail.com>

Make that image your signature...

On 12 December 2012 11:15, Brian Foster <bdfoster at endigotech.com> wrote:

> If you're talking about calls being interrupted due to a negative balance,
> yes that's a feature that can be configured.
>
>
> http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_Call_When_the_Balance_Is_Depleted
>
> [image: image.jpeg]
>
> Sent from my iPhone
>
> On Dec 7, 2012, at 3:52 AM, virendra bhati <virbhati at gmail.com> wrote:
>
> Hi team,
>
> I have configure nibblebill with my freeswitch and it's working. But I am
> facing an issue with billing. Balance goes to -ve and after that calls also
> throw as well....
>
> Is that configuration issue or bug in nibblebill ?
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-9250078532
> Asterisk Developer
> E-mail-: virbhati at gmail.com
> Skype id:- virbhati2
> New Delhi(India)
> [image: View my profile on LinkedIn]<http://in.linkedin.com/pub/virendra-bhati/6/a30/755>
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 yehavi.bourvine at gmail.com  Wed Dec 12 15:46:51 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 12 Dec 2012 14:46:51 +0200
Subject: [Freeswitch-users] Counterpath's Bria-3 and BLF
Message-ID: <CAAtWKeHd3F8j-_vBMN8gTybrh3xPAkeJEsYjzUquKWsDQi9brg@mail.gmail.com>

Hello,

  Is it possible to use Bria-3 contacts as BLF's with Freeswitch? I've
enabled presence, I see the contact's status showing that it is ringing,
but I have no way of picking up the call by pressing the contact...

Dialling manually ***extension-number* picks up the call.

                   Thanks! __Yehavi:
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From a.venugopan at mundio.com  Wed Dec 12 17:43:59 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Wed, 12 Dec 2012 14:43:59 +0000
Subject: [Freeswitch-users] make error
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233B48C@Mail-Kilo.squay.com>

Hi ,

Am trying to install freeswitch in new server. While running make command am facing with the below error. In google I could not get a clue on this error. Can anyone have any idea about this error please?

*** Warning: Linking the shared library libfreeswitch.la against the
*** static library libs/libedit/src/.libs/libedit.a is not portable!
quiet_libtool: link: g++ -shared -nostdlib /usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../../../lib64/crti.o /usr/lib/gcc/x86_64-redhat-linux/4.4.6/crtbeginS.o  .libs/libfreeswitch_la-switch_apr.o .libs/libfreeswitch_la-switch_buffer.o .libs/libfreeswitch_la-switch_caller.o .libs/libfreeswitch_la-switch_channel.o .libs/libfreeswitch_la-switch_console.o .libs/libfreeswitch_la-switch_mprintf.o .libs/libfreeswitch_la-switch_core_media_bug.o .libs/libfreeswitch_la-switch_core_timer.o .libs/libfreeswitch_la-switch_core_asr.o .libs/libfreeswitch_la-switch_core_event_hook.o .libs/libfreeswitch_la-switch_core_speech.o .libs/libfreeswitch_la-switch_core_memory.o .libs/libfreeswitch_la-switch_core_codec.o .libs/libfreeswitch_la-switch_core_file.o .libs/libfreeswitch_la-switch_core_hash.o .libs/libfreeswitch_la-switch_core_sqldb.o .libs/libfreeswitch_la-switch_core_session.o .libs/libfreeswitch_la-switch_core_directory.o .libs/libfreeswitch_la-switch_core_state_machine.o .libs/libfreeswitch_la-switch_core_io.o .libs/libfreeswitch_la-switch_core_rwlock.o .libs/libfreeswitch_la-switch_core_port_allocator.o .libs/libfreeswitch_la-switch_core.o .libs/libfreeswitch_la-switch_scheduler.o .libs/libfreeswitch_la-switch_core_db.o .libs/libfreeswitch_la-switch_dso.o .libs/libfreeswitch_la-switch_loadable_module.o .libs/libfreeswitch_la-switch_utils.o .libs/libfreeswitch_la-switch_event.o .libs/libfreeswitch_la-switch_resample.o .libs/libfreeswitch_la-switch_regex.o .libs/libfreeswitch_la-switch_rtp.o .libs/libfreeswitch_la-switch_ivr_bridge.o .libs/libfreeswitch_la-switch_ivr_originate.o .libs/libfreeswitch_la-switch_ivr_async.o .libs/libfreeswitch_la-switch_ivr_play_say.o .libs/libfreeswitch_la-switch_ivr_say.o .libs/libfreeswitch_la-switch_ivr_menu.o .libs/libfreeswitch_la-switch_ivr.o .libs/libfreeswitch_la-switch_stun.o .libs/libfreeswitch_la-switch_nat.o .libs/libfreeswitch_la-switch_log.o .libs/libfreeswitch_la-switch_xml.o .libs/libfreeswitch_la-switch_xml_config.o .libs/libfreeswitch_la-switch_config.o .libs/libfreeswitch_la-switch_time.o .libs/libfreeswitch_la-switch_odbc.o .libs/libfreeswitch_la-switch_limit.o .libs/libfreeswitch_la-g711.o .libs/libfreeswitch_la-switch_pcm.o .libs/libfreeswitch_la-switch_profile.o .libs/libfreeswitch_la-switch_json.o .libs/libfreeswitch_la-switch_curl.o .libs/libfreeswitch_la-tpl.o .libs/libfreeswitch_la-stfu.o .libs/libfreeswitch_la-libteletone_detect.o .libs/libfreeswitch_la-libteletone_generate.o .libs/libfreeswitch_la-miniwget.o .libs/libfreeswitch_la-minixml.o .libs/libfreeswitch_la-igd_desc_parse.o .libs/libfreeswitch_la-minisoap.o .libs/libfreeswitch_la-miniupnpc.o .libs/libfreeswitch_la-upnpreplyparse.o .libs/libfreeswitch_la-upnpcommands.o .libs/libfreeswitch_la-minissdpc.o .libs/libfreeswitch_la-upnperrors.o .libs/libfreeswitch_la-natpmp.o .libs/libfreeswitch_la-getgateway.o .libs/libfreeswitch_la-plc.o .libs/libfreeswitch_la-bit_operations.o .libs/switch_cpp.o   -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs -L/usr/local/src/freeswitch/libs/apr/.libs libs/apr-util/.libs/libaprutil-1.a -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a libs/apr/.libs/libapr-1.a libs/sqlite/.libs/libsqlite3.a -lpthread libs/pcre/.libs/libpcre.a libs/speex/libspeex/.libs/libspeexdsp.a libs/srtp/.libs/libsrtp.a libs/libedit/src/.libs/libedit.a libs/curl/lib/.libs/libcurl.a -lz -ldl -lcrypt -lrt -lncurses -ljpeg -L/usr/lib/gcc/x86_64-redhat-linux/4.4.6 -L/usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../../../lib64 -L/lib/../lib64 -L/usr/lib/../lib64 -L/usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../.. -lstdc++ -lm -lc -lgcc_s /usr/lib/gcc/x86_64-redhat-linux/4.4.6/crtendS.o /usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../../../lib64/crtn.o    -pthread -Wl,-soname -Wl,libfreeswitch.so.1 -o .libs/libfreeswitch.so.1.0.0
libs/apr-util/.libs/libaprutil-1.a: could not read symbols: File in wrong format
collect2: ld returned 1 exit status
make[1]: *** [libfreeswitch.la] Error 1
make: *** [all] Error 2

Regards,
Archana

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From kbdfck at gmail.com  Wed Dec 12 18:03:05 2012
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 12 Dec 2012 19:03:05 +0400
Subject: [Freeswitch-users] Console FreeTDM commands get executed only once,
	following ftdm hang console
Message-ID: <CA+GmxbZBZ7Wm96qF5avNW0vmrDx+FDWk=UD=kAXHOeeCDaOpvQ@mail.gmail.com>

Hi all!

I have Digium TDM400P with DAHDI, FREETDM and libpri built and installed on
git head. There is one librpi-based span configured. The problem is when I
issue any ftdm-command it works once, and all subsequent ftdm commands hang
and I get no response.

I kill fs_cli, start it again and can do anything but ftdm commands - it
hangs after any ftdm command.

After that I can't unload mod_freetdm, as FS says it is in use.

Is this a bug? What information should I collect to open JIRA?

freeswitch at internal> ftdm list
+OK
span: 1 (trunk1)
type: isdn
physical_status: ok
signaling_status: DOWN
chan_count: 31
dialplan: XML
context: from_trunk1
dial_regex:
fail_dial_regex:
hold_music:
analog_options: none

freeswitch at internal>
freeswitch at internal> ftdm list
<no response>




-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From bdfoster at endigotech.com  Wed Dec 12 19:00:24 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 12 Dec 2012 11:00:24 -0500
Subject: [Freeswitch-users] gateway tcp
In-Reply-To: <50C86249.2040908@softnet.si>
References: <50C85FCC.5030405@softnet.si> <50C86249.2040908@softnet.si>
Message-ID: <0E10BF8E-1E2D-4163-847C-2D7E99D336F8@endigotech.com>



> Please ignore this as I was misguided       by same lync administrator...

That would explain a lot.

Media should be sent via RTP.

> sorry for posting this.
> 
> Br,
> Miha
> 
> Dne 12/12/2012 11:43 AM, pi?e Miha:
>> Hi,
>> 
>> I configured gw like this:
>> 
>> <include>
>>         <gateway name="lync_telprom">
>>                 <param name="username" value="none"/>
>>                 <param name="password" value="none"/>
>>                 <param name="proxy" value="xxx.xxx.xxx.xxx;transport=tcp"/>
>>                 <param name="register-transport" value="tcp"/>
>>                 <param name="register" value="false"/>
>>                 <param name="caller-id-in-from" value="true"/>
>>         </gateway>
>> </include>
>> 
>> I noticed that FS sends to trunk trafic via tcp but suddenly it sends one part of media via upd. Is this normal?
>> 
>> I am using 1.0.6 version of fs, should upgrade to latest stable verion of Fs solve this?
>> 
>> br,
>> 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 bdfoster at endigotech.com  Wed Dec 12 19:02:58 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 12 Dec 2012 11:02:58 -0500
Subject: [Freeswitch-users] make error
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233B48C@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233B48C@Mail-Kilo.squay.com>
Message-ID: <834BA749-FAB6-4C6A-A87A-44EDED20BEEB@endigotech.com>



Sent from my iPhone

On Dec 12, 2012, at 9:43 AM, Archana Venugopan <a.venugopan at mundio.com> wrote:

> Hi ,
>  
> Am trying to install freeswitch in new server. While running make command am facing with the below error. In google I could not get a clue on this error. Can anyone have any idea about this error please?
>  
> *** Warning: Linking the shared library libfreeswitch.la against the
> *** static library libs/libedit/src/.libs/libedit.a is not portable!
> quiet_libtool: link: g++ -shared -nostdlib /usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../../../lib64/crti.o /usr/lib/gcc/x86_64-redhat-linux/4.4.6/crtbeginS.o  .libs/libfreeswitch_la-switch_apr.o .libs/libfreeswitch_la-switch_buffer.o .libs/libfreeswitch_la-switch_caller.o .libs/libfreeswitch_la-switch_channel.o .libs/libfreeswitch_la-switch_console.o .libs/libfreeswitch_la-switch_mprintf.o .libs/libfreeswitch_la-switch_core_media_bug.o .libs/libfreeswitch_la-switch_core_timer.o .libs/libfreeswitch_la-switch_core_asr.o .libs/libfreeswitch_la-switch_core_event_hook.o .libs/libfreeswitch_la-switch_core_speech.o .libs/libfreeswitch_la-switch_core_memory.o .libs/libfreeswitch_la-switch_core_codec.o .libs/libfreeswitch_la-switch_core_file.o .libs/libfreeswitch_la-switch_core_hash.o .libs/libfreeswitch_la-switch_core_sqldb.o .libs/libfreeswitch_la-switch_core_session.o .libs/libfreeswitch_la-switch_core_directory.o .libs/libfreeswitch_la-switch_core_state_machine.o .libs/libfreeswitch_la-switch_core_io.o .libs/libfreeswitch_la-switch_core_rwlock.o .libs/libfreeswitch_la-switch_core_port_allocator.o .libs/libfreeswitch_la-switch_core.o .libs/libfreeswitch_la-switch_scheduler.o .libs/libfreeswitch_la-switch_core_db.o .libs/libfreeswitch_la-switch_dso.o .libs/libfreeswitch_la-switch_loadable_module.o .libs/libfreeswitch_la-switch_utils.o .libs/libfreeswitch_la-switch_event.o .libs/libfreeswitch_la-switch_resample.o .libs/libfreeswitch_la-switch_regex.o .libs/libfreeswitch_la-switch_rtp.o .libs/libfreeswitch_la-switch_ivr_bridge.o .libs/libfreeswitch_la-switch_ivr_originate.o .libs/libfreeswitch_la-switch_ivr_async.o .libs/libfreeswitch_la-switch_ivr_play_say.o  .libs/libfreeswitch_la-switch_ivr_say.o .libs/libfreeswitch_la-switch_ivr_menu.o .libs/libfreeswitch_la-switch_ivr.o .libs/libfreeswitch_la-switch_stun.o .libs/libfreeswitch_la-switch_nat.o .libs/libfreeswitch_la-switch_log.o .libs/libfreeswitch_la-switch_xml.o .libs/libfreeswitch_la-switch_xml_config.o .libs/libfreeswitch_la-switch_config.o .libs/libfreeswitch_la-switch_time.o .libs/libfreeswitch_la-switch_odbc.o .libs/libfreeswitch_la-switch_limit.o .libs/libfreeswitch_la-g711.o .libs/libfreeswitch_la-switch_pcm.o .libs/libfreeswitch_la-switch_profile.o .libs/libfreeswitch_la-switch_json.o .libs/libfreeswitch_la-switch_curl.o .libs/libfreeswitch_la-tpl.o .libs/libfreeswitch_la-stfu.o .libs/libfreeswitch_la-libteletone_detect.o .libs/libfreeswitch_la-libteletone_generate.o  .libs/libfreeswitch_la-miniwget.o .libs/libfreeswitch_la-minixml.o .libs/libfreeswitch_la-igd_desc_parse.o .libs/libfreeswitch_la-minisoap.o .libs/libfreeswitch_la-miniupnpc.o .libs/libfreeswitch_la-upnpreplyparse.o .libs/libfreeswitch_la-upnpcommands.o .libs/libfreeswitch_la-minissdpc.o .libs/libfreeswitch_la-upnperrors.o .libs/libfreeswitch_la-natpmp.o .libs/libfreeswitch_la-getgateway.o .libs/libfreeswitch_la-plc.o .libs/libfreeswitch_la-bit_operations.o .libs/switch_cpp.o   -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs -L/usr/local/src/freeswitch/libs/apr/.libs libs/apr-util/.libs/libaprutil-1.a -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a libs/apr/.libs/libapr-1.a libs/sqlite/.libs/libsqlite3.a -lpthread libs/pcre/.libs/libpcre.a libs/speex/libspeex/.libs/libspeexdsp.a libs/srtp/.libs/libsrtp.a libs/libedit/src/.libs/libedit.a libs/curl/lib/.libs/libcurl.a -lz -ldl -lcrypt -lrt -lncurses -ljpeg -L/usr/lib/gcc/x86_64-redhat-linux/4.4.6 -L/usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../../../lib64 -L/lib/../lib64 -L/usr/lib/../lib64 -L/usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../.. -lstdc++ -lm -lc -lgcc_s /usr/lib/gcc/x86_64-redhat-linux/4.4.6/crtendS.o /usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../../../lib64/crtn.o    -pthread -Wl,-soname -Wl,libfreeswitch.so.1 -o .libs/libfreeswitch.so.1.0.0
> libs/apr-util/.libs/libaprutil-1.a: could not read symbols: File in wrong format
> collect2: ld returned 1 exit status
> make[1]: *** [libfreeswitch.la] Error 1
> make: *** [all] Error 2
>  
> Regards,
> Archana
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 krice at freeswitch.org  Wed Dec 12 19:03:39 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 12 Dec 2012 10:03:39 -0600
Subject: [Freeswitch-users] make error
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233B48C@Mail-Kilo.squay.com>
Message-ID: <CCEE06FB.61E80%krice@freeswitch.org>

Sounds like your build environment is screwed up...

What platform are you building on?

Also try the following if you are getting the code from git

# git clean ?fdx && git reset ?hard && git pull

Then start over at bootstrap.sh and go from there


On 12/12/12 8:43 AM, "Archana Venugopan" <a.venugopan at mundio.com> wrote:

> Hi ,
>  
> Am trying to install freeswitch in new server. While running make command am
> facing with the below error. In google I could not get a clue on this error.
> Can anyone have any idea about this error please?
>  
> *** Warning: Linking the shared library libfreeswitch.la against the
> *** static library libs/libedit/src/.libs/libedit.a is not portable!
> quiet_libtool: link: g++ -shared -nostdlib
> /usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../../../lib64/crti.o
> /usr/lib/gcc/x86_64-redhat-linux/4.4.6/crtbeginS.o
> .libs/libfreeswitch_la-switch_apr.o .libs/libfreeswitch_la-switch_buffer.o
> .libs/libfreeswitch_la-switch_caller.o .libs/libfreeswitch_la-switch_channel.o
> .libs/libfreeswitch_la-switch_console.o
> .libs/libfreeswitch_la-switch_mprintf.o
> .libs/libfreeswitch_la-switch_core_media_bug.o
> .libs/libfreeswitch_la-switch_core_timer.o
> .libs/libfreeswitch_la-switch_core_asr.o
> .libs/libfreeswitch_la-switch_core_event_hook.o
> .libs/libfreeswitch_la-switch_core_speech.o
> .libs/libfreeswitch_la-switch_core_memory.o
> .libs/libfreeswitch_la-switch_core_codec.o
> .libs/libfreeswitch_la-switch_core_file.o
> .libs/libfreeswitch_la-switch_core_hash.o
> .libs/libfreeswitch_la-switch_core_sqldb.o
> .libs/libfreeswitch_la-switch_core_session.o
> .libs/libfreeswitch_la-switch_core_directory.o
> .libs/libfreeswitch_la-switch_core_state_machine.o
> .libs/libfreeswitch_la-switch_core_io.o
> .libs/libfreeswitch_la-switch_core_rwlock.o
> .libs/libfreeswitch_la-switch_core_port_allocator.o
> .libs/libfreeswitch_la-switch_core.o .libs/libfreeswitch_la-switch_scheduler.o
> .libs/libfreeswitch_la-switch_core_db.o .libs/libfreeswitch_la-switch_dso.o
> .libs/libfreeswitch_la-switch_loadable_module.o
> .libs/libfreeswitch_la-switch_utils.o .libs/libfreeswitch_la-switch_event.o
> .libs/libfreeswitch_la-switch_resample.o .libs/libfreeswitch_la-switch_regex.o
> .libs/libfreeswitch_la-switch_rtp.o .libs/libfreeswitch_la-switch_ivr_bridge.o
> .libs/libfreeswitch_la-switch_ivr_originate.o
> .libs/libfreeswitch_la-switch_ivr_async.o
> .libs/libfreeswitch_la-switch_ivr_play_say.o
> .libs/libfreeswitch_la-switch_ivr_say.o
> .libs/libfreeswitch_la-switch_ivr_menu.o .libs/libfreeswitch_la-switch_ivr.o
> .libs/libfreeswitch_la-switch_stun.o .libs/libfreeswitch_la-switch_nat.o
> .libs/libfreeswitch_la-switch_log.o .libs/libfreeswitch_la-switch_xml.o
> .libs/libfreeswitch_la-switch_xml_config.o
> .libs/libfreeswitch_la-switch_config.o .libs/libfreeswitch_la-switch_time.o
> .libs/libfreeswitch_la-switch_odbc.o .libs/libfreeswitch_la-switch_limit.o
> .libs/libfreeswitch_la-g711.o .libs/libfreeswitch_la-switch_pcm.o
> .libs/libfreeswitch_la-switch_profile.o .libs/libfreeswitch_la-switch_json.o
> .libs/libfreeswitch_la-switch_curl.o .libs/libfreeswitch_la-tpl.o
> .libs/libfreeswitch_la-stfu.o .libs/libfreeswitch_la-libteletone_detect.o
> .libs/libfreeswitch_la-libteletone_generate.o
> .libs/libfreeswitch_la-miniwget.o .libs/libfreeswitch_la-minixml.o
> .libs/libfreeswitch_la-igd_desc_parse.o .libs/libfreeswitch_la-minisoap.o
> .libs/libfreeswitch_la-miniupnpc.o .libs/libfreeswitch_la-upnpreplyparse.o
> .libs/libfreeswitch_la-upnpcommands.o .libs/libfreeswitch_la-minissdpc.o
> .libs/libfreeswitch_la-upnperrors.o .libs/libfreeswitch_la-natpmp.o
> .libs/libfreeswitch_la-getgateway.o .libs/libfreeswitch_la-plc.o
> .libs/libfreeswitch_la-bit_operations.o .libs/switch_cpp.o
> -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs
> -L/usr/local/src/freeswitch/libs/apr/.libs libs/apr-util/.libs/libaprutil-1.a
> -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib
> /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
> /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a libs/apr/.libs/libapr-1.a
> libs/sqlite/.libs/libsqlite3.a -lpthread libs/pcre/.libs/libpcre.a
> libs/speex/libspeex/.libs/libspeexdsp.a libs/srtp/.libs/libsrtp.a
> libs/libedit/src/.libs/libedit.a libs/curl/lib/.libs/libcurl.a -lz -ldl
> -lcrypt -lrt -lncurses -ljpeg -L/usr/lib/gcc/x86_64-redhat-linux/4.4.6
> -L/usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../../../lib64 -L/lib/../lib64
> -L/usr/lib/../lib64 -L/usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../.. -lstdc++
> -lm -lc -lgcc_s /usr/lib/gcc/x86_64-redhat-linux/4.4.6/crtendS.o
> /usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../../../lib64/crtn.o    -pthread
> -Wl,-soname -Wl,libfreeswitch.so.1 -o .libs/libfreeswitch.so.1.0.0
> libs/apr-util/.libs/libaprutil-1.a: could not read symbols: File in wrong
> format
> collect2: ld returned 1 exit status
> make[1]: *** [libfreeswitch.la] Error 1
> make: *** [all] Error 2
>  
> Regards,
> Archana
>  
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From kris at kriskinc.com  Wed Dec 12 19:07:28 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 12 Dec 2012 11:07:28 -0500
Subject: [Freeswitch-users] SRTP + hardware crypto
Message-ID: <CAKDfjgesQGS+EK=8Dur6xVJZO16eP-51ctEH9bf8WWJLzrU65Q@mail.gmail.com>

Hello All,

  Has anyone ever looked at enabling hardware acceleration of SIP+TLS
(and more importantly) SRTP?

  Hardware engines are pretty well defined in OpenSSL at this point so
it shouldn't be too hard to get Sofia to use them.  SRTP, on the other
hand, doesn't use OpenSSL...

  So the next question is: How hard would it be to get libsrtp to use
OpenSSL AES so we could take advantage of the various hardware engines
available?  Would it actually help?  Has anyone looked at this?

Thanks!

-- 
Kristian Kielhofner


From anthony.minessale at gmail.com  Wed Dec 12 19:38:32 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 Dec 2012 10:38:32 -0600
Subject: [Freeswitch-users] SRTP + hardware crypto
In-Reply-To: <CAKDfjgesQGS+EK=8Dur6xVJZO16eP-51ctEH9bf8WWJLzrU65Q@mail.gmail.com>
References: <CAKDfjgesQGS+EK=8Dur6xVJZO16eP-51ctEH9bf8WWJLzrU65Q@mail.gmail.com>
Message-ID: <CAKbxfG9xmZWBr4tMoYkLDpXGJmp8KkcGXFqzR_0kMkd-tNsp0A@mail.gmail.com>

Yes actually.  pressureman in IRC was saying he's working on that.  If
he's successful we'll surely be pushing that into tree,



On Wed, Dec 12, 2012 at 10:07 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Hello All,
>
>   Has anyone ever looked at enabling hardware acceleration of SIP+TLS
> (and more importantly) SRTP?
>
>   Hardware engines are pretty well defined in OpenSSL at this point so
> it shouldn't be too hard to get Sofia to use them.  SRTP, on the other
> hand, doesn't use OpenSSL...
>
>   So the next question is: How hard would it be to get libsrtp to use
> OpenSSL AES so we could take advantage of the various hardware engines
> available?  Would it actually help?  Has anyone looked at this?
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
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From msc at freeswitch.org  Wed Dec 12 19:54:31 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Dec 2012 08:54:31 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxXbXubRj4id1ND0ODUgf-8jUQT4fJxBy2RYHLxo_Dqegg@mail.gmail.com>

Hello community!

Today's conference call agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_12

We will be doing a community scrum today. If he's available, Dave Kompel
(IRC: drk__) will be talking about techniques for gathering data when
FreeSWITCH crashes when running under Windows. Talk to you soon!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From kris at kriskinc.com  Wed Dec 12 20:31:23 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 12 Dec 2012 12:31:23 -0500
Subject: [Freeswitch-users] SRTP + hardware crypto
In-Reply-To: <CAKbxfG9xmZWBr4tMoYkLDpXGJmp8KkcGXFqzR_0kMkd-tNsp0A@mail.gmail.com>
References: <CAKDfjgesQGS+EK=8Dur6xVJZO16eP-51ctEH9bf8WWJLzrU65Q@mail.gmail.com>
	<CAKbxfG9xmZWBr4tMoYkLDpXGJmp8KkcGXFqzR_0kMkd-tNsp0A@mail.gmail.com>
Message-ID: <CAKDfjgeEDphmmwgeBcYFGOhG2qdiLyOim39DkcvAaCvdaqWrhQ@mail.gmail.com>

Nice!

IRC...  Does "pressureman" have a real name or e-mail address?

On Wed, Dec 12, 2012 at 11:38 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Yes actually.  pressureman in IRC was saying he's working on that.  If he's
> successful we'll surely be pushing that into tree,
>

-- 
Kristian Kielhofner


From anthony.minessale at gmail.com  Wed Dec 12 20:37:13 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 Dec 2012 11:37:13 -0600
Subject: [Freeswitch-users] SRTP + hardware crypto
In-Reply-To: <CAKDfjgeEDphmmwgeBcYFGOhG2qdiLyOim39DkcvAaCvdaqWrhQ@mail.gmail.com>
References: <CAKDfjgesQGS+EK=8Dur6xVJZO16eP-51ctEH9bf8WWJLzrU65Q@mail.gmail.com>
	<CAKbxfG9xmZWBr4tMoYkLDpXGJmp8KkcGXFqzR_0kMkd-tNsp0A@mail.gmail.com>
	<CAKDfjgeEDphmmwgeBcYFGOhG2qdiLyOim39DkcvAaCvdaqWrhQ@mail.gmail.com>
Message-ID: <CAKbxfG_YFmvzyvLRZ17y_Kah54WJs5gBkOEEUi0k+71gagUAZw@mail.gmail.com>

Yah, I forgot what it was, I'll send him your way when I see him again.


On Wed, Dec 12, 2012 at 11:31 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Nice!
>
> IRC...  Does "pressureman" have a real name or e-mail address?
>
> On Wed, Dec 12, 2012 at 11:38 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > Yes actually.  pressureman in IRC was saying he's working on that.  If
> he's
> > successful we'll surely be pushing that into tree,
> >
>
> --
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
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From itispip-qq at hotmail.com  Wed Dec 12 20:52:29 2012
From: itispip-qq at hotmail.com (=?gb2312?B?zfXA7Q==?=)
Date: Thu, 13 Dec 2012 01:52:29 +0800
Subject: [Freeswitch-users] Email notification when user unregistered?
Message-ID: <BLU169-W7BDC7C356B8C56E4FC8E68C4F0@phx.gbl>


Hi Freeswtich Master,
I'm in a situation where I need to proactively moniter whether some device/extension is on in my local network; If for some reason the devices are offline, I need to be noticed immediately so that I can restart the device;
Previously I  use xlite client to suscribe to the presence notification of the devices, but recently added some devices which not supporting presence notification, so I want FreeSwitch to send an email notification when some user/extention are getting unregistered;
How can I capture a Feeswtich user unregister event & launch a LUA script then?
Happy days!
/Brgds, Alex 		 	   		  
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From anthony.minessale at gmail.com  Wed Dec 12 22:02:37 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 Dec 2012 13:02:37 -0600
Subject: [Freeswitch-users] Console FreeTDM commands get executed only
 once, following ftdm hang console
In-Reply-To: <CA+GmxbZBZ7Wm96qF5avNW0vmrDx+FDWk=UD=kAXHOeeCDaOpvQ@mail.gmail.com>
References: <CA+GmxbZBZ7Wm96qF5avNW0vmrDx+FDWk=UD=kAXHOeeCDaOpvQ@mail.gmail.com>
Message-ID: <CAKbxfG-MaLkmQnkquRJEu9g4puOHf51bv4iT8cv3WV87aYd8iA@mail.gmail.com>

When it's stuck get a gcore of the running process and get a backtrace.
http://wiki.freeswitch.org/wiki/Reporting_Bugs#gcore
attach that to jira


On Wed, Dec 12, 2012 at 9:03 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> Hi all!
>
> I have Digium TDM400P with DAHDI, FREETDM and libpri built and installed
> on git head. There is one librpi-based span configured. The problem is when
> I issue any ftdm-command it works once, and all subsequent ftdm commands
> hang and I get no response.
>
> I kill fs_cli, start it again and can do anything but ftdm commands - it
> hangs after any ftdm command.
>
> After that I can't unload mod_freetdm, as FS says it is in use.
>
> Is this a bug? What information should I collect to open JIRA?
>
> freeswitch at internal> ftdm list
> +OK
> span: 1 (trunk1)
> type: isdn
> physical_status: ok
> signaling_status: DOWN
> chan_count: 31
> dialplan: XML
> context: from_trunk1
> dial_regex:
> fail_dial_regex:
> hold_music:
> analog_options: none
>
> freeswitch at internal>
> freeswitch at internal> ftdm list
> <no response>
>
>
>
>
> --
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
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From steveayre at gmail.com  Wed Dec 12 22:47:07 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 12 Dec 2012 19:47:07 +0000
Subject: [Freeswitch-users] Email notification when user unregistered?
In-Reply-To: <BLU169-W7BDC7C356B8C56E4FC8E68C4F0@phx.gbl>
References: <BLU169-W7BDC7C356B8C56E4FC8E68C4F0@phx.gbl>
Message-ID: <CAFiqYum8ncL3xvNF9vuvx1bWO8MSOcy+Qzur+_-otKkb4kq-DQ@mail.gmail.com>

You'll need to manage your expectations here.

SIP registrations start when the client sends a REGISTER, with an expiry
time. They then periodically send additional REGISTERs to remain
registered. They can be spread quite a long way apart - several minutes.

A client explicitly unregisters by sending a REGISTER with a 0 expiry time,
ie the registration expires immediately.

If the client has gone offline because of reboot, network problem etc it
won't send that final REGISTER. You won't see the client unregister until
the end of the expiry time when the client hasn't renewed its registration.

Since that's several minutes later, there's no way to see that they have
unregistered *immediately*. You may be able to reduce the expiry time to
detect they're offline quicker, but that will increase bandwidth and load.



As for actually how to detect the unregistration... I suspect the 2 events
you want to listen for are sofia::unregister and sofia::expire. The names
suggest the 1st will be for an explicit unregister and the 2nd for when the
client fails to reregister within the expiry time. That means the 2nd'll be
the one you'd see if the client goes unexpectedly offline.


-Steve




On 12 December 2012 17:52, ?? <itispip-qq at hotmail.com> wrote:

>  Hi Freeswtich Master,
>
> I'm in a situation where I need to proactively moniter whether some
> device/extension is on in my local network; If for some reason the devices
> are offline, I need to be noticed immediately so that I can restart the
> device;
>
> Previously I  use xlite client to suscribe to the presence notification of
> the devices, but recently added some devices which not supporting presence
> notification, so I want FreeSwitch to send an email notification when some
> user/extention are getting unregistered;
>
> How can I capture a Feeswtich user unregister event & launch a LUA script
> then?
>
> Happy days!
>
> /Brgds, 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Dec 12 22:55:32 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 12 Dec 2012 19:55:32 +0000
Subject: [Freeswitch-users] Email notification when user unregistered?
In-Reply-To: <CAFiqYum8ncL3xvNF9vuvx1bWO8MSOcy+Qzur+_-otKkb4kq-DQ@mail.gmail.com>
References: <BLU169-W7BDC7C356B8C56E4FC8E68C4F0@phx.gbl>
	<CAFiqYum8ncL3xvNF9vuvx1bWO8MSOcy+Qzur+_-otKkb4kq-DQ@mail.gmail.com>
Message-ID: <CAFiqYumVhy2VTU=ow1B=U96PpaSg-OG-V0nHk7roSqVr7=0ZXw@mail.gmail.com>

Reading the source, FS will only periodically scan the database for expired
registrations and fire the sofia::expire event then. The period between
polling the database will cause a delay between the registration expiring
and the event being raised.

The default is to check every 30s, but can be adjusted with the sofia
profile param registration-thread-frequency. Remember lower values may
raise the event sooner, but will increase load because you're checking the
DB more often. Since the expiry is already going to be a certain amount
after the device disappears a few seconds probably won't make much
difference - 10s vs 30s may sometimes save up to 20s but if the expiry
period was 30 minutes that's not going to be enough to care about since it
may already have been offline for almost half an hour at that point.

The registration expiry time comes from the client, but you can override it
from FS:
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#sip-force-expires

-Steve


On 12 December 2012 19:47, Steven Ayre <steveayre at gmail.com> wrote:

> You'll need to manage your expectations here.
>
> SIP registrations start when the client sends a REGISTER, with an expiry
> time. They then periodically send additional REGISTERs to remain
> registered. They can be spread quite a long way apart - several minutes.
>
> A client explicitly unregisters by sending a REGISTER with a 0 expiry
> time, ie the registration expires immediately.
>
> If the client has gone offline because of reboot, network problem etc it
> won't send that final REGISTER. You won't see the client unregister until
> the end of the expiry time when the client hasn't renewed its registration.
>
> Since that's several minutes later, there's no way to see that they have
> unregistered *immediately*. You may be able to reduce the expiry time to
> detect they're offline quicker, but that will increase bandwidth and load.
>
>
>
> As for actually how to detect the unregistration... I suspect the 2 events
> you want to listen for are sofia::unregister and sofia::expire. The names
> suggest the 1st will be for an explicit unregister and the 2nd for when the
> client fails to reregister within the expiry time. That means the 2nd'll be
> the one you'd see if the client goes unexpectedly offline.
>
>
> -Steve
>
>
>
>
> On 12 December 2012 17:52, ?? <itispip-qq at hotmail.com> wrote:
>
>>  Hi Freeswtich Master,
>>
>> I'm in a situation where I need to proactively moniter whether some
>> device/extension is on in my local network; If for some reason the devices
>> are offline, I need to be noticed immediately so that I can restart the
>> device;
>>
>> Previously I  use xlite client to suscribe to the presence notification
>> of the devices, but recently added some devices which not supporting
>> presence notification, so I want FreeSwitch to send an email notification
>> when some user/extention are getting unregistered;
>>
>> How can I capture a Feeswtich user unregister event & launch a LUA script
>> then?
>>
>> Happy days!
>>
>> /Brgds, 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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  Wed Dec 12 22:57:47 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 12 Dec 2012 19:57:47 +0000
Subject: [Freeswitch-users] Console FreeTDM commands get executed only
 once, following ftdm hang console
In-Reply-To: <CAKbxfG-MaLkmQnkquRJEu9g4puOHf51bv4iT8cv3WV87aYd8iA@mail.gmail.com>
References: <CA+GmxbZBZ7Wm96qF5avNW0vmrDx+FDWk=UD=kAXHOeeCDaOpvQ@mail.gmail.com>
	<CAKbxfG-MaLkmQnkquRJEu9g4puOHf51bv4iT8cv3WV87aYd8iA@mail.gmail.com>
Message-ID: <CAFiqYukeFi4AwwR_mL7ExJpU0dN=wRA4Y1JBfMoG6dje5riCXQ@mail.gmail.com>

Dmitry, also see
http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29
That section explains in detail how to collect the backtrace from the file
gcore will produce.


On 12 December 2012 19:02, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> When it's stuck get a gcore of the running process and get a backtrace.
> http://wiki.freeswitch.org/wiki/Reporting_Bugs#gcore
> attach that to jira
>
>
> On Wed, Dec 12, 2012 at 9:03 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>
>> Hi all!
>>
>> I have Digium TDM400P with DAHDI, FREETDM and libpri built and installed
>> on git head. There is one librpi-based span configured. The problem is when
>> I issue any ftdm-command it works once, and all subsequent ftdm commands
>> hang and I get no response.
>>
>> I kill fs_cli, start it again and can do anything but ftdm commands - it
>> hangs after any ftdm command.
>>
>> After that I can't unload mod_freetdm, as FS says it is in use.
>>
>> Is this a bug? What information should I collect to open JIRA?
>>
>> freeswitch at internal> ftdm list
>> +OK
>> span: 1 (trunk1)
>> type: isdn
>> physical_status: ok
>> signaling_status: DOWN
>> chan_count: 31
>> dialplan: XML
>> context: from_trunk1
>> dial_regex:
>> fail_dial_regex:
>> hold_music:
>> analog_options: none
>>
>> freeswitch at internal>
>> freeswitch at internal> ftdm list
>> <no response>
>>
>>
>>
>>
>> --
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Dec 12 23:05:00 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 12 Dec 2012 20:05:00 +0000
Subject: [Freeswitch-users] ILBC takes precedence regardless of the
	codec prefs
In-Reply-To: <D37F8F9F-301A-4B29-9147-7C5E3E54C9DF@kavun.ch>
References: <D37F8F9F-301A-4B29-9147-7C5E3E54C9DF@kavun.ch>
Message-ID: <CAFiqYukUs23XqM8hhSxKAQPm9q8YcjHgb=CvH_xQ67dWK65z4Q@mail.gmail.com>

In your example FS is selecting the G711 alaw (PCMA) '8' codec. The client
offers it, the FS reply only returns the codec it has selected.

Try a debug level log of the call. It will show you a description of the
codec negotiation.

If you're not expecting the G711 codec to be accepted, check your codec
preferences. Inbound and outbound are separate parameters, and if you're
setting it from vars.conf.xml that sets a variable used in the sofia
profile... the profile could be using a static value that's different.

There are also other parameters that'll have an effect on a bridged call
such as late negotiation which would leave codec selection it up to the
bleg.



-Steve




On 11 December 2012 22:02, Emrah <lists at kavun.ch> wrote:

> Hi all,
>
> I am using FreeSWITCH Version 1.2.5.3+git~20121206T050429Z~91eef34d5c (git
> 91eef34 2012-12-06 05:04:29Z).
> When mod_ilbc is loaded, ILBC is used in any session where the client
> offers it, regardless of my codec preferences. This gets even stranger when
> you look at the SIP trace? FS doesn't offer ILBC. This is brand new, I
> wanted to enable ILBC just in case, and found out that I prefer it to G729.
>
> Here's some output of a trace:
> Client:
> v=0.
> o=- 3564251257 3564251257 IN IP4 10.0.0.102.
> s=pjmedia.
> c=IN IP4 10.0.0.102.
> t=0 0.
> a=X-nat:0.
> m=audio 4000 RTP/AVP 103 102 104 109 3 0 8 9 101.
> a=rtcp:4001 IN IP4 10.0.1.132.
> a=rtpmap:103 speex/16000.
> a=rtpmap:102 speex/8000.
> a=rtpmap:104 speex/32000.
> a=rtpmap:109 iLBC/8000.
> a=fmtp:109 mode=30.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:9 G722/8000.
> a=sendrecv.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
>
> FS:
> v=0.
> o=FreeSWITCH 1355242302 1355242303 IN IP4 1.2.3.4.
> s=FreeSWITCH.
> c=IN IP4 1.2.3.4.
> t=0 0.
> m=audio 20042 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
>
>
> Your input is greatly appreciated, as always.
>
> All the best,
> Emrah
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 avi at avimarcus.net  Wed Dec 12 23:12:43 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 12 Dec 2012 22:12:43 +0200
Subject: [Freeswitch-users] Email notification when user unregistered?
In-Reply-To: <CAFiqYumVhy2VTU=ow1B=U96PpaSg-OG-V0nHk7roSqVr7=0ZXw@mail.gmail.com>
References: <BLU169-W7BDC7C356B8C56E4FC8E68C4F0@phx.gbl>
	<CAFiqYum8ncL3xvNF9vuvx1bWO8MSOcy+Qzur+_-otKkb4kq-DQ@mail.gmail.com>
	<CAFiqYumVhy2VTU=ow1B=U96PpaSg-OG-V0nHk7roSqVr7=0ZXw@mail.gmail.com>
Message-ID: <CALNZuwnzvKFnHQ3hXc-KAk1nkZqVb6r1Q-ToHEkSs3YVz8R3Ag@mail.gmail.com>

That expiry event is something to watch.

Also, you might want to monitor your CDRs if a call to a registered device
doesn't respond:
if variables->progress_mediamsec + variables->progressmsec = 0, then that
means the remote destination never sent a progress media. Although, that
seems to also occur if it replies with a "user_busy" so I haven't fully
nailed this one down yet. Let me know if you figure that one out.

-Avi


On Wed, Dec 12, 2012 at 9:55 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Reading the source, FS will only periodically scan the database for
> expired registrations and fire the sofia::expire event then. The period
> between polling the database will cause a delay between the registration
> expiring and the event being raised.
>
> The default is to check every 30s, but can be adjusted with the sofia
> profile param registration-thread-frequency. Remember lower values may
> raise the event sooner, but will increase load because you're checking the
> DB more often. Since the expiry is already going to be a certain amount
> after the device disappears a few seconds probably won't make much
> difference - 10s vs 30s may sometimes save up to 20s but if the expiry
> period was 30 minutes that's not going to be enough to care about since it
> may already have been offline for almost half an hour at that point.
>
> The registration expiry time comes from the client, but you can override
> it from FS:
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#sip-force-expires
>
> -Steve
>
>
> On 12 December 2012 19:47, Steven Ayre <steveayre at gmail.com> wrote:
>
>> You'll need to manage your expectations here.
>>
>> SIP registrations start when the client sends a REGISTER, with an expiry
>> time. They then periodically send additional REGISTERs to remain
>> registered. They can be spread quite a long way apart - several minutes.
>>
>> A client explicitly unregisters by sending a REGISTER with a 0 expiry
>> time, ie the registration expires immediately.
>>
>> If the client has gone offline because of reboot, network problem etc it
>> won't send that final REGISTER. You won't see the client unregister until
>> the end of the expiry time when the client hasn't renewed its registration.
>>
>> Since that's several minutes later, there's no way to see that they have
>> unregistered *immediately*. You may be able to reduce the expiry time to
>> detect they're offline quicker, but that will increase bandwidth and load.
>>
>>
>>
>> As for actually how to detect the unregistration... I suspect the 2
>> events you want to listen for are sofia::unregister and sofia::expire. The
>> names suggest the 1st will be for an explicit unregister and the 2nd for
>> when the client fails to reregister within the expiry time. That means the
>> 2nd'll be the one you'd see if the client goes unexpectedly offline.
>>
>>
>> -Steve
>>
>>
>>
>>
>> On 12 December 2012 17:52, ?? <itispip-qq at hotmail.com> wrote:
>>
>>>  Hi Freeswtich Master,
>>>
>>> I'm in a situation where I need to proactively moniter whether some
>>> device/extension is on in my local network; If for some reason the devices
>>> are offline, I need to be noticed immediately so that I can restart the
>>> device;
>>>
>>> Previously I  use xlite client to suscribe to the presence notification
>>> of the devices, but recently added some devices which not supporting
>>> presence notification, so I want FreeSwitch to send an email notification
>>> when some user/extention are getting unregistered;
>>>
>>> How can I capture a Feeswtich user unregister event & launch a LUA
>>> script then?
>>>
>>> Happy days!
>>>
>>> /Brgds, 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 anthony.minessale at gmail.com  Wed Dec 12 23:13:29 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 Dec 2012 14:13:29 -0600
Subject: [Freeswitch-users] git IPv6 down?
In-Reply-To: <50C77E7D.1000600@communicatefreely.net>
References: <1FFF97C269757C458224B7C895F35F151E3575@cantor.std.visionutv.se>
	<50C77E7D.1000600@communicatefreely.net>
Message-ID: <CAKbxfG9xnptXBNNjpEEjhmaU0QkJxf1wT9mceDxAOwWjWrRUTQ@mail.gmail.com>

It's still administratively down from our providers since the snafu with
the DDoS.  It will probably resurface soon.


On Tue, Dec 11, 2012 at 12:42 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Likewise - I get "connection refused" at the IP address
> 2606:d900:0:24:1024:ff:fe00:1234
>
>
>
> Peter Olsson wrote:
> > Hi all,
> >
> > Ever since the DoS attack a couple of weeks ago, I can't access git over
> IPv6 anymore. Are there still issues that need to be resolved, or is it
> just me?
> >
> > It fallbacks to IPv4, so it works just fine, I just wanted to let you
> know.
> >
> > /Peter
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
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From sdevoy at bizfocused.com  Wed Dec 12 23:43:01 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 12 Dec 2012 15:43:01 -0500
Subject: [Freeswitch-users] Email notification when user unregistered?
In-Reply-To: <CALNZuwnzvKFnHQ3hXc-KAk1nkZqVb6r1Q-ToHEkSs3YVz8R3Ag@mail.gmail.com>
References: <BLU169-W7BDC7C356B8C56E4FC8E68C4F0@phx.gbl>	<CAFiqYum8ncL3xvNF9vuvx1bWO8MSOcy+Qzur+_-otKkb4kq-DQ@mail.gmail.com>	<CAFiqYumVhy2VTU=ow1B=U96PpaSg-OG-V0nHk7roSqVr7=0ZXw@mail.gmail.com>
	<CALNZuwnzvKFnHQ3hXc-KAk1nkZqVb6r1Q-ToHEkSs3YVz8R3Ag@mail.gmail.com>
Message-ID: <060301cdd8a9$47a794d0$d6f6be70$@bizfocused.com>

I have looked into this a few times, sip messaging does not lend itself well to detecting a missing client ? until you actually try to reach it!  I have pondered some kind of a SIP message to use like an IP ?ping?, but I have nothing yet.  Maybe someone who actually knows ?sip messaging? could suggest an innocuous sip message to send as a means of testing availability.

 

In the meantime, my best suggestion is a DIALPLAN that has some redundancy for failed bridge connections. That is if the initial bridge fails (continue on fail), send yourself an email indicating that EXT is DOWN and redirect the call with another bridge statement. It is Reactive rather than Proactive, but it is very simple to setup and extremely reliable.  You can even include playing a message like ?There is a problem reaching that extension, please hold while we redirect your call.?

 

Hope that helps.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus
Sent: Wednesday, December 12, 2012 3:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Email notification when user unregistered?

 

That expiry event is something to watch.

 

Also, you might want to monitor your CDRs if a call to a registered device doesn't respond:

if variables->progress_mediamsec + variables->progressmsec = 0, then that means the remote destination never sent a progress media. Although, that seems to also occur if it replies with a "user_busy" so I haven't fully nailed this one down yet. Let me know if you figure that one out.




-Avi

 

On Wed, Dec 12, 2012 at 9:55 PM, Steven Ayre <steveayre at gmail.com> wrote:

Reading the source, FS will only periodically scan the database for expired registrations and fire the sofia::expire event then. The period between polling the database will cause a delay between the registration expiring and the event being raised.

 

The default is to check every 30s, but can be adjusted with the sofia profile param registration-thread-frequency. Remember lower values may raise the event sooner, but will increase load because you're checking the DB more often. Since the expiry is already going to be a certain amount after the device disappears a few seconds probably won't make much difference - 10s vs 30s may sometimes save up to 20s but if the expiry period was 30 minutes that's not going to be enough to care about since it may already have been offline for almost half an hour at that point.

 

The registration expiry time comes from the client, but you can override it from FS:

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#sip-force-expires

 

-Steve

 

 

On 12 December 2012 19:47, Steven Ayre <steveayre at gmail.com> wrote:

You'll need to manage your expectations here.

 

SIP registrations start when the client sends a REGISTER, with an expiry time. They then periodically send additional REGISTERs to remain registered. They can be spread quite a long way apart - several minutes.

 

A client explicitly unregisters by sending a REGISTER with a 0 expiry time, ie the registration expires immediately.

 

If the client has gone offline because of reboot, network problem etc it won't send that final REGISTER. You won't see the client unregister until the end of the expiry time when the client hasn't renewed its registration.

 

Since that's several minutes later, there's no way to see that they have unregistered immediately. You may be able to reduce the expiry time to detect they're offline quicker, but that will increase bandwidth and load.

 

 

 

As for actually how to detect the unregistration... I suspect the 2 events you want to listen for are sofia::unregister and sofia::expire. The names suggest the 1st will be for an explicit unregister and the 2nd for when the client fails to reregister within the expiry time. That means the 2nd'll be the one you'd see if the client goes unexpectedly offline.

 

 

-Steve

 

 

 

 

On 12 December 2012 17:52, ?? <itispip-qq at hotmail.com> wrote:

Hi Freeswtich Master,

 

I'm in a situation where I need to proactively moniter whether some device/extension is on in my local network; If for some reason the devices are offline, I need to be noticed immediately so that I can restart the device;

 

Previously I  use xlite client to suscribe to the presence notification of the devices, but recently added some devices which not supporting presence notification, so I want FreeSwitch to send an email notification when some user/extention are getting unregistered;

 

How can I capture a Feeswtich user unregister event & launch a LUA script then?

 

Happy days!

 

/Brgds, Alex

 

_________________________________________________________________________
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/> 

FreeSWITCH-users mailing list
FreeSWITCH-users at 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 <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/> 

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

 

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From yungwei at resolvity.com  Thu Dec 13 00:29:42 2012
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 12 Dec 2012 16:29:42 -0500
Subject: [Freeswitch-users] Unable to install freeswitch-spidermonkey on
	CentOS 5 using yum
Message-ID: <33095823FD21DF429B481B5163264B799F3A97C073@VMBX102.ihostexchange.net>

Hi,

I just installed freeswitch-1.2.5.1-1.i386 on CentOS 5 using yum.
Now I have trouble installing freeswitch-spidermonkey.
How can I fix the problem? Thanks.

Here's the console output:
[root at templ0 freeswitch]# yum install freeswitch-spidermonkey --nogpgcheck
Loaded plugins: fastestmirror
Loading mirror speeds from cached hostfile
 * base: mirrors.einstein.yu.edu
 * updates: mirror.cogentco.com
 * extras: yum.singlehop.com
Setting up Install Process
Parsing package install arguments
Resolving Dependencies
--> Running transaction check
---> Package freeswitch-spidermonkey.i386 0:1.2.5.1-1 set to be updated
--> Finished Dependency Resolution

Dependencies Resolved

===============================================================================================================================================================================================================
 Package                                                      Arch                                      Version                                          Repository                                       Size
===============================================================================================================================================================================================================
Installing:
 freeswitch-spidermonkey                                      i386                                      1.2.5.1-1                                        freeswitch                                      2.0 M

Transaction Summary
===============================================================================================================================================================================================================
Install      1 Package(s)
Update       0 Package(s)
Remove       0 Package(s)

Total size: 2.0 M
Is this ok [y/N]: y
Downloading Packages:
Running rpm_check_debug
Running Transaction Test
Finished Transaction Test


Transaction Check Error:
  file /usr/lib/libnspr4.so from install of freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package nspr-4.9.1-4.el5_8.i386
  file /usr/lib/libplc4.so from install of freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package nspr-4.9.1-4.el5_8.i386
  file /usr/lib/libplds4.so from install of freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package nspr-4.9.1-4.el5_8.i386

Error Summary
-------------

Here's the version of nspr installed the machine.
[root at templ0 freeswitch]# rpm -q nspr
nspr-4.9.1-4.el5_8

Here's a list of things I installed earlier:
rpm -Uvh http://files.freeswitch.org/freeswitch-release-1-0.noarch.rpm
yum install freeswitch-config-vanilla --nogpgcheck 



From anthony.minessale at gmail.com  Thu Dec 13 00:42:52 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 Dec 2012 15:42:52 -0600
Subject: [Freeswitch-users] mod_perl - Can't call method "serialize" on
 an undefined value
In-Reply-To: <ADD8CA27-CB8D-4D61-BE4F-F03DFB354746@kavun.ch>
References: <ADD8CA27-CB8D-4D61-BE4F-F03DFB354746@kavun.ch>
Message-ID: <CAKbxfG-vmXgt4rAuAGBVXOrigssqMFvjChaZoTSw=vqN-y=2SQ@mail.gmail.com>

$env only exists when you call it from the cli or FSAPI via the "perl"
function.



On Tue, Dec 11, 2012 at 6:17 PM, Emrah <lists at kavun.ch> wrote:

>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-November/036404.html




-- 
Anthony Minessale II

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
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From yungwei at resolvity.com  Thu Dec 13 00:47:58 2012
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 12 Dec 2012 16:47:58 -0500
Subject: [Freeswitch-users] Unable to install freeswitch-spidermonkey on
 CentOS 5 using yum
In-Reply-To: <33095823FD21DF429B481B5163264B799F3A97C073@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B799F3A97C073@VMBX102.ihostexchange.net>
Message-ID: <33095823FD21DF429B481B5163264B799F3A97C078@VMBX102.ihostexchange.net>

I found an existing bug for this.
http://jira.freeswitch.org/browse/FS-4653

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen
Sent: Wednesday, December 12, 2012 3:30 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Unable to install freeswitch-spidermonkey on CentOS 5 using yum

Hi,

I just installed freeswitch-1.2.5.1-1.i386 on CentOS 5 using yum.
Now I have trouble installing freeswitch-spidermonkey.
How can I fix the problem? Thanks.

Here's the console output:
[root at templ0 freeswitch]# yum install freeswitch-spidermonkey --nogpgcheck
Loaded plugins: fastestmirror
Loading mirror speeds from cached hostfile
 * base: mirrors.einstein.yu.edu
 * updates: mirror.cogentco.com
 * extras: yum.singlehop.com
Setting up Install Process
Parsing package install arguments
Resolving Dependencies
--> Running transaction check
---> Package freeswitch-spidermonkey.i386 0:1.2.5.1-1 set to be updated
--> Finished Dependency Resolution

Dependencies Resolved

===============================================================================================================================================================================================================
 Package                                                      Arch                                      Version                                          Repository                                       Size
===============================================================================================================================================================================================================
Installing:
 freeswitch-spidermonkey                                      i386                                      1.2.5.1-1                                        freeswitch                                      2.0 M

Transaction Summary
===============================================================================================================================================================================================================
Install      1 Package(s)
Update       0 Package(s)
Remove       0 Package(s)

Total size: 2.0 M
Is this ok [y/N]: y
Downloading Packages:
Running rpm_check_debug
Running Transaction Test
Finished Transaction Test


Transaction Check Error:
  file /usr/lib/libnspr4.so from install of freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package nspr-4.9.1-4.el5_8.i386
  file /usr/lib/libplc4.so from install of freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package nspr-4.9.1-4.el5_8.i386
  file /usr/lib/libplds4.so from install of freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package nspr-4.9.1-4.el5_8.i386

Error Summary
-------------

Here's the version of nspr installed the machine.
[root at templ0 freeswitch]# rpm -q nspr
nspr-4.9.1-4.el5_8

Here's a list of things I installed earlier:
rpm -Uvh http://files.freeswitch.org/freeswitch-release-1-0.noarch.rpm
yum install freeswitch-config-vanilla --nogpgcheck 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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


From vetali100 at gmail.com  Thu Dec 13 00:49:34 2012
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Wed, 12 Dec 2012 13:49:34 -0800
Subject: [Freeswitch-users] Email notification when user unregistered?
In-Reply-To: <060301cdd8a9$47a794d0$d6f6be70$@bizfocused.com>
References: <BLU169-W7BDC7C356B8C56E4FC8E68C4F0@phx.gbl>
	<CAFiqYum8ncL3xvNF9vuvx1bWO8MSOcy+Qzur+_-otKkb4kq-DQ@mail.gmail.com>
	<CAFiqYumVhy2VTU=ow1B=U96PpaSg-OG-V0nHk7roSqVr7=0ZXw@mail.gmail.com>
	<CALNZuwnzvKFnHQ3hXc-KAk1nkZqVb6r1Q-ToHEkSs3YVz8R3Ag@mail.gmail.com>
	<060301cdd8a9$47a794d0$d6f6be70$@bizfocused.com>
Message-ID: <CACc3D0WbM-WXO870HHOOcNjKOomC7Wtg+7Wq1GR-bOK64BLK=Q@mail.gmail.com>

Redirect where? :)
(Assuming user did not setup any additional phone numbers where he can be
reached).


2012/12/12 Sean Devoy <sdevoy at bizfocused.com>

> I have looked into this a few times, sip messaging does not lend itself
> well to detecting a missing client ? until you actually try to reach it!  I
> have pondered some kind of a SIP message to use like an IP ?ping?, but I
> have nothing yet.  Maybe someone who actually knows ?sip messaging? could
> suggest an innocuous sip message to send as a means of testing availability.
> ****
>
> ** **
>
> In the meantime, my best suggestion is a DIALPLAN that has some redundancy
> for failed bridge connections. That is if the initial bridge fails
> (continue on fail), send yourself an email indicating that EXT is DOWN and
> redirect the call with another bridge statement. It is Reactive rather than
> Proactive, but it is very simple to setup and extremely reliable.  You can
> even include playing a message like ?There is a problem reaching that
> extension, please hold while we redirect your call.?****
>
> ** **
>
> Hope that helps.****
>
> ** **
>
> Sean****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
> *Sent:* Wednesday, December 12, 2012 3:13 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Email notification when user
> unregistered?****
>
> ** **
>
> That expiry event is something to watch.****
>
> ** **
>
> Also, you might want to monitor your CDRs if a call to a registered device
> doesn't respond:****
>
> if variables->progress_mediamsec + variables->progressmsec = 0, then that
> means the remote destination never sent a progress media. Although, that
> seems to also occur if it replies with a "user_busy" so I haven't fully
> nailed this one down yet. Let me know if you figure that one out.****
>
>
> ****
>
> -Avi****
>
> ** **
>
> On Wed, Dec 12, 2012 at 9:55 PM, Steven Ayre <steveayre at gmail.com> wrote:*
> ***
>
> Reading the source, FS will only periodically scan the database for
> expired registrations and fire the sofia::expire event then. The period
> between polling the database will cause a delay between the registration
> expiring and the event being raised.****
>
> ** **
>
> The default is to check every 30s, but can be adjusted with the sofia
> profile param registration-thread-frequency. Remember lower values may
> raise the event sooner, but will increase load because you're checking the
> DB more often. Since the expiry is already going to be a certain amount
> after the device disappears a few seconds probably won't make much
> difference - 10s vs 30s may sometimes save up to 20s but if the expiry
> period was 30 minutes that's not going to be enough to care about since it
> may already have been offline for almost half an hour at that point.****
>
> ** **
>
> The registration expiry time comes from the client, but you can override
> it from FS:****
>
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#sip-force-expires
> ****
>
> ** **
>
> -Steve****
>
> ** **
>
> ** **
>
> On 12 December 2012 19:47, Steven Ayre <steveayre at gmail.com> wrote:****
>
> You'll need to manage your expectations here.****
>
> ** **
>
> SIP registrations start when the client sends a REGISTER, with an expiry
> time. They then periodically send additional REGISTERs to remain
> registered. They can be spread quite a long way apart - several minutes.**
> **
>
> ** **
>
> A client explicitly unregisters by sending a REGISTER with a 0 expiry
> time, ie the registration expires immediately.****
>
> ** **
>
> If the client has gone offline because of reboot, network problem etc it
> won't send that final REGISTER. You won't see the client unregister until
> the end of the expiry time when the client hasn't renewed its registration.
> ****
>
> ** **
>
> Since that's several minutes later, there's no way to see that they have
> unregistered *immediately*. You may be able to reduce the expiry time to
> detect they're offline quicker, but that will increase bandwidth and load.
> ****
>
> ** **
>
> ** **
>
> ** **
>
> As for actually how to detect the unregistration... I suspect the 2 events
> you want to listen for are sofia::unregister and sofia::expire. The names
> suggest the 1st will be for an explicit unregister and the 2nd for when the
> client fails to reregister within the expiry time. That means the 2nd'll be
> the one you'd see if the client goes unexpectedly offline.****
>
> ** **
>
> ** **
>
> -Steve****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> On 12 December 2012 17:52, ?? <itispip-qq at hotmail.com> wrote:****
>
> Hi Freeswtich Master,****
>
> ** **
>
> I'm in a situation where I need to proactively moniter whether some
> device/extension is on in my local network; If for some reason the devices
> are offline, I need to be noticed immediately so that I can restart the
> device;****
>
> ** **
>
> Previously I  use xlite client to suscribe to the presence notification of
> the devices, but recently added some devices which not supporting presence
> notification, so I want FreeSwitch to send an email notification when some
> user/extention are getting unregistered;****
>
> ** **
>
> How can I capture a Feeswtich user unregister event & launch a LUA script
> then?****
>
> ** **
>
> Happy days!****
>
> ** **
>
> /Brgds, 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> ** **
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 13 00:56:30 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 12 Dec 2012 21:56:30 +0000
Subject: [Freeswitch-users] Email notification when user unregistered?
In-Reply-To: <060301cdd8a9$47a794d0$d6f6be70$@bizfocused.com>
References: <BLU169-W7BDC7C356B8C56E4FC8E68C4F0@phx.gbl>
	<CAFiqYum8ncL3xvNF9vuvx1bWO8MSOcy+Qzur+_-otKkb4kq-DQ@mail.gmail.com>
	<CAFiqYumVhy2VTU=ow1B=U96PpaSg-OG-V0nHk7roSqVr7=0ZXw@mail.gmail.com>
	<CALNZuwnzvKFnHQ3hXc-KAk1nkZqVb6r1Q-ToHEkSs3YVz8R3Ag@mail.gmail.com>
	<060301cdd8a9$47a794d0$d6f6be70$@bizfocused.com>
Message-ID: <CAFiqYunL_sMJhF9F+k2m3z3pfNQQPWBw6sbc6D55M5_eTf-7eQ@mail.gmail.com>

Sofia's gateways achieve this by doing a SIP OPTIONS ping. Gateway is
marked as down if several timeout or return errors in a row (1 on its own
isn't enough as it's a single UDP packet - you don't want to mark it as
down if a single packet is lost).

That's only for outbound registrations though, AFAIK FS cannot do the same
for inbound registrations. Though it could, in theory.

The time to detect they're offline will still not be immediate though - you
need to periodically do the OPTIONS ping and have several fail before
marking it as unregistered - so it'd still be say half a minute before FS
saw it had gone.

-Steve




On 12 December 2012 20:43, Sean Devoy <sdevoy at bizfocused.com> wrote:

> I have looked into this a few times, sip messaging does not lend itself
> well to detecting a missing client ? until you actually try to reach it!  I
> have pondered some kind of a SIP message to use like an IP ?ping?, but I
> have nothing yet.  Maybe someone who actually knows ?sip messaging? could
> suggest an innocuous sip message to send as a means of testing availability.
> ****
>
> ** **
>
> In the meantime, my best suggestion is a DIALPLAN that has some redundancy
> for failed bridge connections. That is if the initial bridge fails
> (continue on fail), send yourself an email indicating that EXT is DOWN and
> redirect the call with another bridge statement. It is Reactive rather than
> Proactive, but it is very simple to setup and extremely reliable.  You can
> even include playing a message like ?There is a problem reaching that
> extension, please hold while we redirect your call.?****
>
> ** **
>
> Hope that helps.****
>
> ** **
>
> Sean****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
> *Sent:* Wednesday, December 12, 2012 3:13 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Email notification when user
> unregistered?****
>
> ** **
>
> That expiry event is something to watch.****
>
> ** **
>
> Also, you might want to monitor your CDRs if a call to a registered device
> doesn't respond:****
>
> if variables->progress_mediamsec + variables->progressmsec = 0, then that
> means the remote destination never sent a progress media. Although, that
> seems to also occur if it replies with a "user_busy" so I haven't fully
> nailed this one down yet. Let me know if you figure that one out.****
>
>
> ****
>
> -Avi****
>
> ** **
>
> On Wed, Dec 12, 2012 at 9:55 PM, Steven Ayre <steveayre at gmail.com> wrote:*
> ***
>
> Reading the source, FS will only periodically scan the database for
> expired registrations and fire the sofia::expire event then. The period
> between polling the database will cause a delay between the registration
> expiring and the event being raised.****
>
> ** **
>
> The default is to check every 30s, but can be adjusted with the sofia
> profile param registration-thread-frequency. Remember lower values may
> raise the event sooner, but will increase load because you're checking the
> DB more often. Since the expiry is already going to be a certain amount
> after the device disappears a few seconds probably won't make much
> difference - 10s vs 30s may sometimes save up to 20s but if the expiry
> period was 30 minutes that's not going to be enough to care about since it
> may already have been offline for almost half an hour at that point.****
>
> ** **
>
> The registration expiry time comes from the client, but you can override
> it from FS:****
>
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#sip-force-expires
> ****
>
> ** **
>
> -Steve****
>
> ** **
>
> ** **
>
> On 12 December 2012 19:47, Steven Ayre <steveayre at gmail.com> wrote:****
>
> You'll need to manage your expectations here.****
>
> ** **
>
> SIP registrations start when the client sends a REGISTER, with an expiry
> time. They then periodically send additional REGISTERs to remain
> registered. They can be spread quite a long way apart - several minutes.**
> **
>
> ** **
>
> A client explicitly unregisters by sending a REGISTER with a 0 expiry
> time, ie the registration expires immediately.****
>
> ** **
>
> If the client has gone offline because of reboot, network problem etc it
> won't send that final REGISTER. You won't see the client unregister until
> the end of the expiry time when the client hasn't renewed its registration.
> ****
>
> ** **
>
> Since that's several minutes later, there's no way to see that they have
> unregistered *immediately*. You may be able to reduce the expiry time to
> detect they're offline quicker, but that will increase bandwidth and load.
> ****
>
> ** **
>
> ** **
>
> ** **
>
> As for actually how to detect the unregistration... I suspect the 2 events
> you want to listen for are sofia::unregister and sofia::expire. The names
> suggest the 1st will be for an explicit unregister and the 2nd for when the
> client fails to reregister within the expiry time. That means the 2nd'll be
> the one you'd see if the client goes unexpectedly offline.****
>
> ** **
>
> ** **
>
> -Steve****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> On 12 December 2012 17:52, ?? <itispip-qq at hotmail.com> wrote:****
>
> Hi Freeswtich Master,****
>
> ** **
>
> I'm in a situation where I need to proactively moniter whether some
> device/extension is on in my local network; If for some reason the devices
> are offline, I need to be noticed immediately so that I can restart the
> device;****
>
> ** **
>
> Previously I  use xlite client to suscribe to the presence notification of
> the devices, but recently added some devices which not supporting presence
> notification, so I want FreeSwitch to send an email notification when some
> user/extention are getting unregistered;****
>
> ** **
>
> How can I capture a Feeswtich user unregister event & launch a LUA script
> then?****
>
> ** **
>
> Happy days!****
>
> ** **
>
> /Brgds, 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> ** **
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 13 01:15:42 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 12 Dec 2012 17:15:42 -0500
Subject: [Freeswitch-users] Email notification when user unregistered?
In-Reply-To: <CACc3D0WbM-WXO870HHOOcNjKOomC7Wtg+7Wq1GR-bOK64BLK=Q@mail.gmail.com>
References: <BLU169-W7BDC7C356B8C56E4FC8E68C4F0@phx.gbl>	<CAFiqYum8ncL3xvNF9vuvx1bWO8MSOcy+Qzur+_-otKkb4kq-DQ@mail.gmail.com>	<CAFiqYumVhy2VTU=ow1B=U96PpaSg-OG-V0nHk7roSqVr7=0ZXw@mail.gmail.com>	<CALNZuwnzvKFnHQ3hXc-KAk1nkZqVb6r1Q-ToHEkSs3YVz8R3Ag@mail.gmail.com>	<060301cdd8a9$47a794d0$d6f6be70$@bizfocused.com>
	<CACc3D0WbM-WXO870HHOOcNjKOomC7Wtg+7Wq1GR-bOK64BLK=Q@mail.gmail.com>
Message-ID: <070801cdd8b6$3980eb10$ac82c130$@bizfocused.com>

In our case we can redirect to someone else in that department, if that fails, we redirect to a specific cell phone and if that fails this customer has us redirect to an answering service!  This customer REALY wants his callers to speak to a real person.

 

If you really have no one else who could handle the call, you could ?say? that extension is currently unavailable and take a message, but in the meantime text, page or email an emergency contact.  What do you do if the extension is unattended (no answer) or busy.

 

HTH

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vitalie Colosov
Sent: Wednesday, December 12, 2012 4:50 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Email notification when user unregistered?

 

Redirect where? :)

(Assuming user did not setup any additional phone numbers where he can be reached).

 

2012/12/12 Sean Devoy <sdevoy at bizfocused.com>

I have looked into this a few times, sip messaging does not lend itself well to detecting a missing client ? until you actually try to reach it!  I have pondered some kind of a SIP message to use like an IP ?ping?, but I have nothing yet.  Maybe someone who actually knows ?sip messaging? could suggest an innocuous sip message to send as a means of testing availability.

 

In the meantime, my best suggestion is a DIALPLAN that has some redundancy for failed bridge connections. That is if the initial bridge fails (continue on fail), send yourself an email indicating that EXT is DOWN and redirect the call with another bridge statement. It is Reactive rather than Proactive, but it is very simple to setup and extremely reliable.  You can even include playing a message like ?There is a problem reaching that extension, please hold while we redirect your call.?

 

Hope that helps.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus
Sent: Wednesday, December 12, 2012 3:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Email notification when user unregistered?

 

That expiry event is something to watch.

 

Also, you might want to monitor your CDRs if a call to a registered device doesn't respond:

if variables->progress_mediamsec + variables->progressmsec = 0, then that means the remote destination never sent a progress media. Although, that seems to also occur if it replies with a "user_busy" so I haven't fully nailed this one down yet. Let me know if you figure that one out.




-Avi

 

On Wed, Dec 12, 2012 at 9:55 PM, Steven Ayre <steveayre at gmail.com> wrote:

Reading the source, FS will only periodically scan the database for expired registrations and fire the sofia::expire event then. The period between polling the database will cause a delay between the registration expiring and the event being raised.

 

The default is to check every 30s, but can be adjusted with the sofia profile param registration-thread-frequency. Remember lower values may raise the event sooner, but will increase load because you're checking the DB more often. Since the expiry is already going to be a certain amount after the device disappears a few seconds probably won't make much difference - 10s vs 30s may sometimes save up to 20s but if the expiry period was 30 minutes that's not going to be enough to care about since it may already have been offline for almost half an hour at that point.

 

The registration expiry time comes from the client, but you can override it from FS:

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#sip-force-expires

 

-Steve

 

 

On 12 December 2012 19:47, Steven Ayre <steveayre at gmail.com> wrote:

You'll need to manage your expectations here.

 

SIP registrations start when the client sends a REGISTER, with an expiry time. They then periodically send additional REGISTERs to remain registered. They can be spread quite a long way apart - several minutes.

 

A client explicitly unregisters by sending a REGISTER with a 0 expiry time, ie the registration expires immediately.

 

If the client has gone offline because of reboot, network problem etc it won't send that final REGISTER. You won't see the client unregister until the end of the expiry time when the client hasn't renewed its registration.

 

Since that's several minutes later, there's no way to see that they have unregistered immediately. You may be able to reduce the expiry time to detect they're offline quicker, but that will increase bandwidth and load.

 

 

 

As for actually how to detect the unregistration... I suspect the 2 events you want to listen for are sofia::unregister and sofia::expire. The names suggest the 1st will be for an explicit unregister and the 2nd for when the client fails to reregister within the expiry time. That means the 2nd'll be the one you'd see if the client goes unexpectedly offline.

 

 

-Steve

 

 

 

 

On 12 December 2012 17:52, ?? <itispip-qq at hotmail.com> wrote:

Hi Freeswtich Master,

 

I'm in a situation where I need to proactively moniter whether some device/extension is on in my local network; If for some reason the devices are offline, I need to be noticed immediately so that I can restart the device;

 

Previously I  use xlite client to suscribe to the presence notification of the devices, but recently added some devices which not supporting presence notification, so I want FreeSwitch to send an email notification when some user/extention are getting unregistered;

 

How can I capture a Feeswtich user unregister event & launch a LUA script then?

 

Happy days!

 

/Brgds, Alex

 

_________________________________________________________________________
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/> 

FreeSWITCH-users mailing list
FreeSWITCH-users at 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 <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/> 

FreeSWITCH-users mailing list
FreeSWITCH-users at 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

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

 

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From krice at freeswitch.org  Thu Dec 13 01:29:25 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 12 Dec 2012 16:29:25 -0600
Subject: [Freeswitch-users] Unable to install freeswitch-spidermonkey on
 CentOS 5 using yum
In-Reply-To: <33095823FD21DF429B481B5163264B799F3A97C078@VMBX102.ihostexchange.net>
Message-ID: <CCEE6165.61EB3%krice@freeswitch.org>

Do you actually use mod_spidermonkey? (you will know if you do)

If not then there is not worry about not installing it. It is a known
conflict with the nspr as we need a customized version of those libraries
for FreeSWITCH...

At some point we may change that but at this time, I don't see that conflict
going away.

I would venture to say that it is recommended to use something like mod_lua
where you would want to use the javascript stuff for a number of a reasons


On 12/12/12 3:47 PM, "Yungwei Chen" <yungwei at resolvity.com> wrote:

> I found an existing bug for this.
> http://jira.freeswitch.org/browse/FS-4653
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei
> Chen
> Sent: Wednesday, December 12, 2012 3:30 PM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Unable to install freeswitch-spidermonkey on
> CentOS 5 using yum
> 
> Hi,
> 
> I just installed freeswitch-1.2.5.1-1.i386 on CentOS 5 using yum.
> Now I have trouble installing freeswitch-spidermonkey.
> How can I fix the problem? Thanks.
> 
> Here's the console output:
> [root at templ0 freeswitch]# yum install freeswitch-spidermonkey --nogpgcheck
> Loaded plugins: fastestmirror
> Loading mirror speeds from cached hostfile
>  * base: mirrors.einstein.yu.edu
>  * updates: mirror.cogentco.com
>  * extras: yum.singlehop.com
> Setting up Install Process
> Parsing package install arguments
> Resolving Dependencies
> --> Running transaction check
> ---> Package freeswitch-spidermonkey.i386 0:1.2.5.1-1 set to be updated
> --> Finished Dependency Resolution
> 
> Dependencies Resolved
> 
> ==============================================================================
> ==============================================================================
> ===================================================
>  Package                                                      Arch
> Version                                          Repository
> Size
> ==============================================================================
> ==============================================================================
> ===================================================
> Installing:
>  freeswitch-spidermonkey                                      i386
> 1.2.5.1-1                                        freeswitch
> 2.0 M
> 
> Transaction Summary
> ==============================================================================
> ==============================================================================
> ===================================================
> Install      1 Package(s)
> Update       0 Package(s)
> Remove       0 Package(s)
> 
> Total size: 2.0 M
> Is this ok [y/N]: y
> Downloading Packages:
> Running rpm_check_debug
> Running Transaction Test
> Finished Transaction Test
> 
> 
> Transaction Check Error:
>   file /usr/lib/libnspr4.so from install of
> freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package
> nspr-4.9.1-4.el5_8.i386
>   file /usr/lib/libplc4.so from install of
> freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package
> nspr-4.9.1-4.el5_8.i386
>   file /usr/lib/libplds4.so from install of
> freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package
> nspr-4.9.1-4.el5_8.i386
> 
> Error Summary
> -------------
> 
> Here's the version of nspr installed the machine.
> [root at templ0 freeswitch]# rpm -q nspr
> nspr-4.9.1-4.el5_8
> 
> Here's a list of things I installed earlier:
> rpm -Uvh http://files.freeswitch.org/freeswitch-release-1-0.noarch.rpm
> yum install freeswitch-config-vanilla --nogpgcheck
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From spencer at 5ninesolutions.com  Thu Dec 13 01:40:45 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 12 Dec 2012 14:40:45 -0800
Subject: [Freeswitch-users] [OT] Media Gateways in Hawaii
Message-ID: <90F01A96-D656-4C57-AC34-C4FA6279D7B6@5ninesolutions.com>

Hey Guys,
I know this is a bit off topic but we have a few clients in Hawaii and we're trying to reduce latency.  If anyone knows of a decent SIP termination provider with media gateways in Hawaii, please let me know.  Currently the traffic gets backhauled across the US to the PSTN and then back to Hawaii for a local call!

Thanks in advance!
Spencer



From yungwei at resolvity.com  Thu Dec 13 01:54:38 2012
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 12 Dec 2012 17:54:38 -0500
Subject: [Freeswitch-users] Unable to install freeswitch-spidermonkey on
 CentOS 5 using yum
In-Reply-To: <CCEE6165.61EB3%krice@freeswitch.org>
References: <33095823FD21DF429B481B5163264B799F3A97C078@VMBX102.ihostexchange.net>
	<CCEE6165.61EB3%krice@freeswitch.org>
Message-ID: <33095823FD21DF429B481B5163264B799F3A97C08F@VMBX102.ihostexchange.net>

Unfortunately I do.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Wednesday, December 12, 2012 4:29 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Unable to install freeswitch-spidermonkey on CentOS 5 using yum

Do you actually use mod_spidermonkey? (you will know if you do)

If not then there is not worry about not installing it. It is a known
conflict with the nspr as we need a customized version of those libraries
for FreeSWITCH...

At some point we may change that but at this time, I don't see that conflict
going away.

I would venture to say that it is recommended to use something like mod_lua
where you would want to use the javascript stuff for a number of a reasons


On 12/12/12 3:47 PM, "Yungwei Chen" <yungwei at resolvity.com> wrote:

> I found an existing bug for this.
> http://jira.freeswitch.org/browse/FS-4653
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei
> Chen
> Sent: Wednesday, December 12, 2012 3:30 PM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Unable to install freeswitch-spidermonkey on
> CentOS 5 using yum
> 
> Hi,
> 
> I just installed freeswitch-1.2.5.1-1.i386 on CentOS 5 using yum.
> Now I have trouble installing freeswitch-spidermonkey.
> How can I fix the problem? Thanks.
> 
> Here's the console output:
> [root at templ0 freeswitch]# yum install freeswitch-spidermonkey --nogpgcheck
> Loaded plugins: fastestmirror
> Loading mirror speeds from cached hostfile
>  * base: mirrors.einstein.yu.edu
>  * updates: mirror.cogentco.com
>  * extras: yum.singlehop.com
> Setting up Install Process
> Parsing package install arguments
> Resolving Dependencies
> --> Running transaction check
> ---> Package freeswitch-spidermonkey.i386 0:1.2.5.1-1 set to be updated
> --> Finished Dependency Resolution
> 
> Dependencies Resolved
> 
> ==============================================================================
> ==============================================================================
> ===================================================
>  Package                                                      Arch
> Version                                          Repository
> Size
> ==============================================================================
> ==============================================================================
> ===================================================
> Installing:
>  freeswitch-spidermonkey                                      i386
> 1.2.5.1-1                                        freeswitch
> 2.0 M
> 
> Transaction Summary
> ==============================================================================
> ==============================================================================
> ===================================================
> Install      1 Package(s)
> Update       0 Package(s)
> Remove       0 Package(s)
> 
> Total size: 2.0 M
> Is this ok [y/N]: y
> Downloading Packages:
> Running rpm_check_debug
> Running Transaction Test
> Finished Transaction Test
> 
> 
> Transaction Check Error:
>   file /usr/lib/libnspr4.so from install of
> freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package
> nspr-4.9.1-4.el5_8.i386
>   file /usr/lib/libplc4.so from install of
> freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package
> nspr-4.9.1-4.el5_8.i386
>   file /usr/lib/libplds4.so from install of
> freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package
> nspr-4.9.1-4.el5_8.i386
> 
> Error Summary
> -------------
> 
> Here's the version of nspr installed the machine.
> [root at templ0 freeswitch]# rpm -q nspr
> nspr-4.9.1-4.el5_8
> 
> Here's a list of things I installed earlier:
> rpm -Uvh http://files.freeswitch.org/freeswitch-release-1-0.noarch.rpm
> yum install freeswitch-config-vanilla --nogpgcheck
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #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

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


From krice at freeswitch.org  Thu Dec 13 02:55:47 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 12 Dec 2012 17:55:47 -0600
Subject: [Freeswitch-users] Unable to install freeswitch-spidermonkey on
 CentOS 5 using yum
In-Reply-To: <33095823FD21DF429B481B5163264B799F3A97C08F@VMBX102.ihostexchange.net>
Message-ID: <CCEE75A3.61EB9%krice@freeswitch.org>

The only viable work around at this point is to build FS from source, and
install everything into like /usr/local/freeswitch as is the default when
building from source...


On 12/12/12 4:54 PM, "Yungwei Chen" <yungwei at resolvity.com> wrote:

> Unfortunately I do.
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
> Sent: Wednesday, December 12, 2012 4:29 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Unable to install freeswitch-spidermonkey on
> CentOS 5 using yum
> 
> Do you actually use mod_spidermonkey? (you will know if you do)
> 
> If not then there is not worry about not installing it. It is a known
> conflict with the nspr as we need a customized version of those libraries
> for FreeSWITCH...
> 
> At some point we may change that but at this time, I don't see that conflict
> going away.
> 
> I would venture to say that it is recommended to use something like mod_lua
> where you would want to use the javascript stuff for a number of a reasons
> 
> 
> On 12/12/12 3:47 PM, "Yungwei Chen" <yungwei at resolvity.com> wrote:
> 
>> I found an existing bug for this.
>> http://jira.freeswitch.org/browse/FS-4653
>> 
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei
>> Chen
>> Sent: Wednesday, December 12, 2012 3:30 PM
>> To: FreeSWITCH Users Help
>> Subject: [Freeswitch-users] Unable to install freeswitch-spidermonkey on
>> CentOS 5 using yum
>> 
>> Hi,
>> 
>> I just installed freeswitch-1.2.5.1-1.i386 on CentOS 5 using yum.
>> Now I have trouble installing freeswitch-spidermonkey.
>> How can I fix the problem? Thanks.
>> 
>> Here's the console output:
>> [root at templ0 freeswitch]# yum install freeswitch-spidermonkey --nogpgcheck
>> Loaded plugins: fastestmirror
>> Loading mirror speeds from cached hostfile
>>  * base: mirrors.einstein.yu.edu
>>  * updates: mirror.cogentco.com
>>  * extras: yum.singlehop.com
>> Setting up Install Process
>> Parsing package install arguments
>> Resolving Dependencies
>> --> Running transaction check
>> ---> Package freeswitch-spidermonkey.i386 0:1.2.5.1-1 set to be updated
>> --> Finished Dependency Resolution
>> 
>> Dependencies Resolved
>> 
>> 
=============================================================================>>
=
>> 
=============================================================================>>
=
>> ===================================================
>>  Package                                                      Arch
>> Version                                          Repository
>> Size
>> 
=============================================================================>>
=
>> 
=============================================================================>>
=
>> ===================================================
>> Installing:
>>  freeswitch-spidermonkey                                      i386
>> 1.2.5.1-1                                        freeswitch
>> 2.0 M
>> 
>> Transaction Summary
>> 
=============================================================================>>
=
>> 
=============================================================================>>
=
>> ===================================================
>> Install      1 Package(s)
>> Update       0 Package(s)
>> Remove       0 Package(s)
>> 
>> Total size: 2.0 M
>> Is this ok [y/N]: y
>> Downloading Packages:
>> Running rpm_check_debug
>> Running Transaction Test
>> Finished Transaction Test
>> 
>> 
>> Transaction Check Error:
>>   file /usr/lib/libnspr4.so from install of
>> freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package
>> nspr-4.9.1-4.el5_8.i386
>>   file /usr/lib/libplc4.so from install of
>> freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package
>> nspr-4.9.1-4.el5_8.i386
>>   file /usr/lib/libplds4.so from install of
>> freeswitch-spidermonkey-1.2.5.1-1.i386 conflicts with file from package
>> nspr-4.9.1-4.el5_8.i386
>> 
>> Error Summary
>> -------------
>> 
>> Here's the version of nspr installed the machine.
>> [root at templ0 freeswitch]# rpm -q nspr
>> nspr-4.9.1-4.el5_8
>> 
>> Here's a list of things I installed earlier:
>> rpm -Uvh http://files.freeswitch.org/freeswitch-release-1-0.noarch.rpm
>> yum install freeswitch-config-vanilla --nogpgcheck
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From covici at ccs.covici.com  Thu Dec 13 05:29:48 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 12 Dec 2012 21:29:48 -0500
Subject: [Freeswitch-users] mod conference keeps core dumping
Message-ID: <13830.1355365788@ccs.covici.com>

Hi.  Every time I start a conference, eventually core is dumped.  Here
is the back trace -- could someone look at this, please?

Thanks.
http://pastebin.freeswitch.org/20324

-- 
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 jeff at jefflenk.com  Thu Dec 13 06:15:59 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 12 Dec 2012 19:15:59 -0800 (PST)
Subject: [Freeswitch-users] mod conference keeps core dumping
In-Reply-To: <13830.1355365788@ccs.covici.com>
References: <13830.1355365788@ccs.covici.com>
Message-ID: <1355368559688-7585476.post@n2.nabble.com>

John,

Fixed in git head

Jeff



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-conference-keeps-core-dumping-tp7585475p7585476.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From covici at ccs.covici.com  Thu Dec 13 06:30:06 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 12 Dec 2012 22:30:06 -0500
Subject: [Freeswitch-users] mod conference keeps core dumping
In-Reply-To: <1355368559688-7585476.post@n2.nabble.com>
References: <13830.1355365788@ccs.covici.com>
	<1355368559688-7585476.post@n2.nabble.com>
Message-ID: <4136.1355369406@ccs.covici.com>

Thanks much.

Jeff Lenk <jeff at jefflenk.com> wrote:

> John,
> 
> Fixed in git head
> 
> Jeff
> 
> 
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-conference-keeps-core-dumping-tp7585475p7585476.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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 nandy1925 at gmail.com  Thu Dec 13 06:44:57 2012
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 13 Dec 2012 11:44:57 +0800
Subject: [Freeswitch-users] Mutiple GSMopen USB modem setup
Message-ID: <CAHKqMb6OBRVtOkEepF-a5fVXz_x-NbvhDijvU+dgCBsGcQ5sKg@mail.gmail.com>

Hello!

I have run GSMOpen on CentOS 5.7 defying the Wiki advisory. Questions:
1. What is the maximum number of USB modems tested? Can we get the numbers
and the CPU used?
2. I'll be installing multiple modems each connected to a different mobile
network. Is the /dev/ttyUSB assignments  constant for every modem? Meaning
it doesn't change if I plug it on different USB jacks.

Thanks,
/Nandy
================================================
www.magicbox.ph - *the better magic* VoIP phone for Filipinos
*Lapulapu City, Phils
Phone: +63-32-3401807,
Mobile: +63-920-6373450
*USA# :* (646)547-1226
*Worldwide:* [**462 + 17772930540 + #*] via any 200+ Access Numbers (37
Countries) <http://www.snipurl.com/magicbox-access>
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From freeswitch-list at puzzled.xs4all.nl  Thu Dec 13 07:08:01 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 13 Dec 2012 05:08:01 +0100
Subject: [Freeswitch-users] SRTP + hardware crypto
In-Reply-To: <CAKDfjgesQGS+EK=8Dur6xVJZO16eP-51ctEH9bf8WWJLzrU65Q@mail.gmail.com>
References: <CAKDfjgesQGS+EK=8Dur6xVJZO16eP-51ctEH9bf8WWJLzrU65Q@mail.gmail.com>
Message-ID: <50C954A1.7020302@puzzled.xs4all.nl>

On 12/12/2012 05:07 PM, Kristian Kielhofner wrote:
> Hello All,
>
>    Has anyone ever looked at enabling hardware acceleration of SIP+TLS
> (and more importantly) SRTP?

There's support for AES in 2010 and later Intel cpu's. Maybe that could 
possibly be used for this acceleration?

http://software.intel.com/en-us/articles/intel-advanced-encryption-standard-aes-instructions-set

An overview of cpu's with AES can be found here:
http://ark.intel.com/search/advanced/?s=t&AESTech=true

>    Hardware engines are pretty well defined in OpenSSL at this point so
> it shouldn't be too hard to get Sofia to use them.  SRTP, on the other
> hand, doesn't use OpenSSL...

Last time I looked the price of those hardware cards is considerable so 
it should really bump max SRTP calls to justify the CAPEX.

>    So the next question is: How hard would it be to get libsrtp to use
> OpenSSL AES so we could take advantage of the various hardware engines
> available?  Would it actually help?  Has anyone looked at this?

Apparently pressureman is.

> Thanks!

Regards,
Patrick




From itispip-qq at hotmail.com  Thu Dec 13 07:17:02 2012
From: itispip-qq at hotmail.com (=?gb2312?B?zfXA7Q==?=)
Date: Thu, 13 Dec 2012 12:17:02 +0800
Subject: [Freeswitch-users] Email notification when user unregistered?
In-Reply-To: <CAFiqYum8ncL3xvNF9vuvx1bWO8MSOcy+Qzur+_-otKkb4kq-DQ@mail.gmail.com>
References: <BLU169-W7BDC7C356B8C56E4FC8E68C4F0@phx.gbl>,
	<CAFiqYum8ncL3xvNF9vuvx1bWO8MSOcy+Qzur+_-otKkb4kq-DQ@mail.gmail.com>
Message-ID: <BLU169-W91F75BA907D5947572A59C8C4E0@phx.gbl>


Thanks Steve, I think  sofia::expire is what I want. 30s refresh frequency is fair enough.
/brgds, Alex

From: steveayre at gmail.com
Date: Wed, 12 Dec 2012 19:47:07 +0000
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Email notification when user unregistered?

You'll need to manage your expectations here.
SIP registrations start when the client sends a REGISTER, with an expiry time. They then periodically send additional REGISTERs to remain registered. They can be spread quite a long way apart - several minutes.


A client explicitly unregisters by sending a REGISTER with a 0 expiry time, ie the registration expires immediately.
If the client has gone offline because of reboot, network problem etc it won't send that final REGISTER. You won't see the client unregister until the end of the expiry time when the client hasn't renewed its registration.


Since that's several minutes later, there's no way to see that they have unregistered immediately. You may be able to reduce the expiry time to detect they're offline quicker, but that will increase bandwidth and load.




As for actually how to detect the unregistration... I suspect the 2 events you want to listen for are sofia::unregister and sofia::expire. The names suggest the 1st will be for an explicit unregister and the 2nd for when the client fails to reregister within the expiry time. That means the 2nd'll be the one you'd see if the client goes unexpectedly offline.



-Steve



On 12 December 2012 17:52, ?? <itispip-qq at hotmail.com> wrote:






Hi Freeswtich Master,
I'm in a situation where I need to proactively moniter whether some device/extension is on in my local network; If for some reason the devices are offline, I need to be noticed immediately so that I can restart the device;


Previously I  use xlite client to suscribe to the presence notification of the devices, but recently added some devices which not supporting presence notification, so I want FreeSwitch to send an email notification when some user/extention are getting unregistered;


How can I capture a Feeswtich user unregister event & launch a LUA script then?
Happy days!
/Brgds, 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



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

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

http://www.freeswitch.org





_________________________________________________________________________
Professional FreeSWITCH Consulting 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 fs-list at communicatefreely.net  Thu Dec 13 07:58:45 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 12 Dec 2012 23:58:45 -0500
Subject: [Freeswitch-users] Email notification when user unregistered?
In-Reply-To: <BLU169-W7BDC7C356B8C56E4FC8E68C4F0@phx.gbl>
References: <BLU169-W7BDC7C356B8C56E4FC8E68C4F0@phx.gbl>
Message-ID: <50C96085.1010507@communicatefreely.net>

If you enable unregister-on-option-fail in the SIP profile, you could
then watch for the unregister events.  I know that on the console, a
warning is fired if this happens, and I'm sure there is an associated
event you could listen for.  You would have to have some sort of external
script subscribe to events through the socket, then generate the e-mail.

The default options-ping time is 30s, so you would know within 30s of an
endpoint disappearing.  This only sends an event if it was registered and
then goes away for an unknown reason.  You may still want to periodically
check the registrations database against what you expect to be there.



?? wrote:
> Hi Freeswtich Master,
> 
> I'm in a situation where I need to proactively moniter whether some 
> device/extension is on in my local network; If for some reason the 
> devices are offline, I need to be noticed immediately so that I can 
> restart the device;
> 
> Previously I  use xlite client to suscribe to the presence notification 
> of the devices, but recently added some devices which not supporting 
> presence notification, so I want FreeSwitch to send an email 
> notification when some user/extention are getting unregistered;
> 
> How can I capture a Feeswtich user unregister event & launch a LUA 
> script then?
> 
> Happy days!
> 
> /Brgds, 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From govoiper at gmail.com  Thu Dec 13 08:01:07 2012
From: govoiper at gmail.com (SamyGo)
Date: Thu, 13 Dec 2012 10:01:07 +0500
Subject: [Freeswitch-users] Nibblebill negative balance
In-Reply-To: <CAMbW9oJKi==P+RT3OQ+nB02=EVJk_D4DuhpJ1PMh679+=wUwrg@mail.gmail.com>
References: <CANNhuhfyXQTao-4OriaS3Av_MC2=nm+GOLvfwJQAeJ76jRmVNg@mail.gmail.com>
	<F5794F00-C367-40A6-8FAC-D5DD8130B2EF@endigotech.com>
	<CAMbW9oJKi==P+RT3OQ+nB02=EVJk_D4DuhpJ1PMh679+=wUwrg@mail.gmail.com>
Message-ID: <CAJUJwthNCo4MX41RpV9rMEpZ3FpmYfbVbg8xuxjmUy808gseig@mail.gmail.com>

Hi,

I've some doubts as well on Nibblebill, I have setup all the things as
mentioned in the wiki and test calls gets stopped on negative balance. My
observation is that the heartbeat process gets kind of dragged/delayed for
longer calls as well as for large number of calls. For example, I could see
in the CLI that it says -67 seconds since last heartbeat or any figure that
is larger than the heartbeat interval of that session starts appearing and
either the caller gets some extra time or extra cash is deducted from the
account - which leads to negative balance.

How can this delayed response from heartbeat be tweaked ? I think I must've
missed some Freeswitch or linux tweaking before I say that nibblebill is
not for production grade realtime billing.

Thanks
Sammy




On Wed, Dec 12, 2012 at 4:44 PM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> Make that image your signature...
>
>
> On 12 December 2012 11:15, Brian Foster <bdfoster at endigotech.com> wrote:
>
>> If you're talking about calls being interrupted due to a negative
>> balance, yes that's a feature that can be configured.
>>
>>
>> http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_Call_When_the_Balance_Is_Depleted
>>
>> [image: image.jpeg]
>>
>> Sent from my iPhone
>>
>> On Dec 7, 2012, at 3:52 AM, virendra bhati <virbhati at gmail.com> wrote:
>>
>> Hi team,
>>
>> I have configure nibblebill with my freeswitch and it's working. But I am
>> facing an issue with billing. Balance goes to -ve and after that calls also
>> throw as well....
>>
>> Is that configuration issue or bug in nibblebill ?
>>
>> --
>>
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-9250078532
>> Asterisk Developer
>> E-mail-: virbhati at gmail.com
>> Skype id:- virbhati2
>> New Delhi(India)
>> [image: View my profile on LinkedIn]<http://in.linkedin.com/pub/virendra-bhati/6/a30/755>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lists at kavun.ch  Thu Dec 13 08:53:08 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 13 Dec 2012 00:53:08 -0500
Subject: [Freeswitch-users] mod_perl - Can't call method "serialize" on
	an undefined value
In-Reply-To: <CAKbxfG-vmXgt4rAuAGBVXOrigssqMFvjChaZoTSw=vqN-y=2SQ@mail.gmail.com>
References: <ADD8CA27-CB8D-4D61-BE4F-F03DFB354746@kavun.ch>
	<CAKbxfG-vmXgt4rAuAGBVXOrigssqMFvjChaZoTSw=vqN-y=2SQ@mail.gmail.com>
Message-ID: <8091451F-955E-450E-80C4-6FA7A8A8077D@kavun.ch>

Got it, thanks a bunch for your reply.

Cheers,
Emrah
On Dec 12, 2012, at 4:42 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> $env only exists when you call it from the cli or FSAPI via the "perl" function.
> 
> 
> 
> On Tue, Dec 11, 2012 at 6:17 PM, Emrah <lists at kavun.ch> wrote:
> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-November/036404.html
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From lists at kavun.ch  Thu Dec 13 08:55:18 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 13 Dec 2012 00:55:18 -0500
Subject: [Freeswitch-users] ILBC takes precedence regardless of the
	codec prefs
In-Reply-To: <CAFiqYukUs23XqM8hhSxKAQPm9q8YcjHgb=CvH_xQ67dWK65z4Q@mail.gmail.com>
References: <D37F8F9F-301A-4B29-9147-7C5E3E54C9DF@kavun.ch>
	<CAFiqYukUs23XqM8hhSxKAQPm9q8YcjHgb=CvH_xQ67dWK65z4Q@mail.gmail.com>
Message-ID: <848F5ABB-AF9A-41FC-89FB-1D0BDCD7C778@kavun.ch>

Hey Steven,

I operate in late negotiation mode and work with the dialplan, no global values on any call.
I finally fixed it using absolute_codec_string. 

Thanks a lot for your reply.

E
On Dec 12, 2012, at 3:05 PM, Steven Ayre <steveayre at gmail.com> wrote:

> In your example FS is selecting the G711 alaw (PCMA) '8' codec. The client offers it, the FS reply only returns the codec it has selected.
> 
> Try a debug level log of the call. It will show you a description of the codec negotiation.
> 
> If you're not expecting the G711 codec to be accepted, check your codec preferences. Inbound and outbound are separate parameters, and if you're setting it from vars.conf.xml that sets a variable used in the sofia profile... the profile could be using a static value that's different.
> 
> There are also other parameters that'll have an effect on a bridged call such as late negotiation which would leave codec selection it up to the bleg.
> 
> 
> 
> -Steve
> 
> 
> 
> 
> On 11 December 2012 22:02, Emrah <lists at kavun.ch> wrote:
> Hi all,
> 
> I am using FreeSWITCH Version 1.2.5.3+git~20121206T050429Z~91eef34d5c (git 91eef34 2012-12-06 05:04:29Z).
> When mod_ilbc is loaded, ILBC is used in any session where the client offers it, regardless of my codec preferences. This gets even stranger when you look at the SIP trace? FS doesn't offer ILBC. This is brand new, I wanted to enable ILBC just in case, and found out that I prefer it to G729.
> 
> Here's some output of a trace:
> Client:
> v=0.
> o=- 3564251257 3564251257 IN IP4 10.0.0.102.
> s=pjmedia.
> c=IN IP4 10.0.0.102.
> t=0 0.
> a=X-nat:0.
> m=audio 4000 RTP/AVP 103 102 104 109 3 0 8 9 101.
> a=rtcp:4001 IN IP4 10.0.1.132.
> a=rtpmap:103 speex/16000.
> a=rtpmap:102 speex/8000.
> a=rtpmap:104 speex/32000.
> a=rtpmap:109 iLBC/8000.
> a=fmtp:109 mode=30.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:9 G722/8000.
> a=sendrecv.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> 
> FS:
> v=0.
> o=FreeSWITCH 1355242302 1355242303 IN IP4 1.2.3.4.
> s=FreeSWITCH.
> c=IN IP4 1.2.3.4.
> t=0 0.
> m=audio 20042 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> 
> 
> Your input is greatly appreciated, as always.
> 
> All the best,
> Emrah
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From krice at freeswitch.org  Thu Dec 13 09:01:23 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 13 Dec 2012 00:01:23 -0600
Subject: [Freeswitch-users] ILBC takes precedence regardless of the
 codec prefs
In-Reply-To: <848F5ABB-AF9A-41FC-89FB-1D0BDCD7C778@kavun.ch>
Message-ID: <CCEECB53.61ED9%krice@freeswitch.org>

Keep in mind there are settings for who's codec list to follow for
preference order, FreeSWITCH's or the endpoints...

K


On 12/12/12 11:55 PM, "Emrah" <lists at kavun.ch> wrote:

> Hey Steven,
> 
> I operate in late negotiation mode and work with the dialplan, no global
> values on any call.
> I finally fixed it using absolute_codec_string.
> 
> Thanks a lot for your reply.
> 
> E
> On Dec 12, 2012, at 3:05 PM, Steven Ayre <steveayre at gmail.com> wrote:
> 
>> In your example FS is selecting the G711 alaw (PCMA) '8' codec. The client
>> offers it, the FS reply only returns the codec it has selected.
>> 
>> Try a debug level log of the call. It will show you a description of the
>> codec negotiation.
>> 
>> If you're not expecting the G711 codec to be accepted, check your codec
>> preferences. Inbound and outbound are separate parameters, and if you're
>> setting it from vars.conf.xml that sets a variable used in the sofia
>> profile... the profile could be using a static value that's different.
>> 
>> There are also other parameters that'll have an effect on a bridged call such
>> as late negotiation which would leave codec selection it up to the bleg.
>> 
>> 
>> 
>> -Steve
>> 
>> 
>> 
>> 
>> On 11 December 2012 22:02, Emrah <lists at kavun.ch> wrote:
>> Hi all,
>> 
>> I am using FreeSWITCH Version 1.2.5.3+git~20121206T050429Z~91eef34d5c (git
>> 91eef34 2012-12-06 05:04:29Z).
>> When mod_ilbc is loaded, ILBC is used in any session where the client offers
>> it, regardless of my codec preferences. This gets even stranger when you look
>> at the SIP trace? FS doesn't offer ILBC. This is brand new, I wanted to
>> enable ILBC just in case, and found out that I prefer it to G729.
>> 
>> Here's some output of a trace:
>> Client:
>> v=0.
>> o=- 3564251257 3564251257 IN IP4 10.0.0.102.
>> s=pjmedia.
>> c=IN IP4 10.0.0.102.
>> t=0 0.
>> a=X-nat:0.
>> m=audio 4000 RTP/AVP 103 102 104 109 3 0 8 9 101.
>> a=rtcp:4001 IN IP4 10.0.1.132.
>> a=rtpmap:103 speex/16000.
>> a=rtpmap:102 speex/8000.
>> a=rtpmap:104 speex/32000.
>> a=rtpmap:109 iLBC/8000.
>> a=fmtp:109 mode=30.
>> a=rtpmap:3 GSM/8000.
>> a=rtpmap:0 PCMU/8000.
>> a=rtpmap:8 PCMA/8000.
>> a=rtpmap:9 G722/8000.
>> a=sendrecv.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>> 
>> FS:
>> v=0.
>> o=FreeSWITCH 1355242302 1355242303 IN IP4 1.2.3.4.
>> s=FreeSWITCH.
>> c=IN IP4 1.2.3.4.
>> t=0 0.
>> m=audio 20042 RTP/AVP 8 101.
>> a=rtpmap:8 PCMA/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=silenceSupp:off - - - -.
>> a=ptime:20.
>> 
>> 
>> Your input is greatly appreciated, as always.
>> 
>> All the best,
>> Emrah
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From lists at kavun.ch  Thu Dec 13 09:15:55 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 13 Dec 2012 01:15:55 -0500
Subject: [Freeswitch-users] ILBC takes precedence regardless of the
	codec prefs
In-Reply-To: <CCEECB53.61ED9%krice@freeswitch.org>
References: <CCEECB53.61ED9%krice@freeswitch.org>
Message-ID: <7E11280E-3CFF-4015-AA20-0E12119A4913@kavun.ch>

thanks.
How do you make FS's codec choice take precedence on outbound negotiations?
I already have inbound-codec-negotiation = scrooge.

Cheers
On Dec 13, 2012, at 1:01 AM, Ken Rice <krice at freeswitch.org> wrote:

> Keep in mind there are settings for who's codec list to follow for
> preference order, FreeSWITCH's or the endpoints...
> 
> K
> 
> 
> On 12/12/12 11:55 PM, "Emrah" <lists at kavun.ch> wrote:
> 
>> Hey Steven,
>> 
>> I operate in late negotiation mode and work with the dialplan, no global
>> values on any call.
>> I finally fixed it using absolute_codec_string.
>> 
>> Thanks a lot for your reply.
>> 
>> E
>> On Dec 12, 2012, at 3:05 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> 
>>> In your example FS is selecting the G711 alaw (PCMA) '8' codec. The client
>>> offers it, the FS reply only returns the codec it has selected.
>>> 
>>> Try a debug level log of the call. It will show you a description of the
>>> codec negotiation.
>>> 
>>> If you're not expecting the G711 codec to be accepted, check your codec
>>> preferences. Inbound and outbound are separate parameters, and if you're
>>> setting it from vars.conf.xml that sets a variable used in the sofia
>>> profile... the profile could be using a static value that's different.
>>> 
>>> There are also other parameters that'll have an effect on a bridged call such
>>> as late negotiation which would leave codec selection it up to the bleg.
>>> 
>>> 
>>> 
>>> -Steve
>>> 
>>> 
>>> 
>>> 
>>> On 11 December 2012 22:02, Emrah <lists at kavun.ch> wrote:
>>> Hi all,
>>> 
>>> I am using FreeSWITCH Version 1.2.5.3+git~20121206T050429Z~91eef34d5c (git
>>> 91eef34 2012-12-06 05:04:29Z).
>>> When mod_ilbc is loaded, ILBC is used in any session where the client offers
>>> it, regardless of my codec preferences. This gets even stranger when you look
>>> at the SIP trace? FS doesn't offer ILBC. This is brand new, I wanted to
>>> enable ILBC just in case, and found out that I prefer it to G729.
>>> 
>>> Here's some output of a trace:
>>> Client:
>>> v=0.
>>> o=- 3564251257 3564251257 IN IP4 10.0.0.102.
>>> s=pjmedia.
>>> c=IN IP4 10.0.0.102.
>>> t=0 0.
>>> a=X-nat:0.
>>> m=audio 4000 RTP/AVP 103 102 104 109 3 0 8 9 101.
>>> a=rtcp:4001 IN IP4 10.0.1.132.
>>> a=rtpmap:103 speex/16000.
>>> a=rtpmap:102 speex/8000.
>>> a=rtpmap:104 speex/32000.
>>> a=rtpmap:109 iLBC/8000.
>>> a=fmtp:109 mode=30.
>>> a=rtpmap:3 GSM/8000.
>>> a=rtpmap:0 PCMU/8000.
>>> a=rtpmap:8 PCMA/8000.
>>> a=rtpmap:9 G722/8000.
>>> a=sendrecv.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-15.
>>> 
>>> FS:
>>> v=0.
>>> o=FreeSWITCH 1355242302 1355242303 IN IP4 1.2.3.4.
>>> s=FreeSWITCH.
>>> c=IN IP4 1.2.3.4.
>>> t=0 0.
>>> m=audio 20042 RTP/AVP 8 101.
>>> a=rtpmap:8 PCMA/8000.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-16.
>>> a=silenceSupp:off - - - -.
>>> a=ptime:20.
>>> 
>>> 
>>> Your input is greatly appreciated, as always.
>>> 
>>> All the best,
>>> Emrah
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From lists at kavun.ch  Thu Dec 13 09:30:29 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 13 Dec 2012 01:30:29 -0500
Subject: [Freeswitch-users] ILBC takes precedence regardless of the
	codec prefs
In-Reply-To: <7E11280E-3CFF-4015-AA20-0E12119A4913@kavun.ch>
References: <CCEECB53.61ED9%krice@freeswitch.org>
	<7E11280E-3CFF-4015-AA20-0E12119A4913@kavun.ch>
Message-ID: <BF8D7E18-0659-4A5A-A8D2-6BD3AA6EB1D7@kavun.ch>

Just answering my own question.
I was already doing it by using absolute_codec_string.

Thanks Ken
On Dec 13, 2012, at 1:15 AM, Emrah <lists at kavun.ch> wrote:

> thanks.
> How do you make FS's codec choice take precedence on outbound negotiations?
> I already have inbound-codec-negotiation = scrooge.
> 
> Cheers
> On Dec 13, 2012, at 1:01 AM, Ken Rice <krice at freeswitch.org> wrote:
> 
>> Keep in mind there are settings for who's codec list to follow for
>> preference order, FreeSWITCH's or the endpoints...
>> 
>> K
>> 
>> 
>> On 12/12/12 11:55 PM, "Emrah" <lists at kavun.ch> wrote:
>> 
>>> Hey Steven,
>>> 
>>> I operate in late negotiation mode and work with the dialplan, no global
>>> values on any call.
>>> I finally fixed it using absolute_codec_string.
>>> 
>>> Thanks a lot for your reply.
>>> 
>>> E
>>> On Dec 12, 2012, at 3:05 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>> 
>>>> In your example FS is selecting the G711 alaw (PCMA) '8' codec. The client
>>>> offers it, the FS reply only returns the codec it has selected.
>>>> 
>>>> Try a debug level log of the call. It will show you a description of the
>>>> codec negotiation.
>>>> 
>>>> If you're not expecting the G711 codec to be accepted, check your codec
>>>> preferences. Inbound and outbound are separate parameters, and if you're
>>>> setting it from vars.conf.xml that sets a variable used in the sofia
>>>> profile... the profile could be using a static value that's different.
>>>> 
>>>> There are also other parameters that'll have an effect on a bridged call such
>>>> as late negotiation which would leave codec selection it up to the bleg.
>>>> 
>>>> 
>>>> 
>>>> -Steve
>>>> 
>>>> 
>>>> 
>>>> 
>>>> On 11 December 2012 22:02, Emrah <lists at kavun.ch> wrote:
>>>> Hi all,
>>>> 
>>>> I am using FreeSWITCH Version 1.2.5.3+git~20121206T050429Z~91eef34d5c (git
>>>> 91eef34 2012-12-06 05:04:29Z).
>>>> When mod_ilbc is loaded, ILBC is used in any session where the client offers
>>>> it, regardless of my codec preferences. This gets even stranger when you look
>>>> at the SIP trace? FS doesn't offer ILBC. This is brand new, I wanted to
>>>> enable ILBC just in case, and found out that I prefer it to G729.
>>>> 
>>>> Here's some output of a trace:
>>>> Client:
>>>> v=0.
>>>> o=- 3564251257 3564251257 IN IP4 10.0.0.102.
>>>> s=pjmedia.
>>>> c=IN IP4 10.0.0.102.
>>>> t=0 0.
>>>> a=X-nat:0.
>>>> m=audio 4000 RTP/AVP 103 102 104 109 3 0 8 9 101.
>>>> a=rtcp:4001 IN IP4 10.0.1.132.
>>>> a=rtpmap:103 speex/16000.
>>>> a=rtpmap:102 speex/8000.
>>>> a=rtpmap:104 speex/32000.
>>>> a=rtpmap:109 iLBC/8000.
>>>> a=fmtp:109 mode=30.
>>>> a=rtpmap:3 GSM/8000.
>>>> a=rtpmap:0 PCMU/8000.
>>>> a=rtpmap:8 PCMA/8000.
>>>> a=rtpmap:9 G722/8000.
>>>> a=sendrecv.
>>>> a=rtpmap:101 telephone-event/8000.
>>>> a=fmtp:101 0-15.
>>>> 
>>>> FS:
>>>> v=0.
>>>> o=FreeSWITCH 1355242302 1355242303 IN IP4 1.2.3.4.
>>>> s=FreeSWITCH.
>>>> c=IN IP4 1.2.3.4.
>>>> t=0 0.
>>>> m=audio 20042 RTP/AVP 8 101.
>>>> a=rtpmap:8 PCMA/8000.
>>>> a=rtpmap:101 telephone-event/8000.
>>>> a=fmtp:101 0-16.
>>>> a=silenceSupp:off - - - -.
>>>> a=ptime:20.
>>>> 
>>>> 
>>>> Your input is greatly appreciated, as always.
>>>> 
>>>> All the best,
>>>> Emrah
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>> 
>> -- 
>> Ken
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> irc.freenode.net #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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.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  Thu Dec 13 10:14:51 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 13 Dec 2012 02:14:51 -0500
Subject: [Freeswitch-users] Nibblebill negative balance
In-Reply-To: <CAJUJwthNCo4MX41RpV9rMEpZ3FpmYfbVbg8xuxjmUy808gseig@mail.gmail.com>
References: <CANNhuhfyXQTao-4OriaS3Av_MC2=nm+GOLvfwJQAeJ76jRmVNg@mail.gmail.com>
	<F5794F00-C367-40A6-8FAC-D5DD8130B2EF@endigotech.com>
	<CAMbW9oJKi==P+RT3OQ+nB02=EVJk_D4DuhpJ1PMh679+=wUwrg@mail.gmail.com>
	<CAJUJwthNCo4MX41RpV9rMEpZ3FpmYfbVbg8xuxjmUy808gseig@mail.gmail.com>
Message-ID: <7C342CBA-F49A-4A60-934E-640C5EEFA00B@endigotech.com>



Guys, this is getting too easy.

-BDF

Sent from my iPhone

On Dec 13, 2012, at 12:01 AM, SamyGo <govoiper at gmail.com> wrote:

> Hi,
> 
> I've some doubts as well on Nibblebill, I have setup all the things as mentioned in the wiki and test calls gets stopped on negative balance. My observation is that the heartbeat process gets kind of dragged/delayed for longer calls as well as for large number of calls. For example, I could see in the CLI that it says -67 seconds since last heartbeat or any figure that is larger than the heartbeat interval of that session starts appearing and either the caller gets some extra time or extra cash is deducted from the account - which leads to negative balance.
> 
> How can this delayed response from heartbeat be tweaked ? I think I must've missed some Freeswitch or linux tweaking before I say that nibblebill is not for production grade realtime billing.
> 
> Thanks
> Sammy
> 
> 
> 
> 
> On Wed, Dec 12, 2012 at 4:44 PM, Andrew Cassidy <andrew at cassidywebservices.co.uk> wrote:
>> Make that image your signature...
>> 
>> 
>> On 12 December 2012 11:15, Brian Foster <bdfoster at endigotech.com> wrote:
>>> If you're talking about calls being interrupted due to a negative balance, yes that's a feature that can be configured. 
>>> 
>>> http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_Call_When_the_Balance_Is_Depleted
>>> 
>>> <image.jpeg>
>>> 
>>> Sent from my iPhone
>>> 
>>> On Dec 7, 2012, at 3:52 AM, virendra bhati <virbhati at gmail.com> wrote:
>>> 
>>>> Hi team,
>>>> 
>>>> I have configure nibblebill with my freeswitch and it's working. But I am facing an issue with billing. Balance goes to -ve and after that calls also throw as well....
>>>> 
>>>> Is that configuration issue or bug in nibblebill ? 
>>>> 
>>>> -- 
>>>> 
>>>> Thanks and regards
>>>> 
>>>>  Virendra Bhati
>>>> +91-9250078532
>>>> Asterisk Developer
>>>> E-mail-: virbhati at gmail.com
>>>> Skype id:- virbhati2
>>>> New Delhi(India)
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 avi at avimarcus.net  Thu Dec 13 10:36:37 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 13 Dec 2012 09:36:37 +0200
Subject: [Freeswitch-users] Email notification when user unregistered?
In-Reply-To: <50C96085.1010507@communicatefreely.net>
References: <BLU169-W7BDC7C356B8C56E4FC8E68C4F0@phx.gbl>
	<50C96085.1010507@communicatefreely.net>
Message-ID: <CALNZuwkg83s2OqQfHN=syJME8C2hLruD7EVGwkgYz14_vL+xVg@mail.gmail.com>

Be careful with unregister on fail - it seems if there's just an
intermittent internet interruption, it will unregister the box, which won't
know it's been unregistered and you could lose your registration.
Perhaps an option could be added to the code for "event-on-fail" so you
could use that information accordingly...

-Avi

On Thu, Dec 13, 2012 at 6:58 AM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> If you enable unregister-on-option-fail in the SIP profile, you could
> then watch for the unregister events.  I know that on the console, a
> warning is fired if this happens, and I'm sure there is an associated
> event you could listen for.  You would have to have some sort of external
> script subscribe to events through the socket, then generate the e-mail.
>
> The default options-ping time is 30s, so you would know within 30s of an
> endpoint disappearing.  This only sends an event if it was registered and
> then goes away for an unknown reason.  You may still want to periodically
> check the registrations database against what you expect to be there.
>
>
>
> ?? wrote:
> > Hi Freeswtich Master,
> >
> > I'm in a situation where I need to proactively moniter whether some
> > device/extension is on in my local network; If for some reason the
> > devices are offline, I need to be noticed immediately so that I can
> > restart the device;
> >
> > Previously I  use xlite client to suscribe to the presence notification
> > of the devices, but recently added some devices which not supporting
> > presence notification, so I want FreeSwitch to send an email
> > notification when some user/extention are getting unregistered;
> >
> > How can I capture a Feeswtich user unregister event & launch a LUA
> > script then?
> >
> > Happy days!
> >
> > /Brgds, 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 govoiper at gmail.com  Thu Dec 13 10:56:35 2012
From: govoiper at gmail.com (SamyGo)
Date: Thu, 13 Dec 2012 12:56:35 +0500
Subject: [Freeswitch-users] Nibblebill negative balance
In-Reply-To: <7C342CBA-F49A-4A60-934E-640C5EEFA00B@endigotech.com>
References: <CANNhuhfyXQTao-4OriaS3Av_MC2=nm+GOLvfwJQAeJ76jRmVNg@mail.gmail.com>
	<F5794F00-C367-40A6-8FAC-D5DD8130B2EF@endigotech.com>
	<CAMbW9oJKi==P+RT3OQ+nB02=EVJk_D4DuhpJ1PMh679+=wUwrg@mail.gmail.com>
	<CAJUJwthNCo4MX41RpV9rMEpZ3FpmYfbVbg8xuxjmUy808gseig@mail.gmail.com>
	<7C342CBA-F49A-4A60-934E-640C5EEFA00B@endigotech.com>
Message-ID: <CAJUJwtgA52GmTcXtwJwTO_7gJnG6AH+CdpyRqoGhj_1=bqQrrg@mail.gmail.com>

Sir,

I'd love to open up a Jira for this only IF I am sure that this IS a bug.
So the purpose of this mailing list is best served when we ask questions
about things we lack understanding of or need to know if there are steps
missing.

Thanks,
Sammy



On Thu, Dec 13, 2012 at 12:14 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> [image: image.png]
>
> Guys, this is getting too easy.
>
> -BDF
>
> Sent from my iPhone
>
> On Dec 13, 2012, at 12:01 AM, SamyGo <govoiper at gmail.com> wrote:
>
> Hi,
>
> I've some doubts as well on Nibblebill, I have setup all the things as
> mentioned in the wiki and test calls gets stopped on negative balance. My
> observation is that the heartbeat process gets kind of dragged/delayed for
> longer calls as well as for large number of calls. For example, I could see
> in the CLI that it says -67 seconds since last heartbeat or any figure that
> is larger than the heartbeat interval of that session starts appearing and
> either the caller gets some extra time or extra cash is deducted from the
> account - which leads to negative balance.
>
> How can this delayed response from heartbeat be tweaked ? I think I
> must've missed some Freeswitch or linux tweaking before I say that
> nibblebill is not for production grade realtime billing.
>
> Thanks
> Sammy
>
>
>
>
> On Wed, Dec 12, 2012 at 4:44 PM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
>> Make that image your signature...
>>
>>
>> On 12 December 2012 11:15, Brian Foster <bdfoster at endigotech.com> wrote:
>>
>>> If you're talking about calls being interrupted due to a negative
>>> balance, yes that's a feature that can be configured.
>>>
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_Call_When_the_Balance_Is_Depleted
>>>
>>> <image.jpeg>
>>>
>>> Sent from my iPhone
>>>
>>> On Dec 7, 2012, at 3:52 AM, virendra bhati <virbhati at gmail.com> wrote:
>>>
>>> Hi team,
>>>
>>> I have configure nibblebill with my freeswitch and it's working. But I
>>> am facing an issue with billing. Balance goes to -ve and after that calls
>>> also throw as well....
>>>
>>> Is that configuration issue or bug in nibblebill ?
>>>
>>> --
>>>
>>> Thanks and regards
>>>
>>>  Virendra Bhati
>>> +91-9250078532
>>> Asterisk Developer
>>> E-mail-: virbhati at gmail.com
>>> Skype id:- virbhati2
>>> New Delhi(India)
>>> [image: View my profile on LinkedIn]<http://in.linkedin.com/pub/virendra-bhati/6/a30/755>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 13 14:02:22 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 13 Dec 2012 12:02:22 +0100
Subject: [Freeswitch-users] Mutiple GSMopen USB modem setup
In-Reply-To: <CAHKqMb6OBRVtOkEepF-a5fVXz_x-NbvhDijvU+dgCBsGcQ5sKg@mail.gmail.com>
References: <CAHKqMb6OBRVtOkEepF-a5fVXz_x-NbvhDijvU+dgCBsGcQ5sKg@mail.gmail.com>
Message-ID: <CALXCt0pT2UoMWZ0dUPC2zKTbm_7zd2Z_2UAr1e8i78JO24uXTw@mail.gmail.com>

On Thu, Dec 13, 2012 at 4:44 AM, Nandy Dagondon <nandy1925 at gmail.com> wrote:
> I have run GSMOpen on CentOS 5.7 defying the Wiki advisory. Questions:

I would invite you to go for the OS distros detailed in the wiki page.
You'll probably encounter problems with different distros, and you're
on your own to solve it.

> 1. What is the maximum number of USB modems tested? Can we get the numbers
> and the CPU used?

I've heard about 48 concurrent, and 64. Me personally have tested with
5. No CPU consumption. The critical part is the USB BUS. So use
cascading and POWERED good usb 2.0 hubs

> 2. I'll be installing multiple modems each connected to a different mobile
> network. Is the /dev/ttyUSB assignments  constant for every modem? Meaning
> it doesn't change if I plug it on different USB jacks.

it will change not only if you change USB port, but also randomly if
you stay on the same USB port and reboot (and sometimes also without
rebooting). That's a "feature" of Linux distros (a demented one,
cannot understand why they choose this behavior).

Soon or later I'll look into this, and come out with a solution (I've
made some preliminary research and reasoning about in the past).

If you have a commercial interest in that, and a real budget for it,
contact me in private as consultant, or put a public bounty on it.

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From royce3 at gmail.com  Thu Dec 13 16:23:03 2012
From: royce3 at gmail.com (Royce Mitchell III)
Date: Thu, 13 Dec 2012 07:23:03 -0600
Subject: [Freeswitch-users] Originate against dial plan?
Message-ID: <2870919548404989049@unknownmsgid>

I'm trying to avoid duplication of logic. I have a dial plan setup
with two sets of gateways, one set is a route to a pair of adtrans
connecting to the customer downstream telephony system, the other set
of gateways is another pair of adtrans leading to the PSTN. I have all
this setup and working in the dial plan. I would like to do something
like this: originate 8005551212 XML default &txfax(foo), but I can't
figure it out. Is there a way I could, say, generate a one-leg txfax,
then transfer to the destination?

Royce Mitchell
IT Consultant
ITAS Solutions
royce3 at itas-solutions.com


From kbdfck at gmail.com  Thu Dec 13 16:51:44 2012
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 13 Dec 2012 17:51:44 +0400
Subject: [Freeswitch-users] Console FreeTDM commands get executed only
 once, following ftdm hang console
In-Reply-To: <CAFiqYukeFi4AwwR_mL7ExJpU0dN=wRA4Y1JBfMoG6dje5riCXQ@mail.gmail.com>
References: <CA+GmxbZBZ7Wm96qF5avNW0vmrDx+FDWk=UD=kAXHOeeCDaOpvQ@mail.gmail.com>
	<CAKbxfG-MaLkmQnkquRJEu9g4puOHf51bv4iT8cv3WV87aYd8iA@mail.gmail.com>
	<CAFiqYukeFi4AwwR_mL7ExJpU0dN=wRA4Y1JBfMoG6dje5riCXQ@mail.gmail.com>
Message-ID: <CA+GmxbbBR0Tiu1ivvVp5a7qJtCQ2S_J_wekYScuNACK6J-y=ng@mail.gmail.com>

I created http://jira.freeswitch.org/browse/OPENZAP-204

After turning PRI debug on it seems this issue appears when freetdm itself
gets stuck somewhere.

2012/12/12 Steven Ayre <steveayre at gmail.com>

> Dmitry, also see
>
> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29
> That section explains in detail how to collect the backtrace from the file
> gcore will produce.
>
>
> On 12 December 2012 19:02, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>
>> When it's stuck get a gcore of the running process and get a backtrace.
>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#gcore
>> attach that to jira
>>
>>
>> On Wed, Dec 12, 2012 at 9:03 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>
>>> Hi all!
>>>
>>> I have Digium TDM400P with DAHDI, FREETDM and libpri built and installed
>>> on git head. There is one librpi-based span configured. The problem is when
>>> I issue any ftdm-command it works once, and all subsequent ftdm commands
>>> hang and I get no response.
>>>
>>> I kill fs_cli, start it again and can do anything but ftdm commands - it
>>> hangs after any ftdm command.
>>>
>>> After that I can't unload mod_freetdm, as FS says it is in use.
>>>
>>> Is this a bug? What information should I collect to open JIRA?
>>>
>>> freeswitch at internal> ftdm list
>>> +OK
>>> span: 1 (trunk1)
>>> type: isdn
>>> physical_status: ok
>>> signaling_status: DOWN
>>> chan_count: 31
>>> dialplan: XML
>>> context: from_trunk1
>>> dial_regex:
>>> fail_dial_regex:
>>> hold_music:
>>> analog_options: none
>>>
>>> freeswitch at internal>
>>> freeswitch at internal> ftdm list
>>> <no response>
>>>
>>>
>>>
>>>
>>> --
>>> 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From kbdfck at gmail.com  Thu Dec 13 16:55:39 2012
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 13 Dec 2012 17:55:39 +0400
Subject: [Freeswitch-users] Originate against dial plan?
In-Reply-To: <2870919548404989049@unknownmsgid>
References: <2870919548404989049@unknownmsgid>
Message-ID: <CA+Gmxba+0wiVrXGwjEo4O1BbH0btSsMQjTTxdvrE0v2FFxwk=A@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Loopback

There are issues with t.38 according to wiki, although I haven't tested
T.38 on loopback  by myself.

See also http://wiki.freeswitch.org/wiki/Variable_loopback_bowout_on_execute
There was a thread few month ago in mailing list where Anthony explained
things about loopback channels.


2012/12/13 Royce Mitchell III <royce3 at gmail.com>

> I'm trying to avoid duplication of logic. I have a dial plan setup
> with two sets of gateways, one set is a route to a pair of adtrans
> connecting to the customer downstream telephony system, the other set
> of gateways is another pair of adtrans leading to the PSTN. I have all
> this setup and working in the dial plan. I would like to do something
> like this: originate 8005551212 XML default &txfax(foo), but I can't
> figure it out. Is there a way I could, say, generate a one-leg txfax,
> then transfer to the destination?
>
> Royce Mitchell
> IT Consultant
> ITAS Solutions
> royce3 at itas-solutions.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bdfoster at endigotech.com  Thu Dec 13 17:23:24 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 13 Dec 2012 09:23:24 -0500
Subject: [Freeswitch-users] Nibblebill negative balance
In-Reply-To: <CAJUJwtgA52GmTcXtwJwTO_7gJnG6AH+CdpyRqoGhj_1=bqQrrg@mail.gmail.com>
References: <CANNhuhfyXQTao-4OriaS3Av_MC2=nm+GOLvfwJQAeJ76jRmVNg@mail.gmail.com>
	<F5794F00-C367-40A6-8FAC-D5DD8130B2EF@endigotech.com>
	<CAMbW9oJKi==P+RT3OQ+nB02=EVJk_D4DuhpJ1PMh679+=wUwrg@mail.gmail.com>
	<CAJUJwthNCo4MX41RpV9rMEpZ3FpmYfbVbg8xuxjmUy808gseig@mail.gmail.com>
	<7C342CBA-F49A-4A60-934E-640C5EEFA00B@endigotech.com>
	<CAJUJwtgA52GmTcXtwJwTO_7gJnG6AH+CdpyRqoGhj_1=bqQrrg@mail.gmail.com>
Message-ID: <B36C07E7-06FC-4860-A561-C3AEDB12D06D@endigotech.com>

From what I'm gathering on your other email, it's not the intended behavior. I'd file a JIRA. 

Sent from my iPhone

On Dec 13, 2012, at 2:56 AM, SamyGo <govoiper at gmail.com> wrote:

> Sir,
> 
> I'd love to open up a Jira for this only IF I am sure that this IS a bug. So the purpose of this mailing list is best served when we ask questions about things we lack understanding of or need to know if there are steps missing.
> 
> Thanks,
> Sammy
> 
> 
> 
> On Thu, Dec 13, 2012 at 12:14 PM, Brian Foster <bdfoster at endigotech.com> wrote:
>> <image.png>
>> 
>> Guys, this is getting too easy.
>> 
>> -BDF
>> 
>> Sent from my iPhone
>> 
>> On Dec 13, 2012, at 12:01 AM, SamyGo <govoiper at gmail.com> wrote:
>> 
>>> Hi,
>>> 
>>> I've some doubts as well on Nibblebill, I have setup all the things as mentioned in the wiki and test calls gets stopped on negative balance. My observation is that the heartbeat process gets kind of dragged/delayed for longer calls as well as for large number of calls. For example, I could see in the CLI that it says -67 seconds since last heartbeat or any figure that is larger than the heartbeat interval of that session starts appearing and either the caller gets some extra time or extra cash is deducted from the account - which leads to negative balance.
>>> 
>>> How can this delayed response from heartbeat be tweaked ? I think I must've missed some Freeswitch or linux tweaking before I say that nibblebill is not for production grade realtime billing.
>>> 
>>> Thanks
>>> Sammy
>>> 
>>> 
>>> 
>>> 
>>> On Wed, Dec 12, 2012 at 4:44 PM, Andrew Cassidy <andrew at cassidywebservices.co.uk> wrote:
>>>> Make that image your signature...
>>>> 
>>>> 
>>>> On 12 December 2012 11:15, Brian Foster <bdfoster at endigotech.com> wrote:
>>>>> If you're talking about calls being interrupted due to a negative balance, yes that's a feature that can be configured. 
>>>>> 
>>>>> http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_Call_When_the_Balance_Is_Depleted
>>>>> 
>>>>> <image.jpeg>
>>>>> 
>>>>> Sent from my iPhone
>>>>> 
>>>>> On Dec 7, 2012, at 3:52 AM, virendra bhati <virbhati at gmail.com> wrote:
>>>>> 
>>>>>> Hi team,
>>>>>> 
>>>>>> I have configure nibblebill with my freeswitch and it's working. But I am facing an issue with billing. Balance goes to -ve and after that calls also throw as well....
>>>>>> 
>>>>>> Is that configuration issue or bug in nibblebill ? 
>>>>>> 
>>>>>> -- 
>>>>>> 
>>>>>> Thanks and regards
>>>>>> 
>>>>>>  Virendra Bhati
>>>>>> +91-9250078532
>>>>>> Asterisk Developer
>>>>>> E-mail-: virbhati at gmail.com
>>>>>> Skype id:- virbhati2
>>>>>> New Delhi(India)
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting 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
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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 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
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 covici at ccs.covici.com  Thu Dec 13 17:53:43 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 13 Dec 2012 09:53:43 -0500
Subject: [Freeswitch-users] what can I have in ${} construction
Message-ID: <30097.1355410423@ccs.covici.com>

Hi.  I was experimenting with ${variable} and was wondering what could
be in there before the }?  I can do a substring if I say variable:2, but
I wonder what else you can do?  The bash manual has all kinds of things
you can have, but how much of this will fs do?

Thanks in advance for any suggestions.

-- 
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 yungwei at resolvity.com  Thu Dec 13 19:06:30 2012
From: yungwei at resolvity.com (Yungwei Chen)
Date: Thu, 13 Dec 2012 11:06:30 -0500
Subject: [Freeswitch-users] Missing bootstrap.sh in the snapshots
Message-ID: <33095823FD21DF429B481B5163264B799F3A97C133@VMBX102.ihostexchange.net>

Hi,

I found that bootstrap.sh is missing in some tar balls, such as freeswitch-1.2.1.tar.bz2 and freeswitch-1.2.5.3.tar.bz2. 
Are they already bootstraped?
Thanks.


From sdevoy at bizfocused.com  Thu Dec 13 19:58:24 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 13 Dec 2012 11:58:24 -0500
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
Message-ID: <0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>

How do I get sofia sip trace to go in the Freeswitch log?  Do I have to keep
separate logs and try to match them up?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 10, 2012 12:33 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

The SERVICE_NOT_IMPLEMENTED is the q.850 translation of the 406 you received
over SIP.

If you turn on the siptrace with "sofia global siptrace on" you may get more
details.

 

Also leg_delay_start does not do much good in enterprise originate as each
url in the list separated by :_: is an entire originate string so you are
effectively doing 5 concurrent calls to originate and only supplying one
leg.  I don't believe that feature works with only one leg supplied.

 

 

On Mon, Dec 10, 2012 at 10:57 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

HI,

 

I have a bridge statement used dozens and dozens of times a day:

EXECUTE sofia/external_noauth/anonymous at 64.136.174.30
bridge(user/300 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/301 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/203 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/302 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/303 at fs_mbri2.bizfocused.com)

 

Very occasionally, recently extensions (at least 300 and 302) have returned
[CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

All of these extensions are at the same remote location using the same wan
links/routers, etc.

 

What does that mean?  

 

Pastebin: http://pastebin.freeswitch.org/pastebin.php?dl=20306

 

Also, the 8 second delay appears to be ignored (I think, I have never asked
the customer or been on site to test).  Any 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

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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From sdevoy at bizfocused.com  Thu Dec 13 20:00:58 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 13 Dec 2012 12:00:58 -0500
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
Message-ID: <0d3e01cdd953$6cab48d0$4601da70$@bizfocused.com>

NEVERMIND -  RTFM!

 

Found it.

Sorry.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 10, 2012 12:33 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

The SERVICE_NOT_IMPLEMENTED is the q.850 translation of the 406 you received
over SIP.

If you turn on the siptrace with "sofia global siptrace on" you may get more
details.

 

Also leg_delay_start does not do much good in enterprise originate as each
url in the list separated by :_: is an entire originate string so you are
effectively doing 5 concurrent calls to originate and only supplying one
leg.  I don't believe that feature works with only one leg supplied.

 

 

On Mon, Dec 10, 2012 at 10:57 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

HI,

 

I have a bridge statement used dozens and dozens of times a day:

EXECUTE sofia/external_noauth/anonymous at 64.136.174.30
bridge(user/300 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/301 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/203 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/302 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/303 at fs_mbri2.bizfocused.com)

 

Very occasionally, recently extensions (at least 300 and 302) have returned
[CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

All of these extensions are at the same remote location using the same wan
links/routers, etc.

 

What does that mean?  

 

Pastebin: http://pastebin.freeswitch.org/pastebin.php?dl=20306

 

Also, the 8 second delay appears to be ignored (I think, I have never asked
the customer or been on site to test).  Any 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

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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From anthony.minessale at gmail.com  Thu Dec 13 20:04:08 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Dec 2012 11:04:08 -0600
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
Message-ID: <CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>

edit logfile.conf.xml

make sure the map line looks like:

<map name="all" value="debug,info,notice,warning,err,crit,alert,console"/>



On Thu, Dec 13, 2012 at 10:58 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> How do I get sofia sip trace to go in the Freeswitch log?  Do I have to
> keep separate logs and try to match them up?****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 10, 2012 12:33 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
> ** **
>
> The SERVICE_NOT_IMPLEMENTED is the q.850 translation of the 406 you
> received over SIP.****
>
> If you turn on the siptrace with "sofia global siptrace on" you may get
> more details.****
>
> ** **
>
> Also leg_delay_start does not do much good in enterprise originate as each
> url in the list separated by :_: is an entire originate string so you are
> effectively doing 5 concurrent calls to originate and only supplying one
> leg.  I don't believe that feature works with only one leg supplied.****
>
> ** **
>
> ** **
>
> On Mon, Dec 10, 2012 at 10:57 AM, Sean Devoy <sdevoy at bizfocused.com>
> wrote:****
>
> HI,****
>
>  ****
>
> I have a bridge statement used dozens and dozens of times a day:****
>
> EXECUTE sofia/external_noauth/anonymous at 64.136.174.30 bridge(user/
> 300 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> 301 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> 203 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> 302 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> 303 at fs_mbri2.bizfocused.com)****
>
>  ****
>
> Very occasionally, recently extensions (at least 300 and 302) have
> returned [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> All of these extensions are at the same remote location using the same wan
> links/routers, etc.****
>
>  ****
>
> What does that mean?  ****
>
>  ****
>
> Pastebin: http://pastebin.freeswitch.org/pastebin.php?dl=20306****
>
>  ****
>
> Also, the 8 second delay appears to be ignored (I think, I have never
> asked the customer or been on site to test).  Any 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
> ** **
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
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From curriegrad2004 at gmail.com  Thu Dec 13 20:05:29 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 13 Dec 2012 09:05:29 -0800
Subject: [Freeswitch-users] Missing bootstrap.sh in the snapshots
In-Reply-To: <33095823FD21DF429B481B5163264B799F3A97C133@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B799F3A97C133@VMBX102.ihostexchange.net>
Message-ID: <CAMWiMy_jkCSU6Tp1As0XocAXWS7bu78E_+9swFGGJ4cwtwGzOA@mail.gmail.com>

If you see the configure script in the root of the directory then they
are already pre-bootstrapped.

On Thu, Dec 13, 2012 at 8:06 AM, Yungwei Chen <yungwei at resolvity.com> wrote:
> Hi,
>
> I found that bootstrap.sh is missing in some tar balls, such as freeswitch-1.2.1.tar.bz2 and freeswitch-1.2.5.3.tar.bz2.
> Are they already bootstraped?
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Dec 13 20:26:17 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 13 Dec 2012 12:26:17 -0500
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
Message-ID: <0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>

It is set that way Anthony.

I got it going to the log now, but it is also going to the console.  I have
not found a way to get it in the file and not the console.

 

I followed this, but got no output anywhere!

http://lists.freeswitch.org/pipermail/freeswitch-users/2010-February/053516.
html

 

Thanks again.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Thursday, December 13, 2012 12:04 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

edit logfile.conf.xml

 

make sure the map line looks like:

 

<map name="all" value="debug,info,notice,warning,err,crit,alert,console"/>

 

 

On Thu, Dec 13, 2012 at 10:58 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

How do I get sofia sip trace to go in the Freeswitch log?  Do I have to keep
separate logs and try to match them up?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 10, 2012 12:33 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

The SERVICE_NOT_IMPLEMENTED is the q.850 translation of the 406 you received
over SIP.

If you turn on the siptrace with "sofia global siptrace on" you may get more
details.

 

Also leg_delay_start does not do much good in enterprise originate as each
url in the list separated by :_: is an entire originate string so you are
effectively doing 5 concurrent calls to originate and only supplying one
leg.  I don't believe that feature works with only one leg supplied.

 

 

On Mon, Dec 10, 2012 at 10:57 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

HI,

 

I have a bridge statement used dozens and dozens of times a day:

EXECUTE sofia/external_noauth/anonymous at 64.136.174.30
bridge(user/300 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/301 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/203 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/302 at fs_mbri2.bizfocused.com :_:
[leg_delay_start=8]user/303 at fs_mbri2.bizfocused.com)

 

Very occasionally, recently extensions (at least 300 and 302) have returned
[CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

All of these extensions are at the same remote location using the same wan
links/routers, etc.

 

What does that mean?  

 

Pastebin: http://pastebin.freeswitch.org/pastebin.php?dl=20306

 

Also, the 8 second delay appears to be ignored (I think, I have never asked
the customer or been on site to test).  Any 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

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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900> 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From anthony.minessale at gmail.com  Thu Dec 13 20:35:24 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Dec 2012 11:35:24 -0600
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>
Message-ID: <CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>

You can set tracelevel under global settings in sofia.conf.xml

<param name="tracelevel" value="debug"/>

or from cli

sofia tracelevel debug

that will make the traces in debug level instead of console so then if your
console level is less than debug you won't see them but they will still go
in the log.




On Thu, Dec 13, 2012 at 11:26 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-February/053516.html




-- 
Anthony Minessale II

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
-------------- next part --------------
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From krice at freeswitch.org  Thu Dec 13 20:41:47 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 13 Dec 2012 11:41:47 -0600
Subject: [Freeswitch-users] Missing bootstrap.sh in the snapshots
In-Reply-To: <CAMWiMy_jkCSU6Tp1As0XocAXWS7bu78E_+9swFGGJ4cwtwGzOA@mail.gmail.com>
Message-ID: <CCEF6F7B.61F9B%krice@freeswitch.org>

The Source Tarballs are always pre-bootstrapped... If you have a problem
theres a rebootstrap script...


On 12/13/12 11:05 AM, "curriegrad2004" <curriegrad2004 at gmail.com> wrote:

> If you see the configure script in the root of the directory then they
> are already pre-bootstrapped.
> 
> On Thu, Dec 13, 2012 at 8:06 AM, Yungwei Chen <yungwei at resolvity.com> wrote:
>> Hi,
>> 
>> I found that bootstrap.sh is missing in some tar balls, such as
>> freeswitch-1.2.1.tar.bz2 and freeswitch-1.2.5.3.tar.bz2.
>> Are they already bootstraped?
>> 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From covici at ccs.covici.com  Thu Dec 13 20:42:32 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 13 Dec 2012 12:42:32 -0500
Subject: [Freeswitch-users] console directive not working (was Re:
	[CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED])
In-Reply-To: <CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
Message-ID: <24739.1355420552@ccs.covici.com>

Hi.  I was tracing some events in the console, but they did not appear
in the log file at all, even though I had the map  directive exactly as
stated.  Should I file a jira?

Anthony Minessale <anthony.minessale at gmail.com> wrote:

> edit logfile.conf.xml
> 
> make sure the map line looks like:
> 
> <map name="all" value="debug,info,notice,warning,err,crit,alert,console"/>
> 
> 
> 
> On Thu, Dec 13, 2012 at 10:58 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> 
> > How do I get sofia sip trace to go in the Freeswitch log?  Do I have to
> > keep separate logs and try to match them up?****
> >
> > ** **
> >
> > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> > Minessale
> > *Sent:* Monday, December 10, 2012 12:33 PM
> > *To:* FreeSWITCH Users Help
> > *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> > [SERVICE_NOT_IMPLEMENTED]****
> >
> > ** **
> >
> > The SERVICE_NOT_IMPLEMENTED is the q.850 translation of the 406 you
> > received over SIP.****
> >
> > If you turn on the siptrace with "sofia global siptrace on" you may get
> > more details.****
> >
> > ** **
> >
> > Also leg_delay_start does not do much good in enterprise originate as each
> > url in the list separated by :_: is an entire originate string so you are
> > effectively doing 5 concurrent calls to originate and only supplying one
> > leg.  I don't believe that feature works with only one leg supplied.****
> >
> > ** **
> >
> > ** **
> >
> > On Mon, Dec 10, 2012 at 10:57 AM, Sean Devoy <sdevoy at bizfocused.com>
> > wrote:****
> >
> > HI,****
> >
> >  ****
> >
> > I have a bridge statement used dozens and dozens of times a day:****
> >
> > EXECUTE sofia/external_noauth/anonymous at 64.136.174.30 bridge(user/
> > 300 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> > 301 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> > 203 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> > 302 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> > 303 at fs_mbri2.bizfocused.com)****
> >
> >  ****
> >
> > Very occasionally, recently extensions (at least 300 and 302) have
> > returned [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]****
> >
> >  ****
> >
> > All of these extensions are at the same remote location using the same wan
> > links/routers, etc.****
> >
> >  ****
> >
> > What does that mean?  ****
> >
> >  ****
> >
> > Pastebin: http://pastebin.freeswitch.org/pastebin.php?dl=20306****
> >
> >  ****
> >
> > Also, the 8 second delay appears to be ignored (I think, I have never
> > asked the customer or been on site to test).  Any 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org****
> >
> >
> >
> > ****
> >
> > ** **
> >
> > --
> > Anthony Minessale II
> >
> > 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> -- 
> Anthony Minessale II
> 
> 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
> 
> ----------------------------------------------------
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 anthony.minessale at gmail.com  Thu Dec 13 20:45:02 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Dec 2012 11:45:02 -0600
Subject: [Freeswitch-users] what can I have in ${} construction
In-Reply-To: <30097.1355410423@ccs.covici.com>
References: <30097.1355410423@ccs.covici.com>
Message-ID: <CAKbxfG8LEc3f6s+-PyTF1rF2RYR0tF=QMOiXURp2M-Uaekngpw@mail.gmail.com>

It supports the substring with positive and negative vals so:

${foo:3:4} would start at char 3 in the string and eval to the next 4 chars
${foo:-3:2} would start at the end and go back 3 chars then print the next
2 chars

If the var name is followed by a ( or a space, it will pass the values to
the FSAPI and expand the result inline.

${sofia_contact 1004}

equiv of...

${sofia_contact(1004)}







On Thu, Dec 13, 2012 at 8:53 AM, <covici at ccs.covici.com> wrote:

> Hi.  I was experimenting with ${variable} and was wondering what could
> be in there before the }?  I can do a substring if I say variable:2, but
> I wonder what else you can do?  The bash manual has all kinds of things
> you can have, but how much of this will fs do?
>
> Thanks in advance for any suggestions.
>
> --
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
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From jaasmailing at gmail.com  Thu Dec 13 20:50:56 2012
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Thu, 13 Dec 2012 18:50:56 +0100
Subject: [Freeswitch-users] Problem with outbound calls,
	transfers and bypass_media
Message-ID: <50CA1580.10800@gmail.com>

Hi all,

I have a problemin dealing outbound calls and a transfer with bypass_media.
Let me explain with an example(outbound call):
1) alocal extension (200) calls an external number
2) 200 transfers the call to another local extension 202
3) 202 and the external number are bridged with bypass_mediaandthere is 
no audio (of course as they are under NAT).

FS dialplan beetween local extension is:

  <extension name="local_extension" >
<condition field="username" expression="(^\d{2,5}$)" break="never" >
<action application="set" data="bypass_media=true" />
</condition>
    [...]

If I call from an external number to a local extension and than transfer 
the call to another extension, all works fine.
Indeed, in the external profile I have:
inbound-proxy-media=false
inbound-bypass-media=false
inbound-late-negotiation=true
(as I think the bypass_media does not overwrite the global configuration 
in the external profile)


Is there a global configuration for no-media mode also for the outbound 
calls?
In other word, I would like to have bypass_media just beetween local 
extensions.


Best regards,
Carlo

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From anthony.minessale at gmail.com  Thu Dec 13 20:54:05 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Dec 2012 11:54:05 -0600
Subject: [Freeswitch-users] console directive not working (was Re:
 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED])
In-Reply-To: <24739.1355420552@ccs.covici.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
	<24739.1355420552@ccs.covici.com>
Message-ID: <CAKbxfG8a230nK143wNNcRiuFLUQq0Q3dFhevs6SnSUs3AfXJ=A@mail.gmail.com>

tracing events is a fs_cli operation.  The console stuff we are discussing
is in the actual FS console from the terminal it was started from.  Those
events are not actually on the console only in fs_cli over event_socket


On Thu, Dec 13, 2012 at 11:42 AM, <covici at ccs.covici.com> wrote:

> Hi.  I was tracing some events in the console, but they did not appear
> in the log file at all, even though I had the map  directive exactly as
> stated.  Should I file a jira?
>
> Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
> > edit logfile.conf.xml
> >
> > make sure the map line looks like:
> >
> > <map name="all"
> value="debug,info,notice,warning,err,crit,alert,console"/>
> >
> >
> >
> > On Thu, Dec 13, 2012 at 10:58 AM, Sean Devoy <sdevoy at bizfocused.com>
> wrote:
> >
> > > How do I get sofia sip trace to go in the Freeswitch log?  Do I have to
> > > keep separate logs and try to match them up?****
> > >
> > > ** **
> > >
> > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> > > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> > > Minessale
> > > *Sent:* Monday, December 10, 2012 12:33 PM
> > > *To:* FreeSWITCH Users Help
> > > *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> > > [SERVICE_NOT_IMPLEMENTED]****
> > >
> > > ** **
> > >
> > > The SERVICE_NOT_IMPLEMENTED is the q.850 translation of the 406 you
> > > received over SIP.****
> > >
> > > If you turn on the siptrace with "sofia global siptrace on" you may get
> > > more details.****
> > >
> > > ** **
> > >
> > > Also leg_delay_start does not do much good in enterprise originate as
> each
> > > url in the list separated by :_: is an entire originate string so you
> are
> > > effectively doing 5 concurrent calls to originate and only supplying
> one
> > > leg.  I don't believe that feature works with only one leg
> supplied.****
> > >
> > > ** **
> > >
> > > ** **
> > >
> > > On Mon, Dec 10, 2012 at 10:57 AM, Sean Devoy <sdevoy at bizfocused.com>
> > > wrote:****
> > >
> > > HI,****
> > >
> > >  ****
> > >
> > > I have a bridge statement used dozens and dozens of times a day:****
> > >
> > > EXECUTE sofia/external_noauth/anonymous at 64.136.174.30 bridge(user/
> > > 300 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> > > 301 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> > > 203 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> > > 302 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> > > 303 at fs_mbri2.bizfocused.com)****
> > >
> > >  ****
> > >
> > > Very occasionally, recently extensions (at least 300 and 302) have
> > > returned [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]****
> > >
> > >  ****
> > >
> > > All of these extensions are at the same remote location using the same
> wan
> > > links/routers, etc.****
> > >
> > >  ****
> > >
> > > What does that mean?  ****
> > >
> > >  ****
> > >
> > > Pastebin: http://pastebin.freeswitch.org/pastebin.php?dl=20306****
> > >
> > >  ****
> > >
> > > Also, the 8 second delay appears to be ignored (I think, I have never
> > > asked the customer or been on site to test).  Any 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
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org****
> > >
> > >
> > >
> > > ****
> > >
> > > ** **
> > >
> > > --
> > > Anthony Minessale II
> > >
> > > 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
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> >
> > --
> > Anthony Minessale II
> >
> > 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
> >
> > ----------------------------------------------------
> > 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
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From steveayre at gmail.com  Thu Dec 13 20:54:59 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 13 Dec 2012 17:54:59 +0000
Subject: [Freeswitch-users] mod conference keeps core dumping
In-Reply-To: <4136.1355369406@ccs.covici.com>
References: <13830.1355365788@ccs.covici.com>
	<1355368559688-7585476.post@n2.nabble.com>
	<4136.1355369406@ccs.covici.com>
Message-ID: <CAFiqYukcnB8WRmZou8mbh6bk0BPBzoUeHJN+4O=PWWKi0JCsUg@mail.gmail.com>

For future reference, bugs are filed at http://jira.freeswitch.org/, not on
the list. A segfault (coredump) is always a bug.

It makes it much easier to track things, and gives bugs reference numbers
that can be put into the git commit logs to indicate commits that fix bugs.

-Steve


On 13 December 2012 03:30, <covici at ccs.covici.com> wrote:

> Thanks much.
>
> Jeff Lenk <jeff at jefflenk.com> wrote:
>
> > John,
> >
> > Fixed in git head
> >
> > Jeff
> >
> >
> >
> > --
> > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/mod-conference-keeps-core-dumping-tp7585475p7585476.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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.venugopan at mundio.com  Thu Dec 13 21:01:09 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Thu, 13 Dec 2012 18:01:09 +0000
Subject: [Freeswitch-users] SIP error
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233B7F5@Mail-Kilo.squay.com>

Hi team,

I am facing with the below error after installing freeswitch and registering.

2012-12-13 17:55:15.573019 [ERR] sofia.c:1347 Error Creating SIP UA for profile: internal

Guess this is checking /usr/local/freeswitch/conf/sip_profiles/internal.xml

I have attached my internal.xml as well. Please let me know. Thanks.

Regards,
Archana.V

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From covici at ccs.covici.com  Thu Dec 13 21:12:33 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 13 Dec 2012 13:12:33 -0500
Subject: [Freeswitch-users] what can I have in ${} construction
In-Reply-To: <CAKbxfG8LEc3f6s+-PyTF1rF2RYR0tF=QMOiXURp2M-Uaekngpw@mail.gmail.com>
References: <30097.1355410423@ccs.covici.com>
	<CAKbxfG8LEc3f6s+-PyTF1rF2RYR0tF=QMOiXURp2M-Uaekngpw@mail.gmail.com>
Message-ID: <29257.1355422353@ccs.covici.com>

Thanks much.

Anthony Minessale <anthony.minessale at gmail.com> wrote:

> It supports the substring with positive and negative vals so:
> 
> ${foo:3:4} would start at char 3 in the string and eval to the next 4 chars
> ${foo:-3:2} would start at the end and go back 3 chars then print the next
> 2 chars
> 
> If the var name is followed by a ( or a space, it will pass the values to
> the FSAPI and expand the result inline.
> 
> ${sofia_contact 1004}
> 
> equiv of...
> 
> ${sofia_contact(1004)}
> 
> 
> 
> 
> 
> 
> 
> On Thu, Dec 13, 2012 at 8:53 AM, <covici at ccs.covici.com> wrote:
> 
> > Hi.  I was experimenting with ${variable} and was wondering what could
> > be in there before the }?  I can do a substring if I say variable:2, but
> > I wonder what else you can do?  The bash manual has all kinds of things
> > you can have, but how much of this will fs do?
> >
> > Thanks in advance for any suggestions.
> >
> > --
> > 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> 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
> 
> ----------------------------------------------------
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 covici at ccs.covici.com  Thu Dec 13 21:15:17 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 13 Dec 2012 13:15:17 -0500
Subject: [Freeswitch-users] console directive not working (was Re:
	[CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED])
In-Reply-To: <CAKbxfG8a230nK143wNNcRiuFLUQq0Q3dFhevs6SnSUs3AfXJ=A@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
	<24739.1355420552@ccs.covici.com>
	<CAKbxfG8a230nK143wNNcRiuFLUQq0Q3dFhevs6SnSUs3AfXJ=A@mail.gmail.com>
Message-ID: <29701.1355422517@ccs.covici.com>

OK, thanks.

Anthony Minessale <anthony.minessale at gmail.com> wrote:

> tracing events is a fs_cli operation.  The console stuff we are discussing
> is in the actual FS console from the terminal it was started from.  Those
> events are not actually on the console only in fs_cli over event_socket
> 
> 
> On Thu, Dec 13, 2012 at 11:42 AM, <covici at ccs.covici.com> wrote:
> 
> > Hi.  I was tracing some events in the console, but they did not appear
> > in the log file at all, even though I had the map  directive exactly as
> > stated.  Should I file a jira?
> >
> > Anthony Minessale <anthony.minessale at gmail.com> wrote:
> >
> > > edit logfile.conf.xml
> > >
> > > make sure the map line looks like:
> > >
> > > <map name="all"
> > value="debug,info,notice,warning,err,crit,alert,console"/>
> > >
> > >
> > >
> > > On Thu, Dec 13, 2012 at 10:58 AM, Sean Devoy <sdevoy at bizfocused.com>
> > wrote:
> > >
> > > > How do I get sofia sip trace to go in the Freeswitch log?  Do I have to
> > > > keep separate logs and try to match them up?****
> > > >
> > > > ** **
> > > >
> > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> > > > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> > > > Minessale
> > > > *Sent:* Monday, December 10, 2012 12:33 PM
> > > > *To:* FreeSWITCH Users Help
> > > > *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> > > > [SERVICE_NOT_IMPLEMENTED]****
> > > >
> > > > ** **
> > > >
> > > > The SERVICE_NOT_IMPLEMENTED is the q.850 translation of the 406 you
> > > > received over SIP.****
> > > >
> > > > If you turn on the siptrace with "sofia global siptrace on" you may get
> > > > more details.****
> > > >
> > > > ** **
> > > >
> > > > Also leg_delay_start does not do much good in enterprise originate as
> > each
> > > > url in the list separated by :_: is an entire originate string so you
> > are
> > > > effectively doing 5 concurrent calls to originate and only supplying
> > one
> > > > leg.  I don't believe that feature works with only one leg
> > supplied.****
> > > >
> > > > ** **
> > > >
> > > > ** **
> > > >
> > > > On Mon, Dec 10, 2012 at 10:57 AM, Sean Devoy <sdevoy at bizfocused.com>
> > > > wrote:****
> > > >
> > > > HI,****
> > > >
> > > >  ****
> > > >
> > > > I have a bridge statement used dozens and dozens of times a day:****
> > > >
> > > > EXECUTE sofia/external_noauth/anonymous at 64.136.174.30 bridge(user/
> > > > 300 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> > > > 301 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> > > > 203 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> > > > 302 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> > > > 303 at fs_mbri2.bizfocused.com)****
> > > >
> > > >  ****
> > > >
> > > > Very occasionally, recently extensions (at least 300 and 302) have
> > > > returned [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]****
> > > >
> > > >  ****
> > > >
> > > > All of these extensions are at the same remote location using the same
> > wan
> > > > links/routers, etc.****
> > > >
> > > >  ****
> > > >
> > > > What does that mean?  ****
> > > >
> > > >  ****
> > > >
> > > > Pastebin: http://pastebin.freeswitch.org/pastebin.php?dl=20306****
> > > >
> > > >  ****
> > > >
> > > > Also, the 8 second delay appears to be ignored (I think, I have never
> > > > asked the customer or been on site to test).  Any 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
> > > >
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org****
> > > >
> > > >
> > > >
> > > > ****
> > > >
> > > > ** **
> > > >
> > > > --
> > > > Anthony Minessale II
> > > >
> > > > 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
> > > >
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > > >
> > > >
> > >
> > >
> > > --
> > > Anthony Minessale II
> > >
> > > 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
> > >
> > > ----------------------------------------------------
> > > 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
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> 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
> 
> ----------------------------------------------------
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 covici at ccs.covici.com  Thu Dec 13 21:17:45 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 13 Dec 2012 13:17:45 -0500
Subject: [Freeswitch-users] mod conference keeps core dumping
In-Reply-To: <CAFiqYukcnB8WRmZou8mbh6bk0BPBzoUeHJN+4O=PWWKi0JCsUg@mail.gmail.com>
References: <13830.1355365788@ccs.covici.com>
	<1355368559688-7585476.post@n2.nabble.com>
	<4136.1355369406@ccs.covici.com>
	<CAFiqYukcnB8WRmZou8mbh6bk0BPBzoUeHJN+4O=PWWKi0JCsUg@mail.gmail.com>
Message-ID: <30081.1355422665@ccs.covici.com>

I know, but I wanted someone to take a look at it, in case it was
something I did.  Jeff fixed it very quickly -- very nice of him to do
that.


Steven Ayre <steveayre at gmail.com> wrote:

> For future reference, bugs are filed at http://jira.freeswitch.org/, not on
> the list. A segfault (coredump) is always a bug.
> 
> It makes it much easier to track things, and gives bugs reference numbers
> that can be put into the git commit logs to indicate commits that fix bugs.
> 
> -Steve
> 
> 
> On 13 December 2012 03:30, <covici at ccs.covici.com> wrote:
> 
> > Thanks much.
> >
> > Jeff Lenk <jeff at jefflenk.com> wrote:
> >
> > > John,
> > >
> > > Fixed in git head
> > >
> > > Jeff
> > >
> > >
> > >
> > > --
> > > View this message in context:
> > http://freeswitch-users.2379917.n2.nabble.com/mod-conference-keeps-core-dumping-tp7585475p7585476.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
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 anthony.minessale at gmail.com  Thu Dec 13 21:27:26 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Dec 2012 12:27:26 -0600
Subject: [Freeswitch-users] mod conference keeps core dumping
In-Reply-To: <30081.1355422665@ccs.covici.com>
References: <13830.1355365788@ccs.covici.com>
	<1355368559688-7585476.post@n2.nabble.com>
	<4136.1355369406@ccs.covici.com>
	<CAFiqYukcnB8WRmZou8mbh6bk0BPBzoUeHJN+4O=PWWKi0JCsUg@mail.gmail.com>
	<30081.1355422665@ccs.covici.com>
Message-ID: <CAKbxfG8TFemtbcsdn+YV6vrxmbJ1GKf_hW55XBV7R=A4TNO8Vg@mail.gmail.com>

No need to feel guilty about opening a Jira that turns out to not be a bug.
 Those are the most fun to close!
Either way they end up on our plate only with jira we can tie the commit
against the problem description without repeating it.



On Thu, Dec 13, 2012 at 12:17 PM, <covici at ccs.covici.com> wrote:

> I know, but I wanted someone to take a look at it, in case it was
> something I did.  Jeff fixed it very quickly -- very nice of him to do
> that.
>
>
> Steven Ayre <steveayre at gmail.com> wrote:
>
> > For future reference, bugs are filed at http://jira.freeswitch.org/,
> not on
> > the list. A segfault (coredump) is always a bug.
> >
> > It makes it much easier to track things, and gives bugs reference numbers
> > that can be put into the git commit logs to indicate commits that fix
> bugs.
> >
> > -Steve
> >
> >
> > On 13 December 2012 03:30, <covici at ccs.covici.com> wrote:
> >
> > > Thanks much.
> > >
> > > Jeff Lenk <jeff at jefflenk.com> wrote:
> > >
> > > > John,
> > > >
> > > > Fixed in git head
> > > >
> > > > Jeff
> > > >
> > > >
> > > >
> > > > --
> > > > View this message in context:
> > >
> http://freeswitch-users.2379917.n2.nabble.com/mod-conference-keeps-core-dumping-tp7585475p7585476.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
> > > >
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at 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
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
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From anthony.minessale at gmail.com  Thu Dec 13 21:30:29 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Dec 2012 12:30:29 -0600
Subject: [Freeswitch-users] SIP error
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233B7F5@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233B7F5@Mail-Kilo.squay.com>
Message-ID: <CAKbxfG-JwGOY590XzmM-LoJsw2On=Ucg197t7AqdjgvZf6ijvQ@mail.gmail.com>

1) You have both sip-ip and ext-sip-ip set to the same val, if you have it
in sip-ip and its a public addr, you should leave ext-sip-ip empty.  If the
ip is a public ip and you are on a lan addr, you should put the real lan
addr in sip-ip

That error comes from a fail to bind to the port.  Usual causes are the ip
is not actually on the box or something is already listening on it.

The error actually points this out so by looking at your post, you probably
have older code that you should update.




On Thu, Dec 13, 2012 at 12:01 PM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi team,****
>
> ** **
>
> I am facing with the below error after installing freeswitch and
> registering.****
>
> ** **
>
> 2012-12-13 17:55:15.573019 [ERR] sofia.c:1347 Error Creating SIP UA for
> profile: internal****
>
> ** **
>
> Guess this is checking /usr/local/freeswitch/conf/sip_profiles/internal.xml
> ****
>
> ** **
>
> I have attached my internal.xml as well. Please let me know. Thanks.****
>
> ** **
>
> Regards,****
>
> Archana.V****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Anthony Minessale II

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
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From jpablolorenzetti at hotmail.com  Thu Dec 13 21:31:03 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Thu, 13 Dec 2012 18:31:03 +0000
Subject: [Freeswitch-users] callcenter with external agents
Message-ID: <DUB002-W3915B9DAC4F64D9FD8AE3ED94E0@phx.gbl>

Hi, i m trying to set up a callcenter but the lines for the agents are mobile phones in a mobile network and are not attached to freeswitch, i m trying to set it up with the following:
<agent name="1000 at default" type="callback" contact="[call_timeout=10]sofia/external-huawei_gw/XXXXXX" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />
XXXXX being the mobile number but i m getting the following error:
Member 2018 <2018> in queue 'callcenter at default' reached max wait of 0 sec. with no agent plus join grace period of 5 sec.
and i see that freeswitch does not even try to dial out to the trunk. 
i m wondering if configuring it the way i need it is even possible as i think the problem may reside in the fact that freeswitch does not actually see the agent registered. thanks!! 		 	   		  
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From kris at kriskinc.com  Thu Dec 13 22:06:04 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 13 Dec 2012 14:06:04 -0500
Subject: [Freeswitch-users] SRTP + hardware crypto
In-Reply-To: <50C954A1.7020302@puzzled.xs4all.nl>
References: <CAKDfjgesQGS+EK=8Dur6xVJZO16eP-51ctEH9bf8WWJLzrU65Q@mail.gmail.com>
	<50C954A1.7020302@puzzled.xs4all.nl>
Message-ID: <CAKDfjgdMbjoeKTrY7Z5tEHAsrmeuadXQpmGqG12k0ShAfcuwPw@mail.gmail.com>

On Wed, Dec 12, 2012 at 11:08 PM, Patrick Lists
<freeswitch-list at puzzled.xs4all.nl> wrote:
>
> There's support for AES in 2010 and later Intel cpu's. Maybe that could
> possibly be used for this acceleration?
>
> http://software.intel.com/en-us/articles/intel-advanced-encryption-standard-aes-instructions-set
>
> An overview of cpu's with AES can be found here:
> http://ark.intel.com/search/advanced/?s=t&AESTech=true

  ...and VIA Padlock, Geode AES, etc, etc.

> Last time I looked the price of those hardware cards is considerable so
> it should really bump max SRTP calls to justify the CAPEX.

  Hardware (even CPUs mentioned above) accelerated crypto is exposed
in OpenSSL through engine.  It doesn't have to be those expensive
hardware cards...

-- 
Kristian Kielhofner


From covici at ccs.covici.com  Thu Dec 13 22:44:09 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 13 Dec 2012 14:44:09 -0500
Subject: [Freeswitch-users] mod conference keeps core dumping
In-Reply-To: <CAKbxfG8TFemtbcsdn+YV6vrxmbJ1GKf_hW55XBV7R=A4TNO8Vg@mail.gmail.com>
References: <13830.1355365788@ccs.covici.com>
	<1355368559688-7585476.post@n2.nabble.com>
	<4136.1355369406@ccs.covici.com>
	<CAFiqYukcnB8WRmZou8mbh6bk0BPBzoUeHJN+4O=PWWKi0JCsUg@mail.gmail.com>
	<30081.1355422665@ccs.covici.com>
	<CAKbxfG8TFemtbcsdn+YV6vrxmbJ1GKf_hW55XBV7R=A4TNO8Vg@mail.gmail.com>
Message-ID: <9701.1355427849@ccs.covici.com>

I will be more able to open jiras when we get that Perl script or
whatever to make it easier, right now its a pain in the neck.

Anthony Minessale <anthony.minessale at gmail.com> wrote:

> No need to feel guilty about opening a Jira that turns out to not be a bug.
>  Those are the most fun to close!
> Either way they end up on our plate only with jira we can tie the commit
> against the problem description without repeating it.
> 
> 
> 
> On Thu, Dec 13, 2012 at 12:17 PM, <covici at ccs.covici.com> wrote:
> 
> > I know, but I wanted someone to take a look at it, in case it was
> > something I did.  Jeff fixed it very quickly -- very nice of him to do
> > that.
> >
> >
> > Steven Ayre <steveayre at gmail.com> wrote:
> >
> > > For future reference, bugs are filed at http://jira.freeswitch.org/,
> > not on
> > > the list. A segfault (coredump) is always a bug.
> > >
> > > It makes it much easier to track things, and gives bugs reference numbers
> > > that can be put into the git commit logs to indicate commits that fix
> > bugs.
> > >
> > > -Steve
> > >
> > >
> > > On 13 December 2012 03:30, <covici at ccs.covici.com> wrote:
> > >
> > > > Thanks much.
> > > >
> > > > Jeff Lenk <jeff at jefflenk.com> wrote:
> > > >
> > > > > John,
> > > > >
> > > > > Fixed in git head
> > > > >
> > > > > Jeff
> > > > >
> > > > >
> > > > >
> > > > > --
> > > > > View this message in context:
> > > >
> > http://freeswitch-users.2379917.n2.nabble.com/mod-conference-keeps-core-dumping-tp7585475p7585476.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
> > > > >
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at 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
> > > >
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at 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
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> 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
> 
> ----------------------------------------------------
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 anthony.minessale at gmail.com  Thu Dec 13 23:08:11 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Dec 2012 14:08:11 -0600
Subject: [Freeswitch-users] mod conference keeps core dumping
In-Reply-To: <9701.1355427849@ccs.covici.com>
References: <13830.1355365788@ccs.covici.com>
	<1355368559688-7585476.post@n2.nabble.com>
	<4136.1355369406@ccs.covici.com>
	<CAFiqYukcnB8WRmZou8mbh6bk0BPBzoUeHJN+4O=PWWKi0JCsUg@mail.gmail.com>
	<30081.1355422665@ccs.covici.com>
	<CAKbxfG8TFemtbcsdn+YV6vrxmbJ1GKf_hW55XBV7R=A4TNO8Vg@mail.gmail.com>
	<9701.1355427849@ccs.covici.com>
Message-ID: <CAKbxfG9wfyaB_bVbfWTwpDuM6zBV_-fgP+oJKvMSPZ-TMqF=CQ@mail.gmail.com>

Yes I understand your circumstances.  I still like to comment as much as
possible on the topic since I probably have to mention it 10 times a week
and the more I document it the more people may stumble upon it.

As long as someone is making a tool:

How about something that joins the list and routes all the emails to
procmail so we can send it magic commands like hit reply and transform an
email thread into a JIRA on the fly ;)




On Thu, Dec 13, 2012 at 1:44 PM, <covici at ccs.covici.com> wrote:

> I will be more able to open jiras when we get that Perl script or
> whatever to make it easier, right now its a pain in the neck.
>
> Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
> > No need to feel guilty about opening a Jira that turns out to not be a
> bug.
> >  Those are the most fun to close!
> > Either way they end up on our plate only with jira we can tie the commit
> > against the problem description without repeating it.
> >
> >
> >
> > On Thu, Dec 13, 2012 at 12:17 PM, <covici at ccs.covici.com> wrote:
> >
> > > I know, but I wanted someone to take a look at it, in case it was
> > > something I did.  Jeff fixed it very quickly -- very nice of him to do
> > > that.
> > >
> > >
> > > Steven Ayre <steveayre at gmail.com> wrote:
> > >
> > > > For future reference, bugs are filed at http://jira.freeswitch.org/,
> > > not on
> > > > the list. A segfault (coredump) is always a bug.
> > > >
> > > > It makes it much easier to track things, and gives bugs reference
> numbers
> > > > that can be put into the git commit logs to indicate commits that fix
> > > bugs.
> > > >
> > > > -Steve
> > > >
> > > >
> > > > On 13 December 2012 03:30, <covici at ccs.covici.com> wrote:
> > > >
> > > > > Thanks much.
> > > > >
> > > > > Jeff Lenk <jeff at jefflenk.com> wrote:
> > > > >
> > > > > > John,
> > > > > >
> > > > > > Fixed in git head
> > > > > >
> > > > > > Jeff
> > > > > >
> > > > > >
> > > > > >
> > > > > > --
> > > > > > View this message in context:
> > > > >
> > >
> http://freeswitch-users.2379917.n2.nabble.com/mod-conference-keeps-core-dumping-tp7585475p7585476.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
> > > > > >
> > > > > > FreeSWITCH-users mailing list
> > > > > > FreeSWITCH-users at 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
> > > > >
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at 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
> > > >
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at 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
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > 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
> >
> > ----------------------------------------------------
> > 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
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From jaasmailing at gmail.com  Fri Dec 14 00:09:17 2012
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Thu, 13 Dec 2012 22:09:17 +0100
Subject: [Freeswitch-users] Call transfer changes media mode to bypass mode
Message-ID: <50CA43FD.1030604@gmail.com>

Hi guys,

I would like to know why acall transfer could change media mode to 
bypass mode.
In detail, If I perform an outbound call (through a voip gateway)in 
media mode and then transfer this call to a local extension (my dialplan 
forces bypass_mode between local extensions), the call doesn't maintain 
the original mode (media).
Unfortunately, as I am ina NAT environment this breaks the comunication.

In inbound calls (with inbound-proxy-media=false and 
inbound-bypass-media=false), the call after the transfer is still in 
media mode.

Is this an expected behaviour?


Best regards,
Carlo

[P.s. Irewrited my last mail for a direct approach to the problem]
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From bdfoster at endigotech.com  Fri Dec 14 01:37:38 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 13 Dec 2012 17:37:38 -0500
Subject: [Freeswitch-users] mod conference keeps core dumping
In-Reply-To: <CAKbxfG9wfyaB_bVbfWTwpDuM6zBV_-fgP+oJKvMSPZ-TMqF=CQ@mail.gmail.com>
References: <13830.1355365788@ccs.covici.com>
	<1355368559688-7585476.post@n2.nabble.com>
	<4136.1355369406@ccs.covici.com>
	<CAFiqYukcnB8WRmZou8mbh6bk0BPBzoUeHJN+4O=PWWKi0JCsUg@mail.gmail.com>
	<30081.1355422665@ccs.covici.com>
	<CAKbxfG8TFemtbcsdn+YV6vrxmbJ1GKf_hW55XBV7R=A4TNO8Vg@mail.gmail.com>
	<9701.1355427849@ccs.covici.com>
	<CAKbxfG9wfyaB_bVbfWTwpDuM6zBV_-fgP+oJKvMSPZ-TMqF=CQ@mail.gmail.com>
Message-ID: <C5ED3435-D286-4745-B97A-FE6A99D87528@endigotech.com>

Then everybody would use it :)

Maybe doing it on a separate email address and giving it out to those who need it privately? Might be easier that way too instead of trying to deal with mailing list.

Sent from my iPhone

On Dec 13, 2012, at 3:08 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> Yes I understand your circumstances.  I still like to comment as much as possible on the topic since I probably have to mention it 10 times a week and the more I document it the more people may stumble upon it.
> 
> As long as someone is making a tool:
> 
> How about something that joins the list and routes all the emails to procmail so we can send it magic commands like hit reply and transform an email thread into a JIRA on the fly ;)
> 
> 
> 
> 
> On Thu, Dec 13, 2012 at 1:44 PM, <covici at ccs.covici.com> wrote:
>> I will be more able to open jiras when we get that Perl script or
>> whatever to make it easier, right now its a pain in the neck.
>> 
>> Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> 
>> > No need to feel guilty about opening a Jira that turns out to not be a bug.
>> >  Those are the most fun to close!
>> > Either way they end up on our plate only with jira we can tie the commit
>> > against the problem description without repeating it.
>> >
>> >
>> >
>> > On Thu, Dec 13, 2012 at 12:17 PM, <covici at ccs.covici.com> wrote:
>> >
>> > > I know, but I wanted someone to take a look at it, in case it was
>> > > something I did.  Jeff fixed it very quickly -- very nice of him to do
>> > > that.
>> > >
>> > >
>> > > Steven Ayre <steveayre at gmail.com> wrote:
>> > >
>> > > > For future reference, bugs are filed at http://jira.freeswitch.org/,
>> > > not on
>> > > > the list. A segfault (coredump) is always a bug.
>> > > >
>> > > > It makes it much easier to track things, and gives bugs reference numbers
>> > > > that can be put into the git commit logs to indicate commits that fix
>> > > bugs.
>> > > >
>> > > > -Steve
>> > > >
>> > > >
>> > > > On 13 December 2012 03:30, <covici at ccs.covici.com> wrote:
>> > > >
>> > > > > Thanks much.
>> > > > >
>> > > > > Jeff Lenk <jeff at jefflenk.com> wrote:
>> > > > >
>> > > > > > John,
>> > > > > >
>> > > > > > Fixed in git head
>> > > > > >
>> > > > > > Jeff
>> > > > > >
>> > > > > >
>> > > > > >
>> > > > > > --
>> > > > > > View this message in context:
>> > > > >
>> > > http://freeswitch-users.2379917.n2.nabble.com/mod-conference-keeps-core-dumping-tp7585475p7585476.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
>> > > > > >
>> > > > > > FreeSWITCH-users mailing list
>> > > > > > FreeSWITCH-users at 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
>> > > > >
>> > > > > FreeSWITCH-users mailing list
>> > > > > FreeSWITCH-users at 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
>> > > >
>> > > > FreeSWITCH-users mailing list
>> > > > FreeSWITCH-users at 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
>> > >
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > 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
>> >
>> > ----------------------------------------------------
>> > 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
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Fri Dec 14 03:14:57 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Dec 2012 18:14:57 -0600
Subject: [Freeswitch-users] mod conference keeps core dumping
In-Reply-To: <C5ED3435-D286-4745-B97A-FE6A99D87528@endigotech.com>
References: <13830.1355365788@ccs.covici.com>
	<1355368559688-7585476.post@n2.nabble.com>
	<4136.1355369406@ccs.covici.com>
	<CAFiqYukcnB8WRmZou8mbh6bk0BPBzoUeHJN+4O=PWWKi0JCsUg@mail.gmail.com>
	<30081.1355422665@ccs.covici.com>
	<CAKbxfG8TFemtbcsdn+YV6vrxmbJ1GKf_hW55XBV7R=A4TNO8Vg@mail.gmail.com>
	<9701.1355427849@ccs.covici.com>
	<CAKbxfG9wfyaB_bVbfWTwpDuM6zBV_-fgP+oJKvMSPZ-TMqF=CQ@mail.gmail.com>
	<C5ED3435-D286-4745-B97A-FE6A99D87528@endigotech.com>
Message-ID: <CAKbxfG-PLpiRDxseGEavf1i0NWYHjxzu-G0f2M-Ka0K3Mjv-Ag@mail.gmail.com>

Yah true but then we could hook it up to c888 and let him auto correct ppl
too lol.



On Thu, Dec 13, 2012 at 4:37 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> Then everybody would use it :)
>
> Maybe doing it on a separate email address and giving it out to those who
> need it privately? Might be easier that way too instead of trying to deal
> with mailing list.
>
> Sent from my iPhone
>
> On Dec 13, 2012, at 3:08 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> Yes I understand your circumstances.  I still like to comment as much as
> possible on the topic since I probably have to mention it 10 times a week
> and the more I document it the more people may stumble upon it.
>
> As long as someone is making a tool:
>
> How about something that joins the list and routes all the emails to
> procmail so we can send it magic commands like hit reply and transform an
> email thread into a JIRA on the fly ;)
>
>
>
>
> On Thu, Dec 13, 2012 at 1:44 PM, <covici at ccs.covici.com> wrote:
>
>> I will be more able to open jiras when we get that Perl script or
>> whatever to make it easier, right now its a pain in the neck.
>>
>> Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>
>> > No need to feel guilty about opening a Jira that turns out to not be a
>> bug.
>> >  Those are the most fun to close!
>> > Either way they end up on our plate only with jira we can tie the commit
>> > against the problem description without repeating it.
>> >
>> >
>> >
>> > On Thu, Dec 13, 2012 at 12:17 PM, <covici at ccs.covici.com> wrote:
>> >
>> > > I know, but I wanted someone to take a look at it, in case it was
>> > > something I did.  Jeff fixed it very quickly -- very nice of him to do
>> > > that.
>> > >
>> > >
>> > > Steven Ayre <steveayre at gmail.com> wrote:
>> > >
>> > > > For future reference, bugs are filed at http://jira.freeswitch.org/
>> ,
>> > > not on
>> > > > the list. A segfault (coredump) is always a bug.
>> > > >
>> > > > It makes it much easier to track things, and gives bugs reference
>> numbers
>> > > > that can be put into the git commit logs to indicate commits that
>> fix
>> > > bugs.
>> > > >
>> > > > -Steve
>> > > >
>> > > >
>> > > > On 13 December 2012 03:30, <covici at ccs.covici.com> wrote:
>> > > >
>> > > > > Thanks much.
>> > > > >
>> > > > > Jeff Lenk <jeff at jefflenk.com> wrote:
>> > > > >
>> > > > > > John,
>> > > > > >
>> > > > > > Fixed in git head
>> > > > > >
>> > > > > > Jeff
>> > > > > >
>> > > > > >
>> > > > > >
>> > > > > > --
>> > > > > > View this message in context:
>> > > > >
>> > >
>> http://freeswitch-users.2379917.n2.nabble.com/mod-conference-keeps-core-dumping-tp7585475p7585476.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
>> > > > > >
>> > > > > > FreeSWITCH-users mailing list
>> > > > > > FreeSWITCH-users at 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
>> > > > >
>> > > > > FreeSWITCH-users mailing list
>> > > > > FreeSWITCH-users at 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
>> > > >
>> > > > FreeSWITCH-users mailing list
>> > > > FreeSWITCH-users at 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
>> > >
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > 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
>> >
>> > ----------------------------------------------------
>> > 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
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Anthony Minessale II

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
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From bdfoster at endigotech.com  Fri Dec 14 05:04:00 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 13 Dec 2012 21:04:00 -0500
Subject: [Freeswitch-users] callcenter with external agents
In-Reply-To: <DUB002-W3915B9DAC4F64D9FD8AE3ED94E0@phx.gbl>
References: <DUB002-W3915B9DAC4F64D9FD8AE3ED94E0@phx.gbl>
Message-ID: <4521E504-D44A-47D9-B3EE-3EE363A74558@endigotech.com>

+1 I don't really know if this is possible. We've tried to do it this way but we ended up using some LUA scripts and mod_fifo. Unfortunately I can't release the scripts because the client won't allow me. 

Sent from my iPhone

On Dec 13, 2012, at 1:31 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:

> Hi, i m trying to set up a callcenter but the lines for the agents are mobile phones in a mobile network and are not attached to freeswitch, i m trying to set it up with the following:
> 
> <agent name="1000 at default" type="callback" contact="[call_timeout=10]sofia/external-huawei_gw/XXXXXX" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />
> 
> XXXXX being the mobile number but i m getting the following error:
> 
> Member 2018 <2018> in queue 'callcenter at default' reached max wait of 0 sec. with no agent plus join grace period of 5 sec.
> 
> and i see that freeswitch does not even try to dial out to the trunk. 
> 
> i m wondering if configuring it the way i need it is even possible as i think the problem may reside in the fact that freeswitch does not actually see the agent registered. 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From raimund.sacherer at logitravel.com  Fri Dec 14 14:06:05 2012
From: raimund.sacherer at logitravel.com (Raimund Sacherer)
Date: Fri, 14 Dec 2012 12:06:05 +0100 (CET)
Subject: [Freeswitch-users] callcenter with external agents
In-Reply-To: <4202784.3855.1355477198903.JavaMail.javamailuser@localhost>
Message-ID: <26969612.4086.1355483161058.JavaMail.javamailuser@localhost>


Hello, 

we have done this sort of thing in our Brazilian Call center, once for the Carnival (because nobody could come to work) and once because the region was flooded (and also nobody could come to work). 

It worked very well for us, but you should increase the call_timeout, we used 30 seconds, as sometimes call_setup can be quite slow. 


Did you get the setup working for internal phones? If not paste your Queue configuration, "max wait of 0 sec" does not sound right, it might just be that you did not have configured the max_wait_time and max_wait_time_with_no_agent correctly. We have 300 seconds (5 minutes) wait time configured, which, in our case, is almost never reached, 

Best regards, 
Raimund 


----- Original Message -----

From: "Juan Pablo L." <jpablolorenzetti at hotmail.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Jueves, 13 de Diciembre 2012 19:31:03 
Subject: [Freeswitch-users] callcenter with external agents 


Hi, i m trying to set up a callcenter but the lines for the agents are mobile phones in a mobile network and are not attached to freeswitch, i m trying to set it up with the following: 



<agent name="1000 at default" type="callback" contact="[call_timeout=10]sofia/external-huawei_gw/XXXXXX" status="Available" max-no-answer="3" wrap-up-t ime="10" reject-delay-time="10" busy-delay-time="60" /> 


XXXXX being the mobile number but i m getting the following error: 


Member 2018 <2018> in queue 'callcenter at default' reached max wait of 0 sec. with no agent plus join grace period of 5 sec. 


and i see that freeswitch does not even try to dial out to the trunk. 


i m wondering if configuring it the way i need it is even possible as i think the problem may reside in the fact that freeswitch does not actually see the agent registered. 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 

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

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From regis.freeswitch.org at tornad.net  Fri Dec 14 14:20:22 2012
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Fri, 14 Dec 2012 12:20:22 +0100
Subject: [Freeswitch-users] callcenter with external agents
In-Reply-To: <4521E504-D44A-47D9-B3EE-3EE363A74558@endigotech.com>
References: <DUB002-W3915B9DAC4F64D9FD8AE3ED94E0@phx.gbl>
	<4521E504-D44A-47D9-B3EE-3EE363A74558@endigotech.com>
Message-ID: <CAHWtr1RNJa2oJD67xtQYZvYPNdWDyehZ4QLpoMRUt3bwvo6kWA@mail.gmail.com>

It works, we're doing it on a production system.
The message seems to be more a agent State problem instead of gateway
problem. Your agent doesn't seems to been "Waiting" or you don't correctly
affect it in a tier with you queue. There's 2 agents thing to check Status
and State. and Tier association
Check the configuration by make call working with a "normal" sip agent and
then, you could try by changing his contact parameter to gateway outside
the fs callcenter box.

Regards


2012/12/14 Brian Foster <bdfoster at endigotech.com>

> +1 I don't really know if this is possible. We've tried to do it this way
> but we ended up using some LUA scripts and mod_fifo. Unfortunately I can't
> release the scripts because the client won't allow me.
>
> Sent from my iPhone
>
> On Dec 13, 2012, at 1:31 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com>
> wrote:
>
> Hi, i m trying to set up a callcenter but the lines for the agents are
> mobile phones in a mobile network and are not attached to freeswitch, i m
> trying to set it up with the following:
>
> <agent name="1000 at default" type="callback"
> contact="[call_timeout=10]sofia/external-huawei_gw/XXXXXX"
> status="Available" max-no-answer="3" wrap-up-time="10"
> reject-delay-time="10" busy-delay-time="60" />
>
> XXXXX being the mobile number but i m getting the following error:
>
> Member 2018 <2018> in queue 'callcenter at default' reached max wait of 0
> sec. with no agent plus join grace period of 5 sec.
>
> and i see that freeswitch does not even try to dial out to the trunk.
>
> i m wondering if configuring it the way i need it is even possible as i
> think the problem may reside in the fact that freeswitch does not actually
> see the agent registered. 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 steveayre at gmail.com  Fri Dec 14 14:21:07 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 14 Dec 2012 11:21:07 +0000
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <0d3e01cdd953$6cab48d0$4601da70$@bizfocused.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3e01cdd953$6cab48d0$4601da70$@bizfocused.com>
Message-ID: <CAFiqYunSKmi+gF=iGzi=JZ06R2Utqzt6tyT2s1cdnvopvm39MQ@mail.gmail.com>

Runtime:
"sofia global siptrace on" or "sofia profile <profilename> siptrace on"
"sofia tracelevel debug" (or any other loglevel name or number)

Configuration to always log:
<global_settings>
  ...
  <param name="tracelevel" value="DEBUG"/>
  ...
</global_settings>
<profiles>
  ...
  <profile name="...">
    ...
    <param name="sip-trace" value="yes"/>
    ...
  </profile>
  ...
</profiles>






On 13 December 2012 17:00, Sean Devoy <sdevoy at bizfocused.com> wrote:

> NEVERMIND -  RTFM!****
>
> ** **
>
> Found it.****
>
> Sorry.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 10, 2012 12:33 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
> ** **
>
> The SERVICE_NOT_IMPLEMENTED is the q.850 translation of the 406 you
> received over SIP.****
>
> If you turn on the siptrace with "sofia global siptrace on" you may get
> more details.****
>
> ** **
>
> Also leg_delay_start does not do much good in enterprise originate as each
> url in the list separated by :_: is an entire originate string so you are
> effectively doing 5 concurrent calls to originate and only supplying one
> leg.  I don't believe that feature works with only one leg supplied.****
>
> ** **
>
> ** **
>
> On Mon, Dec 10, 2012 at 10:57 AM, Sean Devoy <sdevoy at bizfocused.com>
> wrote:****
>
> HI,****
>
>  ****
>
> I have a bridge statement used dozens and dozens of times a day:****
>
> EXECUTE sofia/external_noauth/anonymous at 64.136.174.30 bridge(user/
> 300 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> 301 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> 203 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> 302 at fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/
> 303 at fs_mbri2.bizfocused.com)****
>
>  ****
>
> Very occasionally, recently extensions (at least 300 and 302) have
> returned [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> All of these extensions are at the same remote location using the same wan
> links/routers, etc.****
>
>  ****
>
> What does that mean?  ****
>
>  ****
>
> Pastebin: http://pastebin.freeswitch.org/pastebin.php?dl=20306****
>
>  ****
>
> Also, the 8 second delay appears to be ignored (I think, I have never
> asked the customer or been on site to test).  Any 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
> ** **
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Dec 14 14:23:00 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 14 Dec 2012 11:23:00 +0000
Subject: [Freeswitch-users] SIP error
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233B7F5@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233B7F5@Mail-Kilo.squay.com>
Message-ID: <CAFiqYunmcHU-j0e_t=i2-4FjN4QnKEeXzVamdXrD-6pjkRFfpw@mail.gmail.com>

The most likely reason is that something already is listening on the
IP:port, each profile must be on its own distinct IP+port combination.

"netstat -anp | grep <ip> | grep <port>" will show if another program is
already using the port.

-Steve



On 13 December 2012 18:01, Archana Venugopan <a.venugopan at mundio.com> wrote:

>  Hi team,****
>
> ** **
>
> I am facing with the below error after installing freeswitch and
> registering.****
>
> ** **
>
> 2012-12-13 17:55:15.573019 [ERR] sofia.c:1347 Error Creating SIP UA for
> profile: internal****
>
> ** **
>
> Guess this is checking /usr/local/freeswitch/conf/sip_profiles/internal.xml
> ****
>
> ** **
>
> I have attached my internal.xml as well. Please let me know. Thanks.****
>
> ** **
>
> Regards,****
>
> Archana.V****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 a.venugopan at mundio.com  Fri Dec 14 14:36:07 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Fri, 14 Dec 2012 11:36:07 +0000
Subject: [Freeswitch-users] SIP error
In-Reply-To: <CAFiqYunmcHU-j0e_t=i2-4FjN4QnKEeXzVamdXrD-6pjkRFfpw@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF233B7F5@Mail-Kilo.squay.com>
	<CAFiqYunmcHU-j0e_t=i2-4FjN4QnKEeXzVamdXrD-6pjkRFfpw@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233B918@Mail-Kilo.squay.com>

Hi Steven,
Thanks. When I gave netstat I see the following in my freeswitch server(82.113.72.113)

[root at VECTONE-CLOUDE sip_profiles]# netstat -lnp | grep 5060
tcp        0      0 82.113.72.113:5060          0.0.0.0:*                   LISTEN      19085/freeswitch
tcp        0      0 ::1:5060                                     :::*                        LISTEN      19085/freeswitch
udp        0    0 82.113.72.113:5060          0.0.0.0:*                                       19085/freeswitch
udp        0    0 ::1:5060                                         :::*                                         19085/freeswitch

Is there something wrong in the above?

Regards,
Archana.V

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 14 December 2012 11:23
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP error

The most likely reason is that something already is listening on the IP:port, each profile must be on its own distinct IP+port combination.

"netstat -anp | grep <ip> | grep <port>" will show if another program is already using the port.

-Steve


On 13 December 2012 18:01, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi team,

I am facing with the below error after installing freeswitch and registering.

2012-12-13 17:55:15.573019 [ERR] sofia.c:1347 Error Creating SIP UA for profile: internal

Guess this is checking /usr/local/freeswitch/conf/sip_profiles/internal.xml

I have attached my internal.xml as well. Please let me know. Thanks.

Regards,
Archana.V


_________________________________________________________________________
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

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From steveayre at gmail.com  Fri Dec 14 15:05:01 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 14 Dec 2012 12:05:01 +0000
Subject: [Freeswitch-users] SIP error
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233B918@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233B7F5@Mail-Kilo.squay.com>
	<CAFiqYunmcHU-j0e_t=i2-4FjN4QnKEeXzVamdXrD-6pjkRFfpw@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233B918@Mail-Kilo.squay.com>
Message-ID: <CAFiqYukGf5-zSYq-s+iyfAKWfr061RWAUBo8mn0eG6n=_vvZsQ@mail.gmail.com>

This looks like it might be fine in itself...
freeswitch is listening for both tcp and udp on 82.113.72.113:5060, that'll
be one profile.
Looks like you also have a profile listening on the IPv6 loopback ::1:5060
- it's up to your use-case whether you want that or not... since that won't
be reachable from any other machine.

You need to compare that to what profiles you have configured though, to
see whether there are any profiles failing to start on another ip:port.

Multiple running copies of freeswitch could also give a problem - if 19085
is the PID of a different instance of freeswitch then that would stop a 2nd
FS instance listening on those ip:ports.

-Steve




On 14 December 2012 11:36, Archana Venugopan <a.venugopan at mundio.com> wrote:

>  Hi Steven,****
>
> Thanks. When I gave netstat I see the following in my freeswitch
> server(82.113.72.113)****
>
> ** **
>
> [root at VECTONE-CLOUDE sip_profiles]# netstat -lnp | grep 5060****
>
> tcp        0      0 82.113.72.113:5060          0.0.0.0:*
> LISTEN      19085/freeswitch****
>
> tcp        0      0 ::1:5060
>                :::*                        LISTEN      19085/freeswitch***
> *
>
> udp        0    0 82.113.72.113:5060          0.0.0.0:*
>         19085/freeswitch****
>
> udp        0    0 ::1:5060
>   :::*                                         19085/freeswitch****
>
> ** **
>
> Is there something wrong in the above?****
>
> ** **
>
> Regards,****
>
> Archana.V****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 14 December 2012 11:23
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP error****
>
> ** **
>
> The most likely reason is that something already is listening on the
> IP:port, each profile must be on its own distinct IP+port combination.****
>
> ** **
>
> "netstat -anp | grep <ip> | grep <port>" will show if another program is
> already using the port.****
>
> ** **
>
> -Steve****
>
> ** **
>
> ** **
>
> On 13 December 2012 18:01, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi team,****
>
>  ****
>
> I am facing with the below error after installing freeswitch and
> registering.****
>
>  ****
>
> 2012-12-13 17:55:15.573019 [ERR] sofia.c:1347 Error Creating SIP UA for
> profile: internal****
>
>  ****
>
> Guess this is checking /usr/local/freeswitch/conf/sip_profiles/internal.xml
> ****
>
>  ****
>
> I have attached my internal.xml as well. Please let me know. Thanks.****
>
>  ****
>
> Regards,****
>
> Archana.V****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.venugopan at mundio.com  Fri Dec 14 15:21:51 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Fri, 14 Dec 2012 12:21:51 +0000
Subject: [Freeswitch-users] FW:  SIP error
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233B971@Mail-Kilo.squay.com>

Hi,

There is only 1 instance of freeswitch running.

[root at VECTONE-CLOUDE log]# ps -ef | grep -i free
root     19173     1  1 12:03 ?        00:00:10 /usr/local/freeswitch/bin/freeswitch -nc
root     19223 21809  0 12:19 pts/0    00:00:00 grep -i free

Am not sure how this ::1:5060  was picked and am not sure how to kill this alone.

Regards,
Archana.V

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 14 December 2012 12:05
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP error

This looks like it might be fine in itself...
freeswitch is listening for both tcp and udp on 82.113.72.113:5060<http://82.113.72.113:5060>, that'll be one profile.
Looks like you also have a profile listening on the IPv6 loopback ::1:5060 - it's up to your use-case whether you want that or not... since that won't be reachable from any other machine.

You need to compare that to what profiles you have configured though, to see whether there are any profiles failing to start on another ip:port.

Multiple running copies of freeswitch could also give a problem - if 19085 is the PID of a different instance of freeswitch then that would stop a 2nd FS instance listening on those ip:ports.

-Steve




On 14 December 2012 11:36, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi Steven,
Thanks. When I gave netstat I see the following in my freeswitch server(82.113.72.113)

[root at VECTONE-CLOUDE sip_profiles]# netstat -lnp | grep 5060
tcp        0      0 82.113.72.113:5060<http://82.113.72.113:5060>          0.0.0.0:*                   LISTEN      19085/freeswitch
tcp        0      0 ::1:5060                                     :::*                        LISTEN      19085/freeswitch
udp        0    0 82.113.72.113:5060<http://82.113.72.113:5060>          0.0.0.0:*                                       19085/freeswitch
udp        0    0 ::1:5060                                         :::*                                         19085/freeswitch

Is there something wrong in the above?

Regards,
Archana.V

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 Steven Ayre
Sent: 14 December 2012 11:23
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP error

The most likely reason is that something already is listening on the IP:port, each profile must be on its own distinct IP+port combination.

"netstat -anp | grep <ip> | grep <port>" will show if another program is already using the port.

-Steve


On 13 December 2012 18:01, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi team,

I am facing with the below error after installing freeswitch and registering.

2012-12-13 17:55:15.573019 [ERR] sofia.c:1347 Error Creating SIP UA for profile: internal

Guess this is checking /usr/local/freeswitch/conf/sip_profiles/internal.xml

I have attached my internal.xml as well. Please let me know. Thanks.

Regards,
Archana.V


_________________________________________________________________________
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

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 fdelawarde at wirelessmundi.com  Fri Dec 14 15:49:08 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Fri, 14 Dec 2012 13:49:08 +0100
Subject: [Freeswitch-users] callcenter with external agents
In-Reply-To: <CAHWtr1RNJa2oJD67xtQYZvYPNdWDyehZ4QLpoMRUt3bwvo6kWA@mail.gmail.com>
References: <DUB002-W3915B9DAC4F64D9FD8AE3ED94E0@phx.gbl>
	<4521E504-D44A-47D9-B3EE-3EE363A74558@endigotech.com>
	<CAHWtr1RNJa2oJD67xtQYZvYPNdWDyehZ4QLpoMRUt3bwvo6kWA@mail.gmail.com>
Message-ID: <1355489348.7162.28.camel@luna.madrid.commsmundi.com>

It should work fine, heck it even works with loopback channels for me!

Just in case, verify that your dialstring is correct. If
"external-huawei_gw" is a gateway, use:

sofia/gateway/external-huawei_gw/XXXXXX

instead of:

sofia/external-huawei_gw/XXXXXX

Regards,
Fran?ois.

On Fri, 2012-12-14 at 12:20 +0100, Regis M wrote:
> It works, we're doing it on a production system.
> The message seems to be more a agent State problem instead of gateway
> problem. Your agent doesn't seems to been "Waiting" or you don't
> correctly affect it in a tier with you queue. There's 2 agents thing
> to check Status and State. and Tier association
> Check the configuration by make call working with a "normal" sip agent
> and then, you could try by changing his contact parameter to gateway
> outside the fs callcenter box.
> 
> Regards
> 
> 
> 2012/12/14 Brian Foster <bdfoster at endigotech.com>
>         +1 I don't really know if this is possible. We've tried to do
>         it this way but we ended up using some LUA scripts and
>         mod_fifo. Unfortunately I can't release the scripts because
>         the client won't allow me. 
>         
>         Sent from my iPhone 
>         
>         On Dec 13, 2012, at 1:31 PM, Juan Pablo L.
>         <jpablolorenzetti at hotmail.com> wrote:
>         
>         
>         > Hi, i m trying to set up a callcenter but the lines for the
>         > agents are mobile phones in a mobile network and are not
>         > attached to freeswitch, i m trying to set it up with the
>         > following: 
>         > 
>         > 
>         > <agent name="1000 at default" type="callback"
>         > contact="[call_timeout=10]sofia/external-huawei_gw/XXXXXX"
>         > status="Available" max-no-answer="3" wrap-up-time="10"
>         > reject-delay-time="10" busy-delay-time="60" /> 
>         > 
>         > 
>         > XXXXX being the mobile number but i m getting the following
>         > error: 
>         > 
>         > 
>         > Member 2018 <2018> in queue 'callcenter at default' reached max
>         > wait of 0 sec. with no agent plus join grace period of 5
>         > sec. 
>         > 
>         > 
>         > and i see that freeswitch does not even try to dial out to
>         > the trunk.  
>         > 
>         > 
>         > i m wondering if configuring it the way i need it is even
>         > possible as i think the problem may reside in the fact that
>         > freeswitch does not actually see the agent registered.
>         > 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
>         > 
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>         > 
>         
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Fri Dec 14 16:16:03 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 14 Dec 2012 13:16:03 +0000
Subject: [Freeswitch-users] FW: SIP error
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233B971@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233B971@Mail-Kilo.squay.com>
Message-ID: <CAFiqYumRcsQOWXe33Lyjhp2BFQUF2fcvAE80LA4TJHHBJOz7hQ@mail.gmail.com>

What SIP profiles do you have defined, and what are their sip-ip and port
settings?


On 14 December 2012 12:21, Archana Venugopan <a.venugopan at mundio.com> wrote:

>  Hi,****
>
> ** **
>
> There is only 1 instance of freeswitch running.****
>
> ** **
>
> [root at VECTONE-CLOUDE log]# ps -ef | grep -i free****
>
> root     19173     1  1 12:03 ?        00:00:10
> /usr/local/freeswitch/bin/freeswitch -nc****
>
> root     19223 21809  0 12:19 pts/0    00:00:00 grep -i free****
>
> ** **
>
> Am not sure how this ::1:5060  was picked and am not sure how to kill
> this alone.    ****
>
> ** **
>
> Regards,****
>
> Archana.V****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 14 December 2012 12:05
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP error****
>
>  ** **
>
> This looks like it might be fine in itself...****
>
> freeswitch is listening for both tcp and udp on 82.113.72.113:5060,
> that'll be one profile.****
>
> Looks like you also have a profile listening on the IPv6 loopback ::1:5060
> - it's up to your use-case whether you want that or not... since that won't
> be reachable from any other machine.****
>
> ** **
>
> You need to compare that to what profiles you have configured though, to
> see whether there are any profiles failing to start on another ip:port.***
> *
>
> ** **
>
> Multiple running copies of freeswitch could also give a problem - if 19085
> is the PID of a different instance of freeswitch then that would stop a 2nd
> FS instance listening on those ip:ports.****
>
> ** **
>
> -Steve****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> On 14 December 2012 11:36, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi Steven,****
>
> Thanks. When I gave netstat I see the following in my freeswitch
> server(82.113.72.113)****
>
>  ****
>
> [root at VECTONE-CLOUDE sip_profiles]# netstat -lnp | grep 5060****
>
> tcp        0      0 82.113.72.113:5060          0.0.0.0:*
> LISTEN      19085/freeswitch****
>
> tcp        0      0 ::1:5060
>                :::*                        LISTEN      19085/freeswitch***
> *
>
> udp        0    0 82.113.72.113:5060          0.0.0.0:*
>         19085/freeswitch****
>
> udp        0    0 ::1:5060
>   :::*                                         19085/freeswitch****
>
>  ****
>
> Is there something wrong in the above?****
>
>  ****
>
> Regards,****
>
> Archana.V****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 14 December 2012 11:23
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP error****
>
>  ****
>
> The most likely reason is that something already is listening on the
> IP:port, each profile must be on its own distinct IP+port combination.****
>
>  ****
>
> "netstat -anp | grep <ip> | grep <port>" will show if another program is
> already using the port.****
>
>  ****
>
> -Steve****
>
>  ****
>
>  ****
>
> On 13 December 2012 18:01, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi team,****
>
>  ****
>
> I am facing with the below error after installing freeswitch and
> registering.****
>
>  ****
>
> 2012-12-13 17:55:15.573019 [ERR] sofia.c:1347 Error Creating SIP UA for
> profile: internal****
>
>  ****
>
> Guess this is checking /usr/local/freeswitch/conf/sip_profiles/internal.xml
> ****
>
>  ****
>
> I have attached my internal.xml as well. Please let me know. Thanks.****
>
>  ****
>
> Regards,****
>
> Archana.V****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 a.venugopan at mundio.com  Fri Dec 14 16:50:33 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Fri, 14 Dec 2012 13:50:33 +0000
Subject: [Freeswitch-users] FW:  FW: SIP error
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233B992@Mail-Kilo.squay.com>

Here are the ones.

  <!-- Internal SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5080"/>
  <X-PRE-PROCESS cmd="set" data="internal_tls_port=5081"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>

  <!-- External SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
  <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/>


Regards,
Archana.V

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 14 December 2012 13:16
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FW: SIP error

What SIP profiles do you have defined, and what are their sip-ip and port settings?

On 14 December 2012 12:21, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

There is only 1 instance of freeswitch running.

[root at VECTONE-CLOUDE log]# ps -ef | grep -i free
root     19173     1  1 12:03 ?        00:00:10 /usr/local/freeswitch/bin/freeswitch -nc
root     19223 21809  0 12:19 pts/0    00:00:00 grep -i free

Am not sure how this ::1:5060  was picked and am not sure how to kill this alone.

Regards,
Archana.V

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 Steven Ayre
Sent: 14 December 2012 12:05

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP error

This looks like it might be fine in itself...
freeswitch is listening for both tcp and udp on 82.113.72.113:5060<http://82.113.72.113:5060>, that'll be one profile.
Looks like you also have a profile listening on the IPv6 loopback ::1:5060 - it's up to your use-case whether you want that or not... since that won't be reachable from any other machine.

You need to compare that to what profiles you have configured though, to see whether there are any profiles failing to start on another ip:port.

Multiple running copies of freeswitch could also give a problem - if 19085 is the PID of a different instance of freeswitch then that would stop a 2nd FS instance listening on those ip:ports.

-Steve




On 14 December 2012 11:36, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi Steven,
Thanks. When I gave netstat I see the following in my freeswitch server(82.113.72.113)

[root at VECTONE-CLOUDE sip_profiles]# netstat -lnp | grep 5060
tcp        0      0 82.113.72.113:5060<http://82.113.72.113:5060>          0.0.0.0:*                   LISTEN      19085/freeswitch
tcp        0      0 ::1:5060                                     :::*                        LISTEN      19085/freeswitch
udp        0    0 82.113.72.113:5060<http://82.113.72.113:5060>          0.0.0.0:*                                       19085/freeswitch
udp        0    0 ::1:5060                                         :::*                                         19085/freeswitch

Is there something wrong in the above?

Regards,
Archana.V

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 Steven Ayre
Sent: 14 December 2012 11:23
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP error

The most likely reason is that something already is listening on the IP:port, each profile must be on its own distinct IP+port combination.

"netstat -anp | grep <ip> | grep <port>" will show if another program is already using the port.

-Steve


On 13 December 2012 18:01, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi team,

I am facing with the below error after installing freeswitch and registering.

2012-12-13 17:55:15.573019 [ERR] sofia.c:1347 Error Creating SIP UA for profile: internal

Guess this is checking /usr/local/freeswitch/conf/sip_profiles/internal.xml

I have attached my internal.xml as well. Please let me know. Thanks.

Regards,
Archana.V


_________________________________________________________________________
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
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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

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

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 steveayre at gmail.com  Fri Dec 14 17:32:14 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 14 Dec 2012 14:32:14 +0000
Subject: [Freeswitch-users] FW: FW: SIP error
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233B992@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233B992@Mail-Kilo.squay.com>
Message-ID: <CAFiqYu=Ec4N9U=oNMYFritMKYRKaG-Gh3tH7UDkstg1_y6cCLA@mail.gmail.com>

Those are global variables, not parameters in the sip profiles... you
should check the profiles themselves. They might be using the variables,
but they might not be too.

In particular nothing you posted there set the ip used in the sip-ip params.


On 14 December 2012 13:50, Archana Venugopan <a.venugopan at mundio.com> wrote:

>  Here are the ones.****
>
> ** **
>
>   <!-- Internal SIP Profile -->****
>
>   <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>****
>
>   <X-PRE-PROCESS cmd="set" data="internal_sip_port=5080"/>****
>
>   <X-PRE-PROCESS cmd="set" data="internal_tls_port=5081"/>****
>
>   <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>****
>
>   <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>
> ****
>
> ** **
>
>   <!-- External SIP Profile -->****
>
>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>****
>
>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>****
>
>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>****
>
>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>****
>
>   <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/>
> ****
>
> ** **
>
> ** **
>
> Regards,****
>
> Archana.V****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 14 December 2012 13:16
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] FW: SIP error****
>
> ** **
>
> What SIP profiles do you have defined, and what are their sip-ip and port
> settings?****
>
> ** **
>
> On 14 December 2012 12:21, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
>  ****
>
> There is only 1 instance of freeswitch running.****
>
>  ****
>
> [root at VECTONE-CLOUDE log]# ps -ef | grep -i free****
>
> root     19173     1  1 12:03 ?        00:00:10
> /usr/local/freeswitch/bin/freeswitch -nc****
>
> root     19223 21809  0 12:19 pts/0    00:00:00 grep -i free****
>
>  ****
>
> Am not sure how this ::1:5060  was picked and am not sure how to kill this
> alone.    ****
>
>  ****
>
> Regards,****
>
> Archana.V****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 14 December 2012 12:05****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP error****
>
>  ****
>
> This looks like it might be fine in itself...****
>
> freeswitch is listening for both tcp and udp on 82.113.72.113:5060,
> that'll be one profile.****
>
> Looks like you also have a profile listening on the IPv6 loopback ::1:5060
> - it's up to your use-case whether you want that or not... since that won't
> be reachable from any other machine.****
>
>  ****
>
> You need to compare that to what profiles you have configured though, to
> see whether there are any profiles failing to start on another ip:port.***
> *
>
>  ****
>
> Multiple running copies of freeswitch could also give a problem - if 19085
> is the PID of a different instance of freeswitch then that would stop a 2nd
> FS instance listening on those ip:ports.****
>
>  ****
>
> -Steve****
>
>  ****
>
>  ****
>
>  ****
>
>  ****
>
> On 14 December 2012 11:36, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi Steven,****
>
> Thanks. When I gave netstat I see the following in my freeswitch
> server(82.113.72.113)****
>
>  ****
>
> [root at VECTONE-CLOUDE sip_profiles]# netstat -lnp | grep 5060****
>
> tcp        0      0 82.113.72.113:5060          0.0.0.0:*
> LISTEN      19085/freeswitch****
>
> tcp        0      0 ::1:5060
>                :::*                        LISTEN      19085/freeswitch***
> *
>
> udp        0    0 82.113.72.113:5060          0.0.0.0:*
>         19085/freeswitch****
>
> udp        0    0 ::1:5060
>   :::*                                         19085/freeswitch****
>
>  ****
>
> Is there something wrong in the above?****
>
>  ****
>
> Regards,****
>
> Archana.V****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 14 December 2012 11:23
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP error****
>
>  ****
>
> The most likely reason is that something already is listening on the
> IP:port, each profile must be on its own distinct IP+port combination.****
>
>  ****
>
> "netstat -anp | grep <ip> | grep <port>" will show if another program is
> already using the port.****
>
>  ****
>
> -Steve****
>
>  ****
>
>  ****
>
> On 13 December 2012 18:01, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi team,****
>
>  ****
>
> I am facing with the below error after installing freeswitch and
> registering.****
>
>  ****
>
> 2012-12-13 17:55:15.573019 [ERR] sofia.c:1347 Error Creating SIP UA for
> profile: internal****
>
>  ****
>
> Guess this is checking /usr/local/freeswitch/conf/sip_profiles/internal.xml
> ****
>
>  ****
>
> I have attached my internal.xml as well. Please let me know. Thanks.****
>
>  ****
>
> Regards,****
>
> Archana.V****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Dec 14 19:38:20 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 14 Dec 2012 16:38:20 +0000
Subject: [Freeswitch-users] Originate against dial plan?
In-Reply-To: <2870919548404989049@unknownmsgid>
References: <2870919548404989049@unknownmsgid>
Message-ID: <CAFiqYu=_OB0UTvk8mDAO_9quT6vLL3o7u6Mq5dhLe4rkWQGp7g@mail.gmail.com>

There are 3 things you can use:

1. 'transfer' application
2. 'execute_extension'
3. loopback endpoint (higher overhead because it creates additional
channels, which means more legs for cdrs... use unroll=true to decrease
overhead once the call is setup)

I combine transfer and variables.

-Steve





On 13 December 2012 13:23, Royce Mitchell III <royce3 at gmail.com> wrote:

> I'm trying to avoid duplication of logic. I have a dial plan setup
> with two sets of gateways, one set is a route to a pair of adtrans
> connecting to the customer downstream telephony system, the other set
> of gateways is another pair of adtrans leading to the PSTN. I have all
> this setup and working in the dial plan. I would like to do something
> like this: originate 8005551212 XML default &txfax(foo), but I can't
> figure it out. Is there a way I could, say, generate a one-leg txfax,
> then transfer to the destination?
>
> Royce Mitchell
> IT Consultant
> ITAS Solutions
> royce3 at itas-solutions.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Dec 15 01:46:39 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Fri, 14 Dec 2012 22:46:39 +0000
Subject: [Freeswitch-users] callcenter with external agents
In-Reply-To: <1355489348.7162.28.camel@luna.madrid.commsmundi.com>
References: <DUB002-W3915B9DAC4F64D9FD8AE3ED94E0@phx.gbl>,
	<4521E504-D44A-47D9-B3EE-3EE363A74558@endigotech.com>,
	<CAHWtr1RNJa2oJD67xtQYZvYPNdWDyehZ4QLpoMRUt3bwvo6kWA@mail.gmail.com>,
	<1355489348.7162.28.camel@luna.madrid.commsmundi.com>
Message-ID: <DUB002-W848BC9DC7196903EB35AE2D94D0@phx.gbl>

Hi All, thank you very much for your answers, it is a relieve to know that what i m trying to accomplish is possible, i have not done it before so i m not sure what to expect, it is likely that i m configuring something wrong ..... i will try what was suggested in these responses and get back to you guys asap. thanks!!

> From: fdelawarde at wirelessmundi.com
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 14 Dec 2012 13:49:08 +0100
> Subject: Re: [Freeswitch-users] callcenter with external agents
> 
> It should work fine, heck it even works with loopback channels for me!
> 
> Just in case, verify that your dialstring is correct. If
> "external-huawei_gw" is a gateway, use:
> 
> sofia/gateway/external-huawei_gw/XXXXXX
> 
> instead of:
> 
> sofia/external-huawei_gw/XXXXXX
> 
> Regards,
> Fran?ois.
> 
> On Fri, 2012-12-14 at 12:20 +0100, Regis M wrote:
> > It works, we're doing it on a production system.
> > The message seems to be more a agent State problem instead of gateway
> > problem. Your agent doesn't seems to been "Waiting" or you don't
> > correctly affect it in a tier with you queue. There's 2 agents thing
> > to check Status and State. and Tier association
> > Check the configuration by make call working with a "normal" sip agent
> > and then, you could try by changing his contact parameter to gateway
> > outside the fs callcenter box.
> > 
> > Regards
> > 
> > 
> > 2012/12/14 Brian Foster <bdfoster at endigotech.com>
> >         +1 I don't really know if this is possible. We've tried to do
> >         it this way but we ended up using some LUA scripts and
> >         mod_fifo. Unfortunately I can't release the scripts because
> >         the client won't allow me. 
> >         
> >         Sent from my iPhone 
> >         
> >         On Dec 13, 2012, at 1:31 PM, Juan Pablo L.
> >         <jpablolorenzetti at hotmail.com> wrote:
> >         
> >         
> >         > Hi, i m trying to set up a callcenter but the lines for the
> >         > agents are mobile phones in a mobile network and are not
> >         > attached to freeswitch, i m trying to set it up with the
> >         > following: 
> >         > 
> >         > 
> >         > <agent name="1000 at default" type="callback"
> >         > contact="[call_timeout=10]sofia/external-huawei_gw/XXXXXX"
> >         > status="Available" max-no-answer="3" wrap-up-time="10"
> >         > reject-delay-time="10" busy-delay-time="60" /> 
> >         > 
> >         > 
> >         > XXXXX being the mobile number but i m getting the following
> >         > error: 
> >         > 
> >         > 
> >         > Member 2018 <2018> in queue 'callcenter at default' reached max
> >         > wait of 0 sec. with no agent plus join grace period of 5
> >         > sec. 
> >         > 
> >         > 
> >         > and i see that freeswitch does not even try to dial out to
> >         > the trunk.  
> >         > 
> >         > 
> >         > i m wondering if configuring it the way i need it is even
> >         > possible as i think the problem may reside in the fact that
> >         > freeswitch does not actually see the agent registered.
> >         > 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
> >         > 
> >         > FreeSWITCH-users mailing list
> >         > FreeSWITCH-users at lists.freeswitch.org
> >         > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > http://www.freeswitch.org
> >         > 
> >         
> >         _________________________________________________________________________
> >         Professional FreeSWITCH Consulting 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
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 jpablolorenzetti at hotmail.com  Sat Dec 15 02:00:27 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Fri, 14 Dec 2012 23:00:27 +0000
Subject: [Freeswitch-users] callcenter with external agents
In-Reply-To: <DUB002-W848BC9DC7196903EB35AE2D94D0@phx.gbl>
References: <DUB002-W3915B9DAC4F64D9FD8AE3ED94E0@phx.gbl>, ,
	<4521E504-D44A-47D9-B3EE-3EE363A74558@endigotech.com>, ,
	<CAHWtr1RNJa2oJD67xtQYZvYPNdWDyehZ4QLpoMRUt3bwvo6kWA@mail.gmail.com>,
	, <1355489348.7162.28.camel@luna.madrid.commsmundi.com>,
	<DUB002-W848BC9DC7196903EB35AE2D94D0@phx.gbl>
Message-ID: <DUB002-W73F9E83A51542D9A07C2FFD94D0@phx.gbl>

I have change somethings around as suggested but still freeswitch does not even make the attempt to dial out, this is the config i m using http://pastebin.freeswitch.org/20326 and this is the console logs http://pastebin.freeswitch.org/20327.
the name of the gateway i m trying to send the call to is huawei_csoft, in my original post i sent the name of the external profile whichis wrong but i was making lots of combinations trying to make it work .... thanks for any help ... 

From: jpablolorenzetti at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 14 Dec 2012 22:46:39 +0000
Subject: Re: [Freeswitch-users] callcenter with external agents




Hi All, thank you very much for your answers, it is a relieve to know that what i m trying to accomplish is possible, i have not done it before so i m not sure what to expect, it is likely that i m configuring something wrong ..... i will try what was suggested in these responses and get back to you guys asap. thanks!!

> From: fdelawarde at wirelessmundi.com
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 14 Dec 2012 13:49:08 +0100
> Subject: Re: [Freeswitch-users] callcenter with external agents
> 
> It should work fine, heck it even works with loopback channels for me!
> 
> Just in case, verify that your dialstring is correct. If
> "external-huawei_gw" is a gateway, use:
> 
> sofia/gateway/external-huawei_gw/XXXXXX
> 
> instead of:
> 
> sofia/external-huawei_gw/XXXXXX
> 
> Regards,
> Fran?ois.
> 
> On Fri, 2012-12-14 at 12:20 +0100, Regis M wrote:
> > It works, we're doing it on a production system.
> > The message seems to be more a agent State problem instead of gateway
> > problem. Your agent doesn't seems to been "Waiting" or you don't
> > correctly affect it in a tier with you queue. There's 2 agents thing
> > to check Status and State. and Tier association
> > Check the configuration by make call working with a "normal" sip agent
> > and then, you could try by changing his contact parameter to gateway
> > outside the fs callcenter box.
> > 
> > Regards
> > 
> > 
> > 2012/12/14 Brian Foster <bdfoster at endigotech.com>
> >         +1 I don't really know if this is possible. We've tried to do
> >         it this way but we ended up using some LUA scripts and
> >         mod_fifo. Unfortunately I can't release the scripts because
> >         the client won't allow me. 
> >         
> >         Sent from my iPhone 
> >         
> >         On Dec 13, 2012, at 1:31 PM, Juan Pablo L.
> >         <jpablolorenzetti at hotmail.com> wrote:
> >         
> >         
> >         > Hi, i m trying to set up a callcenter but the lines for the
> >         > agents are mobile phones in a mobile network and are not
> >         > attached to freeswitch, i m trying to set it up with the
> >         > following: 
> >         > 
> >         > 
> >         > <agent name="1000 at default" type="callback"
> >         > contact="[call_timeout=10]sofia/external-huawei_gw/XXXXXX"
> >         > status="Available" max-no-answer="3" wrap-up-time="10"
> >         > reject-delay-time="10" busy-delay-time="60" /> 
> >         > 
> >         > 
> >         > XXXXX being the mobile number but i m getting the following
> >         > error: 
> >         > 
> >         > 
> >         > Member 2018 <2018> in queue 'callcenter at default' reached max
> >         > wait of 0 sec. with no agent plus join grace period of 5
> >         > sec. 
> >         > 
> >         > 
> >         > and i see that freeswitch does not even try to dial out to
> >         > the trunk.  
> >         > 
> >         > 
> >         > i m wondering if configuring it the way i need it is even
> >         > possible as i think the problem may reside in the fact that
> >         > freeswitch does not actually see the agent registered.
> >         > 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
> >         > 
> >         > FreeSWITCH-users mailing list
> >         > FreeSWITCH-users at lists.freeswitch.org
> >         > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > http://www.freeswitch.org
> >         > 
> >         
> >         _________________________________________________________________________
> >         Professional FreeSWITCH Consulting 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
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From ahmed at netelsat.net  Sat Dec 15 02:52:21 2012
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Sat, 15 Dec 2012 04:52:21 +0500
Subject: [Freeswitch-users] PostgreSQL in the core
Message-ID: <CALFFxt13ePTSk_GFnAvK6GD7tRo2qn1mv3irsE+d1ZSu=SH08A@mail.gmail.com>

Hi all,
i am just trying to use 1.2.5.3 to use Postgresql as core.
Followed by http://wiki.freeswitch.org/wiki/PostgreSQL_in_the_core
but its giving error like :

switch_pgsql.c:492 invalid connection option "application_name"

and

[CRIT] switch_core_sqldb.c:504 Failure to connect to PGSQL
hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password=''
options='-c client_min_messages=NOTICE' application_name='freeswitch'!


Postgresql version is 8.4.13.

can any one please help where i am missing something ?

Thanking you all
Ahmed
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From dujinfang at gmail.com  Sat Dec 15 05:21:25 2012
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 15 Dec 2012 10:21:25 +0800
Subject: [Freeswitch-users] PostgreSQL in the core
In-Reply-To: <CALFFxt13ePTSk_GFnAvK6GD7tRo2qn1mv3irsE+d1ZSu=SH08A@mail.gmail.com>
References: <CALFFxt13ePTSk_GFnAvK6GD7tRo2qn1mv3irsE+d1ZSu=SH08A@mail.gmail.com>
Message-ID: <B89E0118B7FD4A9CA453643D5B1675B9@gmail.com>

Try to remove the application_name option ? 

-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Saturday, December 15, 2012 at 7:52 AM, Ahmed Sboor wrote:

> Hi all,
> i am just trying to use 1.2.5.3 to use Postgresql as core.
> Followed by http://wiki.freeswitch.org/wiki/PostgreSQL_in_the_core 
> but its giving error like :
> 
> switch_pgsql.c:492 invalid connection option "application_name"
> 
> and 
> 
> [CRIT] switch_core_sqldb.c:504 Failure to connect to PGSQL hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE' application_name='freeswitch'! 
> 
> 
> Postgresql version is 8.4.13. 
> 
> can any one please help where i am missing something ?
> 
> Thanking you all
> Ahmed
> 
> _________________________________________________________________________
> 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
> 
> 


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From neilp at cs.stanford.edu  Sat Dec 15 05:26:42 2012
From: neilp at cs.stanford.edu (Neil Patel)
Date: Sat, 15 Dec 2012 07:56:42 +0530
Subject: [Freeswitch-users] Sangoma PRI/Freetdm issue: LOSE_RACE error on
	first call after FS startup
Message-ID: <CAGy1GmMTFj6_jGEBheSjP7gUyGjGeObbgWV7OJjLFuJLZ+jGpw@mail.gmail.com>

Hi All,

I recently upgraded my box to Ubuntu 12.04 LTS and re-installed FS from
latest git. I'm using a Sangoma A108DE card, interfacing with FS with
freetdm. Since upgrade, I'm getting a weird behavior: whenever I first
start up freeswitch, the *first call* I make to any of my PRI lines does
not go through. The error is pasted below. Subsequent calls go through
fine, without the error. So this problem only occurs anytime I have to
restart FS; I have to call each of the lines manually to get each of them
to "wake up".

Judging by the error, my first thought is that this is a
Sangoma/FreeTDM/Wanpipe driver issue. Can a real expert weigh in? Happy to
provide more logs/output. Thanks!
Neil

===================
2012-12-15 07:38:01.241594 [INFO] ftmod_sangoma_isdn_stack_rcv.c:96
[s4c1][4:1] Received SETUP (suId:1 suInstId:0 spInstId:59)
2012-12-15 07:38:05.241594 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:862
[SNGISDN Q931] s4: Protocol: Unknown Event Code(2): Incomp Msg(276)
2012-12-15 07:38:09.241595 [INFO] ftmod_sangoma_isdn_stack_rcv.c:232
[s4c1][4:1] Received DISCONNECT (suId:1 suInstId:0 spInstId:59)
2012-12-15 07:38:09.241595 [INFO] ftmod_sangoma_isdn_stack_hndl.c:153
[s4c1][4:1] Incoming call: Called No:[61608300] Calling No:[919586550654]
2012-12-15 07:38:09.241595 [NOTICE] switch_channel.c:968 New Channel
FreeTDM/4:1/61608300 [458fb2e2-465c-11e2-b550-cfadc9bb1b74]
2012-12-15 07:38:09.241595 [NOTICE] mod_freetdm.c:425 Hangup
FreeTDM/4:1/61608300 [CS_INIT] [LOSE_RACE]
2012-12-15 07:38:09.241595 [INFO] ftmod_sangoma_isdn_stack_out.c:441
[s4c1][4:1] Sending RELEASE/RELEASE COMPLETE (suId:1 suInstId:59
spInstId:59)
2012-12-15 07:38:09.241595 [NOTICE] switch_core_session.c:1506 Session 59
(FreeTDM/4:1/61608300) Ended
2012-12-15 07:38:09.241595 [NOTICE] switch_core_session.c:1510 Close
Channel FreeTDM/4:1/61608300 [CS_DESTROY]
2012-12-15 07:38:09.281595 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267
[s4c1][4:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:59
spInstId:59)
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From ahmed at netelsat.net  Sat Dec 15 05:33:54 2012
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Sat, 15 Dec 2012 07:33:54 +0500
Subject: [Freeswitch-users] PostgreSQL in the core
In-Reply-To: <B89E0118B7FD4A9CA453643D5B1675B9@gmail.com>
References: <CALFFxt13ePTSk_GFnAvK6GD7tRo2qn1mv3irsE+d1ZSu=SH08A@mail.gmail.com>
	<B89E0118B7FD4A9CA453643D5B1675B9@gmail.com>
Message-ID: <CALFFxt3zWP22vYJMPiLuvQzfj9aqwJ+MeCZvH2efV4QWtAfQ5Q@mail.gmail.com>

Did that , then it says invalid option " client_min_messages"

On Sat, Dec 15, 2012 at 7:21 AM, Seven Du <dujinfang at gmail.com> wrote:

> Try to remove the application_name option ?
>
> --
> Seven Du
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Saturday, December 15, 2012 at 7:52 AM, Ahmed Sboor wrote:
>
> Hi all,
> i am just trying to use 1.2.5.3 to use Postgresql as core.
> Followed by http://wiki.freeswitch.org/wiki/PostgreSQL_in_the_core
> but its giving error like :
>
> switch_pgsql.c:492 invalid connection option "application_name"
>
> and
>
> [CRIT] switch_core_sqldb.c:504 Failure to connect to PGSQL
> hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password=''
> options='-c client_min_messages=NOTICE' application_name='freeswitch'!
>
>
> Postgresql version is 8.4.13.
>
> can any one please help where i am missing something ?
>
> Thanking you all
> Ahmed
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Dec 15 07:43:06 2012
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 15 Dec 2012 12:43:06 +0800
Subject: [Freeswitch-users] fifo questions
Message-ID: <E75D158C1EE546CEBF617224942B26FF@gmail.com>

Hi, 


I'm using mod_fifo with onhook agents, and when a caller in it will ring all agents. The problem is that if an agent is placing an outbound call and the fifo still ring it regardless it's "busy". I manually set on the phone to accept only one channel solved the problem. But mod_fifo still try to ring it, is it possible to not ring the "busy" agent? I found fifo_track_calls, might work with this?

also, what's the purpose of fifo_add_outbound? should it have a difference with fifo_member add ?


Thanks.


-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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From dujinfang at gmail.com  Sat Dec 15 14:03:23 2012
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 15 Dec 2012 19:03:23 +0800
Subject: [Freeswitch-users] PostgreSQL in the core
In-Reply-To: <CALFFxt3zWP22vYJMPiLuvQzfj9aqwJ+MeCZvH2efV4QWtAfQ5Q@mail.gmail.com>
References: <CALFFxt13ePTSk_GFnAvK6GD7tRo2qn1mv3irsE+d1ZSu=SH08A@mail.gmail.com>
	<B89E0118B7FD4A9CA453643D5B1675B9@gmail.com>
	<CALFFxt3zWP22vYJMPiLuvQzfj9aqwJ+MeCZvH2efV4QWtAfQ5Q@mail.gmail.com>
Message-ID: <20F2980066FC4F198FF78D66271ECF47@gmail.com>

I would try also remove options. 

-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Saturday, December 15, 2012 at 10:33 AM, Ahmed Sboor wrote:

> Did that , then it says invalid option " client_min_messages" 
> 
> On Sat, Dec 15, 2012 at 7:21 AM, Seven Du <dujinfang at gmail.com (mailto:dujinfang at gmail.com)> wrote:
> > Try to remove the application_name option ? 
> > 
> > -- 
> > Seven Du
> > Sent with Sparrow (http://www.sparrowmailapp.com/?sig)
> > 
> > 
> > On Saturday, December 15, 2012 at 7:52 AM, Ahmed Sboor wrote:
> > 
> > 
> > 
> > > Hi all,
> > > i am just trying to use 1.2.5.3 to use Postgresql as core.
> > > Followed by http://wiki.freeswitch.org/wiki/PostgreSQL_in_the_core 
> > > but its giving error like :
> > > 
> > > switch_pgsql.c:492 invalid connection option "application_name"
> > > 
> > > and 
> > > 
> > > [CRIT] switch_core_sqldb.c:504 Failure to connect to PGSQL hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE' application_name='freeswitch'! 
> > > 
> > > 
> > > Postgresql version is 8.4.13. 
> > > 
> > > can any one please help where i am missing something ?
> > > 
> > > Thanking you all
> > > Ahmed
> > > 
> > > _________________________________________________________________________
> > > 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
> > 
> > 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
> 
> 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 vbvbrj at gmail.com  Sat Dec 15 15:00:09 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Sat, 15 Dec 2012 14:00:09 +0200
Subject: [Freeswitch-users] Natting question.
In-Reply-To: <33fa01cdd710$f0168a30$d0439e90$@bizfocused.com>
References: <50C5EAAA.6040100@gmail.com>
	<33fa01cdd710$f0168a30$d0439e90$@bizfocused.com>
Message-ID: <50CC6649.1010606@gmail.com>

On 10.12.2012 22:00, Sean Devoy wrote:
> Hi Mimiko,
>
> I have had hours and hours of problems with NAT, so I can at least tell you
> what helped me get past them.
>
> All of my problems have been in the configuration     PHONE > NAT > Internet
>> FS  .

I've gived up using NAT on FS server. So I put server on public IP, 
because if FS is after a firewall then there is too much complexity like 
different cases how a phone can connect:

For example, FS firewall is on public ip: IP-PUB1
A phone is registering with this IP-PUB1, but the client can be in 
different location:

1) client -> local-net -> FS firewall -> NAT -> FS firewall -> local-net 
-> local-ip of FS. Ie, client and FS is on same network, but client uses 
FS public natted IP-PUB1.
2) client -> other-local-nat (wifi) -> router -> local-net -> FS 
firewall -> NAT -> FS firewall -> local-net -> local-ip of FS. Ie client 
is behind router located in same network as FS and is connecting to FS 
IP-PUB1. Triple nat.
3) client -> internet -> FS firewall -> local-net -> local-ip of FS.
4) client -> router -> internet -> FS firewall -> local-net -> local-ip 
of FS.
5) client -> VPN -> local-net -> FS firewall -> NAT -> FS firewall -> 
local-net -> local-ip of FS.

If in all cases except 3 and 5 is using local-ip of FS, then its ok. But 
to handle or 5 cases with public IP is troublesome.

-- 
Mimiko desu.


From vbvbrj at gmail.com  Sat Dec 15 15:09:22 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Sat, 15 Dec 2012 14:09:22 +0200
Subject: [Freeswitch-users] External profile, client before nat.
Message-ID: <50CC6872.1080307@gmail.com>

Hello.

External profile is bind to port 5080 on public IP of FS server 
connected directly to server. Server has a public IP for serving 
external connections to some local services like IVR without any 
registration.

client -> internet -> FS -- is ok.
client -> nat -> internet -> FS -- no sound.

I don't need any client registration for this calls, so no directory 
users are used. The external sip profile is from git. Only sip-ip and 
rtp-ip is equal to ext-sip-ip and ext-rtp-ip and is equal to external IP 
of FS server.

When a client behind a nat calls, in FS log remote SDP shows client's 
local lan ip, not public IP of the router thru which client get access 
to internet. Also channel name is sofia/external/internal-ip. I've tried 
aggressive-nat-detection, NDLB-force-rport, NDLB-sendrecv-in-session in 
sofia profile "external" ad even:

<extension name="sip_nat_detected" continue="true">
   <condition field="${sip_nat_detected}" expression="^true$">
     <action application="set" inline="true" 
data="sip-force-contact=NDLB-connectile-dysfunction"/>
   </condition>
</extension>

in public.xml to no available. I tried two different soft phones. One 
does not get audio at all, other softphone get audio after about 5 sec, 
and in log I see that remote SDP IP had changed to client's public ip.

Interesting thing that the same softphone which never gets audio, does 
not have problem to connect to FS conference server.

What I am missing?
-- 
Mimiko desu.


From a.venugopan at mundio.com  Sat Dec 15 18:57:55 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Sat, 15 Dec 2012 15:57:55 +0000
Subject: [Freeswitch-users] Dbh error
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233BA77@Mail-Kilo.squay.com>

Hi,

When I re-start freeswitch am facing with the below issue irrespective of my database connection is there

2012-12-15 15:13:40.434648 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/directory.lua:164: attempt to call field 'Dbh' (a nil value)
2012-12-15 15:13:40.434664 [ERR] mod_lua.cpp:248 LUA script parse/execute error!

isql -v smepbx smepbx smeswitch
+---------------------------------------+
| Connected!                            |
|                                       |
| sql-statement                         |
| help [tablename]                      |
| quit                                  |
|                                       |
+---------------------------------------+
SQL>

In directory.lua script in 164 line the lines below were present

local dsn = "smepbx"
local dbh = freeswitch.Dbh(dsn, "smepbx", "smeswitch")

Any reason why is it throwing this error in dbh?

Regards,
Archana.V

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From moises.silva at gmail.com  Sat Dec 15 23:33:18 2012
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 15 Dec 2012 15:33:18 -0500
Subject: [Freeswitch-users] Sangoma PRI/Freetdm issue: LOSE_RACE error
 on first call after FS startup
In-Reply-To: <CAGy1GmMTFj6_jGEBheSjP7gUyGjGeObbgWV7OJjLFuJLZ+jGpw@mail.gmail.com>
References: <CAGy1GmMTFj6_jGEBheSjP7gUyGjGeObbgWV7OJjLFuJLZ+jGpw@mail.gmail.com>
Message-ID: <CAA4nhyD-GhPk_S62BYQfUWbbN=w4kucPOcO_7Yvk03O=++dyaQ@mail.gmail.com>

Hello Neil,

On Fri, Dec 14, 2012 at 9:26 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Hi All,
>
> I recently upgraded my box to Ubuntu 12.04 LTS and re-installed FS from
> latest git. I'm using a Sangoma A108DE card, interfacing with FS with
> freetdm. Since upgrade, I'm getting a weird behavior: whenever I first
> start up freeswitch, the *first call* I make to any of my PRI lines does
> not go through. The error is pasted below. Subsequent calls go through
> fine, without the error. So this problem only occurs anytime I have to
> restart FS; I have to call each of the lines manually to get each of them
> to "wake up".
>

It would seem we receive a disconnect just after the setup message. In
order to determine if the disconnect really comes from the network, we need
protocol traces.

Try enabling Q.931 and Q.921 traces just after you restart freeswitch.

fscli> ftdm sangoma_isdn trace q921 <freetdm-span-name>
fscli> ftdm sangoma_isdn trace q931 <freetdm-span-name>

The <freetdm-span-name> argument is the span name you configured in
freetdm.conf.xml

Then also enable some pcap traces at the kernel driver level.

wanpipemon  -i w1g1 -pcap -pcap_file w1g1.pcap -prot ISDN -full -systime -c
trd

This assumes your span is the first one (w1g1), adjust as necessary.

Start all the traces just after you started FreeSWITCH. Then place a single
call in one of your lines (the one you report always fail), then wait a few
seconds, and place the second call that should succeed. Finally stop the
traces and send them to techdesk at sangoma.com with your description of the
problem. Tell them you already talked to me on the FreeSWITCH mailing lists.

Cheers,

*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>







>
> Judging by the error, my first thought is that this is a
> Sangoma/FreeTDM/Wanpipe driver issue. Can a real expert weigh in? Happy to
> provide more logs/output. Thanks!
> Neil
>
> ===================
> 2012-12-15 07:38:01.241594 [INFO] ftmod_sangoma_isdn_stack_rcv.c:96
> [s4c1][4:1] Received SETUP (suId:1 suInstId:0 spInstId:59)
> 2012-12-15 07:38:05.241594 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:862
> [SNGISDN Q931] s4: Protocol: Unknown Event Code(2): Incomp Msg(276)
> 2012-12-15 07:38:09.241595 [INFO] ftmod_sangoma_isdn_stack_rcv.c:232
> [s4c1][4:1] Received DISCONNECT (suId:1 suInstId:0 spInstId:59)
> 2012-12-15 07:38:09.241595 [INFO] ftmod_sangoma_isdn_stack_hndl.c:153
> [s4c1][4:1] Incoming call: Called No:[61608300] Calling No:[919586550654]
> 2012-12-15 07:38:09.241595 [NOTICE] switch_channel.c:968 New Channel
> FreeTDM/4:1/61608300 [458fb2e2-465c-11e2-b550-cfadc9bb1b74]
> 2012-12-15 07:38:09.241595 [NOTICE] mod_freetdm.c:425 Hangup
> FreeTDM/4:1/61608300 [CS_INIT] [LOSE_RACE]
> 2012-12-15 07:38:09.241595 [INFO] ftmod_sangoma_isdn_stack_out.c:441
> [s4c1][4:1] Sending RELEASE/RELEASE COMPLETE (suId:1 suInstId:59
> spInstId:59)
> 2012-12-15 07:38:09.241595 [NOTICE] switch_core_session.c:1506 Session 59
> (FreeTDM/4:1/61608300) Ended
> 2012-12-15 07:38:09.241595 [NOTICE] switch_core_session.c:1510 Close
> Channel FreeTDM/4:1/61608300 [CS_DESTROY]
> 2012-12-15 07:38:09.281595 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267
> [s4c1][4:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:59
> spInstId:59)
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 vbvbrj at gmail.com  Sat Dec 15 23:50:38 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Sat, 15 Dec 2012 22:50:38 +0200
Subject: [Freeswitch-users] Multihomed server.
Message-ID: <50CCE29E.6010804@gmail.com>

Hello.

A problem. I need internal profile to listen to two IPs out of 5. As one 
profile can be set or on "auto" ip, or specific local IP, it is needed 
to create a profile for each IP. But a problem arise when one phone 
registers to first profile and another phone registers to the second 
profile with the same credentials. When calling this specific extension, 
only one phone rings.

How to correctly setup one profile with two different IP on multihomed 
server?

-- 
Mimiko desu.


From steveayre at gmail.com  Sun Dec 16 00:19:37 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 15 Dec 2012 21:19:37 +0000
Subject: [Freeswitch-users] Multihomed server.
In-Reply-To: <50CCE29E.6010804@gmail.com>
References: <50CCE29E.6010804@gmail.com>
Message-ID: <5742042A-24EA-44C7-B44E-2C65FF548FFE@gmail.com>

Impossible... One profile can only listen to a single ip:port - It needs to know an IP to put inside the SIP packets for Contact/SDP. But this isn't your problem.

Rather your problem is coming from the fact they're both registering with the same credentials.

Can you get each phone to register with its own user account?

Also see what string sofia_contact returns - it might be its returning multiple destinations in the generated dial string, but that they're either not a forked dial or you need to set ignore_early_media=true so that a successful bridge happens on the first phone to answer, not the first phone to ring.

I suspect simply prefixing your bridge dial string with {ignore_early_media=true} will get the behaviour you desire.

Sent from my iPad




On 15 Dec 2012, at 20:50, Mimiko <vbvbrj at gmail.com> wrote:

> Hello.
> 
> A problem. I need internal profile to listen to two IPs out of 5. As one 
> profile can be set or on "auto" ip, or specific local IP, it is needed 
> to create a profile for each IP. But a problem arise when one phone 
> registers to first profile and another phone registers to the second 
> profile with the same credentials. When calling this specific extension, 
> only one phone rings.
> 
> How to correctly setup one profile with two different IP on multihomed 
> server?
> 
> -- 
> Mimiko desu.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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


From steveayre at gmail.com  Sun Dec 16 00:30:33 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 15 Dec 2012 21:30:33 +0000
Subject: [Freeswitch-users] External profile, client before nat.
In-Reply-To: <50CC6872.1080307@gmail.com>
References: <50CC6872.1080307@gmail.com>
Message-ID: <E26B9953-DA98-4D1B-B573-605ADF90F103@gmail.com>

The remote SDP will always show what the client sends, that can only be corrected at the client side.

The NDLB options all refer to the return ip/port for sip, generally sending sip replies to where the packets came from. Unfortunately you can't do the same for RTP because it runs on a different port, which will be randomised by most NAT implementations, and may not even be the same ip. There's simply no way to guess it from the FS end.

The mechanism FS does try to use in this situation is that when the client starts to send audio to a FS ip:port (the one in the local SDP), FS will see where the audio is coming from and auto-adjust to using that ip:port to send back to. The log shows when that happens.

Perhaps this is what is happening on the conference server, but not on your own?

It requires that the call be answered by FS, until then the client won't be sending audio. That does mean you won't hear ringing.

Properly correcting the SDP IP is only possible on the client, or failing that on their NAT router (the latter is less preferable as the SIP ALG may interfere with other clients that do it correctly).

The client can learn its external RTP ip:port through STUN - look to see if your client has any options to enable this.

Sent from my iPad



On 15 Dec 2012, at 12:09, Mimiko <vbvbrj at gmail.com> wrote:

> Hello.
> 
> External profile is bind to port 5080 on public IP of FS server 
> connected directly to server. Server has a public IP for serving 
> external connections to some local services like IVR without any 
> registration.
> 
> client -> internet -> FS -- is ok.
> client -> nat -> internet -> FS -- no sound.
> 
> I don't need any client registration for this calls, so no directory 
> users are used. The external sip profile is from git. Only sip-ip and 
> rtp-ip is equal to ext-sip-ip and ext-rtp-ip and is equal to external IP 
> of FS server.
> 
> When a client behind a nat calls, in FS log remote SDP shows client's 
> local lan ip, not public IP of the router thru which client get access 
> to internet. Also channel name is sofia/external/internal-ip. I've tried 
> aggressive-nat-detection, NDLB-force-rport, NDLB-sendrecv-in-session in 
> sofia profile "external" ad even:
> 
> <extension name="sip_nat_detected" continue="true">
>   <condition field="${sip_nat_detected}" expression="^true$">
>     <action application="set" inline="true" 
> data="sip-force-contact=NDLB-connectile-dysfunction"/>
>   </condition>
> </extension>
> 
> in public.xml to no available. I tried two different soft phones. One 
> does not get audio at all, other softphone get audio after about 5 sec, 
> and in log I see that remote SDP IP had changed to client's public ip.
> 
> Interesting thing that the same softphone which never gets audio, does 
> not have problem to connect to FS conference server.
> 
> What I am missing?
> -- 
> Mimiko desu.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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


From krice at freeswitch.org  Sun Dec 16 00:32:22 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 15 Dec 2012 15:32:22 -0600
Subject: [Freeswitch-users] Multihomed server.
In-Reply-To: <5742042A-24EA-44C7-B44E-2C65FF548FFE@gmail.com>
Message-ID: <CCF24886.6211C%krice@freeswitch.org>

Why not just setup 2 profiles sharing the same domain?

This allows the phones to use one set of credentials on either network... If
you are having problems, then why not simply use split horizon dns so that
endpoints on any of the attached networks get proper host pointers...

Its beyond the scope of FreeSWITCH to tell you how to your DNS for this, but
there is a pile of documentation on this.

K 



On 12/15/12 3:19 PM, "Steven Ayre" <steveayre at gmail.com> wrote:

> Impossible... One profile can only listen to a single ip:port - It needs to
> know an IP to put inside the SIP packets for Contact/SDP. But this isn't your
> problem.
> 
> Rather your problem is coming from the fact they're both registering with the
> same credentials.
> 
> Can you get each phone to register with its own user account?
> 
> Also see what string sofia_contact returns - it might be its returning
> multiple destinations in the generated dial string, but that they're either
> not a forked dial or you need to set ignore_early_media=true so that a
> successful bridge happens on the first phone to answer, not the first phone to
> ring.
> 
> I suspect simply prefixing your bridge dial string with
> {ignore_early_media=true} will get the behaviour you desire.
> 
> Sent from my iPad
> 
> 
> 
> 
> On 15 Dec 2012, at 20:50, Mimiko <vbvbrj at gmail.com> wrote:
> 
>> Hello.
>> 
>> A problem. I need internal profile to listen to two IPs out of 5. As one
>> profile can be set or on "auto" ip, or specific local IP, it is needed
>> to create a profile for each IP. But a problem arise when one phone
>> registers to first profile and another phone registers to the second
>> profile with the same credentials. When calling this specific extension,
>> only one phone rings.
>> 
>> How to correctly setup one profile with two different IP on multihomed
>> server?
>> 
>> -- 
>> Mimiko desu.
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From freeswitch at orresta.no-ip.com  Sun Dec 16 02:51:18 2012
From: freeswitch at orresta.no-ip.com (Jakob)
Date: Sun, 16 Dec 2012 00:51:18 +0100
Subject: [Freeswitch-users] conference with inbound portaudio fails on
	second call
Message-ID: <50CD0CF6.3060402@orresta.no-ip.com>

Hi,

I'm trying to get a conference running with a portaudio inbound leg.

this is my dialplan config that fails on the second call to the 
conference, what am I doing wrong?

<condition field="destination_number" expression="^(3001)$" break="on-true">
                         <!-- Right auditorium -->
                         <action application="set" 
data="conference_auto_outcall_flags=unmute"/>
                         <action 
application="conference_set_auto_outcall" 
data="portaudio/endpoint/in-right"/>
                         <action application="conference" 
data="bridge:$1++flags{mute|endconf}"/>
</condition>

the logs from the second call:

2012-12-16 00:28:10.805545 [DEBUG] switch_core_session.c:830 Send signal 
sofia/external/anonymous at sip.teleman.com [BREAK]
2012-12-16 00:28:10.805545 [DEBUG] mod_conference.c:3487 Setup timer 
soft success interval: 20  samples: 160
2012-12-16 00:28:10.805545 [DEBUG] mod_conference.c:6835 Launching BG 
Thread for outcall
2012-12-16 00:28:10.815163 [DEBUG] switch_ivr_originate.c:2022 Parsing 
global variables
2012-12-16 00:28:10.815163 [NOTICE] mod_portaudio.c:1294 Close Channel 
N/A [CS_NEW]
2012-12-16 00:28:10.815163 [DEBUG] switch_core_state_machine.c:559 () 
Running State Change CS_DESTROY
2012-12-16 00:28:10.815163 [DEBUG] switch_core_state_machine.c:569 (N/A) 
State DESTROY
2012-12-16 00:28:10.815163 [DEBUG] switch_core_state_machine.c:569 (N/A) 
State DESTROY going to sleep
2012-12-16 00:28:10.815163 [NOTICE] switch_ivr_originate.c:2608 Cannot 
create outgoing channel of type [portaudio] cause: [USER_BUSY]
2012-12-16 00:28:10.815163 [DEBUG] switch_ivr_originate.c:3531 Originate 
Resulted in Error Cause: 17 [USER_BUSY]
2012-12-16 00:28:10.815163 [ERR] mod_conference.c:6634 Cannot create 
outgoing channel, cause: USER_BUSY
2012-12-16 00:28:10.914722 [DEBUG] mod_conference.c:4301 Queueing file 
'tone_stream://%(500,0,640)' for play

Regards
Jakob


From lists at kavun.ch  Sun Dec 16 03:10:02 2012
From: lists at kavun.ch (Emrah)
Date: Sat, 15 Dec 2012 19:10:02 -0500
Subject: [Freeswitch-users] Multihomed server.
In-Reply-To: <50CCE29E.6010804@gmail.com>
References: <50CCE29E.6010804@gmail.com>
Message-ID: <51655B53-8F97-4A8D-85BB-23A6A84A3B1F@kavun.ch>

Hi,

You can create 2 profiles for the same domain and look for the same user on both profiles.
Try:
<action application="bridge" data="${sofia_contact(profile1/user at domain)}:_:${sofia_contact(profile2/user at domain)}" />

Let us know if that works.

Emrah
On Dec 15, 2012, at 3:50 PM, Mimiko <vbvbrj at gmail.com> wrote:

> Hello.
> 
> A problem. I need internal profile to listen to two IPs out of 5. As one 
> profile can be set or on "auto" ip, or specific local IP, it is needed 
> to create a profile for each IP. But a problem arise when one phone 
> registers to first profile and another phone registers to the second 
> profile with the same credentials. When calling this specific extension, 
> only one phone rings.
> 
> How to correctly setup one profile with two different IP on multihomed 
> server?
> 
> -- 
> Mimiko desu.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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



From nandy1925 at gmail.com  Sun Dec 16 05:08:59 2012
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sun, 16 Dec 2012 10:08:59 +0800
Subject: [Freeswitch-users] Mutiple GSMopen USB modem setup
In-Reply-To: <CALXCt0pT2UoMWZ0dUPC2zKTbm_7zd2Z_2UAr1e8i78JO24uXTw@mail.gmail.com>
References: <CAHKqMb6OBRVtOkEepF-a5fVXz_x-NbvhDijvU+dgCBsGcQ5sKg@mail.gmail.com>
	<CALXCt0pT2UoMWZ0dUPC2zKTbm_7zd2Z_2UAr1e8i78JO24uXTw@mail.gmail.com>
Message-ID: <CAHKqMb69d22pUBqDEw16MwEh_EDR-wMG=j78zYKGFq1ubGZ0DA@mail.gmail.com>

Thanks for the feedback Giovanni.  Re fixing the ttyUSB port assignment, I
have searched a guy made a script to scan USB serial numbers using udev,
then created symbolic links. Will try it out later.
/Nandy


On Thu, Dec 13, 2012 at 7:02 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> On Thu, Dec 13, 2012 at 4:44 AM, Nandy Dagondon <nandy1925 at gmail.com>
> wrote:
> > I have run GSMOpen on CentOS 5.7 defying the Wiki advisory. Questions:
>
> I would invite you to go for the OS distros detailed in the wiki page.
> You'll probably encounter problems with different distros, and you're
> on your own to solve it.
>
> > 1. What is the maximum number of USB modems tested? Can we get the
> numbers
> > and the CPU used?
>
> I've heard about 48 concurrent, and 64. Me personally have tested with
> 5. No CPU consumption. The critical part is the USB BUS. So use
> cascading and POWERED good usb 2.0 hubs
>
> > 2. I'll be installing multiple modems each connected to a different
> mobile
> > network. Is the /dev/ttyUSB assignments  constant for every modem?
> Meaning
> > it doesn't change if I plug it on different USB jacks.
>
> it will change not only if you change USB port, but also randomly if
> you stay on the same USB port and reboot (and sometimes also without
> rebooting). That's a "feature" of Linux distros (a demented one,
> cannot understand why they choose this behavior).
>
> Soon or later I'll look into this, and come out with a solution (I've
> made some preliminary research and reasoning about in the past).
>
> If you have a commercial interest in that, and a real budget for it,
> contact me in private as consultant, or put a public bounty on it.
>
> -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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Dec 16 12:01:24 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 16 Dec 2012 11:01:24 +0200
Subject: [Freeswitch-users] Disable media one leg?
Message-ID: <CALNZuwkEfHFyauP=M3_jip27fJN7vhFZ_MiSsOhQdCvQXTHShg@mail.gmail.com>

This is a funny question.
I'm setting up a call-in line to listen to to streaming audio.
On this particular setup, there's no options - no dtmf, nothing at all for
the user to do other than listen or hang up.

So can I disable the RTP on the leg from the caller to my server, to not
waste bandwidth? (Although it's inbound, so I don't actually pay for that.)
e.g. can I set MY inbound RTP ip/port to null?

Even if FS supports this, will this possibly not play well with the carrier?

Thanks!
-Avi
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From b2m at a-cti.com  Sun Dec 16 14:40:49 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Sun, 16 Dec 2012 17:10:49 +0530
Subject: [Freeswitch-users] G729 - [INCOMPATIBLE_DESTINATION]
Message-ID: <CAP2fbbcixeTy4H+KPbxJQvAhCDWmco5a0jF0XQkGHBtE2Qkmdg@mail.gmail.com>

Need help on Codec Negotiation


2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare *
[G729:0:8000:20:64000]/[G729:18:8000:20:8000]*
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[G722:9:8000:20:64000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
send/recv payload to 101


2012-12-16 11:35:52.246422 [DEBUG] switch_channel.c:2852 (sofia/internal/
501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
2012-12-16 11:35:52.246422 [NOTICE] sofia.c:5781 Hangup sofia/internal/
501 at 50.54.12.39 [CS_NEW] *[INCOMPATIBLE_DESTINATION]*

Thanks,
Bala
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From lists at kavun.ch  Sun Dec 16 16:43:37 2012
From: lists at kavun.ch (Emrah)
Date: Sun, 16 Dec 2012 08:43:37 -0500
Subject: [Freeswitch-users] Disable media one leg?
In-Reply-To: <CALNZuwkEfHFyauP=M3_jip27fJN7vhFZ_MiSsOhQdCvQXTHShg@mail.gmail.com>
References: <CALNZuwkEfHFyauP=M3_jip27fJN7vhFZ_MiSsOhQdCvQXTHShg@mail.gmail.com>
Message-ID: <913A55CF-2372-4A41-B4EE-1A39F00B64D2@kavun.ch>

I am trying to make sense of what you are trying to do with no luck.

If FS is going to be streaming the audio the media path is obviously essential, especially if it's going to be doing some transcoding with mod_shout or something similar.
How do you expect the audio to be carried if you drop the RTP? 

If the carrier is playing the stream and all it needs is a session to be acknowledged by FS (e.g.: early media music ringtone) try ring_ready from your dialplan.

Please clarify further.

On Dec 16, 2012, at 4:01 AM, Avi Marcus <avi at avimarcus.net> wrote:

> This is a funny question.
> I'm setting up a call-in line to listen to to streaming audio.
> On this particular setup, there's no options - no dtmf, nothing at all for the user to do other than listen or hang up.
> 
> So can I disable the RTP on the leg from the caller to my server, to not waste bandwidth? (Although it's inbound, so I don't actually pay for that.) e.g. can I set MY inbound RTP ip/port to null?
> 
> Even if FS supports this, will this possibly not play well with the carrier?
> 
> Thanks!
> -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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Dec 16 16:56:24 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 16 Dec 2012 15:56:24 +0200
Subject: [Freeswitch-users] Disable media one leg?
In-Reply-To: <913A55CF-2372-4A41-B4EE-1A39F00B64D2@kavun.ch>
References: <CALNZuwkEfHFyauP=M3_jip27fJN7vhFZ_MiSsOhQdCvQXTHShg@mail.gmail.com>
	<913A55CF-2372-4A41-B4EE-1A39F00B64D2@kavun.ch>
Message-ID: <CALNZuwn5-tde2TA1S4iJdF7Mouwhg4otcKuivNhv8h6E0Za=pw@mail.gmail.com>

I want to drop HALF the RTP. FS -> endpoint should send media, but the
other leg of the RTP inbound should just be dropped and not bother the
interweb.

-Avi


On Sun, Dec 16, 2012 at 3:43 PM, Emrah <lists at kavun.ch> wrote:

> I am trying to make sense of what you are trying to do with no luck.
>
> If FS is going to be streaming the audio the media path is obviously
> essential, especially if it's going to be doing some transcoding with
> mod_shout or something similar.
> How do you expect the audio to be carried if you drop the RTP?
>
> If the carrier is playing the stream and all it needs is a session to be
> acknowledged by FS (e.g.: early media music ringtone) try ring_ready from
> your dialplan.
>
> Please clarify further.
>
> On Dec 16, 2012, at 4:01 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
> > This is a funny question.
> > I'm setting up a call-in line to listen to to streaming audio.
> > On this particular setup, there's no options - no dtmf, nothing at all
> for the user to do other than listen or hang up.
> >
> > So can I disable the RTP on the leg from the caller to my server, to not
> waste bandwidth? (Although it's inbound, so I don't actually pay for that.)
> e.g. can I set MY inbound RTP ip/port to null?
> >
> > Even if FS supports this, will this possibly not play well with the
> carrier?
> >
> > Thanks!
> > -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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 ahmed at netelsat.net  Sun Dec 16 17:08:59 2012
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Sun, 16 Dec 2012 19:08:59 +0500
Subject: [Freeswitch-users] PostgreSQL in the core
In-Reply-To: <20F2980066FC4F198FF78D66271ECF47@gmail.com>
References: <CALFFxt13ePTSk_GFnAvK6GD7tRo2qn1mv3irsE+d1ZSu=SH08A@mail.gmail.com>
	<B89E0118B7FD4A9CA453643D5B1675B9@gmail.com>
	<CALFFxt3zWP22vYJMPiLuvQzfj9aqwJ+MeCZvH2efV4QWtAfQ5Q@mail.gmail.com>
	<20F2980066FC4F198FF78D66271ECF47@gmail.com>
Message-ID: <CALFFxt2BiqKGZcHTARakahGUQRe4fsPZBwjKoQ=E4+-JJAFevg@mail.gmail.com>

Even removing that doesn't work.
Any one using new Postgresql in core without ODBC off course  ?

On Sat, Dec 15, 2012 at 4:03 PM, Seven Du <dujinfang at gmail.com> wrote:

> I would try also remove options.
>
> --
> Seven Du
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Saturday, December 15, 2012 at 10:33 AM, Ahmed Sboor wrote:
>
> Did that , then it says invalid option " client_min_messages"
>
> On Sat, Dec 15, 2012 at 7:21 AM, Seven Du <dujinfang at gmail.com> wrote:
>
> Try to remove the application_name option ?
>
> --
> Seven Du
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Saturday, December 15, 2012 at 7:52 AM, Ahmed Sboor wrote:
>
> Hi all,
> i am just trying to use 1.2.5.3 to use Postgresql as core.
> Followed by http://wiki.freeswitch.org/wiki/PostgreSQL_in_the_core
> but its giving error like :
>
> switch_pgsql.c:492 invalid connection option "application_name"
>
> and
>
> [CRIT] switch_core_sqldb.c:504 Failure to connect to PGSQL
> hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password=''
> options='-c client_min_messages=NOTICE' application_name='freeswitch'!
>
>
> Postgresql version is 8.4.13.
>
> can any one please help where i am missing something ?
>
> Thanking you all
> Ahmed
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jnvines at gmail.com  Sun Dec 16 17:25:01 2012
From: jnvines at gmail.com (Nick Vines)
Date: Sun, 16 Dec 2012 09:25:01 -0500
Subject: [Freeswitch-users] Disable media one leg?
In-Reply-To: <CALNZuwn5-tde2TA1S4iJdF7Mouwhg4otcKuivNhv8h6E0Za=pw@mail.gmail.com>
References: <CALNZuwkEfHFyauP=M3_jip27fJN7vhFZ_MiSsOhQdCvQXTHShg@mail.gmail.com>
	<913A55CF-2372-4A41-B4EE-1A39F00B64D2@kavun.ch>
	<CALNZuwn5-tde2TA1S4iJdF7Mouwhg4otcKuivNhv8h6E0Za=pw@mail.gmail.com>
Message-ID: <CA+NjZ2uz6k1foGgiseS5-O6E=G2u01XzHCL2FM1t4AnxesFwRw@mail.gmail.com>

This might not be the best way, but you could make a 2 person conference
with the listener muted and the controls disabled. Then you could also use rtp
vad <http://wiki.freeswitch.org/wiki/VAD_and_CNG> to reduce the bandwidth.
Again, I'm not sure if these would stack appropriately, but it seems
feasible.

Nick


On Sun, Dec 16, 2012 at 8:56 AM, Avi Marcus <avi at avimarcus.net> wrote:

> If FS is going to be streaming the audio the media path is obviously
>> essential, especially if it's going to be doing some transcoding with
>> mod_shout or something similar.
>>
>
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From curriegrad2004 at gmail.com  Sun Dec 16 18:30:00 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 16 Dec 2012 07:30:00 -0800
Subject: [Freeswitch-users] G729 - [INCOMPATIBLE_DESTINATION]
In-Reply-To: <CAP2fbbcixeTy4H+KPbxJQvAhCDWmco5a0jF0XQkGHBtE2Qkmdg@mail.gmail.com>
References: <CAP2fbbcixeTy4H+KPbxJQvAhCDWmco5a0jF0XQkGHBtE2Qkmdg@mail.gmail.com>
Message-ID: <CAMWiMy8mSvf3jzaqS55BN3OU_UrTKUYefh=HVtuWS_DoYwRySA@mail.gmail.com>

Could you paste the sofia config that handles the internal profile?
I'm suspecting either the G729 codec is missing on the vars.xml file
or missing in there. Also posting the SDP codec negotiation would be
helpful.

On Sun, Dec 16, 2012 at 3:40 AM, Balamurugan Mahendran <b2m at a-cti.com> wrote:
> Need help on Codec Negotiation
>
>
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf send/recv
> payload to 101
>
>
> 2012-12-16 11:35:52.246422 [DEBUG] switch_channel.c:2852
> (sofia/internal/501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
> 2012-12-16 11:35:52.246422 [NOTICE] sofia.c:5781 Hangup
> sofia/internal/501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From cal.leeming at simplicitymedialtd.co.uk  Sun Dec 16 19:15:02 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 16 Dec 2012 16:15:02 +0000
Subject: [Freeswitch-users] NAT traversal - the final say..!
Message-ID: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>

*Any and all feedback on this thread would be much welcomed.*

Hello,
*
*
There seems to be a large number of discussions surrounding NAT traversal,
as well as lots of documentation, but with no concrete answers.

The NAT related wiki documentation is tedious, and depending on the outcome
of this thread, I'd like to spend some time cleaning it up.

The most common problem (the same as ours) was having a router with broken
ALG and a softphone that does not seem to work with STUN.

The following REGISTER is sent from a phone.

REGISTER sip:1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:57787
;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>
To: "foxx"<sip:2000 at 1.2.3.4:5060>
From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448
Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.
CSeq: 7 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 6.0.25732.0
Content-Length: 0

As you can see, the client's public IP is not specified
anywhere. FreeSWITCH offers several ways around this, the main ones being;

* NDLB-connectile-dysfunction
* NDLB-force-rport
* apply-nat-acl
* sip-force-contact

The one that has worked in our case was "NDLB-connectile-dysfunction"
(otherwise known as NAT HACK), however there seems to be a lot of negative
comments about using this.

>From what I can tell, the general argument is that NAT HACK is considered a
non RFC compliant hack, and the SIP phones should be doing a better job of
keeping to the RFCs.

In principle, this is a fair argument - but in practise, it's not a
reasonable assumption that all phones are RFC compliant, and (imho) not a
reasonable argument to have this functionality disabled by default.

So, I'd like to present the following arguments;

* Are there any other negative aspects about
using NDLB-connectile-dysfunction, other than it is a non compliant RFC
hack?

* Why is NDLB-connectile-dysfunction not enabled by default when certain
conditions are met? In the event that FreeSWITCH receives a REGISTER from a
phone specifying a Contact/Via as 192.168.0.0/16, but received on a public
IP, then it should be obvious that NAT is broken and automatically try to
circumvent it.

* People seem to get confused between server side and client side NAT
problems, and that they both need to be resolved in a different way. The
documentation doesn't seem to reflect this clearly.
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From lists at kavun.ch  Sun Dec 16 19:34:06 2012
From: lists at kavun.ch (Emrah)
Date: Sun, 16 Dec 2012 11:34:06 -0500
Subject: [Freeswitch-users] Disable media one leg?
In-Reply-To: <CA+NjZ2uz6k1foGgiseS5-O6E=G2u01XzHCL2FM1t4AnxesFwRw@mail.gmail.com>
References: <CALNZuwkEfHFyauP=M3_jip27fJN7vhFZ_MiSsOhQdCvQXTHShg@mail.gmail.com>
	<913A55CF-2372-4A41-B4EE-1A39F00B64D2@kavun.ch>
	<CALNZuwn5-tde2TA1S4iJdF7Mouwhg4otcKuivNhv8h6E0Za=pw@mail.gmail.com>
	<CA+NjZ2uz6k1foGgiseS5-O6E=G2u01XzHCL2FM1t4AnxesFwRw@mail.gmail.com>
Message-ID: <43C0CE25-09A5-47E2-B121-1E89FDCF9303@kavun.ch>

Ah ha! I get it now.
VAD sounds like what you may be looking for.

http://wiki.freeswitch.org/wiki/VAD_and_CNG
On Dec 16, 2012, at 9:25 AM, Nick Vines <jnvines at gmail.com> wrote:

> This might not be the best way, but you could make a 2 person conference with the listener muted and the controls disabled. Then you could also use rtp vad to reduce the bandwidth. Again, I'm not sure if these would stack appropriately, but it seems feasible. 
> 
> Nick
> 
> 
> On Sun, Dec 16, 2012 at 8:56 AM, Avi Marcus <avi at avimarcus.net> wrote:
> If FS is going to be streaming the audio the media path is obviously essential, especially if it's going to be doing some transcoding with mod_shout or something similar.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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



From sdevoy at bizfocused.com  Sun Dec 16 19:57:26 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Sun, 16 Dec 2012 11:57:26 -0500
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
Message-ID: <480f01cddbae$6d7d67e0$487837a0$@bizfocused.com>

I have spent many hours working on NAT issues on client end, my server has a
public address.  

 

With CISCO brand phones I did not need any non-standards compliant settings,
just turning on all the choices in the CISCO web setup NAT section.
However, with Polycom 335 phones (as of Dec 2012) I could not get registered
or get audio without the following:

* NDLB-connectile-dysfunction

* NDLB-force-rport

* Enable SIP ALG on my FIOS router.

With those setting however, this has worked perfectly.  Also note that when
I turned on SIP ALG, my Cisco phones quite working until I added the NDLB
parameter/variable to the Cisco <user> in the directory.   They seem to be
quite complimentary but seem be requirements for each other.

 

I really tried to stay away from SIP ALG because so many posts were so
negative about it.  Without the NDLB "flags" I could never see any
difference when enabling SIP ALG.  The combination for me has been
fantastic.

 

HOWEVER, since there are so many different versions of "success" in the IRC
and Wiki,  I am pretty sure that other router brands with different SIP ALG
implementations and/or other phone brands or even firmware versions may need
different configurations.  It is almost like we just need a checklist that
says try these combinations until you find one that fits your site.

 

HTH,

sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cal
Leeming [Simplicity Media Ltd]
Sent: Sunday, December 16, 2012 11:15 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] NAT traversal - the final say..!

 

Any and all feedback on this thread would be much welcomed.

 

Hello,

 

There seems to be a large number of discussions surrounding NAT traversal,
as well as lots of documentation, but with no concrete answers. 

 

The NAT related wiki documentation is tedious, and depending on the outcome
of this thread, I'd like to spend some time cleaning it up.

 

The most common problem (the same as ours) was having a router with broken
ALG and a softphone that does not seem to work with STUN.

 

The following REGISTER is sent from a phone.

 

REGISTER sip:1.2.3.4:5060 SIP/2.0

Via: SIP/2.0/UDP
192.168.1.102:57787;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>

To: "foxx"<sip:2000 at 1.2.3.4:5060>

From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448

Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.

CSeq: 7 REGISTER

Expires: 120

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
REFER, INFO, MESSAGE

Supported: replaces

User-Agent: 3CXPhone 6.0.25732.0

Content-Length: 0

 

As you can see, the client's public IP is not specified anywhere. FreeSWITCH
offers several ways around this, the main ones being;

 

* NDLB-connectile-dysfunction

* NDLB-force-rport

* apply-nat-acl

* sip-force-contact

 

The one that has worked in our case was "NDLB-connectile-dysfunction"
(otherwise known as NAT HACK), however there seems to be a lot of negative
comments about using this.

 

>From what I can tell, the general argument is that NAT HACK is considered a
non RFC compliant hack, and the SIP phones should be doing a better job of
keeping to the RFCs.

 

In principle, this is a fair argument - but in practise, it's not a
reasonable assumption that all phones are RFC compliant, and (imho) not a
reasonable argument to have this functionality disabled by default.

 

So, I'd like to present the following arguments;

 

* Are there any other negative aspects about using
NDLB-connectile-dysfunction, other than it is a non compliant RFC hack?

 

* Why is NDLB-connectile-dysfunction not enabled by default when certain
conditions are met? In the event that FreeSWITCH receives a REGISTER from a
phone specifying a Contact/Via as 192.168.0.0/16, but received on a public
IP, then it should be obvious that NAT is broken and automatically try to
circumvent it.

 

* People seem to get confused between server side and client side NAT
problems, and that they both need to be resolved in a different way. The
documentation doesn't seem to reflect this clearly.

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From krice at freeswitch.org  Sun Dec 16 19:58:03 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 16 Dec 2012 10:58:03 -0600
Subject: [Freeswitch-users] Disable media one leg?
In-Reply-To: <43C0CE25-09A5-47E2-B121-1E89FDCF9303@kavun.ch>
Message-ID: <CCF359BB.6228C%krice@freeswitch.org>

If you look at mod conference there is the waste flag, if there is no media
to send to a channel, mod_conference doesn't send media to it... You should
start looking there


On 12/16/12 10:34 AM, "Emrah" <lists at kavun.ch> wrote:

> Ah ha! I get it now.
> VAD sounds like what you may be looking for.
> 
> http://wiki.freeswitch.org/wiki/VAD_and_CNG
> On Dec 16, 2012, at 9:25 AM, Nick Vines <jnvines at gmail.com> wrote:
> 
>> This might not be the best way, but you could make a 2 person conference with
>> the listener muted and the controls disabled. Then you could also use rtp vad
>> to reduce the bandwidth. Again, I'm not sure if these would stack
>> appropriately, but it seems feasible.
>> 
>> Nick
>> 
>> 
>> On Sun, Dec 16, 2012 at 8:56 AM, Avi Marcus <avi at avimarcus.net> wrote:
>> If FS is going to be streaming the audio the media path is obviously
>> essential, especially if it's going to be doing some transcoding with
>> mod_shout or something similar.
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From avi at avimarcus.net  Sun Dec 16 20:09:12 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 16 Dec 2012 19:09:12 +0200
Subject: [Freeswitch-users] Disable media one leg?
In-Reply-To: <43C0CE25-09A5-47E2-B121-1E89FDCF9303@kavun.ch>
References: <CALNZuwkEfHFyauP=M3_jip27fJN7vhFZ_MiSsOhQdCvQXTHShg@mail.gmail.com>
	<913A55CF-2372-4A41-B4EE-1A39F00B64D2@kavun.ch>
	<CALNZuwn5-tde2TA1S4iJdF7Mouwhg4otcKuivNhv8h6E0Za=pw@mail.gmail.com>
	<CA+NjZ2uz6k1foGgiseS5-O6E=G2u01XzHCL2FM1t4AnxesFwRw@mail.gmail.com>
	<43C0CE25-09A5-47E2-B121-1E89FDCF9303@kavun.ch>
Message-ID: <CALNZuw=AkE63L3m50=qZkEbB0UL5sYgZbZWa2zaS73jnacOtWw@mail.gmail.com>

Hmm, it sounds like I want rtp_enable_vad_in - for incoming media to be
reduced, if I understand it correctly .. Where do I set this?

I presume silence threshold set on the opposite end, and I can't set
something very high to ensure no audio at all comes to me.

-Avi



On Sun, Dec 16, 2012 at 6:34 PM, Emrah <lists at kavun.ch> wrote:

> Ah ha! I get it now.
> VAD sounds like what you may be looking for.
>
> http://wiki.freeswitch.org/wiki/VAD_and_CNG
> On Dec 16, 2012, at 9:25 AM, Nick Vines <jnvines at gmail.com> wrote:
>
> > This might not be the best way, but you could make a 2 person conference
> with the listener muted and the controls disabled. Then you could also use
> rtp vad to reduce the bandwidth. Again, I'm not sure if these would stack
> appropriately, but it seems feasible.
> >
> > Nick
> >
> >
> > On Sun, Dec 16, 2012 at 8:56 AM, Avi Marcus <avi at avimarcus.net> wrote:
> > If FS is going to be streaming the audio the media path is obviously
> essential, especially if it's going to be doing some transcoding with
> mod_shout or something similar.
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From cal.leeming at simplicitymedialtd.co.uk  Sun Dec 16 20:15:17 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 16 Dec 2012 17:15:17 +0000
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <480f01cddbae$6d7d67e0$487837a0$@bizfocused.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<480f01cddbae$6d7d67e0$487837a0$@bizfocused.com>
Message-ID: <CAHKQagGETUVwZ+4jV4mXzXt_P-yMFjFEfEY=1DQBf6i8Yo6mfg@mail.gmail.com>

Hi Sean,

Thank you for the detailed reply.

The more info we can get about individual NAT experiences, the better - I'm
hoping others will follow suit!

Cal

On Sun, Dec 16, 2012 at 4:57 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> I have spent many hours working on *NAT issues on client end*, my server
> has a public address.  ****
>
> ** **
>
> With CISCO brand phones I did not need any non-standards compliant
> settings, just turning on all the choices in the CISCO web setup NAT
> section.  However, with Polycom 335 phones (as of Dec 2012) I could not get
> registered or get audio without the following:****
>
> * NDLB-connectile-dysfunction****
>
> * NDLB-force-rport****
>
> * Enable SIP ALG on my FIOS router.****
>
> With those setting however, this has worked perfectly.  Also note that
> when I turned on SIP ALG, my Cisco phones quite working until I added the
> NDLB parameter/variable to the Cisco <user> in the directory.   They seem
> to be quite complimentary but seem be requirements for each other.****
>
> ** **
>
> I really tried to stay away from SIP ALG because so many posts were so
> negative about it.  Without the NDLB ?flags? I could never see any
> difference when enabling SIP ALG.  The combination for me has been
> fantastic.****
>
> ** **
>
> HOWEVER, since there are so many different versions of ?success? in the
> IRC and Wiki,  I am pretty sure that other router brands with different SIP
> ALG implementations and/or other phone brands or even firmware versions may
> need different configurations.  It is almost like we just need a checklist
> that says try these combinations until you find one that fits your site.**
> **
>
> ** **
>
> HTH,****
>
> sean****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal Leeming
> [Simplicity Media Ltd]
> *Sent:* Sunday, December 16, 2012 11:15 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] NAT traversal - the final say..!****
>
> ** **
>
> *Any and all feedback on this thread would be much welcomed.*****
>
> ** **
>
> Hello,****
>
> ** **
>
> There seems to be a large number of discussions surrounding NAT traversal,
> as well as lots of documentation, but with no concrete answers. ****
>
> ** **
>
> The NAT related wiki documentation is tedious, and depending on the
> outcome of this thread, I'd like to spend some time cleaning it up.****
>
> ** **
>
> The most common problem (the same as ours) was having a router with broken
> ALG and a softphone that does not seem to work with STUN.****
>
> ** **
>
> The following REGISTER is sent from a phone.****
>
> ** **
>
> REGISTER sip:1.2.3.4:5060 SIP/2.0****
>
> Via: SIP/2.0/UDP 192.168.1.102:57787
> ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport****
>
> Max-Forwards: 70****
>
> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>****
>
> To: "foxx"<sip:2000 at 1.2.3.4:5060>****
>
> From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448****
>
> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.****
>
> CSeq: 7 REGISTER****
>
> Expires: 120****
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
> REFER, INFO, MESSAGE****
>
> Supported: replaces****
>
> User-Agent: 3CXPhone 6.0.25732.0****
>
> Content-Length: 0****
>
> ** **
>
> As you can see, the client's public IP is not specified
> anywhere. FreeSWITCH offers several ways around this, the main ones being;
> ****
>
> ** **
>
> * NDLB-connectile-dysfunction****
>
> * NDLB-force-rport****
>
> * apply-nat-acl****
>
> * sip-force-contact****
>
> ** **
>
> The one that has worked in our case was "NDLB-connectile-dysfunction"
> (otherwise known as NAT HACK), however there seems to be a lot of negative
> comments about using this.****
>
> ** **
>
> From what I can tell, the general argument is that NAT HACK is considered
> a non RFC compliant hack, and the SIP phones should be doing a better job
> of keeping to the RFCs.****
>
> ** **
>
> In principle, this is a fair argument - but in practise, it's not a
> reasonable assumption that all phones are RFC compliant, and (imho) not a
> reasonable argument to have this functionality disabled by default.****
>
> ** **
>
> So, I'd like to present the following arguments;****
>
> ** **
>
> * Are there any other negative aspects about
> using NDLB-connectile-dysfunction, other than it is a non compliant RFC
> hack?****
>
> ** **
>
> * Why is NDLB-connectile-dysfunction not enabled by default when certain
> conditions are met? In the event that FreeSWITCH receives a REGISTER from a
> phone specifying a Contact/Via as 192.168.0.0/16, but received on a
> public IP, then it should be obvious that NAT is broken and automatically
> try to circumvent it.****
>
> ** **
>
> * People seem to get confused between server side and client side NAT
> problems, and that they both need to be resolved in a different way. The
> documentation doesn't seem to reflect this clearly.****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 avi at avimarcus.net  Sun Dec 16 20:33:56 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 16 Dec 2012 19:33:56 +0200
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAHKQagGETUVwZ+4jV4mXzXt_P-yMFjFEfEY=1DQBf6i8Yo6mfg@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<480f01cddbae$6d7d67e0$487837a0$@bizfocused.com>
	<CAHKQagGETUVwZ+4jV4mXzXt_P-yMFjFEfEY=1DQBf6i8Yo6mfg@mail.gmail.com>
Message-ID: <CALNZuwnTOOavLzOrGVq+zw1CGj+HdkCamCUqnUo6hPsb-YeMJg@mail.gmail.com>

My main experience is with the Linksys/Cisco (sipura) SPA-2102 ATA.

I always disable ALG in the router.

I turn on NAT ping of 15 seconds in the Linksys.
And.. here's the variable part - I also turn on 2-5 of the VIAs. I haven't
really pinned that one down.

This is not strictly NAT related... but has bit me a few times: devices by
default want to use :5060 for their SIP. Not all are smart enough to see
something else is using it and try a different port automatically.

And for your peace of mind, try to never need NAT on the server.

-Avi


On Sun, Dec 16, 2012 at 7:15 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Hi Sean,
>
> Thank you for the detailed reply.
>
> The more info we can get about individual NAT experiences, the better -
> I'm hoping others will follow suit!
>
> Cal
>
> On Sun, Dec 16, 2012 at 4:57 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>
>> I have spent many hours working on *NAT issues on client end*, my server
>> has a public address.  ****
>>
>> ** **
>>
>> With CISCO brand phones I did not need any non-standards compliant
>> settings, just turning on all the choices in the CISCO web setup NAT
>> section.  However, with Polycom 335 phones (as of Dec 2012) I could not get
>> registered or get audio without the following:****
>>
>> * NDLB-connectile-dysfunction****
>>
>> * NDLB-force-rport****
>>
>> * Enable SIP ALG on my FIOS router.****
>>
>> With those setting however, this has worked perfectly.  Also note that
>> when I turned on SIP ALG, my Cisco phones quite working until I added the
>> NDLB parameter/variable to the Cisco <user> in the directory.   They seem
>> to be quite complimentary but seem be requirements for each other.****
>>
>> ** **
>>
>> I really tried to stay away from SIP ALG because so many posts were so
>> negative about it.  Without the NDLB ?flags? I could never see any
>> difference when enabling SIP ALG.  The combination for me has been
>> fantastic.****
>>
>> ** **
>>
>> HOWEVER, since there are so many different versions of ?success? in the
>> IRC and Wiki,  I am pretty sure that other router brands with different SIP
>> ALG implementations and/or other phone brands or even firmware versions may
>> need different configurations.  It is almost like we just need a checklist
>> that says try these combinations until you find one that fits your site.*
>> ***
>>
>> ** **
>>
>> HTH,****
>>
>> sean****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal
>> Leeming [Simplicity Media Ltd]
>> *Sent:* Sunday, December 16, 2012 11:15 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] NAT traversal - the final say..!****
>>
>> ** **
>>
>> *Any and all feedback on this thread would be much welcomed.*****
>>
>> ** **
>>
>> Hello,****
>>
>> ** **
>>
>> There seems to be a large number of discussions surrounding NAT
>> traversal, as well as lots of documentation, but with no concrete answers.
>> ****
>>
>> ** **
>>
>> The NAT related wiki documentation is tedious, and depending on the
>> outcome of this thread, I'd like to spend some time cleaning it up.****
>>
>> ** **
>>
>> The most common problem (the same as ours) was having a router with
>> broken ALG and a softphone that does not seem to work with STUN.****
>>
>> ** **
>>
>> The following REGISTER is sent from a phone.****
>>
>> ** **
>>
>> REGISTER sip:1.2.3.4:5060 SIP/2.0****
>>
>> Via: SIP/2.0/UDP 192.168.1.102:57787
>> ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport****
>>
>> Max-Forwards: 70****
>>
>> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513<https://mail.google.com/_/mail-static/_/js/main/m_i,t,it/rt=h/ver=dQ95YePrryI.en./sv=1/am=!rQczwx1unpD1BO2bNKsLUpXXtiIjaa01SgsJmP23wMtqPKKB37R0dPvFB_9tzlm4wJdbIQ/d=1>
>> >****
>>
>> To: "foxx"<sip:2000 at 1.2.3.4:5060>****
>>
>> From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448****
>>
>> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.****
>>
>> CSeq: 7 REGISTER****
>>
>> Expires: 120****
>>
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
>> REFER, INFO, MESSAGE****
>>
>> Supported: replaces****
>>
>> User-Agent: 3CXPhone 6.0.25732.0****
>>
>> Content-Length: 0****
>>
>> ** **
>>
>> As you can see, the client's public IP is not specified
>> anywhere. FreeSWITCH offers several ways around this, the main ones being;
>> ****
>>
>> ** **
>>
>> * NDLB-connectile-dysfunction****
>>
>> * NDLB-force-rport****
>>
>> * apply-nat-acl****
>>
>> * sip-force-contact****
>>
>> ** **
>>
>> The one that has worked in our case was "NDLB-connectile-dysfunction"
>> (otherwise known as NAT HACK), however there seems to be a lot of negative
>> comments about using this.****
>>
>> ** **
>>
>> From what I can tell, the general argument is that NAT HACK is considered
>> a non RFC compliant hack, and the SIP phones should be doing a better job
>> of keeping to the RFCs.****
>>
>> ** **
>>
>> In principle, this is a fair argument - but in practise, it's not a
>> reasonable assumption that all phones are RFC compliant, and (imho) not a
>> reasonable argument to have this functionality disabled by default.****
>>
>> ** **
>>
>> So, I'd like to present the following arguments;****
>>
>> ** **
>>
>> * Are there any other negative aspects about
>> using NDLB-connectile-dysfunction, other than it is a non compliant RFC
>> hack?****
>>
>> ** **
>>
>> * Why is NDLB-connectile-dysfunction not enabled by default when certain
>> conditions are met? In the event that FreeSWITCH receives a REGISTER from a
>> phone specifying a Contact/Via as 192.168.0.0/16, but received on a
>> public IP, then it should be obvious that NAT is broken and automatically
>> try to circumvent it.****
>>
>> ** **
>>
>> * People seem to get confused between server side and client side NAT
>> problems, and that they both need to be resolved in a different way. The
>> documentation doesn't seem to reflect this clearly.****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Dec 16 20:34:30 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Sun, 16 Dec 2012 12:34:30 -0500
Subject: [Freeswitch-users] LUA - Call a Function through a variable name
Message-ID: <CAGvuWYbg+3kc5t6DxzdB7POmTCYtKdL0oqp4qbw1iiEgPEB7Fg@mail.gmail.com>

Hello

Does lua support , Call a Function through a variable name

Ex:

function *playfile*()

end

variable_a = "*playfile*"

how to call the *variable_a* , that execute function?

Thanks
Lloyd
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From regis.freeswitch.org at tornad.net  Sun Dec 16 20:46:31 2012
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Sun, 16 Dec 2012 18:46:31 +0100
Subject: [Freeswitch-users] callcenter with external agents
In-Reply-To: <DUB002-W73F9E83A51542D9A07C2FFD94D0@phx.gbl>
References: <DUB002-W3915B9DAC4F64D9FD8AE3ED94E0@phx.gbl>
	<4521E504-D44A-47D9-B3EE-3EE363A74558@endigotech.com>
	<CAHWtr1RNJa2oJD67xtQYZvYPNdWDyehZ4QLpoMRUt3bwvo6kWA@mail.gmail.com>
	<1355489348.7162.28.camel@luna.madrid.commsmundi.com>
	<DUB002-W848BC9DC7196903EB35AE2D94D0@phx.gbl>
	<DUB002-W73F9E83A51542D9A07C2FFD94D0@phx.gbl>
Message-ID: <CAHWtr1TE8RM5Xa9adrJz1hfMty8U++56mbFOuV0zN85TK-ztUg@mail.gmail.com>

Hello.
As I said the first time, it seems that there's no agent in your queue.
Check the TIER config
In agent you put :
agent name="1000 at default"

But in TIER
tier agent="1000 at callcenter"

For me, it's not the same agent, Try to put agent at default in TIER config
party

Regards


2012/12/15 Juan Pablo L. <jpablolorenzetti at hotmail.com>

> I have change somethings around as suggested but still freeswitch does not
> even make the attempt to dial out, this is the config i m using
> http://pastebin.freeswitch.org/20326 and this is the console logs
> http://pastebin.freeswitch.org/20327.
>
> the name of the gateway i m trying to send the call to is huawei_csoft, in
> my original post i sent the name of the external profile whichis wrong but
> i was making lots of combinations trying to make it work .... thanks for
> any help ...
>
> ------------------------------
> From: jpablolorenzetti at hotmail.com
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 14 Dec 2012 22:46:39 +0000
>
> Subject: Re: [Freeswitch-users] callcenter with external agents
>
> Hi All, thank you very much for your answers, it is a relieve to know that
> what i m trying to accomplish is possible, i have not done it before so i m
> not sure what to expect, it is likely that i m configuring something wrong
> ..... i will try what was suggested in these responses and get back to you
> guys asap. thanks!!
>
> > From: fdelawarde at wirelessmundi.com
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Fri, 14 Dec 2012 13:49:08 +0100
> > Subject: Re: [Freeswitch-users] callcenter with external agents
> >
> > It should work fine, heck it even works with loopback channels for me!
> >
> > Just in case, verify that your dialstring is correct. If
> > "external-huawei_gw" is a gateway, use:
> >
> > sofia/gateway/external-huawei_gw/XXXXXX
> >
> > instead of:
> >
> > sofia/external-huawei_gw/XXXXXX
> >
> > Regards,
> > Fran?ois.
> >
> > On Fri, 2012-12-14 at 12:20 +0100, Regis M wrote:
> > > It works, we're doing it on a production system.
> > > The message seems to be more a agent State problem instead of gateway
> > > problem. Your agent doesn't seems to been "Waiting" or you don't
> > > correctly affect it in a tier with you queue. There's 2 agents thing
> > > to check Status and State. and Tier association
> > > Check the configuration by make call working with a "normal" sip agent
> > > and then, you could try by changing his contact parameter to gateway
> > > outside the fs callcenter box.
> > >
> > > Regards
> > >
> > >
> > > 2012/12/14 Brian Foster <bdfoster at endigotech.com>
> > > +1 I don't really know if this is possible. We've tried to do
> > > it this way but we ended up using some LUA scripts and
> > > mod_fifo. Unfortunately I can't release the scripts because
> > > the client won't allow me.
> > >
> > > Sent from my iPhone
> > >
> > > On Dec 13, 2012, at 1:31 PM, Juan Pablo L.
> > > <jpablolorenzetti at hotmail.com> wrote:
> > >
> > >
> > > > Hi, i m trying to set up a callcenter but the lines for the
> > > > agents are mobile phones in a mobile network and are not
> > > > attached to freeswitch, i m trying to set it up with the
> > > > following:
> > > >
> > > >
> > > > <agent name="1000 at default" type="callback"
> > > > contact="[call_timeout=10]sofia/external-huawei_gw/XXXXXX"
> > > > status="Available" max-no-answer="3" wrap-up-time="10"
> > > > reject-delay-time="10" busy-delay-time="60" />
> > > >
> > > >
> > > > XXXXX being the mobile number but i m getting the following
> > > > error:
> > > >
> > > >
> > > > Member 2018 <2018> in queue 'callcenter at default' reached max
> > > > wait of 0 sec. with no agent plus join grace period of 5
> > > > sec.
> > > >
> > > >
> > > > and i see that freeswitch does not even try to dial out to
> > > > the trunk.
> > > >
> > > >
> > > > i m wondering if configuring it the way i need it is even
> > > > possible as i think the problem may reside in the fact that
> > > > freeswitch does not actually see the agent registered.
> > > > 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
> > > >
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > > >
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting 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
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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 FreeSWITCH-powered IP PBX: The CudaTel
> Communication Server  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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cal.leeming at simplicitymedialtd.co.uk  Sun Dec 16 21:08:40 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 16 Dec 2012 18:08:40 +0000
Subject: [Freeswitch-users] IVR docs updated - playing remote audio files
Message-ID: <CAHKQagFOUc9_TK6g6h6SNVkt0w-ztuKw0hroZ3yJKfKXjL2JSg@mail.gmail.com>

Hello,

The following documentation has been updated;
http://wiki.freeswitch.org/wiki/IVR_Menu

This now includes instructions on the different ways of playing remote
audio files as a greeting.

Both 'mod_http_cache' and 'mod_shout' have been explained in detail, with
'mod_http_cache' being recommended as the most production suitable approach.

There does appear to be a bug with 'mod_shout' in which menu events are
delayed by exactly 1 second (this is confirmed to NOT be the same problem
as http://jira.freeswitch.org/browse/FS-4924 ), however given that
mod_shout is not production worthy, I have not spent any time investigating
this.

If anyone disagrees with this update or feels it needs further
clarification, please feel free to reply.

Cal
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From andrew at cassidywebservices.co.uk  Sun Dec 16 21:48:26 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sun, 16 Dec 2012 18:48:26 +0000
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CALNZuwnTOOavLzOrGVq+zw1CGj+HdkCamCUqnUo6hPsb-YeMJg@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<480f01cddbae$6d7d67e0$487837a0$@bizfocused.com>
	<CAHKQagGETUVwZ+4jV4mXzXt_P-yMFjFEfEY=1DQBf6i8Yo6mfg@mail.gmail.com>
	<CALNZuwnTOOavLzOrGVq+zw1CGj+HdkCamCUqnUo6hPsb-YeMJg@mail.gmail.com>
Message-ID: <CAMbW9o+1J2R2nnTaV6SBjQeYpf3=o2P0ra_Oo3M5Dku9QGw7_Q@mail.gmail.com>

In my experience there is no 'fix all' procedure, you just have to use sip
traces to diagnose individual setups to get around the problems. More often
than not I find that the NAT routers at the client end are causing the
problems, but different phone/router combinations produce different results.

In my current setup, I have freeswitch 1:1 NAT mapped behind pfSense (on
someone else's network at the moment) and with the ext-sip-ip and
ext-rtp-ip set to stun:stun.freeswitch.org and my phones at home not using
STUN behind an OpenWRT-based router, everything is working fine. I did have
to install the extra connection tracking modules onto my home router,
though.

But that's just one setup.

On 16 December 2012 17:33, Avi Marcus <avi at avimarcus.net> wrote:

> My main experience is with the Linksys/Cisco (sipura) SPA-2102 ATA.
>
> I always disable ALG in the router.
>
> I turn on NAT ping of 15 seconds in the Linksys.
> And.. here's the variable part - I also turn on 2-5 of the VIAs. I haven't
> really pinned that one down.
>
> This is not strictly NAT related... but has bit me a few times: devices by
> default want to use :5060 for their SIP. Not all are smart enough to see
> something else is using it and try a different port automatically.
>
> And for your peace of mind, try to never need NAT on the server.
>
> -Avi
>
>
> On Sun, Dec 16, 2012 at 7:15 PM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> Hi Sean,
>>
>> Thank you for the detailed reply.
>>
>> The more info we can get about individual NAT experiences, the better -
>> I'm hoping others will follow suit!
>>
>> Cal
>>
>> On Sun, Dec 16, 2012 at 4:57 PM, Sean Devoy <sdevoy at bizfocused.com>wrote:
>>
>>> I have spent many hours working on *NAT issues on client end*, my
>>> server has a public address.  ****
>>>
>>> ** **
>>>
>>> With CISCO brand phones I did not need any non-standards compliant
>>> settings, just turning on all the choices in the CISCO web setup NAT
>>> section.  However, with Polycom 335 phones (as of Dec 2012) I could not get
>>> registered or get audio without the following:****
>>>
>>> * NDLB-connectile-dysfunction****
>>>
>>> * NDLB-force-rport****
>>>
>>> * Enable SIP ALG on my FIOS router.****
>>>
>>> With those setting however, this has worked perfectly.  Also note that
>>> when I turned on SIP ALG, my Cisco phones quite working until I added the
>>> NDLB parameter/variable to the Cisco <user> in the directory.   They seem
>>> to be quite complimentary but seem be requirements for each other.****
>>>
>>> ** **
>>>
>>> I really tried to stay away from SIP ALG because so many posts were so
>>> negative about it.  Without the NDLB ?flags? I could never see any
>>> difference when enabling SIP ALG.  The combination for me has been
>>> fantastic.****
>>>
>>> ** **
>>>
>>> HOWEVER, since there are so many different versions of ?success? in the
>>> IRC and Wiki,  I am pretty sure that other router brands with different SIP
>>> ALG implementations and/or other phone brands or even firmware versions may
>>> need different configurations.  It is almost like we just need a checklist
>>> that says try these combinations until you find one that fits your site.
>>> ****
>>>
>>> ** **
>>>
>>> HTH,****
>>>
>>> sean****
>>>
>>> ** **
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal
>>> Leeming [Simplicity Media Ltd]
>>> *Sent:* Sunday, December 16, 2012 11:15 AM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* [Freeswitch-users] NAT traversal - the final say..!****
>>>
>>> ** **
>>>
>>> *Any and all feedback on this thread would be much welcomed.*****
>>>
>>> ** **
>>>
>>> Hello,****
>>>
>>> ** **
>>>
>>> There seems to be a large number of discussions surrounding NAT
>>> traversal, as well as lots of documentation, but with no concrete answers.
>>> ****
>>>
>>> ** **
>>>
>>> The NAT related wiki documentation is tedious, and depending on the
>>> outcome of this thread, I'd like to spend some time cleaning it up.****
>>>
>>> ** **
>>>
>>> The most common problem (the same as ours) was having a router with
>>> broken ALG and a softphone that does not seem to work with STUN.****
>>>
>>> ** **
>>>
>>> The following REGISTER is sent from a phone.****
>>>
>>> ** **
>>>
>>> REGISTER sip:1.2.3.4:5060 SIP/2.0****
>>>
>>> Via: SIP/2.0/UDP 192.168.1.102:57787
>>> ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport****
>>>
>>> Max-Forwards: 70****
>>>
>>> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513<https://mail.google.com/_/mail-static/_/js/main/m_i,t,it/rt=h/ver=dQ95YePrryI.en./sv=1/am=!rQczwx1unpD1BO2bNKsLUpXXtiIjaa01SgsJmP23wMtqPKKB37R0dPvFB_9tzlm4wJdbIQ/d=1>
>>> >****
>>>
>>> To: "foxx"<sip:2000 at 1.2.3.4:5060>****
>>>
>>> From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448****
>>>
>>> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.****
>>>
>>> CSeq: 7 REGISTER****
>>>
>>> Expires: 120****
>>>
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
>>> REFER, INFO, MESSAGE****
>>>
>>> Supported: replaces****
>>>
>>> User-Agent: 3CXPhone 6.0.25732.0****
>>>
>>> Content-Length: 0****
>>>
>>> ** **
>>>
>>> As you can see, the client's public IP is not specified
>>> anywhere. FreeSWITCH offers several ways around this, the main ones being;
>>> ****
>>>
>>> ** **
>>>
>>> * NDLB-connectile-dysfunction****
>>>
>>> * NDLB-force-rport****
>>>
>>> * apply-nat-acl****
>>>
>>> * sip-force-contact****
>>>
>>> ** **
>>>
>>> The one that has worked in our case was "NDLB-connectile-dysfunction"
>>> (otherwise known as NAT HACK), however there seems to be a lot of negative
>>> comments about using this.****
>>>
>>> ** **
>>>
>>> From what I can tell, the general argument is that NAT HACK is
>>> considered a non RFC compliant hack, and the SIP phones should be doing a
>>> better job of keeping to the RFCs.****
>>>
>>> ** **
>>>
>>> In principle, this is a fair argument - but in practise, it's not a
>>> reasonable assumption that all phones are RFC compliant, and (imho) not a
>>> reasonable argument to have this functionality disabled by default.****
>>>
>>> ** **
>>>
>>> So, I'd like to present the following arguments;****
>>>
>>> ** **
>>>
>>> * Are there any other negative aspects about
>>> using NDLB-connectile-dysfunction, other than it is a non compliant RFC
>>> hack?****
>>>
>>> ** **
>>>
>>> * Why is NDLB-connectile-dysfunction not enabled by default when certain
>>> conditions are met? In the event that FreeSWITCH receives a REGISTER from a
>>> phone specifying a Contact/Via as 192.168.0.0/16, but received on a
>>> public IP, then it should be obvious that NAT is broken and automatically
>>> try to circumvent it.****
>>>
>>> ** **
>>>
>>> * People seem to get confused between server side and client side NAT
>>> problems, and that they both need to be resolved in a different way. The
>>> documentation doesn't seem to reflect this clearly.****
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
*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 a.venugopan at mundio.com  Sun Dec 16 22:27:19 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Sun, 16 Dec 2012 19:27:19 +0000
Subject: [Freeswitch-users] latest git version error
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233BB3A@Mail-Kilo.squay.com>

Hi,
When I try to fetch latest git version am facing with the below error and am not able to get the git. From where can I get freeswitch latest version?

[root at VECTONE-CLOUDE src]# git clone git://git.freeswitch.org/freeswitch.git
Initialized empty Git repository in /usr/local/src/freeswitch/.git/
git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out
git.freeswitch.org[0: 2606:d900:0:24:1024:ff:fe00:1234]: errno=Network is unreachable
fatal: unable to connect a socket (Network is unreachable)

Regards,
Archana.V

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From vbvbrj at gmail.com  Sun Dec 16 22:39:11 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Sun, 16 Dec 2012 21:39:11 +0200
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CALNZuwnTOOavLzOrGVq+zw1CGj+HdkCamCUqnUo6hPsb-YeMJg@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<480f01cddbae$6d7d67e0$487837a0$@bizfocused.com>
	<CAHKQagGETUVwZ+4jV4mXzXt_P-yMFjFEfEY=1DQBf6i8Yo6mfg@mail.gmail.com>
	<CALNZuwnTOOavLzOrGVq+zw1CGj+HdkCamCUqnUo6hPsb-YeMJg@mail.gmail.com>
Message-ID: <50CE235F.2050500@gmail.com>

On 16.12.2012 19:33, Avi Marcus wrote:
> And for your peace of mind, try to never need NAT on the server.
>
I've moved FS to public IP, but run into another issue on using multiple 
registration for same user when two phones connect to pubic IP and 
internal IP. Putting two sip profiles to use same alias domain end up 
only the last (in alphabetical order) is aliased. So when calling user, 
only phones connected to public IP in my case are called.

-- 
Mimiko desu.


From jnvines at gmail.com  Sun Dec 16 22:49:05 2012
From: jnvines at gmail.com (Nick Vines)
Date: Sun, 16 Dec 2012 14:49:05 -0500
Subject: [Freeswitch-users] latest git version error
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233BB3A@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233BB3A@Mail-Kilo.squay.com>
Message-ID: <CA+NjZ2vwm3iccENBwapU3vPu5hhHKGsj-xn7Yg4vSdaUp0JqRg@mail.gmail.com>

Just tested and it works for me. Just making sure, does your FS server have
an internet connection?




On Sun, Dec 16, 2012 at 2:27 PM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi,****
>
> When I try to fetch latest git version am facing with the below error and
> am not able to get the git. From where can I get freeswitch latest version?
> ****
>
> ** **
>
> [root at VECTONE-CLOUDE src]# git clone git://
> git.freeswitch.org/freeswitch.git****
>
> Initialized empty Git repository in /usr/local/src/freeswitch/.git/****
>
> git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out****
>
> git.freeswitch.org[0: 2606:d900:0:24:1024:ff:fe00:1234]: errno=Network is
> unreachable****
>
> fatal: unable to connect a socket (Network is unreachable)****
>
> ** **
>
> Regards,****
>
> Archana.V****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 william.king at quentustech.com  Sun Dec 16 22:52:43 2012
From: william.king at quentustech.com (William King)
Date: Sun, 16 Dec 2012 11:52:43 -0800
Subject: [Freeswitch-users] IVR docs updated - playing remote audio files
In-Reply-To: <CAHKQagFOUc9_TK6g6h6SNVkt0w-ztuKw0hroZ3yJKfKXjL2JSg@mail.gmail.com>
References: <CAHKQagFOUc9_TK6g6h6SNVkt0w-ztuKw0hroZ3yJKfKXjL2JSg@mail.gmail.com>
Message-ID: <50CE268B.9070005@quentustech.com>

Have you tried using mod_vlc for playing the remote files?

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 12/16/2012 10:08 AM, Cal Leeming [Simplicity Media Ltd] wrote:
> Hello,
> 
> The following documentation has been updated;
> http://wiki.freeswitch.org/wiki/IVR_Menu
> 
> This now includes instructions on the different ways of playing remote
> audio files as a greeting.
> 
> Both 'mod_http_cache' and 'mod_shout' have been explained in detail,
> with 'mod_http_cache' being recommended as the most production suitable
> approach.
> 
> There does appear to be a bug with 'mod_shout' in which menu events are
> delayed by exactly 1 second (this is confirmed to NOT be the same
> problem as http://jira.freeswitch.org/browse/FS-4924 ), however given
> that mod_shout is not production worthy, I have not spent any time
> investigating this.
> 
> If anyone disagrees with this update or feels it needs further
> clarification, please feel free to reply.
> 
> Cal
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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


From steveayre at gmail.com  Sun Dec 16 22:53:25 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Dec 2012 19:53:25 +0000
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
Message-ID: <CAFiqYu=wL6+yk67AzzfF_HKrMBBreqeb5jybX7r8LjoXBHVs0A@mail.gmail.com>

"There seems to be a large number of discussions surrounding NAT traversal,
as well as lots of documentation, but with no concrete answers. "

Part of the problem is that:

   - Not all NAT implementations function in the same way (eg some rewrite
   ports others do not)
   - Not all SIP ALG implementations work the same/work
   - Not all clients handle NAT in the same way
   - You can encounter other odd situations such as double NAT that further
   complicate matters

So what works in one case might not work in another, so it's hard to give a
concrete 'this is how to do it' that'll work in all cases. And often that
means you need to find a failing client first then put in a NDLB workaround
for that specific client. You could enable them by default, but that then
can cause problems in other cases where the clients handle NAT correctly.

Roll on NAT-less IPv6 for true end-to-end connectivity*. :o)

-Steve



On 16 December 2012 16:15, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> *Any and all feedback on this thread would be much welcomed.*
>
> Hello,
> *
> *
> There seems to be a large number of discussions surrounding NAT traversal,
> as well as lots of documentation, but with no concrete answers.
>
> The NAT related wiki documentation is tedious, and depending on the
> outcome of this thread, I'd like to spend some time cleaning it up.
>
> The most common problem (the same as ours) was having a router with broken
> ALG and a softphone that does not seem to work with STUN.
>
> The following REGISTER is sent from a phone.
>
> REGISTER sip:1.2.3.4:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.102:57787
> ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport
> Max-Forwards: 70
> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>
> To: "foxx"<sip:2000 at 1.2.3.4:5060>
> From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448
> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.
> CSeq: 7 REGISTER
> Expires: 120
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
> REFER, INFO, MESSAGE
> Supported: replaces
> User-Agent: 3CXPhone 6.0.25732.0
> Content-Length: 0
>
> As you can see, the client's public IP is not specified
> anywhere. FreeSWITCH offers several ways around this, the main ones being;
>
> * NDLB-connectile-dysfunction
> * NDLB-force-rport
> * apply-nat-acl
> * sip-force-contact
>
> The one that has worked in our case was "NDLB-connectile-dysfunction"
> (otherwise known as NAT HACK), however there seems to be a lot of negative
> comments about using this.
>
> From what I can tell, the general argument is that NAT HACK is considered
> a non RFC compliant hack, and the SIP phones should be doing a better job
> of keeping to the RFCs.
>
> In principle, this is a fair argument - but in practise, it's not a
> reasonable assumption that all phones are RFC compliant, and (imho) not a
> reasonable argument to have this functionality disabled by default.
>
> So, I'd like to present the following arguments;
>
> * Are there any other negative aspects about
> using NDLB-connectile-dysfunction, other than it is a non compliant RFC
> hack?
>
> * Why is NDLB-connectile-dysfunction not enabled by default when certain
> conditions are met? In the event that FreeSWITCH receives a REGISTER from a
> phone specifying a Contact/Via as 192.168.0.0/16, but received on a
> public IP, then it should be obvious that NAT is broken and automatically
> try to circumvent it.
>
> * People seem to get confused between server side and client side NAT
> problems, and that they both need to be resolved in a different way. The
> documentation doesn't seem to reflect this clearly.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 william.king at quentustech.com  Sun Dec 16 22:55:06 2012
From: william.king at quentustech.com (William King)
Date: Sun, 16 Dec 2012 11:55:06 -0800
Subject: [Freeswitch-users] latest git version error
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233BB3A@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233BB3A@Mail-Kilo.squay.com>
Message-ID: <50CE271A.6090701@quentustech.com>

Your machine is trying to connect to the git server over ipv6. Currently
the ipv6 route is still down. Try the ipv4 route.

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 12/16/2012 11:27 AM, Archana Venugopan wrote:
> Hi,
> 
> When I try to fetch latest git version am facing with the below error
> and am not able to get the git. From where can I get freeswitch latest
> version?
> 
>  
> 
> [root at VECTONE-CLOUDE src]# git clone git://git.freeswitch.org/freeswitch.git
> 
> Initialized empty Git repository in /usr/local/src/freeswitch/.git/
> 
> git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out
> 
> git.freeswitch.org[0: 2606:d900:0:24:1024:ff:fe00:1234]: errno=Network
> is unreachable
> 
> fatal: unable to connect a socket (Network is unreachable)
> 
>  
> 
> Regards,
> 
> Archana.V
> 
>  
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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


From steveayre at gmail.com  Sun Dec 16 23:05:10 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Dec 2012 20:05:10 +0000
Subject: [Freeswitch-users] G729 - [INCOMPATIBLE_DESTINATION]
In-Reply-To: <CAP2fbbcixeTy4H+KPbxJQvAhCDWmco5a0jF0XQkGHBtE2Qkmdg@mail.gmail.com>
References: <CAP2fbbcixeTy4H+KPbxJQvAhCDWmco5a0jF0XQkGHBtE2Qkmdg@mail.gmail.com>
Message-ID: <CAFiqYuk+OwUVPCmy8H1Yq5hW-+Hhy40DS1RdRe8T6oBTjeY2LA@mail.gmail.com>

What G729 module are you using, and what codec is in use on the other leg?

G729 needs licenses due to patents.

mod_g729 is passthrough only but requires no licenses. This is because it
merely forwards the data, but doesn't perform the encoding/transcoding step
that the patents cover. But that means it can't transcode between different
codecs.
mod_com_g729 is the licensed version: http://www.freeswitch.org/node/235

With the passthrough mod_g729 codec, if the other leg has selected a codec
other than g729 you'll see an error in your logs and it'll hangup with that
reason.

You can tweak your codec negotiation to avoid this (eg
late-negotiation=true), or you can use the licensed version.




On 16 December 2012 11:40, Balamurugan Mahendran <b2m at a-cti.com> wrote:

> Need help on Codec Negotiation
>
>
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare *
> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]*
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
> send/recv payload to 101
>
>
> 2012-12-16 11:35:52.246422 [DEBUG] switch_channel.c:2852 (sofia/internal/
> 501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
> 2012-12-16 11:35:52.246422 [NOTICE] sofia.c:5781 Hangup sofia/internal/
> 501 at 50.54.12.39 [CS_NEW] *[INCOMPATIBLE_DESTINATION]*
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Dec 16 23:06:34 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Dec 2012 20:06:34 +0000
Subject: [Freeswitch-users] PostgreSQL in the core
In-Reply-To: <CALFFxt2BiqKGZcHTARakahGUQRe4fsPZBwjKoQ=E4+-JJAFevg@mail.gmail.com>
References: <CALFFxt13ePTSk_GFnAvK6GD7tRo2qn1mv3irsE+d1ZSu=SH08A@mail.gmail.com>
	<B89E0118B7FD4A9CA453643D5B1675B9@gmail.com>
	<CALFFxt3zWP22vYJMPiLuvQzfj9aqwJ+MeCZvH2efV4QWtAfQ5Q@mail.gmail.com>
	<20F2980066FC4F198FF78D66271ECF47@gmail.com>
	<CALFFxt2BiqKGZcHTARakahGUQRe4fsPZBwjKoQ=E4+-JJAFevg@mail.gmail.com>
Message-ID: <CAFiqYumNjr5a894+i0DmhZ7gS1Zv0tCha-j=XWP5finHXt81jA@mail.gmail.com>

Did you enable it in configure with the --enable-core-pgsql-support option?



On 16 December 2012 14:08, Ahmed Sboor <ahmed at netelsat.net> wrote:

> Even removing that doesn't work.
> Any one using new Postgresql in core without ODBC off course  ?
>
>
> On Sat, Dec 15, 2012 at 4:03 PM, Seven Du <dujinfang at gmail.com> wrote:
>
>> I would try also remove options.
>>
>> --
>> Seven Du
>> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>>
>> On Saturday, December 15, 2012 at 10:33 AM, Ahmed Sboor wrote:
>>
>> Did that , then it says invalid option " client_min_messages"
>>
>> On Sat, Dec 15, 2012 at 7:21 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> Try to remove the application_name option ?
>>
>> --
>> Seven Du
>> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>>
>> On Saturday, December 15, 2012 at 7:52 AM, Ahmed Sboor wrote:
>>
>> Hi all,
>> i am just trying to use 1.2.5.3 to use Postgresql as core.
>> Followed by http://wiki.freeswitch.org/wiki/PostgreSQL_in_the_core
>> but its giving error like :
>>
>> switch_pgsql.c:492 invalid connection option "application_name"
>>
>> and
>>
>> [CRIT] switch_core_sqldb.c:504 Failure to connect to PGSQL
>> hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password=''
>> options='-c client_min_messages=NOTICE' application_name='freeswitch'!
>>
>>
>> Postgresql version is 8.4.13.
>>
>> can any one please help where i am missing something ?
>>
>> Thanking you all
>> Ahmed
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 ahmed at netelsat.net  Sun Dec 16 23:11:43 2012
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Mon, 17 Dec 2012 01:11:43 +0500
Subject: [Freeswitch-users] PostgreSQL in the core
In-Reply-To: <CAFiqYumNjr5a894+i0DmhZ7gS1Zv0tCha-j=XWP5finHXt81jA@mail.gmail.com>
References: <CALFFxt13ePTSk_GFnAvK6GD7tRo2qn1mv3irsE+d1ZSu=SH08A@mail.gmail.com>
	<B89E0118B7FD4A9CA453643D5B1675B9@gmail.com>
	<CALFFxt3zWP22vYJMPiLuvQzfj9aqwJ+MeCZvH2efV4QWtAfQ5Q@mail.gmail.com>
	<20F2980066FC4F198FF78D66271ECF47@gmail.com>
	<CALFFxt2BiqKGZcHTARakahGUQRe4fsPZBwjKoQ=E4+-JJAFevg@mail.gmail.com>
	<CAFiqYumNjr5a894+i0DmhZ7gS1Zv0tCha-j=XWP5finHXt81jA@mail.gmail.com>
Message-ID: <CALFFxt3qZHWE3BzzMJQ8z3XxyHMr5c2MEoHj8db-=NWE9XaXcQ@mail.gmail.com>

Yes , I did exactly as mentioned on
http://wiki.freeswitch.org/wiki/PostgreSQL_in_the_core

./configure --enable-core-pgsql-support

And there is no error in compilation .


On Mon, Dec 17, 2012 at 1:06 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Did you enable it in configure with the --enable-core-pgsql-support option?
>
>
>
> On 16 December 2012 14:08, Ahmed Sboor <ahmed at netelsat.net> wrote:
>
>> Even removing that doesn't work.
>> Any one using new Postgresql in core without ODBC off course  ?
>>
>>
>> On Sat, Dec 15, 2012 at 4:03 PM, Seven Du <dujinfang at gmail.com> wrote:
>>
>>> I would try also remove options.
>>>
>>> --
>>> Seven Du
>>> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>>>
>>> On Saturday, December 15, 2012 at 10:33 AM, Ahmed Sboor wrote:
>>>
>>> Did that , then it says invalid option " client_min_messages"
>>>
>>> On Sat, Dec 15, 2012 at 7:21 AM, Seven Du <dujinfang at gmail.com> wrote:
>>>
>>> Try to remove the application_name option ?
>>>
>>> --
>>> Seven Du
>>> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>>>
>>> On Saturday, December 15, 2012 at 7:52 AM, Ahmed Sboor wrote:
>>>
>>> Hi all,
>>> i am just trying to use 1.2.5.3 to use Postgresql as core.
>>> Followed by http://wiki.freeswitch.org/wiki/PostgreSQL_in_the_core
>>> but its giving error like :
>>>
>>> switch_pgsql.c:492 invalid connection option "application_name"
>>>
>>> and
>>>
>>> [CRIT] switch_core_sqldb.c:504 Failure to connect to PGSQL
>>> hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password=''
>>> options='-c client_min_messages=NOTICE' application_name='freeswitch'!
>>>
>>>
>>> Postgresql version is 8.4.13.
>>>
>>> can any one please help where i am missing something ?
>>>
>>> Thanking you all
>>> Ahmed
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.venugopan at mundio.com  Sun Dec 16 23:55:26 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Sun, 16 Dec 2012 20:55:26 +0000
Subject: [Freeswitch-users] ODBC error
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233BB74@Mail-Kilo.squay.com>

Hi,
Can anyone please tell me why I am getting this error inspite 'isql -v smepbx smepbx smeswitch' was working.


[CRIT] switch_core_sqldb.c:433 Failure! ODBC NOT AVAILABLE! Can't connect to DSN smepbx

Regards,
Archana.V

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From curriegrad2004 at gmail.com  Sun Dec 16 23:55:34 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 16 Dec 2012 12:55:34 -0800
Subject: [Freeswitch-users] G729 - [INCOMPATIBLE_DESTINATION]
In-Reply-To: <CAFiqYuk+OwUVPCmy8H1Yq5hW-+Hhy40DS1RdRe8T6oBTjeY2LA@mail.gmail.com>
References: <CAP2fbbcixeTy4H+KPbxJQvAhCDWmco5a0jF0XQkGHBtE2Qkmdg@mail.gmail.com>
	<CAFiqYuk+OwUVPCmy8H1Yq5hW-+Hhy40DS1RdRe8T6oBTjeY2LA@mail.gmail.com>
Message-ID: <CAMWiMy_0f44SwcXZR4CQgP7=RBqrTYW8fQQhAa8m1nR+kKv=oQ@mail.gmail.com>

If he is using mod_g729 then the log would throw an error saying that
the codec only does passthrough mode, however he hasn't shown that
part of the log to us yet

On Sun, Dec 16, 2012 at 12:05 PM, Steven Ayre <steveayre at gmail.com> wrote:
> What G729 module are you using, and what codec is in use on the other leg?
>
> G729 needs licenses due to patents.
>
> mod_g729 is passthrough only but requires no licenses. This is because it
> merely forwards the data, but doesn't perform the encoding/transcoding step
> that the patents cover. But that means it can't transcode between different
> codecs.
> mod_com_g729 is the licensed version: http://www.freeswitch.org/node/235
>
> With the passthrough mod_g729 codec, if the other leg has selected a codec
> other than g729 you'll see an error in your logs and it'll hangup with that
> reason.
>
> You can tweak your codec negotiation to avoid this (eg
> late-negotiation=true), or you can use the licensed version.
>
>
>
>
> On 16 December 2012 11:40, Balamurugan Mahendran <b2m at a-cti.com> wrote:
>>
>> Need help on Codec Negotiation
>>
>>
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
>> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
>> send/recv payload to 101
>>
>>
>> 2012-12-16 11:35:52.246422 [DEBUG] switch_channel.c:2852
>> (sofia/internal/501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
>> 2012-12-16 11:35:52.246422 [NOTICE] sofia.c:5781 Hangup
>> sofia/internal/501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>


From cal.leeming at simplicitymedialtd.co.uk  Mon Dec 17 00:12:15 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 16 Dec 2012 21:12:15 +0000
Subject: [Freeswitch-users] IVR docs updated - playing remote audio files
In-Reply-To: <50CE268B.9070005@quentustech.com>
References: <CAHKQagFOUc9_TK6g6h6SNVkt0w-ztuKw0hroZ3yJKfKXjL2JSg@mail.gmail.com>
	<50CE268B.9070005@quentustech.com>
Message-ID: <CAHKQagGaPXoY0tPD_azpAPv_fJNWO5d2EBfbX0eVkW22=2oo6g@mail.gmail.com>

I haven't tried using mod_vlc yet.

On first looks, it seems this would fall into the same category as
mod_shout, in terms of its production worthy-ness - any thoughts?

Cal

On Sun, Dec 16, 2012 at 7:52 PM, William King
<william.king at quentustech.com>wrote:

> mod_vlc
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From cal.leeming at simplicitymedialtd.co.uk  Mon Dec 17 00:21:14 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 16 Dec 2012 21:21:14 +0000
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAFiqYu=wL6+yk67AzzfF_HKrMBBreqeb5jybX7r8LjoXBHVs0A@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<CAFiqYu=wL6+yk67AzzfF_HKrMBBreqeb5jybX7r8LjoXBHVs0A@mail.gmail.com>
Message-ID: <CAHKQagFbEdS+wLt-CH7t_=B5UESRQkVVnFf3HSt8EB4=Gq=pOA@mail.gmail.com>

It seems that most of us agree there is no single answer to fix NAT
problems - the double-NAT is something I hadn't thought about.

Sean mentioned having a checklist of approaches, which would be good
addition for the documentation fix.

I agree that enabling NAT HACK by default could break clients that are
functioning normally, but not if it is only enabled automatically under
certain conditions (described in the original email).

However - the upside is that it gives another layer of "this just works"..
the downside is that it gives users a reason to not bother looking at why
their NAT is broken in the first place.

With that in mind, I'm thinking that just a documentation fix is the answer
here, to avoid user lazyness.

Cal

On Sun, Dec 16, 2012 at 7:53 PM, Steven Ayre <steveayre at gmail.com> wrote:

> "There seems to be a large number of discussions surrounding NAT
> traversal, as well as lots of documentation, but with no concrete answers. "
>
> Part of the problem is that:
>
>    - Not all NAT implementations function in the same way (eg some
>    rewrite ports others do not)
>    - Not all SIP ALG implementations work the same/work
>    - Not all clients handle NAT in the same way
>    - You can encounter other odd situations such as double NAT that
>    further complicate matters
>
> So what works in one case might not work in another, so it's hard to give
> a concrete 'this is how to do it' that'll work in all cases. And often that
> means you need to find a failing client first then put in a NDLB workaround
> for that specific client. You could enable them by default, but that then
> can cause problems in other cases where the clients handle NAT correctly.
>
> Roll on NAT-less IPv6 for true end-to-end connectivity*. :o)
>
> -Steve
>
>
>
> On 16 December 2012 16:15, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> *Any and all feedback on this thread would be much welcomed.*
>>
>> Hello,
>> *
>> *
>> There seems to be a large number of discussions surrounding NAT
>> traversal, as well as lots of documentation, but with no concrete answers.
>>
>> The NAT related wiki documentation is tedious, and depending on the
>> outcome of this thread, I'd like to spend some time cleaning it up.
>>
>> The most common problem (the same as ours) was having a router with
>> broken ALG and a softphone that does not seem to work with STUN.
>>
>> The following REGISTER is sent from a phone.
>>
>> REGISTER sip:1.2.3.4:5060 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.1.102:57787
>> ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport
>> Max-Forwards: 70
>> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>
>> To: "foxx"<sip:2000 at 1.2.3.4:5060>
>> From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448
>> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.
>> CSeq: 7 REGISTER
>> Expires: 120
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
>> REFER, INFO, MESSAGE
>> Supported: replaces
>> User-Agent: 3CXPhone 6.0.25732.0
>> Content-Length: 0
>>
>> As you can see, the client's public IP is not specified
>> anywhere. FreeSWITCH offers several ways around this, the main ones being;
>>
>> * NDLB-connectile-dysfunction
>> * NDLB-force-rport
>> * apply-nat-acl
>> * sip-force-contact
>>
>> The one that has worked in our case was "NDLB-connectile-dysfunction"
>> (otherwise known as NAT HACK), however there seems to be a lot of negative
>> comments about using this.
>>
>> From what I can tell, the general argument is that NAT HACK is considered
>> a non RFC compliant hack, and the SIP phones should be doing a better job
>> of keeping to the RFCs.
>>
>> In principle, this is a fair argument - but in practise, it's not a
>> reasonable assumption that all phones are RFC compliant, and (imho) not a
>> reasonable argument to have this functionality disabled by default.
>>
>> So, I'd like to present the following arguments;
>>
>> * Are there any other negative aspects about
>> using NDLB-connectile-dysfunction, other than it is a non compliant RFC
>> hack?
>>
>> * Why is NDLB-connectile-dysfunction not enabled by default when certain
>> conditions are met? In the event that FreeSWITCH receives a REGISTER from a
>> phone specifying a Contact/Via as 192.168.0.0/16, but received on a
>> public IP, then it should be obvious that NAT is broken and automatically
>> try to circumvent it.
>>
>> * People seem to get confused between server side and client side NAT
>> problems, and that they both need to be resolved in a different way. The
>> documentation doesn't seem to reflect this clearly.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Dec 17 00:34:11 2012
From: william.king at quentustech.com (William King)
Date: Sun, 16 Dec 2012 13:34:11 -0800
Subject: [Freeswitch-users] IVR docs updated - playing remote audio files
In-Reply-To: <CAHKQagGaPXoY0tPD_azpAPv_fJNWO5d2EBfbX0eVkW22=2oo6g@mail.gmail.com>
References: <CAHKQagFOUc9_TK6g6h6SNVkt0w-ztuKw0hroZ3yJKfKXjL2JSg@mail.gmail.com>
	<50CE268B.9070005@quentustech.com>
	<CAHKQagGaPXoY0tPD_azpAPv_fJNWO5d2EBfbX0eVkW22=2oo6g@mail.gmail.com>
Message-ID: <50CE3E53.7000603@quentustech.com>

libvlc in general is considered very production stable. The current
implementation of mod_vlc may still have some tweaks required to perform
to the expectations of the telephony environment. I'd suggest testing it
with your requirements and provide feedback on how it performs.

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 12/16/2012 01:12 PM, Cal Leeming [Simplicity Media Ltd] wrote:
> I haven't tried using mod_vlc yet.
> 
> On first looks, it seems this would fall into the same category as
> mod_shout, in terms of its production worthy-ness - any thoughts?
> 
> Cal
> 
> On Sun, Dec 16, 2012 at 7:52 PM, William King
> <william.king at quentustech.com <mailto:william.king at quentustech.com>> wrote:
> 
>     mod_vlc
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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


From jmesquita at freeswitch.org  Mon Dec 17 00:52:18 2012
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 16 Dec 2012 18:52:18 -0300
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAHKQagFbEdS+wLt-CH7t_=B5UESRQkVVnFf3HSt8EB4=Gq=pOA@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<CAFiqYu=wL6+yk67AzzfF_HKrMBBreqeb5jybX7r8LjoXBHVs0A@mail.gmail.com>
	<CAHKQagFbEdS+wLt-CH7t_=B5UESRQkVVnFf3HSt8EB4=Gq=pOA@mail.gmail.com>
Message-ID: <CABH8QX0WNTaMHXbFeJ8ip+D7ADn+b45nc2=fSHdf5F8OpWzMNQ@mail.gmail.com>

My take on the subject is that we are trying to tackle 2 different problems
at once.

On one hand we have how NAT works and what problems it create on a VoIP
world. I really was not able to find any condensed documentation on the
internet that would describe with VoIP in mind what a admin need to know
about SIP packets and NAT handling. If there was one, we could just add a
link to the wiki page. Since NAT has no "one configuration" fix, user NEEDS
to know about it in order to fix his own scenario.

On the other hand, we have how FS particularly deals with NAT (client and
server side). I think there is more documentation to be added/created on
that end as well. NAT is a complicated matter indeed and understanding
Sofia profiles alone is a challenge. Add NAT to the mix and the
configuration can look like black magic.

So, question is, who can elaborate/contribute/find a definitive guide to
NAT so we can lecture users and who knows all about the NAT handling
internals so we can document each and every option available?

Jo?o Mesquita



On Sun, Dec 16, 2012 at 6:21 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> It seems that most of us agree there is no single answer to fix NAT
> problems - the double-NAT is something I hadn't thought about.
>
> Sean mentioned having a checklist of approaches, which would be good
> addition for the documentation fix.
>
> I agree that enabling NAT HACK by default could break clients that are
> functioning normally, but not if it is only enabled automatically under
> certain conditions (described in the original email).
>
> However - the upside is that it gives another layer of "this just works"..
> the downside is that it gives users a reason to not bother looking at why
> their NAT is broken in the first place.
>
> With that in mind, I'm thinking that just a documentation fix is the
> answer here, to avoid user lazyness.
>
> Cal
>
> On Sun, Dec 16, 2012 at 7:53 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> "There seems to be a large number of discussions surrounding NAT
>> traversal, as well as lots of documentation, but with no concrete answers. "
>>
>> Part of the problem is that:
>>
>>    - Not all NAT implementations function in the same way (eg some
>>    rewrite ports others do not)
>>    - Not all SIP ALG implementations work the same/work
>>    - Not all clients handle NAT in the same way
>>    - You can encounter other odd situations such as double NAT that
>>    further complicate matters
>>
>> So what works in one case might not work in another, so it's hard to give
>> a concrete 'this is how to do it' that'll work in all cases. And often that
>> means you need to find a failing client first then put in a NDLB workaround
>> for that specific client. You could enable them by default, but that then
>> can cause problems in other cases where the clients handle NAT correctly.
>>
>> Roll on NAT-less IPv6 for true end-to-end connectivity*. :o)
>>
>> -Steve
>>
>>
>>
>> On 16 December 2012 16:15, Cal Leeming [Simplicity Media Ltd] <
>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>
>>> *Any and all feedback on this thread would be much welcomed.*
>>>
>>> Hello,
>>> *
>>> *
>>> There seems to be a large number of discussions surrounding NAT
>>> traversal, as well as lots of documentation, but with no concrete answers.
>>>
>>> The NAT related wiki documentation is tedious, and depending on the
>>> outcome of this thread, I'd like to spend some time cleaning it up.
>>>
>>> The most common problem (the same as ours) was having a router with
>>> broken ALG and a softphone that does not seem to work with STUN.
>>>
>>> The following REGISTER is sent from a phone.
>>>
>>> REGISTER sip:1.2.3.4:5060 SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.1.102:57787
>>> ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport
>>> Max-Forwards: 70
>>> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>
>>> To: "foxx"<sip:2000 at 1.2.3.4:5060>
>>> From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448
>>> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.
>>> CSeq: 7 REGISTER
>>> Expires: 120
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
>>> REFER, INFO, MESSAGE
>>> Supported: replaces
>>> User-Agent: 3CXPhone 6.0.25732.0
>>> Content-Length: 0
>>>
>>> As you can see, the client's public IP is not specified
>>> anywhere. FreeSWITCH offers several ways around this, the main ones being;
>>>
>>> * NDLB-connectile-dysfunction
>>> * NDLB-force-rport
>>> * apply-nat-acl
>>> * sip-force-contact
>>>
>>> The one that has worked in our case was "NDLB-connectile-dysfunction"
>>> (otherwise known as NAT HACK), however there seems to be a lot of negative
>>> comments about using this.
>>>
>>> From what I can tell, the general argument is that NAT HACK is
>>> considered a non RFC compliant hack, and the SIP phones should be doing a
>>> better job of keeping to the RFCs.
>>>
>>> In principle, this is a fair argument - but in practise, it's not a
>>> reasonable assumption that all phones are RFC compliant, and (imho) not a
>>> reasonable argument to have this functionality disabled by default.
>>>
>>> So, I'd like to present the following arguments;
>>>
>>> * Are there any other negative aspects about
>>> using NDLB-connectile-dysfunction, other than it is a non compliant RFC
>>> hack?
>>>
>>> * Why is NDLB-connectile-dysfunction not enabled by default when certain
>>> conditions are met? In the event that FreeSWITCH receives a REGISTER from a
>>> phone specifying a Contact/Via as 192.168.0.0/16, but received on a
>>> public IP, then it should be obvious that NAT is broken and automatically
>>> try to circumvent it.
>>>
>>> * People seem to get confused between server side and client side NAT
>>> problems, and that they both need to be resolved in a different way. The
>>> documentation doesn't seem to reflect this clearly.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 avi at avimarcus.net  Mon Dec 17 00:53:53 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 16 Dec 2012 23:53:53 +0200
Subject: [Freeswitch-users] What is "Recovery On Timer Expire"?
Message-ID: <CALNZuw=z2ANmWmB6Fz3nximTyoPqUSWLJ=-ob55gdT6Q=_qC4Q@mail.gmail.com>

I'm trying to clear up my CDRs from errors, if possible... I see a bunch
of Recovery On Timer Expire and they mostly seem to be after 45 seconds of
ring time (there is progress).
Does that basically mean it rang for 40+ seconds before some timer
somewhere kicked in and ended the call?
So it's basically a "no answer" or "no user response"?

Thanks,
-Avi
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From jmesquita at freeswitch.org  Mon Dec 17 00:57:56 2012
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 16 Dec 2012 18:57:56 -0300
Subject: [Freeswitch-users] Compiling FSComm
In-Reply-To: <1355222761.7628.21.camel@marces.madrid.commsmundi.com>
References: <1354878605.5967.44.camel@vmmarces.vm.marces.com>
	<21EB7A95-BDA4-48FE-A944-DFB5662BBB14@freeswitch.org>
	<1355222761.7628.21.camel@marces.madrid.commsmundi.com>
Message-ID: <CABH8QX3kbZQZDo4_H_pwLPgS3d_31sb4RoGjS5gWT03VzxdFfg@mail.gmail.com>

Sorry Antonio, I was on 4K Conference these last 2 weeks and had no time to
look into it. I will update the ticket this week.

Jo?o Mesquita
FreeSWITCH? Solutions



On Tue, Dec 11, 2012 at 7:46 AM, Antonio <asilva at wirelessmundi.com> wrote:

> **
> Hi Jo?o,
>
> I just open you the jira, http://jira.freeswitch.org/browse/FSCOMM-11
>
>
> Thanks,
> Ant?nio
>
>
>
>
>
> On Sun, 2012-12-09 at 02:22 -0200, Jo?o Mesquita wrote:
>
> Antonio, I am the dev of FSComm and I am glad to hear there is still interest on it. I will try to fix the problem tomorrow and if not Monday as I am out of the country.
>
> Please help us out and file a Jira for it?
>
> Sent from my iPhone
>
> On Dec 7, 2012, at 9:10 AM, Antonio Silva <asilva at wirelessmundi.com> wrote:
>
> > Hi,
> >
> > i'm trying to compile fscomm but i have the following errors:
> >
> > "
> > freeswitch-git/fscomm# qmake
> > freeswitch-git/fscomm# make
> > /usr/bin/uic-qt4 mainwindow.ui -o ui_mainwindow.h
> > /usr/bin/uic-qt4 preferences/prefdialog.ui -o ui_prefdialog.h
> > /usr/bin/uic-qt4 preferences/accountdialog.ui -o ui_accountdialog.h
> > /usr/bin/uic-qt4 widgets/codecwidget.ui -o ui_codecwidget.h
> > /usr/bin/uic-qt4 debugtools/consolewindow.ui -o ui_consolewindow.h
> > /usr/bin/uic-qt4 debugtools/statedebugdialog.ui -o ui_statedebugdialog.h
> > Warning: name layoutWidget is already used
> > Warning: name layoutWidget is already used
> > g++ -c -pipe -O2 -Wall -W -D_REENTRANT -DQT_NO_DEBUG -DQT_XML_LIB -DQT_GUI_LIB -DQT_CORE_LIB -DQT_SHARED -I/usr/share/qt4/mkspecs/linux-g++ -I. -I/usr/include/qt4/QtCore -I/usr/include/qt4/QtGui -I/usr/include/qt4/QtXml -I/usr/include/qt4 -I../src/include -I../libs/apr/include -I../libs/libteletone/src -I. -I. -o main.o main.cpp
> > In file included from mainwindow.h:38,
> >                  from main.cpp:32:
> > ../src/include/switch.h:110:18: error: stfu.h: No such file or directory
> > In file included from ../src/include/switch.h:121,
> >                  from mainwindow.h:38,
> >                  from main.cpp:32:
> > ../src/include/switch_core.h:752: error: expected constructor, destructor, or type conversion before ?*? token
> > In file included from ../src/include/switch_loadable_module.h:46,
> >                  from ../src/include/switch.h:122,
> >                  from mainwindow.h:38,
> >                  from main.cpp:32:
> > ../src/include/switch_module_interfaces.h:121: error: expected initializer before ?*? token
> > ../src/include/switch_module_interfaces.h:162: error: ?switch_io_get_jb_t? does not name a type
> > In file included from ../src/include/switch.h:134,
> >                  from mainwindow.h:38,
> >                  from main.cpp:32:
> > ../src/include/switch_rtp.h:243: error: expected constructor, destructor, or type conversion before ?*? token
> > ./fshost.h:43: warning: ?void eventHandlerCallback(switch_event_t*)? declared ?static? but never defined
> > ./fshost.h:44: warning: ?switch_status_t loggerHandler(const switch_log_node_t*, switch_log_level_t)? declared ?static? but never defined
> > make: *** [main.o] Error 1
> > "
> >
> > watching this error:
> > "../src/include/switch.h:110:18: error: stfu.h: No such file or directory"
> >
> > i manually change switch.h to fix the problem with the include by adding the following:
> >
> > "
> > diff --git a/src/include/switch.h b/src/include/switch.h
> > index c7ea7b0..2847112 100644
> > --- a/src/include/switch.h
> > +++ b/src/include/switch.h
> > @@ -107,7 +107,8 @@
> > #include <errno.h>
> >
> > #ifndef WIN32
> > -#include "stfu.h"
> > +/* #include "stfu.h" */
> > +#include "../../../libs/stfu/stfu.h"
> > #else
> > #include "../../../libs/stfu/stfu.h"
> > #endif
> >
> > "
> >
> > I could compile fscomm, but now i can't start it... i have the following error:
> > "
> > Initializing core...
> > Failed to initialize FreeSWITCH's core: Cannot Open log directory or XML Root!
> > Everything OK, Entering runtime loop ...
> > Segmentation fault
> > "
> > i had try the fix in the wiki: "chmod  644 ~/.fscomm/conf/freeswitch.xml", and even "chmod -R 777 ~/.fscomm",  but no luck...
> >
> > Can you help me to go further...?
> >
> > I'm trying it to install on a debian squeeze, i installed qt4-dev-tools. Is it possible to install in debian squeeze our i should just  give up... and try another distro?
> > The freeswitch-git is the lasted head.
> >
> >
> > Thanks,
> > Ant?nio
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
>
>   --
>
> Un cordial saludo / Best regards,
>
>  _________________________
>
> Ant?nio Silva
>
> E-mail:asilva at wirelessmundi.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cal.leeming at simplicitymedialtd.co.uk  Mon Dec 17 00:58:50 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 16 Dec 2012 21:58:50 +0000
Subject: [Freeswitch-users] IVR docs updated - playing remote audio files
In-Reply-To: <50CE3E53.7000603@quentustech.com>
References: <CAHKQagFOUc9_TK6g6h6SNVkt0w-ztuKw0hroZ3yJKfKXjL2JSg@mail.gmail.com>
	<50CE268B.9070005@quentustech.com>
	<CAHKQagGaPXoY0tPD_azpAPv_fJNWO5d2EBfbX0eVkW22=2oo6g@mail.gmail.com>
	<50CE3E53.7000603@quentustech.com>
Message-ID: <CAHKQagEksZGDZkbXX+1uDi+BZG9j2QqChbNGAF3cKKtq6oifEw@mail.gmail.com>

Sorry, I should re-clarify - by production worthy-ness, I mean specifically
the lack of caching.

Cal

On Sun, Dec 16, 2012 at 9:34 PM, William King
<william.king at quentustech.com>wrote:

> libvlc in general is considered very production stable. The current
> implementation of mod_vlc may still have some tweaks required to perform
> to the expectations of the telephony environment. I'd suggest testing it
> with your requirements and provide feedback on how it performs.
>
> 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 12/16/2012 01:12 PM, Cal Leeming [Simplicity Media Ltd] wrote:
> > I haven't tried using mod_vlc yet.
> >
> > On first looks, it seems this would fall into the same category as
> > mod_shout, in terms of its production worthy-ness - any thoughts?
> >
> > Cal
> >
> > On Sun, Dec 16, 2012 at 7:52 PM, William King
> > <william.king at quentustech.com <mailto:william.king at quentustech.com>>
> wrote:
> >
> >     mod_vlc
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Dec 17 02:00:38 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sun, 16 Dec 2012 23:00:38 +0000
Subject: [Freeswitch-users] What is "Recovery On Timer Expire"?
In-Reply-To: <CALNZuw=z2ANmWmB6Fz3nximTyoPqUSWLJ=-ob55gdT6Q=_qC4Q@mail.gmail.com>
References: <CALNZuw=z2ANmWmB6Fz3nximTyoPqUSWLJ=-ob55gdT6Q=_qC4Q@mail.gmail.com>
Message-ID: <CAMbW9oJS32=e2=r6CrMx6X8n6afRXFudyQq2kAAR_qOp9NCPyg@mail.gmail.com>

Basically, FreeSWITCH sends occasional re-invites to the remote party to
make sure the call is still valid. If no response is received it will
assume the call has ended and drop the call.
This behavior is completely valid. However, this can also show up potential
NAT/firewall issues, where the NAT mapping is being lost.

I have come across people that deliberately block these re-invites,
who obviously have a fetish for zombie calls clogging up their systems...

On 16 December 2012 21:53, Avi Marcus <avi at avimarcus.net> wrote:

> I'm trying to clear up my CDRs from errors, if possible... I see a bunch
> of Recovery On Timer Expire and they mostly seem to be after 45 seconds of
> ring time (there is progress).
> Does that basically mean it rang for 40+ seconds before some timer
> somewhere kicked in and ended the call?
> So it's basically a "no answer" or "no user response"?
>
> Thanks,
> -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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon Dec 17 02:15:21 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 17 Dec 2012 01:15:21 +0200
Subject: [Freeswitch-users] What is "Recovery On Timer Expire"?
In-Reply-To: <CAMbW9oJS32=e2=r6CrMx6X8n6afRXFudyQq2kAAR_qOp9NCPyg@mail.gmail.com>
References: <CALNZuw=z2ANmWmB6Fz3nximTyoPqUSWLJ=-ob55gdT6Q=_qC4Q@mail.gmail.com>
	<CAMbW9oJS32=e2=r6CrMx6X8n6afRXFudyQq2kAAR_qOp9NCPyg@mail.gmail.com>
Message-ID: <CALNZuw=nShpsNpUP1C-ibAHVH+NKxf7keo+Me+jXJgw7gDmanQ@mail.gmail.com>

Not this case.
There is no NAT involved - my FS is on public and it's to a carrier to PSTN.
Also, the call never connected: there's no billsec and the entire length of
the call was ringing.

-Avi


On Mon, Dec 17, 2012 at 1:00 AM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> Basically, FreeSWITCH sends occasional re-invites to the remote party to
> make sure the call is still valid. If no response is received it will
> assume the call has ended and drop the call.
> This behavior is completely valid. However, this can also show up potential
> NAT/firewall issues, where the NAT mapping is being lost.
>
> I have come across people that deliberately block these re-invites,
> who obviously have a fetish for zombie calls clogging up their systems...
>
> On 16 December 2012 21:53, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I'm trying to clear up my CDRs from errors, if possible... I see a bunch
>> of Recovery On Timer Expire and they mostly seem to be after 45 seconds of
>> ring time (there is progress).
>> Does that basically mean it rang for 40+ seconds before some timer
>> somewhere kicked in and ended the call?
>> So it's basically a "no answer" or "no user response"?
>>
>> Thanks,
>> -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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Mon Dec 17 02:27:27 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 16 Dec 2012 18:27:27 -0500
Subject: [Freeswitch-users] What is "Recovery On Timer Expire"?
In-Reply-To: <CALNZuw=nShpsNpUP1C-ibAHVH+NKxf7keo+Me+jXJgw7gDmanQ@mail.gmail.com>
References: <CALNZuw=z2ANmWmB6Fz3nximTyoPqUSWLJ=-ob55gdT6Q=_qC4Q@mail.gmail.com>
	<CAMbW9oJS32=e2=r6CrMx6X8n6afRXFudyQq2kAAR_qOp9NCPyg@mail.gmail.com>
	<CALNZuw=nShpsNpUP1C-ibAHVH+NKxf7keo+Me+jXJgw7gDmanQ@mail.gmail.com>
Message-ID: <2148350.szYfaMxe3S@sos>

I bet there is no SIP 180 or 183 after INVITE within progress_timeout 
interval.

On Monday 17 December 2012 01:15:21 Avi Marcus wrote:
> Not this case.
> There is no NAT involved - my FS is on public and it's to a carrier to PSTN.
> Also, the call never connected: there's no billsec and the entire length of
> the call was ringing.
> 
> -Avi
> 
> 
> On Mon, Dec 17, 2012 at 1:00 AM, Andrew Cassidy <
> 
> andrew at cassidywebservices.co.uk> wrote:
> > Basically, FreeSWITCH sends occasional re-invites to the remote party to
> > make sure the call is still valid. If no response is received it will
> > assume the call has ended and drop the call.
> > This behavior is completely valid. However, this can also show up
> > potential
> > NAT/firewall issues, where the NAT mapping is being lost.
> > 
> > I have come across people that deliberately block these re-invites,
> > who obviously have a fetish for zombie calls clogging up their systems...
> > 
> > On 16 December 2012 21:53, Avi Marcus <avi at avimarcus.net> wrote:
> >> I'm trying to clear up my CDRs from errors, if possible... I see a bunch
> >> of Recovery On Timer Expire and they mostly seem to be after 45 seconds
> >> of
> >> ring time (there is progress).
> >> Does that basically mean it rang for 40+ seconds before some timer
> >> somewhere kicked in and ended the call?
> >> So it's basically a "no answer" or "no user response"?
> >> 
> >> Thanks,
> >> -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
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org


From Sirish.MasurMohan at oa.com.au  Mon Dec 17 02:31:09 2012
From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan)
Date: Mon, 17 Dec 2012 10:31:09 +1100
Subject: [Freeswitch-users] Newbie question: Executing Lua scripts for
 incoming calls, how concurrency is to be handled?
Message-ID: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au>

Hello All,

I am a newbie to FreeSWITCH, please excuse if this question has already been answered.

My requirement is as follows:

*         When an incoming call is received, it needs to be bridged to one of the 4 available destination lines

*         Minimal management of these lines is required: need to keep track of when each line was used, so that, when a new incoming call is received, I can use the oldest used available line to bridge the call to

In order to achieve this, I have:

*         Defined a few global variables in vars.xml, which would keep track of each line's status (available/busy) and last used timestamp

*         Implemented a Lua script which 'chooses' the line based on global variables status and timestamp (using "global_getvar") and update the status of the line to be used (using "global_setvar")

*         Bridge to the line returned by the script

*         Execute another Lua script on hang-up, which updates the global variables

I am now wondering if I would run into issues in cases where I receive concurrent calls (I am using a Sangoma E1 card, and will receive multiple concurrent calls soon)? Would I need to be applying some kind of explicit mutex protection in these scripts for the global variables?

Thanks in advance!

With regards,
Sirish

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From william.king at quentustech.com  Mon Dec 17 02:46:54 2012
From: william.king at quentustech.com (William King)
Date: Sun, 16 Dec 2012 15:46:54 -0800
Subject: [Freeswitch-users] Newbie question: Executing Lua scripts for
 incoming calls, how concurrency is to be handled?
In-Reply-To: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au>
References: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au>
Message-ID: <50CE5D6E.2020401@quentustech.com>

Sounds like you want to take a look into mod_fifo.

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 12/16/2012 03:31 PM, Sirish Masur Mohan wrote:
> Hello All,
> 
>  
> 
> I am a newbie to FreeSWITCH, please excuse if this question has already
> been answered.
> 
>  
> 
> My requirement is as follows:
> 
> ?         When an incoming call is received, it needs to be bridged to
> one of the 4 available destination lines
> 
> ?         Minimal management of these lines is required: need to keep
> track of when each line was used, so that, when a new incoming call is
> received, I can use the oldest used available line to bridge the call to
> 
>  
> 
> In order to achieve this, I have:
> 
> ?         Defined a few global variables in vars.xml, which would keep
> track of each line?s status (available/busy) and last used timestamp
> 
> ?         Implemented a Lua script which ?chooses? the line based on
> global variables status and timestamp (using "global_getvar") and update
> the status of the line to be used (using "global_setvar")
> 
> ?         Bridge to the line returned by the script
> 
> ?         Execute another Lua script on hang-up, which updates the
> global variables
> 
>  
> 
> I am now wondering if I would run into issues in cases where I receive
> concurrent calls (I am using a Sangoma E1 card, and will receive
> multiple concurrent calls soon)? Would I need to be applying some kind
> of explicit mutex protection in these scripts for the global variables?  
> 
>  
> 
> Thanks in advance!
> 
>  
> 
> With regards,
> 
> Sirish
> 
>  
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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


From covici at ccs.covici.com  Mon Dec 17 02:48:36 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 16 Dec 2012 18:48:36 -0500
Subject: [Freeswitch-users] Disable media one leg?
In-Reply-To: <CCF359BB.6228C%krice@freeswitch.org>
References: <CCF359BB.6228C%krice@freeswitch.org>
Message-ID: <32431.1355701716@ccs.covici.com>

I  think waste is now the default.

Ken Rice <krice at freeswitch.org> wrote:

> If you look at mod conference there is the waste flag, if there is no media
> to send to a channel, mod_conference doesn't send media to it... You should
> start looking there
> 
> 
> On 12/16/12 10:34 AM, "Emrah" <lists at kavun.ch> wrote:
> 
> > Ah ha! I get it now.
> > VAD sounds like what you may be looking for.
> > 
> > http://wiki.freeswitch.org/wiki/VAD_and_CNG
> > On Dec 16, 2012, at 9:25 AM, Nick Vines <jnvines at gmail.com> wrote:
> > 
> >> This might not be the best way, but you could make a 2 person conference with
> >> the listener muted and the controls disabled. Then you could also use rtp vad
> >> to reduce the bandwidth. Again, I'm not sure if these would stack
> >> appropriately, but it seems feasible.
> >> 
> >> Nick
> >> 
> >> 
> >> On Sun, Dec 16, 2012 at 8:56 AM, Avi Marcus <avi at avimarcus.net> wrote:
> >> If FS is going to be streaming the audio the media path is obviously
> >> essential, especially if it's going to be doing some transcoding with
> >> mod_shout or something similar.
> >> 
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting 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
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sirish.MasurMohan at oa.com.au  Mon Dec 17 04:41:11 2012
From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan)
Date: Mon, 17 Dec 2012 12:41:11 +1100
Subject: [Freeswitch-users] Newbie question: Executing Lua scripts for
 incoming calls, how concurrency is to be handled?
In-Reply-To: <50CE5D6E.2020401@quentustech.com>
References: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au>
	<50CE5D6E.2020401@quentustech.com>
Message-ID: <965759A53E43FE439E43565A7715E5F058F4156D32@oa-exchange1.oa.com.au>

Hi William,

Thanks for the reply.

My setup is as follows:
Client(Caller)->dialup modem->PBX->FreeSWITCH->PBX-> 4 dialup modems->Server(Receiver)

I basically need FreeSWITCH to bridge the incoming call to the best external destination (out of the 4 available), so that the modem training, connection etc can take place smoothly, before exchange of data. I am not sure if mod_fifo would help me in this scenario, as, I would require an agent to dial in and read the fifo. Could you please clarify?

Thanks! 

With regards,
Sirish

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William King
Sent: Monday, 17 December 2012 10:47 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

Sounds like you want to take a look into mod_fifo.

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




From chad at apartmentlines.com  Mon Dec 17 06:55:24 2012
From: chad at apartmentlines.com (Chad Phillips)
Date: Sun, 16 Dec 2012 21:55:24 -0600
Subject: [Freeswitch-users] LUA - Call a Function through a variable name
In-Reply-To: <CAGvuWYbg+3kc5t6DxzdB7POmTCYtKdL0oqp4qbw1iiEgPEB7Fg@mail.gmail.com>
References: <CAGvuWYbg+3kc5t6DxzdB7POmTCYtKdL0oqp4qbw1iiEgPEB7Fg@mail.gmail.com>
Message-ID: <ECDF822222814AEF8B742F54F1277E3D@gmail.com>

Lua 5.1.4  Copyright (C) 1994-2008 Lua.org, PUC-Rio
> function foo() print "hello" end
> bar = foo
> bar()
hello
> 

or, if you really want to have the function name as a string, there are some suggestions here:

http://stackoverflow.com/questions/1791234/lua-call-function-from-a-string-with-function-name

hunmonk 


On Sunday, December 16, 2012 at 11:34 AM, Lloyd Aloysius wrote:

> Hello 
> 
> Does lua support , Call a Function through a variable name
> 
> Ex:
> 
> function playfile()
> 
> end
> 
> variable_a = "playfile" 
> 
> how to call the variable_a , that execute function? 
> 
> Thanks
> Lloyd
> 
> 
> 
> _________________________________________________________________________
> 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
> 
> 


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From b2m at a-cti.com  Mon Dec 17 10:31:38 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Mon, 17 Dec 2012 13:01:38 +0530
Subject: [Freeswitch-users] G729 - [INCOMPATIBLE_DESTINATION]
In-Reply-To: <CAMWiMy_0f44SwcXZR4CQgP7=RBqrTYW8fQQhAa8m1nR+kKv=oQ@mail.gmail.com>
References: <CAP2fbbcixeTy4H+KPbxJQvAhCDWmco5a0jF0XQkGHBtE2Qkmdg@mail.gmail.com>
	<CAFiqYuk+OwUVPCmy8H1Yq5hW-+Hhy40DS1RdRe8T6oBTjeY2LA@mail.gmail.com>
	<CAMWiMy_0f44SwcXZR4CQgP7=RBqrTYW8fQQhAa8m1nR+kKv=oQ@mail.gmail.com>
Message-ID: <CAP2fbbfczFPcBF5ZF0G_+FL_+n0XZPBcu6KCCSHU1ryny2hBKg@mail.gmail.com>

Thanks for the help!!  Please let me know if I missed anything.

 <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,SILK at 8000h
,SILK at 12000h,SILK at 16000h,SILK at 24000h,G7221 at 32000h,G7221 at 16000h
,G722,PCMU,PCMA,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,SILK at 8000h
,SILK at 12000h,SILK at 16000h,SILK at 24000h,G7221 at 32000h,G7221 at 16000h
,G722,PCMU,PCMA,GSM"/>


freeswitch at internal> g729_info
Permitted G729 channels: 10
Encoders in use: 4
Decoders in use: 3

freeswitch at internal>



recv 742 bytes from udp/[182.79.149.142]:16251 at 07:27:19.212195:

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

   INVITE sip:500 at 50.54.12.39 SIP/2.0

   Via: SIP/2.0/UDP 182.79.149.142:16251;rport;branch=z9hG4bK68891

   Route: <sip:50.54.12.39;transport=udp;lr>

   Max-Forwards: 70

   From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060

   To: <sip:500 at 50.54.12.39>

   Call-ID: 047707180743 at 10.4.2.7

   CSeq: 1 INVITE

   Contact: <sip:501 at 182.79.149.142:16251;transport=udp>

   Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS

   Expires: 3600

   User-Agent: SyncPhone 1.0

   Content-Length: 244

   Content-Type: application/sdp



   v=0

   o=- 1111 1111 IN IP4 182.79.149.142

   s=Session SIP/SDP

   c=IN IP4 182.79.149.142

   t=0 0

   a=sendrecv

   a=rtcp:57395 IN IP4 182.79.149.142

   m=audio 26474 RTP/AVP 0 101

   a=rtpmap:0 G729/8000

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-15

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

send 321 bytes to udp/[182.79.149.142]:16251 at 07:27:19.212520:

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

   SIP/2.0 100 Trying

   Via: SIP/2.0/UDP 182.79.149.142:16251;rport=16251;branch=z9hG4bK68891

   From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060

   To: <sip:500 at 50.54.12.39>

   Call-ID: 047707180743 at 10.4.2.7

   CSeq: 1 INVITE

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
18-32-35 -0600

   Content-Length: 0



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

2012-12-17 07:27:19.206419 [DEBUG] sofia.c:7481 IP 182.79.149.142 Approved
by acl "domains[]". Access Granted.

2012-12-17 07:27:19.206419 [NOTICE] switch_channel.c:930 New Channel
sofia/internal/501 at 50.54.12.39 [30a75cd4-481b-11e2-84b4-e5c28303057e]

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/501 at 50.54.12.39) Running State Change CS_NEW

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:380
(sofia/internal/501 at 50.54.12.39) State NEW

2012-12-17 07:27:19.206419 [DEBUG] sofia.c:5502 Channel sofia/internal/
501 at 50.54.12.39 entering state [received][100]

2012-12-17 07:27:19.206419 [DEBUG] sofia.c:5513 Remote SDP:

v=0

o=- 1111 1111 IN IP4 182.79.149.142

s=Session SIP/SDP

c=IN IP4 182.79.149.142

t=0 0

a=sendrecv

a=rtcp:57395 IN IP4 182.79.149.142

m=audio 26474 RTP/AVP 0 101

a=rtpmap:0 G729/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15


2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[G729:18:8000:20:8000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[SILK:117:8000:20:20000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[SILK:118:12000:20:25000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[SILK:119:16000:20:30000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[SILK:120:24000:20:40000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[G7221:115:32000:20:48000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[G7221:107:16000:20:32000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[G722:9:8000:20:64000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[G729:0:8000:20:64000]/[GSM:3:8000:20:13200]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]

2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
send/recv payload to 101

2012-12-17 07:27:19.206419 [DEBUG] switch_channel.c:2852 (sofia/internal/
501 at 50.54.12.39) Callstate Change DOWN -> HANGUP

2012-12-17 07:27:19.206419 [NOTICE] sofia.c:5781 Hangup sofia/internal/
501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]

2012-12-17 07:27:19.206419 [DEBUG] switch_channel.c:2875 Send signal
sofia/internal/501 at 50.54.12.39 [KILL]

2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send signal
sofia/internal/501 at 50.54.12.39 [BREAK]

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/501 at 50.54.12.39) Running State Change CS_HANGUP

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:602
(sofia/internal/501 at 50.54.12.39) State HANGUP

2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:469 Channel sofia/internal/
501 at 50.54.12.39 hanging up, cause: INCOMPATIBLE_DESTINATION

2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:534 Responding to INVITE
with: 488

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:47
sofia/internal/501 at 50.54.12.39 Standard HANGUP, cause:
INCOMPATIBLE_DESTINATION

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:602
(sofia/internal/501 at 50.54.12.39) State HANGUP going to sleep

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:393
(sofia/internal/501 at 50.54.12.39) State Change CS_HANGUP -> CS_REPORTING

2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send signal
sofia/internal/501 at 50.54.12.39 [BREAK]

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/501 at 50.54.12.39) Running State Change CS_REPORTING

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:662
(sofia/internal/501 at 50.54.12.39) State REPORTING

send 816 bytes to udp/[182.79.149.142]:16251 at 07:27:19.215295:

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

   SIP/2.0 488 Not Acceptable Here

   Via: SIP/2.0/UDP 182.79.149.142:16251;rport=16251;branch=z9hG4bK68891

   From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060

   To: <sip:500 at 50.54.12.39>;tag=8K31Z9S2DymjN

   Call-ID: 047707180743 at 10.4.2.7

   CSeq: 1 INVITE

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
18-32-35 -0600

   Accept: application/sdp

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: timer, precondition, path, replaces

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:79
sofia/internal/501 at 50.54.12.39 Standard REPORTING, cause:
INCOMPATIBLE_DESTINATION

   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer

   Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"2012-12-17
07:27:19.206419 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/
501 at 50.54.12.39) State REPORTING going to sleep


   Content-Length: 0

   Remote-Party-ID: "500" <sip:500 at 50.54.12.39
>;party=calling;privacy=off;screen=no



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

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:387
(sofia/internal/501 at 50.54.12.39) State Change CS_REPORTING -> CS_DESTROY

2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send signal
sofia/internal/501 at 50.54.12.39 [BREAK]

2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1380 Session 204
(sofia/internal/501 at 50.54.12.39) Locked, Waiting on external entities

2012-12-17 07:27:19.206419 [NOTICE] switch_core_session.c:1398 Session 204
(sofia/internal/501 at 50.54.12.39) Ended

2012-12-17 07:27:19.206419 [NOTICE] switch_core_session.c:1400 Close
Channel sofia/internal/501 at 50.54.12.39 [CS_DESTROY]

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:491
(sofia/internal/501 at 50.54.12.39) Callstate Change HANGUP -> DOWN

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:494
(sofia/internal/501 at 50.54.12.39) Running State Change CS_DESTROY

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:504
(sofia/internal/501 at 50.54.12.39) State DESTROY

2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:374 sofia/internal/
501 at 50.54.12.39 SOFIA DESTROY

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:86
sofia/internal/501 at 50.54.12.39 Standard DESTROY

2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:504
(sofia/internal/501 at 50.54.12.39) State DESTROY going to sleep

recv 407 bytes from udp/[182.79.149.142]:16251 at 07:27:19.581401:

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

   ACK sip:500 at 50.54.12.39 SIP/2.0

   Via: SIP/2.0/UDP 182.79.149.142:16251;rport;branch=z9hG4bK68891

   Route: <sip:50.54.12.39;transport=udp;lr>

   Max-Forwards: 70

   From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060

   To: <sip:500 at 50.54.12.39>;tag=8K31Z9S2DymjN

   Call-ID: 047707180743 at 10.4.2.7

   CSeq: 1 ACK

   Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS

   User-Agent: SyncPhone 1.0

   Content-Length: 0



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

recv 465 bytes from udp/[182.79.149.142]:36420 at 07:27:20.672450:

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

   REGISTER sip:50.54.12.39 SIP/2.0

   Via: SIP/2.0/UDP 182.79.149.142:36420;rport;branch=z9hG4bK20451

   Route: <sip:50.54.12.39;transport=udp;lr>

   Max-Forwards: 70

   From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121

   To: <sip:501 at 50.54.12.39>

   Call-ID: 790899967509 at 10.4.2.7

   CSeq: 1 REGISTER

   Contact: <sip:501 at 182.79.149.142:36420;transport=udp>

   Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS

   Expires: 3600

   User-Agent: SyncPhone 1.0

   Content-Length: 0



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

2012-12-17 07:27:20.666425 [DEBUG] sofia_reg.c:1417 Send challenge for [
501 at 50.54.12.39]

2012-12-17 07:27:20.666425 [WARNING] sofia_reg.c:1421 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [501 at 50.54.12.39] from ip
182.79.149.142

send 626 bytes to udp/[182.79.149.142]:36420 at 07:27:20.673200:

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

   SIP/2.0 401 Unauthorized

   Via: SIP/2.0/UDP 182.79.149.142:36420;rport=36420;branch=z9hG4bK20451

   From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121

   To: <sip:501 at 50.54.12.39>;tag=9vvt14a6a7a5g

   Call-ID: 790899967509 at 10.4.2.7

   CSeq: 1 REGISTER

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
18-32-35 -0600

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: timer, precondition, path, replaces

   WWW-Authenticate: Digest realm="50.54.12.39",
nonce="31862298-481b-11e2-84b6-e5c28303057e", algorithm=MD5, qop="auth"

   Content-Length: 0



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

recv 707 bytes from udp/[182.79.149.142]:36420 at 07:27:21.094563:

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

   REGISTER sip:50.54.12.39 SIP/2.0

   Via: SIP/2.0/UDP 182.79.149.142:36420;rport;branch=z9hG4bK84890

   Route: <sip:50.54.12.39;transport=udp;lr>

   Max-Forwards: 70

   From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121

   To: <sip:501 at 50.54.12.39>

   Call-ID: 790899967509 at 10.4.2.7

   CSeq: 2 REGISTER

   Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS

   Expires: 3600

   User-Agent: SyncPhone 1.0

   Contact: <sip:501 at 182.79.149.142:36420;transport=udp>

   Authorization: Digest username="501", realm="50.54.12.39",
nonce="31862298-481b-11e2-84b6-e5c28303057e", uri="sip:50.54.12.39",
algorithm=MD5, qop=auth, nc=00000001, cnonce="a5d0aab2288476b3",
response="eb8ccc768c7f9fee038f0d5dfc599931"

   Content-Length: 0



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

2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
'record_stereo' = 'true'

2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
'default_gateway' = 'example.com'

2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
'default_areacode' = '918'

2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
'transfer_fallback_extension' = 'operator'

2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
'toll_allow' = 'domestic,international,local'

2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
'accountcode' = '501'

2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
'user_context' = 'default'

2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
'effective_caller_id_name' = 'Extension 501'

2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
'effective_caller_id_number' = '501'

2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
'outbound_caller_id_name' = 'FS'

2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
'outbound_caller_id_number' = '9732386040'

2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
'callgroup' = 'techsupport'

2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:1575 Register:

From:    [501 at 10.252.148.21]

Contact: ["user" <sip:501 at 182.79.149.142:36420;transport=udp>]

Expires: [3600]

send 600 bytes to udp/[182.79.149.142]:36420 at 07:27:21.096965:

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

   SIP/2.0 200 OK

   Via: SIP/2.0/UDP 182.79.149.142:36420;rport=36420;branch=z9hG4bK84890

   From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121

   To: <sip:501 at 50.54.12.39>;tag=a6NK3ZU97F1Qc

   Call-ID: 790899967509 at 10.4.2.7

   CSeq: 2 REGISTER

   Contact: <sip:501 at 182.79.149.142:36420;transport=udp>;expires=3600

   Date: Mon, 17 Dec 2012 07:27:21 GMT

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
18-32-35 -0600

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: timer, precondition, path, replaces

   Content-Length: 0



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

send 926 bytes to udp/[182.79.149.142]:36420 at 07:27:21.112029:

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

   NOTIFY sip:501 at 182.79.149.142:36420;transport=udp SIP/2.0

   Via: SIP/2.0/UDP 50.54.12.39;rport;branch=z9hG4bKXXQBeBHUQ5yZp

   Max-Forwards: 70

   From: <sip:501 at 50.54.12.39>;tag=BFFc5tcD5rQar

   To: <sip:501 at 50.54.12.39>

   Call-ID: 09276738-c2be-1230-ec86-22000afc9415

   CSeq: 37528556 NOTIFY

   Contact: <sip:mod_sofia at 50.54.12.39:5060>

   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
18-32-35 -0600

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: timer, precondition, path, replaces

   Event: message-summary

   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer

   Subscription-State: terminated;reason=noresource

   Content-Type: application/simple-message-summary

   Content-Length: 64



   Messages-Waiting: no

   Message-Account: sip:501 at 10.252.148.21



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

recv 295 bytes from udp/[182.79.149.142]:36420 at 07:27:21.531463:

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

   SIP/2.0 405 Method Not Allowed

   Via: SIP/2.0/UDP 50.54.12.39;branch=z9hG4bKXXQBeBHUQ5yZp;rport=5060

   To: <sip:501 at 50.54.12.39>

   From: <sip:501 at 50.54.12.39>;tag=BFFc5tcD5rQar

   Call-ID: 09276738-c2be-1230-ec86-22000afc9415

   CSeq: 37528556 NOTIFY

   Server: SyncPhone 1.0

   Content-Length: 0



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




On Mon, Dec 17, 2012 at 2:25 AM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> If he is using mod_g729 then the log would throw an error saying that
> the codec only does passthrough mode, however he hasn't shown that
> part of the log to us yet
>
> On Sun, Dec 16, 2012 at 12:05 PM, Steven Ayre <steveayre at gmail.com> wrote:
> > What G729 module are you using, and what codec is in use on the other
> leg?
> >
> > G729 needs licenses due to patents.
> >
> > mod_g729 is passthrough only but requires no licenses. This is because it
> > merely forwards the data, but doesn't perform the encoding/transcoding
> step
> > that the patents cover. But that means it can't transcode between
> different
> > codecs.
> > mod_com_g729 is the licensed version: http://www.freeswitch.org/node/235
> >
> > With the passthrough mod_g729 codec, if the other leg has selected a
> codec
> > other than g729 you'll see an error in your logs and it'll hangup with
> that
> > reason.
> >
> > You can tweak your codec negotiation to avoid this (eg
> > late-negotiation=true), or you can use the licensed version.
> >
> >
> >
> >
> > On 16 December 2012 11:40, Balamurugan Mahendran <b2m at a-cti.com> wrote:
> >>
> >> Need help on Codec Negotiation
> >>
> >>
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> >> [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
> >> send/recv payload to 101
> >>
> >>
> >> 2012-12-16 11:35:52.246422 [DEBUG] switch_channel.c:2852
> >> (sofia/internal/501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
> >> 2012-12-16 11:35:52.246422 [NOTICE] sofia.c:5781 Hangup
> >> sofia/internal/501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
> >>
> >> 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
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vbvbrj at gmail.com  Mon Dec 17 12:38:39 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Mon, 17 Dec 2012 11:38:39 +0200
Subject: [Freeswitch-users] Multihomed server.
In-Reply-To: <CCF24886.6211C%krice@freeswitch.org>
References: <CCF24886.6211C%krice@freeswitch.org>
Message-ID: <50CEE81F.6020408@gmail.com>

On 15.12.2012 23:32, Ken Rice wrote:
> Why not just setup 2 profiles sharing the same domain?

In the wiki its not very well described about domains, especially about 
sharing same domain between multiple profiles.

> why not simply use split horizon dns so that
> endpoints on any of the attached networks get proper host pointers...

It's not possible for now, as company does not have a dedicated DNS name 
registered at international DNS servers specifically for this, but it 
will be used when everything is settled down.

On 15.12.2012 23:19, Steven Ayre wrote:
 >
 > Rather your problem is coming from the fact they're both registering 
with the same credentials.
 >
 > Can you get each phone to register with its own user account?

Nope. The idea is to register multiple phone to same account and use 
simple contact to get phones ringing.

 >
 > I suspect simply prefixing your bridge dial string with 
{ignore_early_media=true} will get the behaviour you desire.

I didn't try this, but finally I managed to accomplish the idea. I took 
sip_profiles/internal.xml and changed its name to 
internal_(internal_IP).xml, and in it I put rtp-ip, sip-ip, ext-rtp-ip, 
ext-sip-ip to internal IP of the server and listening port to 5060 UDP. 
Also, made a copy with name internal_(external-IP).xml for IP facing to 
internet and modified rtp-ip, sip-ip, ext-rtp-ip, ext-sip-ip to IP of 
interface facing to internet and listening on port 5060 UDP.

Then in both profiles I changed
<domains>
	<domain name="all" alias="true" parse="false"/>
</domains>
to
<domains>
	<domain name="$${domain}" alias="false" parse="false"/>
</domains>

And now all registered phones for same account via both profiles rings. 
Of course in both profile is a parameter:
<param name="multiple-registrations" value="contact"/>

That is how I managed to get it working.

-- 
Mimiko desu.


From steveayre at gmail.com  Mon Dec 17 14:21:24 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 17 Dec 2012 11:21:24 +0000
Subject: [Freeswitch-users] G729 - [INCOMPATIBLE_DESTINATION]
In-Reply-To: <CAP2fbbfczFPcBF5ZF0G_+FL_+n0XZPBcu6KCCSHU1ryny2hBKg@mail.gmail.com>
References: <CAP2fbbcixeTy4H+KPbxJQvAhCDWmco5a0jF0XQkGHBtE2Qkmdg@mail.gmail.com>
	<CAFiqYuk+OwUVPCmy8H1Yq5hW-+Hhy40DS1RdRe8T6oBTjeY2LA@mail.gmail.com>
	<CAMWiMy_0f44SwcXZR4CQgP7=RBqrTYW8fQQhAa8m1nR+kKv=oQ@mail.gmail.com>
	<CAP2fbbfczFPcBF5ZF0G_+FL_+n0XZPBcu6KCCSHU1ryny2hBKg@mail.gmail.com>
Message-ID: <CAFiqYukVJW5BzbXeRtJmonAXxudRvuW9=h7SwCQjyyTHk3th6g@mail.gmail.com>

>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]

This looks odd, although I'm not sure why it's not working.

The remote endpoint is using the static payload type 0 for G729. Normally
that's reserved for G711 ulaw, while G729 uses 18. But the rtpmap should
allow overriding that.

> G729:0:8000:20:64000


The 64000 rate here also looks odd - it's comparing 64khz to 8khz and I
suspect that's also a reason why it doesn't recognise it. I don't know
where that 64000 is coming from though.

-Steve





On 17 December 2012 07:31, Balamurugan Mahendran <b2m at a-cti.com> wrote:

> Thanks for the help!!  Please let me know if I missed anything.
>
>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,SILK at 8000h
> ,SILK at 12000h,SILK at 16000h,SILK at 24000h,G7221 at 32000h,G7221 at 16000h
> ,G722,PCMU,PCMA,GSM"/>
>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,SILK at 8000h
> ,SILK at 12000h,SILK at 16000h,SILK at 24000h,G7221 at 32000h,G7221 at 16000h
> ,G722,PCMU,PCMA,GSM"/>
>
>
>  freeswitch at internal> g729_info
> Permitted G729 channels: 10
> Encoders in use: 4
> Decoders in use: 3
>
> freeswitch at internal>
>
>
>
> recv 742 bytes from udp/[182.79.149.142]:16251 at 07:27:19.212195:
>
>    ------------------------------------------------------------------------
>
>    INVITE sip:500 at 50.54.12.39 SIP/2.0
>
>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport;branch=z9hG4bK68891
>
>    Route: <sip:50.54.12.39;transport=udp;lr>
>
>    Max-Forwards: 70
>
>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>
>    To: <sip:500 at 50.54.12.39>
>
>    Call-ID: 047707180743 at 10.4.2.7
>
>    CSeq: 1 INVITE
>
>    Contact: <sip:501 at 182.79.149.142:16251;transport=udp>
>
>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>
>    Expires: 3600
>
>    User-Agent: SyncPhone 1.0
>
>    Content-Length: 244
>
>    Content-Type: application/sdp
>
>
>
>    v=0
>
>    o=- 1111 1111 IN IP4 182.79.149.142
>
>    s=Session SIP/SDP
>
>    c=IN IP4 182.79.149.142
>
>    t=0 0
>
>    a=sendrecv
>
>    a=rtcp:57395 IN IP4 182.79.149.142
>
>    m=audio 26474 RTP/AVP 0 101
>
>    a=rtpmap:0 G729/8000
>
>    a=rtpmap:101 telephone-event/8000
>
>    a=fmtp:101 0-15
>
>    ------------------------------------------------------------------------
>
> send 321 bytes to udp/[182.79.149.142]:16251 at 07:27:19.212520:
>
>    ------------------------------------------------------------------------
>
>    SIP/2.0 100 Trying
>
>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport=16251;branch=z9hG4bK68891
>
>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>
>    To: <sip:500 at 50.54.12.39>
>
>    Call-ID: 047707180743 at 10.4.2.7
>
>    CSeq: 1 INVITE
>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
> 18-32-35 -0600
>
>    Content-Length: 0
>
>
>
>    ------------------------------------------------------------------------
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:7481 IP 182.79.149.142 Approved
> by acl "domains[]". Access Granted.
>
> 2012-12-17 07:27:19.206419 [NOTICE] switch_channel.c:930 New Channel
> sofia/internal/501 at 50.54.12.39 [30a75cd4-481b-11e2-84b4-e5c28303057e]
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/501 at 50.54.12.39) Running State Change CS_NEW
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:380
> (sofia/internal/501 at 50.54.12.39) State NEW
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:5502 Channel sofia/internal/
> 501 at 50.54.12.39 entering state [received][100]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:5513 Remote SDP:
>
> v=0
>
> o=- 1111 1111 IN IP4 182.79.149.142
>
> s=Session SIP/SDP
>
> c=IN IP4 182.79.149.142
>
> t=0 0
>
> a=sendrecv
>
> a=rtcp:57395 IN IP4 182.79.149.142
>
> m=audio 26474 RTP/AVP 0 101
>
> a=rtpmap:0 G729/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-15
>
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
>
> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
> send/recv payload to 101
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_channel.c:2852 (sofia/internal/
> 501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
>
> 2012-12-17 07:27:19.206419 [NOTICE] sofia.c:5781 Hangup sofia/internal/
> 501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_channel.c:2875 Send signal
> sofia/internal/501 at 50.54.12.39 [KILL]
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send signal
> sofia/internal/501 at 50.54.12.39 [BREAK]
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/501 at 50.54.12.39) Running State Change CS_HANGUP
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:602
> (sofia/internal/501 at 50.54.12.39) State HANGUP
>
> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:469 Channel sofia/internal/
> 501 at 50.54.12.39 hanging up, cause: INCOMPATIBLE_DESTINATION
>
> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:534 Responding to INVITE
> with: 488
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:47
> sofia/internal/501 at 50.54.12.39 Standard HANGUP, cause:
> INCOMPATIBLE_DESTINATION
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:602
> (sofia/internal/501 at 50.54.12.39) State HANGUP going to sleep
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:393
> (sofia/internal/501 at 50.54.12.39) State Change CS_HANGUP -> CS_REPORTING
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send signal
> sofia/internal/501 at 50.54.12.39 [BREAK]
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/501 at 50.54.12.39) Running State Change CS_REPORTING
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:662
> (sofia/internal/501 at 50.54.12.39) State REPORTING
>
> send 816 bytes to udp/[182.79.149.142]:16251 at 07:27:19.215295:
>
>    ------------------------------------------------------------------------
>
>    SIP/2.0 488 Not Acceptable Here
>
>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport=16251;branch=z9hG4bK68891
>
>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>
>    To: <sip:500 at 50.54.12.39>;tag=8K31Z9S2DymjN
>
>    Call-ID: 047707180743 at 10.4.2.7
>
>    CSeq: 1 INVITE
>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
> 18-32-35 -0600
>
>    Accept: application/sdp
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
>    Supported: timer, precondition, path, replaces
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:79
> sofia/internal/501 at 50.54.12.39 Standard REPORTING, cause:
> INCOMPATIBLE_DESTINATION
>
>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>
>    Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"2012-12-17
> 07:27:19.206419 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/
> 501 at 50.54.12.39) State REPORTING going to sleep
>
>
>    Content-Length: 0
>
>    Remote-Party-ID: "500" <sip:500 at 50.54.12.39
> >;party=calling;privacy=off;screen=no
>
>
>
>    ------------------------------------------------------------------------
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:387
> (sofia/internal/501 at 50.54.12.39) State Change CS_REPORTING -> CS_DESTROY
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send signal
> sofia/internal/501 at 50.54.12.39 [BREAK]
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1380 Session 204
> (sofia/internal/501 at 50.54.12.39) Locked, Waiting on external entities
>
> 2012-12-17 07:27:19.206419 [NOTICE] switch_core_session.c:1398 Session 204
> (sofia/internal/501 at 50.54.12.39) Ended
>
> 2012-12-17 07:27:19.206419 [NOTICE] switch_core_session.c:1400 Close
> Channel sofia/internal/501 at 50.54.12.39 [CS_DESTROY]
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:491
> (sofia/internal/501 at 50.54.12.39) Callstate Change HANGUP -> DOWN
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:494
> (sofia/internal/501 at 50.54.12.39) Running State Change CS_DESTROY
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:504
> (sofia/internal/501 at 50.54.12.39) State DESTROY
>
> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:374 sofia/internal/
> 501 at 50.54.12.39 SOFIA DESTROY
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:86
> sofia/internal/501 at 50.54.12.39 Standard DESTROY
>
> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:504
> (sofia/internal/501 at 50.54.12.39) State DESTROY going to sleep
>
> recv 407 bytes from udp/[182.79.149.142]:16251 at 07:27:19.581401:
>
>    ------------------------------------------------------------------------
>
>    ACK sip:500 at 50.54.12.39 SIP/2.0
>
>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport;branch=z9hG4bK68891
>
>    Route: <sip:50.54.12.39;transport=udp;lr>
>
>    Max-Forwards: 70
>
>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>
>    To: <sip:500 at 50.54.12.39>;tag=8K31Z9S2DymjN
>
>    Call-ID: 047707180743 at 10.4.2.7
>
>    CSeq: 1 ACK
>
>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>
>    User-Agent: SyncPhone 1.0
>
>    Content-Length: 0
>
>
>
>    ------------------------------------------------------------------------
>
> recv 465 bytes from udp/[182.79.149.142]:36420 at 07:27:20.672450:
>
>    ------------------------------------------------------------------------
>
>    REGISTER sip:50.54.12.39 SIP/2.0
>
>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport;branch=z9hG4bK20451
>
>    Route: <sip:50.54.12.39;transport=udp;lr>
>
>    Max-Forwards: 70
>
>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>
>    To: <sip:501 at 50.54.12.39>
>
>    Call-ID: 790899967509 at 10.4.2.7
>
>    CSeq: 1 REGISTER
>
>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>
>
>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>
>    Expires: 3600
>
>    User-Agent: SyncPhone 1.0
>
>    Content-Length: 0
>
>
>
>    ------------------------------------------------------------------------
>
> 2012-12-17 07:27:20.666425 [DEBUG] sofia_reg.c:1417 Send challenge for [
> 501 at 50.54.12.39]
>
> 2012-12-17 07:27:20.666425 [WARNING] sofia_reg.c:1421 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [501 at 50.54.12.39] from ip
> 182.79.149.142
>
> send 626 bytes to udp/[182.79.149.142]:36420 at 07:27:20.673200:
>
>    ------------------------------------------------------------------------
>
>    SIP/2.0 401 Unauthorized
>
>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport=36420;branch=z9hG4bK20451
>
>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>
>    To: <sip:501 at 50.54.12.39>;tag=9vvt14a6a7a5g
>
>    Call-ID: 790899967509 at 10.4.2.7
>
>    CSeq: 1 REGISTER
>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
> 18-32-35 -0600
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
>    Supported: timer, precondition, path, replaces
>
>    WWW-Authenticate: Digest realm="50.54.12.39",
> nonce="31862298-481b-11e2-84b6-e5c28303057e", algorithm=MD5, qop="auth"
>
>    Content-Length: 0
>
>
>
>    ------------------------------------------------------------------------
>
> recv 707 bytes from udp/[182.79.149.142]:36420 at 07:27:21.094563:
>
>    ------------------------------------------------------------------------
>
>    REGISTER sip:50.54.12.39 SIP/2.0
>
>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport;branch=z9hG4bK84890
>
>    Route: <sip:50.54.12.39;transport=udp;lr>
>
>    Max-Forwards: 70
>
>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>
>    To: <sip:501 at 50.54.12.39>
>
>    Call-ID: 790899967509 at 10.4.2.7
>
>    CSeq: 2 REGISTER
>
>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>
>    Expires: 3600
>
>    User-Agent: SyncPhone 1.0
>
>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>
>
>    Authorization: Digest username="501", realm="50.54.12.39",
> nonce="31862298-481b-11e2-84b6-e5c28303057e", uri="sip:50.54.12.39",
> algorithm=MD5, qop=auth, nc=00000001, cnonce="a5d0aab2288476b3",
> response="eb8ccc768c7f9fee038f0d5dfc599931"
>
>    Content-Length: 0
>
>
>
>    ------------------------------------------------------------------------
>
> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
> 'record_stereo' = 'true'
>
> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
> 'default_gateway' = 'example.com'
>
> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
> 'default_areacode' = '918'
>
> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
> 'transfer_fallback_extension' = 'operator'
>
> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
> 'toll_allow' = 'domestic,international,local'
>
> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
> 'accountcode' = '501'
>
> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
> 'user_context' = 'default'
>
> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
> 'effective_caller_id_name' = 'Extension 501'
>
> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
> 'effective_caller_id_number' = '501'
>
> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
> 'outbound_caller_id_name' = 'FS'
>
> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
> 'outbound_caller_id_number' = '9732386040'
>
> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
> 'callgroup' = 'techsupport'
>
> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:1575 Register:
>
> From:    [501 at 10.252.148.21]
>
> Contact: ["user" <sip:501 at 182.79.149.142:36420;transport=udp>]
>
> Expires: [3600]
>
> send 600 bytes to udp/[182.79.149.142]:36420 at 07:27:21.096965:
>
>    ------------------------------------------------------------------------
>
>    SIP/2.0 200 OK
>
>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport=36420;branch=z9hG4bK84890
>
>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>
>    To: <sip:501 at 50.54.12.39>;tag=a6NK3ZU97F1Qc
>
>    Call-ID: 790899967509 at 10.4.2.7
>
>    CSeq: 2 REGISTER
>
>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>;expires=3600
>
>    Date: Mon, 17 Dec 2012 07:27:21 GMT
>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
> 18-32-35 -0600
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
>    Supported: timer, precondition, path, replaces
>
>    Content-Length: 0
>
>
>
>    ------------------------------------------------------------------------
>
> send 926 bytes to udp/[182.79.149.142]:36420 at 07:27:21.112029:
>
>    ------------------------------------------------------------------------
>
>    NOTIFY sip:501 at 182.79.149.142:36420;transport=udp SIP/2.0
>
>    Via: SIP/2.0/UDP 50.54.12.39;rport;branch=z9hG4bKXXQBeBHUQ5yZp
>
>    Max-Forwards: 70
>
>    From: <sip:501 at 50.54.12.39>;tag=BFFc5tcD5rQar
>
>    To: <sip:501 at 50.54.12.39>
>
>    Call-ID: 09276738-c2be-1230-ec86-22000afc9415
>
>    CSeq: 37528556 NOTIFY
>
>    Contact: <sip:mod_sofia at 50.54.12.39:5060>
>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
> 18-32-35 -0600
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
>    Supported: timer, precondition, path, replaces
>
>    Event: message-summary
>
>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>
>    Subscription-State: terminated;reason=noresource
>
>    Content-Type: application/simple-message-summary
>
>    Content-Length: 64
>
>
>
>    Messages-Waiting: no
>
>    Message-Account: sip:501 at 10.252.148.21
>
>
>
>    ------------------------------------------------------------------------
>
> recv 295 bytes from udp/[182.79.149.142]:36420 at 07:27:21.531463:
>
>    ------------------------------------------------------------------------
>
>    SIP/2.0 405 Method Not Allowed
>
>    Via: SIP/2.0/UDP 50.54.12.39;branch=z9hG4bKXXQBeBHUQ5yZp;rport=5060
>
>    To: <sip:501 at 50.54.12.39>
>
>    From: <sip:501 at 50.54.12.39>;tag=BFFc5tcD5rQar
>
>    Call-ID: 09276738-c2be-1230-ec86-22000afc9415
>
>    CSeq: 37528556 NOTIFY
>
>    Server: SyncPhone 1.0
>
>    Content-Length: 0
>
>
>
>    ------------------------------------------------------------------------
>
>
>
>
> On Mon, Dec 17, 2012 at 2:25 AM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:
>
>> If he is using mod_g729 then the log would throw an error saying that
>> the codec only does passthrough mode, however he hasn't shown that
>> part of the log to us yet
>>
>> On Sun, Dec 16, 2012 at 12:05 PM, Steven Ayre <steveayre at gmail.com>
>> wrote:
>> > What G729 module are you using, and what codec is in use on the other
>> leg?
>> >
>> > G729 needs licenses due to patents.
>> >
>> > mod_g729 is passthrough only but requires no licenses. This is because
>> it
>> > merely forwards the data, but doesn't perform the encoding/transcoding
>> step
>> > that the patents cover. But that means it can't transcode between
>> different
>> > codecs.
>> > mod_com_g729 is the licensed version:
>> http://www.freeswitch.org/node/235
>> >
>> > With the passthrough mod_g729 codec, if the other leg has selected a
>> codec
>> > other than g729 you'll see an error in your logs and it'll hangup with
>> that
>> > reason.
>> >
>> > You can tweak your codec negotiation to avoid this (eg
>> > late-negotiation=true), or you can use the licensed version.
>> >
>> >
>> >
>> >
>> > On 16 December 2012 11:40, Balamurugan Mahendran <b2m at a-cti.com> wrote:
>> >>
>> >> Need help on Codec Negotiation
>> >>
>> >>
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>> Compare
>> >> [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
>> >> send/recv payload to 101
>> >>
>> >>
>> >> 2012-12-16 11:35:52.246422 [DEBUG] switch_channel.c:2852
>> >> (sofia/internal/501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
>> >> 2012-12-16 11:35:52.246422 [NOTICE] sofia.c:5781 Hangup
>> >> sofia/internal/501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>> >>
>> >> 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
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 steveayre at gmail.com  Mon Dec 17 14:22:51 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 17 Dec 2012 11:22:51 +0000
Subject: [Freeswitch-users] G729 - [INCOMPATIBLE_DESTINATION]
In-Reply-To: <CAFiqYukVJW5BzbXeRtJmonAXxudRvuW9=h7SwCQjyyTHk3th6g@mail.gmail.com>
References: <CAP2fbbcixeTy4H+KPbxJQvAhCDWmco5a0jF0XQkGHBtE2Qkmdg@mail.gmail.com>
	<CAFiqYuk+OwUVPCmy8H1Yq5hW-+Hhy40DS1RdRe8T6oBTjeY2LA@mail.gmail.com>
	<CAMWiMy_0f44SwcXZR4CQgP7=RBqrTYW8fQQhAa8m1nR+kKv=oQ@mail.gmail.com>
	<CAP2fbbfczFPcBF5ZF0G_+FL_+n0XZPBcu6KCCSHU1ryny2hBKg@mail.gmail.com>
	<CAFiqYukVJW5BzbXeRtJmonAXxudRvuW9=h7SwCQjyyTHk3th6g@mail.gmail.com>
Message-ID: <CAFiqYu=0ShYLYE+Y3rEtUw6eKp1EjSkxPaM2YkqSnYFksrfqEw@mail.gmail.com>

>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28 18-32-35
> -0600


Your version is pretty old now - almost a year. Are you able to reproduce
this problem on the latest head of 1.2.stable and/or master?

It could be this is a bug that's already been resolved in the past 12
months.

-Steve



On 17 December 2012 11:21, Steven Ayre <steveayre at gmail.com> wrote:

> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>
> This looks odd, although I'm not sure why it's not working.
>
> The remote endpoint is using the static payload type 0 for G729. Normally
> that's reserved for G711 ulaw, while G729 uses 18. But the rtpmap should
> allow overriding that.
>
>> G729:0:8000:20:64000
>
>
> The 64000 rate here also looks odd - it's comparing 64khz to 8khz and I
> suspect that's also a reason why it doesn't recognise it. I don't know
> where that 64000 is coming from though.
>
> -Steve
>
>
>
>
>
> On 17 December 2012 07:31, Balamurugan Mahendran <b2m at a-cti.com> wrote:
>
>> Thanks for the help!!  Please let me know if I missed anything.
>>
>>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,SILK at 8000h
>> ,SILK at 12000h,SILK at 16000h,SILK at 24000h,G7221 at 32000h,G7221 at 16000h
>> ,G722,PCMU,PCMA,GSM"/>
>>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,SILK at 8000h
>> ,SILK at 12000h,SILK at 16000h,SILK at 24000h,G7221 at 32000h,G7221 at 16000h
>> ,G722,PCMU,PCMA,GSM"/>
>>
>>
>>  freeswitch at internal> g729_info
>> Permitted G729 channels: 10
>> Encoders in use: 4
>> Decoders in use: 3
>>
>> freeswitch at internal>
>>
>>
>>
>> recv 742 bytes from udp/[182.79.149.142]:16251 at 07:27:19.212195:
>>
>>
>> ------------------------------------------------------------------------
>>
>>    INVITE sip:500 at 50.54.12.39 SIP/2.0
>>
>>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport;branch=z9hG4bK68891
>>
>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>
>>    Max-Forwards: 70
>>
>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>
>>    To: <sip:500 at 50.54.12.39>
>>
>>    Call-ID: 047707180743 at 10.4.2.7
>>
>>    CSeq: 1 INVITE
>>
>>    Contact: <sip:501 at 182.79.149.142:16251;transport=udp>
>>
>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>
>>    Expires: 3600
>>
>>    User-Agent: SyncPhone 1.0
>>
>>    Content-Length: 244
>>
>>    Content-Type: application/sdp
>>
>>
>>
>>    v=0
>>
>>    o=- 1111 1111 IN IP4 182.79.149.142
>>
>>    s=Session SIP/SDP
>>
>>    c=IN IP4 182.79.149.142
>>
>>    t=0 0
>>
>>    a=sendrecv
>>
>>    a=rtcp:57395 IN IP4 182.79.149.142
>>
>>    m=audio 26474 RTP/AVP 0 101
>>
>>    a=rtpmap:0 G729/8000
>>
>>    a=rtpmap:101 telephone-event/8000
>>
>>    a=fmtp:101 0-15
>>
>>
>> ------------------------------------------------------------------------
>>
>> send 321 bytes to udp/[182.79.149.142]:16251 at 07:27:19.212520:
>>
>>
>> ------------------------------------------------------------------------
>>
>>    SIP/2.0 100 Trying
>>
>>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport=16251;branch=z9hG4bK68891
>>
>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>
>>    To: <sip:500 at 50.54.12.39>
>>
>>    Call-ID: 047707180743 at 10.4.2.7
>>
>>    CSeq: 1 INVITE
>>
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>> 18-32-35 -0600
>>
>>    Content-Length: 0
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:7481 IP 182.79.149.142
>> Approved by acl "domains[]". Access Granted.
>>
>> 2012-12-17 07:27:19.206419 [NOTICE] switch_channel.c:930 New Channel
>> sofia/internal/501 at 50.54.12.39 [30a75cd4-481b-11e2-84b4-e5c28303057e]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_NEW
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:380
>> (sofia/internal/501 at 50.54.12.39) State NEW
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:5502 Channel sofia/internal/
>> 501 at 50.54.12.39 entering state [received][100]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:5513 Remote SDP:
>>
>> v=0
>>
>> o=- 1111 1111 IN IP4 182.79.149.142
>>
>> s=Session SIP/SDP
>>
>> c=IN IP4 182.79.149.142
>>
>> t=0 0
>>
>> a=sendrecv
>>
>> a=rtcp:57395 IN IP4 182.79.149.142
>>
>> m=audio 26474 RTP/AVP 0 101
>>
>> a=rtpmap:0 G729/8000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-15
>>
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>> [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
>> send/recv payload to 101
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_channel.c:2852 (sofia/internal/
>> 501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
>>
>> 2012-12-17 07:27:19.206419 [NOTICE] sofia.c:5781 Hangup sofia/internal/
>> 501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_channel.c:2875 Send signal
>> sofia/internal/501 at 50.54.12.39 [KILL]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send signal
>> sofia/internal/501 at 50.54.12.39 [BREAK]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_HANGUP
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:602
>> (sofia/internal/501 at 50.54.12.39) State HANGUP
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:469 Channel sofia/internal/
>> 501 at 50.54.12.39 hanging up, cause: INCOMPATIBLE_DESTINATION
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:534 Responding to INVITE
>> with: 488
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:47
>> sofia/internal/501 at 50.54.12.39 Standard HANGUP, cause:
>> INCOMPATIBLE_DESTINATION
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:602
>> (sofia/internal/501 at 50.54.12.39) State HANGUP going to sleep
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:393
>> (sofia/internal/501 at 50.54.12.39) State Change CS_HANGUP -> CS_REPORTING
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send signal
>> sofia/internal/501 at 50.54.12.39 [BREAK]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_REPORTING
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:662
>> (sofia/internal/501 at 50.54.12.39) State REPORTING
>>
>> send 816 bytes to udp/[182.79.149.142]:16251 at 07:27:19.215295:
>>
>>
>> ------------------------------------------------------------------------
>>
>>    SIP/2.0 488 Not Acceptable Here
>>
>>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport=16251;branch=z9hG4bK68891
>>
>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>
>>    To: <sip:500 at 50.54.12.39>;tag=8K31Z9S2DymjN
>>
>>    Call-ID: 047707180743 at 10.4.2.7
>>
>>    CSeq: 1 INVITE
>>
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>> 18-32-35 -0600
>>
>>    Accept: application/sdp
>>
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>
>>    Supported: timer, precondition, path, replaces
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:79
>> sofia/internal/501 at 50.54.12.39 Standard REPORTING, cause:
>> INCOMPATIBLE_DESTINATION
>>
>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> sla, include-session-description, presence.winfo, message-summary, refer
>>
>>    Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"2012-12-17
>> 07:27:19.206419 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/
>> 501 at 50.54.12.39) State REPORTING going to sleep
>>
>>
>>    Content-Length: 0
>>
>>    Remote-Party-ID: "500" <sip:500 at 50.54.12.39
>> >;party=calling;privacy=off;screen=no
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:387
>> (sofia/internal/501 at 50.54.12.39) State Change CS_REPORTING -> CS_DESTROY
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send signal
>> sofia/internal/501 at 50.54.12.39 [BREAK]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1380 Session 204
>> (sofia/internal/501 at 50.54.12.39) Locked, Waiting on external entities
>>
>> 2012-12-17 07:27:19.206419 [NOTICE] switch_core_session.c:1398 Session
>> 204 (sofia/internal/501 at 50.54.12.39) Ended
>>
>> 2012-12-17 07:27:19.206419 [NOTICE] switch_core_session.c:1400 Close
>> Channel sofia/internal/501 at 50.54.12.39 [CS_DESTROY]
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:491
>> (sofia/internal/501 at 50.54.12.39) Callstate Change HANGUP -> DOWN
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:494
>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_DESTROY
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:504
>> (sofia/internal/501 at 50.54.12.39) State DESTROY
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:374 sofia/internal/
>> 501 at 50.54.12.39 SOFIA DESTROY
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:86
>> sofia/internal/501 at 50.54.12.39 Standard DESTROY
>>
>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:504
>> (sofia/internal/501 at 50.54.12.39) State DESTROY going to sleep
>>
>> recv 407 bytes from udp/[182.79.149.142]:16251 at 07:27:19.581401:
>>
>>
>> ------------------------------------------------------------------------
>>
>>    ACK sip:500 at 50.54.12.39 SIP/2.0
>>
>>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport;branch=z9hG4bK68891
>>
>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>
>>    Max-Forwards: 70
>>
>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>
>>    To: <sip:500 at 50.54.12.39>;tag=8K31Z9S2DymjN
>>
>>    Call-ID: 047707180743 at 10.4.2.7
>>
>>    CSeq: 1 ACK
>>
>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>
>>    User-Agent: SyncPhone 1.0
>>
>>    Content-Length: 0
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> recv 465 bytes from udp/[182.79.149.142]:36420 at 07:27:20.672450:
>>
>>
>> ------------------------------------------------------------------------
>>
>>    REGISTER sip:50.54.12.39 SIP/2.0
>>
>>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport;branch=z9hG4bK20451
>>
>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>
>>    Max-Forwards: 70
>>
>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>
>>    To: <sip:501 at 50.54.12.39>
>>
>>    Call-ID: 790899967509 at 10.4.2.7
>>
>>    CSeq: 1 REGISTER
>>
>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>
>>
>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>
>>    Expires: 3600
>>
>>    User-Agent: SyncPhone 1.0
>>
>>    Content-Length: 0
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> 2012-12-17 07:27:20.666425 [DEBUG] sofia_reg.c:1417 Send challenge for [
>> 501 at 50.54.12.39]
>>
>> 2012-12-17 07:27:20.666425 [WARNING] sofia_reg.c:1421 SIP auth challenge
>> (REGISTER) on sofia profile 'internal' for [501 at 50.54.12.39] from ip
>> 182.79.149.142
>>
>> send 626 bytes to udp/[182.79.149.142]:36420 at 07:27:20.673200:
>>
>>
>> ------------------------------------------------------------------------
>>
>>    SIP/2.0 401 Unauthorized
>>
>>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport=36420;branch=z9hG4bK20451
>>
>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>
>>    To: <sip:501 at 50.54.12.39>;tag=9vvt14a6a7a5g
>>
>>    Call-ID: 790899967509 at 10.4.2.7
>>
>>    CSeq: 1 REGISTER
>>
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>> 18-32-35 -0600
>>
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>
>>    Supported: timer, precondition, path, replaces
>>
>>    WWW-Authenticate: Digest realm="50.54.12.39",
>> nonce="31862298-481b-11e2-84b6-e5c28303057e", algorithm=MD5, qop="auth"
>>
>>    Content-Length: 0
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> recv 707 bytes from udp/[182.79.149.142]:36420 at 07:27:21.094563:
>>
>>
>> ------------------------------------------------------------------------
>>
>>    REGISTER sip:50.54.12.39 SIP/2.0
>>
>>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport;branch=z9hG4bK84890
>>
>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>
>>    Max-Forwards: 70
>>
>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>
>>    To: <sip:501 at 50.54.12.39>
>>
>>    Call-ID: 790899967509 at 10.4.2.7
>>
>>    CSeq: 2 REGISTER
>>
>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>
>>    Expires: 3600
>>
>>    User-Agent: SyncPhone 1.0
>>
>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>
>>
>>    Authorization: Digest username="501", realm="50.54.12.39",
>> nonce="31862298-481b-11e2-84b6-e5c28303057e", uri="sip:50.54.12.39",
>> algorithm=MD5, qop=auth, nc=00000001, cnonce="a5d0aab2288476b3",
>> response="eb8ccc768c7f9fee038f0d5dfc599931"
>>
>>    Content-Length: 0
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>> 'record_stereo' = 'true'
>>
>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>> 'default_gateway' = 'example.com'
>>
>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>> 'default_areacode' = '918'
>>
>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>> 'transfer_fallback_extension' = 'operator'
>>
>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>> 'toll_allow' = 'domestic,international,local'
>>
>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>> 'accountcode' = '501'
>>
>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>> 'user_context' = 'default'
>>
>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>> 'effective_caller_id_name' = 'Extension 501'
>>
>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>> 'effective_caller_id_number' = '501'
>>
>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>> 'outbound_caller_id_name' = 'FS'
>>
>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>> 'outbound_caller_id_number' = '9732386040'
>>
>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>> 'callgroup' = 'techsupport'
>>
>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:1575 Register:
>>
>> From:    [501 at 10.252.148.21]
>>
>> Contact: ["user" <sip:501 at 182.79.149.142:36420;transport=udp>]
>>
>> Expires: [3600]
>>
>> send 600 bytes to udp/[182.79.149.142]:36420 at 07:27:21.096965:
>>
>>
>> ------------------------------------------------------------------------
>>
>>    SIP/2.0 200 OK
>>
>>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport=36420;branch=z9hG4bK84890
>>
>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>
>>    To: <sip:501 at 50.54.12.39>;tag=a6NK3ZU97F1Qc
>>
>>    Call-ID: 790899967509 at 10.4.2.7
>>
>>    CSeq: 2 REGISTER
>>
>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>;expires=3600
>>
>>    Date: Mon, 17 Dec 2012 07:27:21 GMT
>>
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>> 18-32-35 -0600
>>
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>
>>    Supported: timer, precondition, path, replaces
>>
>>    Content-Length: 0
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> send 926 bytes to udp/[182.79.149.142]:36420 at 07:27:21.112029:
>>
>>
>> ------------------------------------------------------------------------
>>
>>    NOTIFY sip:501 at 182.79.149.142:36420;transport=udp SIP/2.0
>>
>>    Via: SIP/2.0/UDP 50.54.12.39;rport;branch=z9hG4bKXXQBeBHUQ5yZp
>>
>>    Max-Forwards: 70
>>
>>    From: <sip:501 at 50.54.12.39>;tag=BFFc5tcD5rQar
>>
>>    To: <sip:501 at 50.54.12.39>
>>
>>    Call-ID: 09276738-c2be-1230-ec86-22000afc9415
>>
>>    CSeq: 37528556 NOTIFY
>>
>>    Contact: <sip:mod_sofia at 50.54.12.39:5060>
>>
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>> 18-32-35 -0600
>>
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>
>>    Supported: timer, precondition, path, replaces
>>
>>    Event: message-summary
>>
>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> sla, include-session-description, presence.winfo, message-summary, refer
>>
>>    Subscription-State: terminated;reason=noresource
>>
>>    Content-Type: application/simple-message-summary
>>
>>    Content-Length: 64
>>
>>
>>
>>    Messages-Waiting: no
>>
>>    Message-Account: sip:501 at 10.252.148.21
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> recv 295 bytes from udp/[182.79.149.142]:36420 at 07:27:21.531463:
>>
>>
>> ------------------------------------------------------------------------
>>
>>    SIP/2.0 405 Method Not Allowed
>>
>>    Via: SIP/2.0/UDP 50.54.12.39;branch=z9hG4bKXXQBeBHUQ5yZp;rport=5060
>>
>>    To: <sip:501 at 50.54.12.39>
>>
>>    From: <sip:501 at 50.54.12.39>;tag=BFFc5tcD5rQar
>>
>>    Call-ID: 09276738-c2be-1230-ec86-22000afc9415
>>
>>    CSeq: 37528556 NOTIFY
>>
>>    Server: SyncPhone 1.0
>>
>>    Content-Length: 0
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>>
>>
>>
>> On Mon, Dec 17, 2012 at 2:25 AM, curriegrad2004 <curriegrad2004 at gmail.com
>> > wrote:
>>
>>> If he is using mod_g729 then the log would throw an error saying that
>>> the codec only does passthrough mode, however he hasn't shown that
>>> part of the log to us yet
>>>
>>> On Sun, Dec 16, 2012 at 12:05 PM, Steven Ayre <steveayre at gmail.com>
>>> wrote:
>>> > What G729 module are you using, and what codec is in use on the other
>>> leg?
>>> >
>>> > G729 needs licenses due to patents.
>>> >
>>> > mod_g729 is passthrough only but requires no licenses. This is because
>>> it
>>> > merely forwards the data, but doesn't perform the encoding/transcoding
>>> step
>>> > that the patents cover. But that means it can't transcode between
>>> different
>>> > codecs.
>>> > mod_com_g729 is the licensed version:
>>> http://www.freeswitch.org/node/235
>>> >
>>> > With the passthrough mod_g729 codec, if the other leg has selected a
>>> codec
>>> > other than g729 you'll see an error in your logs and it'll hangup with
>>> that
>>> > reason.
>>> >
>>> > You can tweak your codec negotiation to avoid this (eg
>>> > late-negotiation=true), or you can use the licensed version.
>>> >
>>> >
>>> >
>>> >
>>> > On 16 December 2012 11:40, Balamurugan Mahendran <b2m at a-cti.com>
>>> wrote:
>>> >>
>>> >> Need help on Codec Negotiation
>>> >>
>>> >>
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>> Compare
>>> >> [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
>>> >> send/recv payload to 101
>>> >>
>>> >>
>>> >> 2012-12-16 11:35:52.246422 [DEBUG] switch_channel.c:2852
>>> >> (sofia/internal/501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
>>> >> 2012-12-16 11:35:52.246422 [NOTICE] sofia.c:5781 Hangup
>>> >> sofia/internal/501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>> >>
>>> >> 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
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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 steveayre at gmail.com  Mon Dec 17 14:25:56 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 17 Dec 2012 11:25:56 +0000
Subject: [Freeswitch-users] latest git version error
In-Reply-To: <50CE271A.6090701@quentustech.com>
References: <592A9CF93E12394E8472A6CC66E66BF233BB3A@Mail-Kilo.squay.com>
	<50CE271A.6090701@quentustech.com>
Message-ID: <CAFiqYu=E+C9oGpMb2r9+LQWa7PoOtAyVnBjOG1cvB6Rz_kx7-g@mail.gmail.com>

He did:

> git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out

It tried v6 after v4 had failed.

git.freeswitch.org resolves to vc-01.ptk.freeswitch.org (198.22.64.222) for
me.

I've just tried pinging this and was getting 100% packet loss, then it
started to reply - so perhaps there is a problem with the server/route? Or
maybe the admins just did a reboot.

-Steve


**


On 16 December 2012 19:55, William King <william.king at quentustech.com>wrote:

> Your machine is trying to connect to the git server over ipv6. Currently
> the ipv6 route is still down. Try the ipv4 route.
>
> 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 12/16/2012 11:27 AM, Archana Venugopan wrote:
> > Hi,
> >
> > When I try to fetch latest git version am facing with the below error
> > and am not able to get the git. From where can I get freeswitch latest
> > version?
> >
> >
> >
> > [root at VECTONE-CLOUDE src]# git clone git://
> git.freeswitch.org/freeswitch.git
> >
> > Initialized empty Git repository in /usr/local/src/freeswitch/.git/
> >
> > git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out
> >
> > git.freeswitch.org[0: 2606:d900:0:24:1024:ff:fe00:1234]: errno=Network
> > is unreachable
> >
> > fatal: unable to connect a socket (Network is unreachable)
> >
> >
> >
> > Regards,
> >
> > Archana.V
> >
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 17 14:27:11 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 17 Dec 2012 11:27:11 +0000
Subject: [Freeswitch-users] latest git version error
In-Reply-To: <CAFiqYu=E+C9oGpMb2r9+LQWa7PoOtAyVnBjOG1cvB6Rz_kx7-g@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF233BB3A@Mail-Kilo.squay.com>
	<50CE271A.6090701@quentustech.com>
	<CAFiqYu=E+C9oGpMb2r9+LQWa7PoOtAyVnBjOG1cvB6Rz_kx7-g@mail.gmail.com>
Message-ID: <CAFiqYun3=6odEQhYa8nLNyJBR0JrUXC7h96XQq8qfX0hbVrm1Q@mail.gmail.com>

In any case... git clone works for me from here (at least now) too.

Archana, can you try again? Perhaps it'll work now.

-Steve



On 17 December 2012 11:25, Steven Ayre <steveayre at gmail.com> wrote:

> He did:
>
>> git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out
>
> It tried v6 after v4 had failed.
>
> git.freeswitch.org resolves to vc-01.ptk.freeswitch.org (198.22.64.222)
> for me.
>
> I've just tried pinging this and was getting 100% packet loss, then it
> started to reply - so perhaps there is a problem with the server/route? Or
> maybe the admins just did a reboot.
>
>  -Steve
>
>
> **
>
>
> On 16 December 2012 19:55, William King <william.king at quentustech.com>wrote:
>
>> Your machine is trying to connect to the git server over ipv6. Currently
>> the ipv6 route is still down. Try the ipv4 route.
>>
>> 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 12/16/2012 11:27 AM, Archana Venugopan wrote:
>> > Hi,
>> >
>> > When I try to fetch latest git version am facing with the below error
>> > and am not able to get the git. From where can I get freeswitch latest
>> > version?
>> >
>> >
>> >
>> > [root at VECTONE-CLOUDE src]# git clone git://
>> git.freeswitch.org/freeswitch.git
>> >
>> > Initialized empty Git repository in /usr/local/src/freeswitch/.git/
>> >
>> > git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out
>> >
>> > git.freeswitch.org[0: 2606:d900:0:24:1024:ff:fe00:1234]: errno=Network
>> > is unreachable
>> >
>> > fatal: unable to connect a socket (Network is unreachable)
>> >
>> >
>> >
>> > Regards,
>> >
>> > Archana.V
>> >
>> >
>> >
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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  Mon Dec 17 14:30:47 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Mon, 17 Dec 2012 17:00:47 +0530
Subject: [Freeswitch-users] G729 - [INCOMPATIBLE_DESTINATION]
In-Reply-To: <CAFiqYu=0ShYLYE+Y3rEtUw6eKp1EjSkxPaM2YkqSnYFksrfqEw@mail.gmail.com>
References: <CAP2fbbcixeTy4H+KPbxJQvAhCDWmco5a0jF0XQkGHBtE2Qkmdg@mail.gmail.com>
	<CAFiqYuk+OwUVPCmy8H1Yq5hW-+Hhy40DS1RdRe8T6oBTjeY2LA@mail.gmail.com>
	<CAMWiMy_0f44SwcXZR4CQgP7=RBqrTYW8fQQhAa8m1nR+kKv=oQ@mail.gmail.com>
	<CAP2fbbfczFPcBF5ZF0G_+FL_+n0XZPBcu6KCCSHU1ryny2hBKg@mail.gmail.com>
	<CAFiqYukVJW5BzbXeRtJmonAXxudRvuW9=h7SwCQjyyTHk3th6g@mail.gmail.com>
	<CAFiqYu=0ShYLYE+Y3rEtUw6eKp1EjSkxPaM2YkqSnYFksrfqEw@mail.gmail.com>
Message-ID: <CAP2fbbeZaQG3ai6+GVyyY08=rJS=L1FJWmNk4wmb_j2J_Lhtbg@mail.gmail.com>

Yes - both are same.  Actually call is from Android SIP client. Trying to
make outbound call from it - Getting incoming call with NO issues.

Thanks,
Bala

On Mon, Dec 17, 2012 at 4:52 PM, Steven Ayre <steveayre at gmail.com> wrote:

> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28 18-32-35
>> -0600
>
>
> Your version is pretty old now - almost a year. Are you able to reproduce
> this problem on the latest head of 1.2.stable and/or master?
>
> It could be this is a bug that's already been resolved in the past 12
> months.
>
> -Steve
>
>
>
> On 17 December 2012 11:21, Steven Ayre <steveayre at gmail.com> wrote:
>
>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>>
>> This looks odd, although I'm not sure why it's not working.
>>
>> The remote endpoint is using the static payload type 0 for G729. Normally
>> that's reserved for G711 ulaw, while G729 uses 18. But the rtpmap should
>> allow overriding that.
>>
>>> G729:0:8000:20:64000
>>
>>
>> The 64000 rate here also looks odd - it's comparing 64khz to 8khz and I
>> suspect that's also a reason why it doesn't recognise it. I don't know
>> where that 64000 is coming from though.
>>
>> -Steve
>>
>>
>>
>>
>>
>> On 17 December 2012 07:31, Balamurugan Mahendran <b2m at a-cti.com> wrote:
>>
>>> Thanks for the help!!  Please let me know if I missed anything.
>>>
>>>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,SILK at 8000h
>>> ,SILK at 12000h,SILK at 16000h,SILK at 24000h,G7221 at 32000h,G7221 at 16000h
>>> ,G722,PCMU,PCMA,GSM"/>
>>>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,SILK at 8000h
>>> ,SILK at 12000h,SILK at 16000h,SILK at 24000h,G7221 at 32000h,G7221 at 16000h
>>> ,G722,PCMU,PCMA,GSM"/>
>>>
>>>
>>>  freeswitch at internal> g729_info
>>> Permitted G729 channels: 10
>>> Encoders in use: 4
>>> Decoders in use: 3
>>>
>>> freeswitch at internal>
>>>
>>>
>>>
>>> recv 742 bytes from udp/[182.79.149.142]:16251 at 07:27:19.212195:
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>>    INVITE sip:500 at 50.54.12.39 SIP/2.0
>>>
>>>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport;branch=z9hG4bK68891
>>>
>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>
>>>    Max-Forwards: 70
>>>
>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>
>>>    To: <sip:500 at 50.54.12.39>
>>>
>>>    Call-ID: 047707180743 at 10.4.2.7
>>>
>>>    CSeq: 1 INVITE
>>>
>>>    Contact: <sip:501 at 182.79.149.142:16251;transport=udp>
>>>
>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>
>>>    Expires: 3600
>>>
>>>    User-Agent: SyncPhone 1.0
>>>
>>>    Content-Length: 244
>>>
>>>    Content-Type: application/sdp
>>>
>>>
>>>
>>>    v=0
>>>
>>>    o=- 1111 1111 IN IP4 182.79.149.142
>>>
>>>    s=Session SIP/SDP
>>>
>>>    c=IN IP4 182.79.149.142
>>>
>>>    t=0 0
>>>
>>>    a=sendrecv
>>>
>>>    a=rtcp:57395 IN IP4 182.79.149.142
>>>
>>>    m=audio 26474 RTP/AVP 0 101
>>>
>>>    a=rtpmap:0 G729/8000
>>>
>>>    a=rtpmap:101 telephone-event/8000
>>>
>>>    a=fmtp:101 0-15
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> send 321 bytes to udp/[182.79.149.142]:16251 at 07:27:19.212520:
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>>    SIP/2.0 100 Trying
>>>
>>>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport=16251;branch=z9hG4bK68891
>>>
>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>
>>>    To: <sip:500 at 50.54.12.39>
>>>
>>>    Call-ID: 047707180743 at 10.4.2.7
>>>
>>>    CSeq: 1 INVITE
>>>
>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>> 18-32-35 -0600
>>>
>>>    Content-Length: 0
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:7481 IP 182.79.149.142
>>> Approved by acl "domains[]". Access Granted.
>>>
>>> 2012-12-17 07:27:19.206419 [NOTICE] switch_channel.c:930 New Channel
>>> sofia/internal/501 at 50.54.12.39 [30a75cd4-481b-11e2-84b4-e5c28303057e]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_NEW
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:380
>>> (sofia/internal/501 at 50.54.12.39) State NEW
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:5502 Channel sofia/internal/
>>> 501 at 50.54.12.39 entering state [received][100]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:5513 Remote SDP:
>>>
>>> v=0
>>>
>>> o=- 1111 1111 IN IP4 182.79.149.142
>>>
>>> s=Session SIP/SDP
>>>
>>> c=IN IP4 182.79.149.142
>>>
>>> t=0 0
>>>
>>> a=sendrecv
>>>
>>> a=rtcp:57395 IN IP4 182.79.149.142
>>>
>>> m=audio 26474 RTP/AVP 0 101
>>>
>>> a=rtpmap:0 G729/8000
>>>
>>> a=rtpmap:101 telephone-event/8000
>>>
>>> a=fmtp:101 0-15
>>>
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>> [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
>>> send/recv payload to 101
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_channel.c:2852 (sofia/internal/
>>> 501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
>>>
>>> 2012-12-17 07:27:19.206419 [NOTICE] sofia.c:5781 Hangup sofia/internal/
>>> 501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_channel.c:2875 Send signal
>>> sofia/internal/501 at 50.54.12.39 [KILL]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send
>>> signal sofia/internal/501 at 50.54.12.39 [BREAK]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_HANGUP
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:602
>>> (sofia/internal/501 at 50.54.12.39) State HANGUP
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:469 Channel
>>> sofia/internal/501 at 50.54.12.39 hanging up, cause:
>>> INCOMPATIBLE_DESTINATION
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:534 Responding to INVITE
>>> with: 488
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:47
>>> sofia/internal/501 at 50.54.12.39 Standard HANGUP, cause:
>>> INCOMPATIBLE_DESTINATION
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:602
>>> (sofia/internal/501 at 50.54.12.39) State HANGUP going to sleep
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:393
>>> (sofia/internal/501 at 50.54.12.39) State Change CS_HANGUP -> CS_REPORTING
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send
>>> signal sofia/internal/501 at 50.54.12.39 [BREAK]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_REPORTING
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:662
>>> (sofia/internal/501 at 50.54.12.39) State REPORTING
>>>
>>> send 816 bytes to udp/[182.79.149.142]:16251 at 07:27:19.215295:
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>>    SIP/2.0 488 Not Acceptable Here
>>>
>>>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport=16251;branch=z9hG4bK68891
>>>
>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>
>>>    To: <sip:500 at 50.54.12.39>;tag=8K31Z9S2DymjN
>>>
>>>    Call-ID: 047707180743 at 10.4.2.7
>>>
>>>    CSeq: 1 INVITE
>>>
>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>> 18-32-35 -0600
>>>
>>>    Accept: application/sdp
>>>
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>
>>>    Supported: timer, precondition, path, replaces
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:79
>>> sofia/internal/501 at 50.54.12.39 Standard REPORTING, cause:
>>> INCOMPATIBLE_DESTINATION
>>>
>>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>> sla, include-session-description, presence.winfo, message-summary, refer
>>>
>>>    Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"2012-12-17
>>> 07:27:19.206419 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/
>>> 501 at 50.54.12.39) State REPORTING going to sleep
>>>
>>>
>>>    Content-Length: 0
>>>
>>>    Remote-Party-ID: "500" <sip:500 at 50.54.12.39
>>> >;party=calling;privacy=off;screen=no
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:387
>>> (sofia/internal/501 at 50.54.12.39) State Change CS_REPORTING -> CS_DESTROY
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send
>>> signal sofia/internal/501 at 50.54.12.39 [BREAK]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1380 Session
>>> 204 (sofia/internal/501 at 50.54.12.39) Locked, Waiting on external
>>> entities
>>>
>>> 2012-12-17 07:27:19.206419 [NOTICE] switch_core_session.c:1398 Session
>>> 204 (sofia/internal/501 at 50.54.12.39) Ended
>>>
>>> 2012-12-17 07:27:19.206419 [NOTICE] switch_core_session.c:1400 Close
>>> Channel sofia/internal/501 at 50.54.12.39 [CS_DESTROY]
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:491
>>> (sofia/internal/501 at 50.54.12.39) Callstate Change HANGUP -> DOWN
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:494
>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_DESTROY
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:504
>>> (sofia/internal/501 at 50.54.12.39) State DESTROY
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:374 sofia/internal/
>>> 501 at 50.54.12.39 SOFIA DESTROY
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:86
>>> sofia/internal/501 at 50.54.12.39 Standard DESTROY
>>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:504
>>> (sofia/internal/501 at 50.54.12.39) State DESTROY going to sleep
>>>
>>> recv 407 bytes from udp/[182.79.149.142]:16251 at 07:27:19.581401:
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>>    ACK sip:500 at 50.54.12.39 SIP/2.0
>>>
>>>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport;branch=z9hG4bK68891
>>>
>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>
>>>    Max-Forwards: 70
>>>
>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>
>>>    To: <sip:500 at 50.54.12.39>;tag=8K31Z9S2DymjN
>>>
>>>    Call-ID: 047707180743 at 10.4.2.7
>>>
>>>    CSeq: 1 ACK
>>>
>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>
>>>    User-Agent: SyncPhone 1.0
>>>
>>>    Content-Length: 0
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> recv 465 bytes from udp/[182.79.149.142]:36420 at 07:27:20.672450:
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>>    REGISTER sip:50.54.12.39 SIP/2.0
>>>
>>>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport;branch=z9hG4bK20451
>>>
>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>
>>>    Max-Forwards: 70
>>>
>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>
>>>    To: <sip:501 at 50.54.12.39>
>>>
>>>    Call-ID: 790899967509 at 10.4.2.7
>>>
>>>    CSeq: 1 REGISTER
>>>
>>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>
>>>
>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>
>>>    Expires: 3600
>>>
>>>    User-Agent: SyncPhone 1.0
>>>
>>>    Content-Length: 0
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> 2012-12-17 07:27:20.666425 [DEBUG] sofia_reg.c:1417 Send challenge for [
>>> 501 at 50.54.12.39]
>>>
>>> 2012-12-17 07:27:20.666425 [WARNING] sofia_reg.c:1421 SIP auth challenge
>>> (REGISTER) on sofia profile 'internal' for [501 at 50.54.12.39] from ip
>>> 182.79.149.142
>>>
>>> send 626 bytes to udp/[182.79.149.142]:36420 at 07:27:20.673200:
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>>    SIP/2.0 401 Unauthorized
>>>
>>>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport=36420;branch=z9hG4bK20451
>>>
>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>
>>>    To: <sip:501 at 50.54.12.39>;tag=9vvt14a6a7a5g
>>>
>>>    Call-ID: 790899967509 at 10.4.2.7
>>>
>>>    CSeq: 1 REGISTER
>>>
>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>> 18-32-35 -0600
>>>
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>
>>>    Supported: timer, precondition, path, replaces
>>>
>>>    WWW-Authenticate: Digest realm="50.54.12.39",
>>> nonce="31862298-481b-11e2-84b6-e5c28303057e", algorithm=MD5, qop="auth"
>>>
>>>    Content-Length: 0
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> recv 707 bytes from udp/[182.79.149.142]:36420 at 07:27:21.094563:
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>>    REGISTER sip:50.54.12.39 SIP/2.0
>>>
>>>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport;branch=z9hG4bK84890
>>>
>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>
>>>    Max-Forwards: 70
>>>
>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>
>>>    To: <sip:501 at 50.54.12.39>
>>>
>>>    Call-ID: 790899967509 at 10.4.2.7
>>>
>>>    CSeq: 2 REGISTER
>>>
>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>
>>>    Expires: 3600
>>>
>>>    User-Agent: SyncPhone 1.0
>>>
>>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>
>>>
>>>    Authorization: Digest username="501", realm="50.54.12.39",
>>> nonce="31862298-481b-11e2-84b6-e5c28303057e", uri="sip:50.54.12.39",
>>> algorithm=MD5, qop=auth, nc=00000001, cnonce="a5d0aab2288476b3",
>>> response="eb8ccc768c7f9fee038f0d5dfc599931"
>>>
>>>    Content-Length: 0
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>> 'record_stereo' = 'true'
>>>
>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>> 'default_gateway' = 'example.com'
>>>
>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>> 'default_areacode' = '918'
>>>
>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>> 'transfer_fallback_extension' = 'operator'
>>>
>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>> 'toll_allow' = 'domestic,international,local'
>>>
>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>> 'accountcode' = '501'
>>>
>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>> 'user_context' = 'default'
>>>
>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>> 'effective_caller_id_name' = 'Extension 501'
>>>
>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>> 'effective_caller_id_number' = '501'
>>>
>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>> 'outbound_caller_id_name' = 'FS'
>>>
>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>> 'outbound_caller_id_number' = '9732386040'
>>>
>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>> 'callgroup' = 'techsupport'
>>>
>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:1575 Register:
>>>
>>> From:    [501 at 10.252.148.21]
>>>
>>> Contact: ["user" <sip:501 at 182.79.149.142:36420;transport=udp>]
>>>
>>> Expires: [3600]
>>>
>>> send 600 bytes to udp/[182.79.149.142]:36420 at 07:27:21.096965:
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>>    SIP/2.0 200 OK
>>>
>>>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport=36420;branch=z9hG4bK84890
>>>
>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>
>>>    To: <sip:501 at 50.54.12.39>;tag=a6NK3ZU97F1Qc
>>>
>>>    Call-ID: 790899967509 at 10.4.2.7
>>>
>>>    CSeq: 2 REGISTER
>>>
>>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>;expires=3600
>>>
>>>    Date: Mon, 17 Dec 2012 07:27:21 GMT
>>>
>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>> 18-32-35 -0600
>>>
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>
>>>    Supported: timer, precondition, path, replaces
>>>
>>>    Content-Length: 0
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> send 926 bytes to udp/[182.79.149.142]:36420 at 07:27:21.112029:
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>>    NOTIFY sip:501 at 182.79.149.142:36420;transport=udp SIP/2.0
>>>
>>>    Via: SIP/2.0/UDP 50.54.12.39;rport;branch=z9hG4bKXXQBeBHUQ5yZp
>>>
>>>    Max-Forwards: 70
>>>
>>>    From: <sip:501 at 50.54.12.39>;tag=BFFc5tcD5rQar
>>>
>>>    To: <sip:501 at 50.54.12.39>
>>>
>>>    Call-ID: 09276738-c2be-1230-ec86-22000afc9415
>>>
>>>    CSeq: 37528556 NOTIFY
>>>
>>>    Contact: <sip:mod_sofia at 50.54.12.39:5060>
>>>
>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>> 18-32-35 -0600
>>>
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>
>>>    Supported: timer, precondition, path, replaces
>>>
>>>    Event: message-summary
>>>
>>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>> sla, include-session-description, presence.winfo, message-summary, refer
>>>
>>>    Subscription-State: terminated;reason=noresource
>>>
>>>    Content-Type: application/simple-message-summary
>>>
>>>    Content-Length: 64
>>>
>>>
>>>
>>>    Messages-Waiting: no
>>>
>>>    Message-Account: sip:501 at 10.252.148.21
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> recv 295 bytes from udp/[182.79.149.142]:36420 at 07:27:21.531463:
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>>    SIP/2.0 405 Method Not Allowed
>>>
>>>    Via: SIP/2.0/UDP 50.54.12.39;branch=z9hG4bKXXQBeBHUQ5yZp;rport=5060
>>>
>>>    To: <sip:501 at 50.54.12.39>
>>>
>>>    From: <sip:501 at 50.54.12.39>;tag=BFFc5tcD5rQar
>>>
>>>    Call-ID: 09276738-c2be-1230-ec86-22000afc9415
>>>
>>>    CSeq: 37528556 NOTIFY
>>>
>>>    Server: SyncPhone 1.0
>>>
>>>    Content-Length: 0
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>>
>>>
>>>
>>> On Mon, Dec 17, 2012 at 2:25 AM, curriegrad2004 <
>>> curriegrad2004 at gmail.com> wrote:
>>>
>>>> If he is using mod_g729 then the log would throw an error saying that
>>>> the codec only does passthrough mode, however he hasn't shown that
>>>> part of the log to us yet
>>>>
>>>> On Sun, Dec 16, 2012 at 12:05 PM, Steven Ayre <steveayre at gmail.com>
>>>> wrote:
>>>> > What G729 module are you using, and what codec is in use on the other
>>>> leg?
>>>> >
>>>> > G729 needs licenses due to patents.
>>>> >
>>>> > mod_g729 is passthrough only but requires no licenses. This is
>>>> because it
>>>> > merely forwards the data, but doesn't perform the
>>>> encoding/transcoding step
>>>> > that the patents cover. But that means it can't transcode between
>>>> different
>>>> > codecs.
>>>> > mod_com_g729 is the licensed version:
>>>> http://www.freeswitch.org/node/235
>>>> >
>>>> > With the passthrough mod_g729 codec, if the other leg has selected a
>>>> codec
>>>> > other than g729 you'll see an error in your logs and it'll hangup
>>>> with that
>>>> > reason.
>>>> >
>>>> > You can tweak your codec negotiation to avoid this (eg
>>>> > late-negotiation=true), or you can use the licensed version.
>>>> >
>>>> >
>>>> >
>>>> >
>>>> > On 16 December 2012 11:40, Balamurugan Mahendran <b2m at a-cti.com>
>>>> wrote:
>>>> >>
>>>> >> Need help on Codec Negotiation
>>>> >>
>>>> >>
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare
>>>> >> [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
>>>> >> send/recv payload to 101
>>>> >>
>>>> >>
>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] switch_channel.c:2852
>>>> >> (sofia/internal/501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
>>>> >> 2012-12-16 11:35:52.246422 [NOTICE] sofia.c:5781 Hangup
>>>> >> sofia/internal/501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>>> >>
>>>> >> 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
>>>> >>
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >
>>>> >
>>>> >
>>>> _________________________________________________________________________
>>>> > Professional FreeSWITCH Consulting 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
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 17 14:34:56 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 17 Dec 2012 11:34:56 +0000
Subject: [Freeswitch-users] What is "Recovery On Timer Expire"?
In-Reply-To: <2148350.szYfaMxe3S@sos>
References: <CALNZuw=z2ANmWmB6Fz3nximTyoPqUSWLJ=-ob55gdT6Q=_qC4Q@mail.gmail.com>
	<CAMbW9oJS32=e2=r6CrMx6X8n6afRXFudyQq2kAAR_qOp9NCPyg@mail.gmail.com>
	<CALNZuw=nShpsNpUP1C-ibAHVH+NKxf7keo+Me+jXJgw7gDmanQ@mail.gmail.com>
	<2148350.szYfaMxe3S@sos>
Message-ID: <CAFiqYumAtHwHUyMgoUVod=ExAqoDy9PyrCS=rXeAQWpBKa=1JQ@mail.gmail.com>

I believe it can also cover no 100 Trying reply to the INVITE.

There's also a no RTP timeout in FS(disabled by default), not sure the
cause it'll raise but it might be that one.

You might also be receiving this cause from upstream - SIP '408 Request
Timeout' response, or ISDN cause set in the Reason header.

-Steve



On 16 December 2012 23:27, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:

> I bet there is no SIP 180 or 183 after INVITE within progress_timeout
> interval.
>
> On Monday 17 December 2012 01:15:21 Avi Marcus wrote:
> > Not this case.
> > There is no NAT involved - my FS is on public and it's to a carrier to
> PSTN.
> > Also, the call never connected: there's no billsec and the entire length
> of
> > the call was ringing.
> >
> > -Avi
> >
> >
> > On Mon, Dec 17, 2012 at 1:00 AM, Andrew Cassidy <
> >
> > andrew at cassidywebservices.co.uk> wrote:
> > > Basically, FreeSWITCH sends occasional re-invites to the remote party
> to
> > > make sure the call is still valid. If no response is received it will
> > > assume the call has ended and drop the call.
> > > This behavior is completely valid. However, this can also show up
> > > potential
> > > NAT/firewall issues, where the NAT mapping is being lost.
> > >
> > > I have come across people that deliberately block these re-invites,
> > > who obviously have a fetish for zombie calls clogging up their
> systems...
> > >
> > > On 16 December 2012 21:53, Avi Marcus <avi at avimarcus.net> wrote:
> > >> I'm trying to clear up my CDRs from errors, if possible... I see a
> bunch
> > >> of Recovery On Timer Expire and they mostly seem to be after 45
> seconds
> > >> of
> > >> ring time (there is progress).
> > >> Does that basically mean it rang for 40+ seconds before some timer
> > >> somewhere kicked in and ended the call?
> > >> So it's basically a "no answer" or "no user response"?
> > >>
> > >> Thanks,
> > >> -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
> > >>
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at 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
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 steveayre at gmail.com  Mon Dec 17 14:42:02 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 17 Dec 2012 11:42:02 +0000
Subject: [Freeswitch-users] ODBC error
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233BB74@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233BB74@Mail-Kilo.squay.com>
Message-ID: <CAFiqYun5nYjUvCTY8=F-29P5hRi76RrkTrzLRmBETK2mrPgo+w@mail.gmail.com>

>
> ODBC NOT AVAILABLE


This means your FS was compiled without ODBC support.

Check you have the unixodbc development package installed when building
(usually unixodbc-dev or unixodbc-devel).

It should be auto-detected if installed, but to be sure run configure with
the --enable-core-odbc-support to force it to look for it, that'll should
then give you an error if it can't find it.



There has also been some changes to DSNs since v1.2.4 of which you should
be aware.

DSNs of format "datasourcename" are no longer supported. They should be
changed to one of:
"datasourcename::"
"datasourcename:username:password"
"odbc://datasourcename"
"odbc://datasourcename::"
"odbc://datasourcename:username:password"

http://wiki.freeswitch.org/wiki/DSN

I prefer using the odbc:// prefix  - it seems less likely that it would
change long-term

-Steve



On 16 December 2012 20:55, Archana Venugopan <a.venugopan at mundio.com> wrote:

>  Hi,****
>
> Can anyone please tell me why I am getting this error inspite ?isql ?v
> smepbx smepbx smeswitch? was working. ****
>
> ** **
>
> ** **
>
> [CRIT] switch_core_sqldb.c:433 Failure! ODBC NOT AVAILABLE! Can't connect
> to DSN smepbx****
>
> ** **
>
> Regards,****
>
> Archana.V****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 a.venugopan at mundio.com  Mon Dec 17 15:32:35 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Mon, 17 Dec 2012 12:32:35 +0000
Subject: [Freeswitch-users] latest git version error
In-Reply-To: <CAFiqYun3=6odEQhYa8nLNyJBR0JrUXC7h96XQq8qfX0hbVrm1Q@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF233BB3A@Mail-Kilo.squay.com>
	<50CE271A.6090701@quentustech.com>
	<CAFiqYu=E+C9oGpMb2r9+LQWa7PoOtAyVnBjOG1cvB6Rz_kx7-g@mail.gmail.com>
	<CAFiqYun3=6odEQhYa8nLNyJBR0JrUXC7h96XQq8qfX0hbVrm1Q@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233BCB5@Mail-Kilo.squay.com>

Thanks Steven. I got the packages:)

Regards,
Archana.V

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 17 December 2012 11:27
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] latest git version error

In any case... git clone works for me from here (at least now) too.

Archana, can you try again? Perhaps it'll work now.

-Steve


On 17 December 2012 11:25, Steven Ayre <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
He did:
git.freeswitch.org<http://git.freeswitch.org/>[0: 198.22.64.222]: errno=Connection timed out
It tried v6 after v4 had failed.

git.freeswitch.org<http://git.freeswitch.org> resolves to vc-01.ptk.freeswitch.org<http://vc-01.ptk.freeswitch.org> (198.22.64.222) for me.

I've just tried pinging this and was getting 100% packet loss, then it started to reply - so perhaps there is a problem with the server/route? Or maybe the admins just did a reboot.

-Steve




On 16 December 2012 19:55, William King <william.king at quentustech.com<mailto:william.king at quentustech.com>> wrote:
Your machine is trying to connect to the git server over ipv6. Currently
the ipv6 route is still down. Try the ipv4 route.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337<tel:%28877%29%20211-9337>
Office: (206) 388-4772<tel:%28206%29%20388-4772>
Cell:   (253) 686-5518<tel:%28253%29%20686-5518>
william.king at quentustech.com<mailto:william.king at quentustech.com>

On 12/16/2012 11:27 AM, Archana Venugopan wrote:
> Hi,
>
> When I try to fetch latest git version am facing with the below error
> and am not able to get the git. From where can I get freeswitch latest
> version?
>
>
>
> [root at VECTONE-CLOUDE src]# git clone git://git.freeswitch.org/freeswitch.git<http://git.freeswitch.org/freeswitch.git>
>
> Initialized empty Git repository in /usr/local/src/freeswitch/.git/
>
> git.freeswitch.org<http://git.freeswitch.org>[0: 198.22.64.222]: errno=Connection timed out
>
> git.freeswitch.org<http://git.freeswitch.org>[0: 2606:d900:0:24:1024:ff:fe00:1234]: errno=Network
> is unreachable
>
> fatal: unable to connect a socket (Network is unreachable)
>
>
>
> Regards,
>
> Archana.V
>
>
>
>
>
> _________________________________________________________________________
> 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

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 a.venugopan at mundio.com  Mon Dec 17 15:32:44 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Mon, 17 Dec 2012 12:32:44 +0000
Subject: [Freeswitch-users] ODBC error
In-Reply-To: <CAFiqYun5nYjUvCTY8=F-29P5hRi76RrkTrzLRmBETK2mrPgo+w@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF233BB74@Mail-Kilo.squay.com>
	<CAFiqYun5nYjUvCTY8=F-29P5hRi76RrkTrzLRmBETK2mrPgo+w@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233BCBB@Mail-Kilo.squay.com>

Thanks again. ODBC was the issue.

Regards,
Archana.V

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 17 December 2012 11:42
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] ODBC error

ODBC NOT AVAILABLE

This means your FS was compiled without ODBC support.

Check you have the unixodbc development package installed when building (usually unixodbc-dev or unixodbc-devel).

It should be auto-detected if installed, but to be sure run configure with the --enable-core-odbc-support to force it to look for it, that'll should then give you an error if it can't find it.



There has also been some changes to DSNs since v1.2.4 of which you should be aware.

DSNs of format "datasourcename" are no longer supported. They should be changed to one of:
"datasourcename::"
"datasourcename:username:password"
"odbc://datasourcename"
"odbc://datasourcename::"
"odbc://datasourcename:username:password"

http://wiki.freeswitch.org/wiki/DSN

I prefer using the odbc:// prefix  - it seems less likely that it would change long-term

-Steve


On 16 December 2012 20:55, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
Can anyone please tell me why I am getting this error inspite 'isql -v smepbx smepbx smeswitch' was working.


[CRIT] switch_core_sqldb.c:433 Failure! ODBC NOT AVAILABLE! Can't connect to DSN smepbx

Regards,
Archana.V


_________________________________________________________________________
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

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From jnvines at gmail.com  Mon Dec 17 16:12:27 2012
From: jnvines at gmail.com (Nick Vines)
Date: Mon, 17 Dec 2012 08:12:27 -0500
Subject: [Freeswitch-users] Conference Calls 404
Message-ID: <CA+NjZ2vwfxyDiG753NPABA9ZKzb7LhqKs03iw+bHvYeoAudAAQ@mail.gmail.com>

The two most recent calls, Nov 28 and Dec 5, torrent links are broken. Does
anyone have the torrent file or the recordings?

Thanks,
Nick
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From cal.leeming at simplicitymedialtd.co.uk  Mon Dec 17 17:17:40 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Mon, 17 Dec 2012 14:17:40 +0000
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CABH8QX0WNTaMHXbFeJ8ip+D7ADn+b45nc2=fSHdf5F8OpWzMNQ@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<CAFiqYu=wL6+yk67AzzfF_HKrMBBreqeb5jybX7r8LjoXBHVs0A@mail.gmail.com>
	<CAHKQagFbEdS+wLt-CH7t_=B5UESRQkVVnFf3HSt8EB4=Gq=pOA@mail.gmail.com>
	<CABH8QX0WNTaMHXbFeJ8ip+D7ADn+b45nc2=fSHdf5F8OpWzMNQ@mail.gmail.com>
Message-ID: <CAHKQagH8uu1nOBvgZ+nr_o0dV4u+G=3wfoaEx9pfxZd30F7YWw@mail.gmail.com>

Any other experiences/thoughts from others on this?

Feedback so far has been great, lets keep it coming guys!

Cal

On Sun, Dec 16, 2012 at 9:52 PM, Jo?o Mesquita <jmesquita at freeswitch.org>wrote:

> My take on the subject is that we are trying to tackle 2 different
> problems at once.
>
> On one hand we have how NAT works and what problems it create on a VoIP
> world. I really was not able to find any condensed documentation on the
> internet that would describe with VoIP in mind what a admin need to know
> about SIP packets and NAT handling. If there was one, we could just add a
> link to the wiki page. Since NAT has no "one configuration" fix, user NEEDS
> to know about it in order to fix his own scenario.
>
> On the other hand, we have how FS particularly deals with NAT (client and
> server side). I think there is more documentation to be added/created on
> that end as well. NAT is a complicated matter indeed and understanding
> Sofia profiles alone is a challenge. Add NAT to the mix and the
> configuration can look like black magic.
>
> So, question is, who can elaborate/contribute/find a definitive guide to
> NAT so we can lecture users and who knows all about the NAT handling
> internals so we can document each and every option available?
>
> Jo?o Mesquita
>
>
>
> On Sun, Dec 16, 2012 at 6:21 PM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> It seems that most of us agree there is no single answer to fix NAT
>> problems - the double-NAT is something I hadn't thought about.
>>
>> Sean mentioned having a checklist of approaches, which would be good
>> addition for the documentation fix.
>>
>> I agree that enabling NAT HACK by default could break clients that are
>> functioning normally, but not if it is only enabled automatically under
>> certain conditions (described in the original email).
>>
>> However - the upside is that it gives another layer of "this just
>> works".. the downside is that it gives users a reason to not bother looking
>> at why their NAT is broken in the first place.
>>
>> With that in mind, I'm thinking that just a documentation fix is the
>> answer here, to avoid user lazyness.
>>
>> Cal
>>
>> On Sun, Dec 16, 2012 at 7:53 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> "There seems to be a large number of discussions surrounding NAT
>>> traversal, as well as lots of documentation, but with no concrete answers. "
>>>
>>> Part of the problem is that:
>>>
>>>    - Not all NAT implementations function in the same way (eg some
>>>    rewrite ports others do not)
>>>    - Not all SIP ALG implementations work the same/work
>>>    - Not all clients handle NAT in the same way
>>>    - You can encounter other odd situations such as double NAT that
>>>    further complicate matters
>>>
>>> So what works in one case might not work in another, so it's hard to
>>> give a concrete 'this is how to do it' that'll work in all cases. And often
>>> that means you need to find a failing client first then put in a NDLB
>>> workaround for that specific client. You could enable them by default, but
>>> that then can cause problems in other cases where the clients handle NAT
>>> correctly.
>>>
>>> Roll on NAT-less IPv6 for true end-to-end connectivity*. :o)
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 16 December 2012 16:15, Cal Leeming [Simplicity Media Ltd] <
>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>
>>>> *Any and all feedback on this thread would be much welcomed.*
>>>>
>>>> Hello,
>>>> *
>>>> *
>>>> There seems to be a large number of discussions surrounding NAT
>>>> traversal, as well as lots of documentation, but with no concrete answers.
>>>>
>>>> The NAT related wiki documentation is tedious, and depending on the
>>>> outcome of this thread, I'd like to spend some time cleaning it up.
>>>>
>>>> The most common problem (the same as ours) was having a router with
>>>> broken ALG and a softphone that does not seem to work with STUN.
>>>>
>>>> The following REGISTER is sent from a phone.
>>>>
>>>> REGISTER sip:1.2.3.4:5060 SIP/2.0
>>>> Via: SIP/2.0/UDP 192.168.1.102:57787
>>>> ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport
>>>> Max-Forwards: 70
>>>> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>
>>>> To: "foxx"<sip:2000 at 1.2.3.4:5060>
>>>> From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448
>>>> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.
>>>> CSeq: 7 REGISTER
>>>> Expires: 120
>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
>>>> REFER, INFO, MESSAGE
>>>> Supported: replaces
>>>> User-Agent: 3CXPhone 6.0.25732.0
>>>> Content-Length: 0
>>>>
>>>> As you can see, the client's public IP is not specified
>>>> anywhere. FreeSWITCH offers several ways around this, the main ones being;
>>>>
>>>> * NDLB-connectile-dysfunction
>>>> * NDLB-force-rport
>>>> * apply-nat-acl
>>>> * sip-force-contact
>>>>
>>>> The one that has worked in our case was "NDLB-connectile-dysfunction"
>>>> (otherwise known as NAT HACK), however there seems to be a lot of negative
>>>> comments about using this.
>>>>
>>>> From what I can tell, the general argument is that NAT HACK is
>>>> considered a non RFC compliant hack, and the SIP phones should be doing a
>>>> better job of keeping to the RFCs.
>>>>
>>>> In principle, this is a fair argument - but in practise, it's not a
>>>> reasonable assumption that all phones are RFC compliant, and (imho) not a
>>>> reasonable argument to have this functionality disabled by default.
>>>>
>>>> So, I'd like to present the following arguments;
>>>>
>>>> * Are there any other negative aspects about
>>>> using NDLB-connectile-dysfunction, other than it is a non compliant RFC
>>>> hack?
>>>>
>>>> * Why is NDLB-connectile-dysfunction not enabled by default when
>>>> certain conditions are met? In the event that FreeSWITCH receives a
>>>> REGISTER from a phone specifying a Contact/Via as 192.168.0.0/16, but
>>>> received on a public IP, then it should be obvious that NAT is broken and
>>>> automatically try to circumvent it.
>>>>
>>>> * People seem to get confused between server side and client side NAT
>>>> problems, and that they both need to be resolved in a different way. The
>>>> documentation doesn't seem to reflect this clearly.
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 17 17:51:15 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 17 Dec 2012 14:51:15 +0000
Subject: [Freeswitch-users] ODBC error
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233BCBB@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233BB74@Mail-Kilo.squay.com>
	<CAFiqYun5nYjUvCTY8=F-29P5hRi76RrkTrzLRmBETK2mrPgo+w@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233BCBB@Mail-Kilo.squay.com>
Message-ID: <CAFiqYunNaRpFu6EyBE7xAaBxq_ZMYtwgi5E=DaWt_w0SefdmJA@mail.gmail.com>

Great  :)

Did that fix the problem?

Just a thought for the core devs...
Since ODBC is now so popular perhaps rather than auto-detecting unixodbc
it'd be better to have it enabled by default and give an error if it's
missing?
For anyone still wanting to build without it (eg for embedded systems) add
a --disable-core-odbc-support option.

-Steve


On 17 December 2012 12:32, Archana Venugopan <a.venugopan at mundio.com> wrote:

>  Thanks again. ODBC was the issue.****
>
> ** **
>
> Regards,****
>
> Archana.V****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 17 December 2012 11:42
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] ODBC error****
>
> ** **
>
> ODBC NOT AVAILABLE****
>
>  ** **
>
> This means your FS was compiled without ODBC support.****
>
> ** **
>
> Check you have the unixodbc development package installed when building
> (usually unixodbc-dev or unixodbc-devel).****
>
> ** **
>
> It should be auto-detected if installed, but to be sure run configure with
> the --enable-core-odbc-support to force it to look for it, that'll should
> then give you an error if it can't find it.****
>
> ** **
>
> ** **
>
> ** **
>
> There has also been some changes to DSNs since v1.2.4 of which you should
> be aware.****
>
> ** **
>
> DSNs of format "datasourcename" are no longer supported. They should be
> changed to one of:****
>
> "datasourcename::"****
>
> "datasourcename:username:password"****
>
> "odbc://datasourcename"****
>
> "odbc://datasourcename::"****
>
> "odbc://datasourcename:username:password"****
>
> ** **
>
> http://wiki.freeswitch.org/wiki/DSN****
>
> ** **
>
> I prefer using the odbc:// prefix  - it seems less likely that it would
> change long-term ****
>
> ** **
>
> -Steve****
>
> ** **
>
> ** **
>
> On 16 December 2012 20:55, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
> Can anyone please tell me why I am getting this error inspite ?isql ?v
> smepbx smepbx smeswitch? was working. ****
>
>  ****
>
>  ****
>
> [CRIT] switch_core_sqldb.c:433 Failure! ODBC NOT AVAILABLE! Can't connect
> to DSN smepbx****
>
>  ****
>
> Regards,****
>
> Archana.V****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.venugopan at mundio.com  Mon Dec 17 17:59:42 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Mon, 17 Dec 2012 14:59:42 +0000
Subject: [Freeswitch-users] ODBC error
In-Reply-To: <CAFiqYunNaRpFu6EyBE7xAaBxq_ZMYtwgi5E=DaWt_w0SefdmJA@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF233BB74@Mail-Kilo.squay.com>
	<CAFiqYun5nYjUvCTY8=F-29P5hRi76RrkTrzLRmBETK2mrPgo+w@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233BCBB@Mail-Kilo.squay.com>
	<CAFiqYunNaRpFu6EyBE7xAaBxq_ZMYtwgi5E=DaWt_w0SefdmJA@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233BD43@Mail-Kilo.squay.com>

Ya it did fixed the problem. Just installed this unixodbc-devel.

Regards,
Archana.V

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 17 December 2012 14:51
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] ODBC error

Great  :)

Did that fix the problem?

Just a thought for the core devs...
Since ODBC is now so popular perhaps rather than auto-detecting unixodbc it'd be better to have it enabled by default and give an error if it's missing?
For anyone still wanting to build without it (eg for embedded systems) add a --disable-core-odbc-support option.

-Steve


On 17 December 2012 12:32, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Thanks again. ODBC was the issue.

Regards,
Archana.V

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 Steven Ayre
Sent: 17 December 2012 11:42
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] ODBC error

ODBC NOT AVAILABLE

This means your FS was compiled without ODBC support.

Check you have the unixodbc development package installed when building (usually unixodbc-dev or unixodbc-devel).

It should be auto-detected if installed, but to be sure run configure with the --enable-core-odbc-support to force it to look for it, that'll should then give you an error if it can't find it.



There has also been some changes to DSNs since v1.2.4 of which you should be aware.

DSNs of format "datasourcename" are no longer supported. They should be changed to one of:
"datasourcename::"
"datasourcename:username:password"
"odbc://datasourcename"
"odbc://datasourcename::"
"odbc://datasourcename:username:password"

http://wiki.freeswitch.org/wiki/DSN

I prefer using the odbc:// prefix  - it seems less likely that it would change long-term

-Steve


On 16 December 2012 20:55, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
Can anyone please tell me why I am getting this error inspite 'isql -v smepbx smepbx smeswitch' was working.


[CRIT] switch_core_sqldb.c:433 Failure! ODBC NOT AVAILABLE! Can't connect to DSN smepbx

Regards,
Archana.V


_________________________________________________________________________
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

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 lists at kavun.ch  Mon Dec 17 18:03:38 2012
From: lists at kavun.ch (Emrah)
Date: Mon, 17 Dec 2012 10:03:38 -0500
Subject: [Freeswitch-users] Video call
Message-ID: <A7B83A68-4703-438A-8A0F-74E669B1DB84@kavun.ch>

Hi there,

I am having a hard time getting video working.

I have mod_h26x loaded.
I tried setting proxy_media=true before bridging.
I tried forcing codecs with absolute_codec_string.
I tried inherit_codec=true.

I get no video no matter what I do.

The codec I need is H264.

What is the procedure to get video calling working on FS?

Cheers and thanks,
Emrah


From daveh at beachdognet.com  Mon Dec 17 19:26:04 2012
From: daveh at beachdognet.com (Dave Horton)
Date: Mon, 17 Dec 2012 11:26:04 -0500
Subject: [Freeswitch-users] Stereo recording of conference - audio on the
	two channels is out of sync
Message-ID: <F3779ADE-F3F6-4EA7-86EF-954409F9044F@beachdognet.com>

I am trying to make a stereo recording of a two-party bridged conference -- i.e. caller calls in, we create a bridged conference and outdial, and then get a stereo recording with caller and called party on different channels.  My dialplan is shown below.  The problem is that the two channels on the resulting recording are completely out of sync with each other.  Does anyone have an idea about why this might be happening?

(In case anyone is wondering why I am recording a 2 party conference, rather than just a bridged outdial, the reason is that I need to play an audio clip to both parties that the call is being recording, and I need that clip to also be captured in the recording, and I can't figure out a way to do that other than to create a conference and play the clip into the conference.  I'm open to other suggestions though if anyone has an alternative approach).

(Note: the lua script in the dialplan below just collects a number to outdial)

<include>
    <extension name="617xxxxxxx" continue="false">
      <condition field="source" expression="mod_sofia"/>
      <condition field="destination_number" expression="617xxxxx">
                <action application="ring_ready"/>
                <action application="answer"/>
                <action application="start_dtmf"/>
                <action application="set" data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/> 
                <action application="export" data="outfile=$${base_dir}/tmp/${call_id}.wav"/> 
                <action application="lua" data="demo_call2.lua"/>
                <action application="export" data="RECORD_STEREO=true"/>
                <action application="export" data="execute_on_media=record_session ${outfile}"/>
                <action application="export" data="api_on_answer=conference ${call_id} play misc/this-call-may-be-recorded.wav async"/>
                <action application="conference" data="bridge: ${call_id}@simple:{origination_caller_id_number=${caller_id_number}}sofia/normal_customer/${outdial}@${egress_gateway}"/>
        </condition>
    </extension>
</include>

From jpablolorenzetti at hotmail.com  Mon Dec 17 19:31:27 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Mon, 17 Dec 2012 16:31:27 +0000
Subject: [Freeswitch-users] callcenter with external agents
In-Reply-To: <CAHWtr1TE8RM5Xa9adrJz1hfMty8U++56mbFOuV0zN85TK-ztUg@mail.gmail.com>
References: <DUB002-W3915B9DAC4F64D9FD8AE3ED94E0@phx.gbl>,
	<4521E504-D44A-47D9-B3EE-3EE363A74558@endigotech.com>,
	<CAHWtr1RNJa2oJD67xtQYZvYPNdWDyehZ4QLpoMRUt3bwvo6kWA@mail.gmail.com>,
	<1355489348.7162.28.camel@luna.madrid.commsmundi.com>,
	<DUB002-W848BC9DC7196903EB35AE2D94D0@phx.gbl>,
	<DUB002-W73F9E83A51542D9A07C2FFD94D0@phx.gbl>,
	<CAHWtr1TE8RM5Xa9adrJz1hfMty8U++56mbFOuV0zN85TK-ztUg@mail.gmail.com>
Message-ID: <DUB002-W716D9B0E0648D2EA7B6CA0D9320@phx.gbl>

Hi Regis, thank you very much, that was the issue, it is calling out now ....
thanks all for helping me get this going ... regards!

Date: Sun, 16 Dec 2012 18:46:31 +0100
From: regis.freeswitch.org at tornad.net
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] callcenter with external agents

Hello.
As I said the first time, it seems that there's no agent in your queue.
Check the TIER config
In agent you put : 
agent name="1000 at default" 

But in TIER
tier agent="1000 at callcenter"


For me, it's not the same agent, Try to put agent at default in TIER config party

Regards


2012/12/15 Juan Pablo L. <jpablolorenzetti at hotmail.com>




I have change somethings around as suggested but still freeswitch does not even make the attempt to dial out, this is the config i m using http://pastebin.freeswitch.org/20326 and this is the console logs http://pastebin.freeswitch.org/20327.

the name of the gateway i m trying to send the call to is huawei_csoft, in my original post i sent the name of the external profile whichis wrong but i was making lots of combinations trying to make it work .... thanks for any help ... 


From: jpablolorenzetti at hotmail.com
To: freeswitch-users at lists.freeswitch.org

Date: Fri, 14 Dec 2012 22:46:39 +0000
Subject: Re: [Freeswitch-users] callcenter with external agents




Hi All, thank you very much for your answers, it is a relieve to know that what i m trying to accomplish is possible, i have not done it before so i m not sure what to expect, it is likely that i m configuring something wrong ..... i will try what was suggested in these responses and get back to you guys asap. thanks!!


> From: fdelawarde at wirelessmundi.com
> To: freeswitch-users at lists.freeswitch.org

> Date: Fri, 14 Dec 2012 13:49:08 +0100
> Subject: Re: [Freeswitch-users] callcenter with external agents
> 
> It should work fine, heck it even works with loopback channels for me!
> 
> Just in case, verify that your dialstring is correct. If

> "external-huawei_gw" is a gateway, use:
> 
> sofia/gateway/external-huawei_gw/XXXXXX
> 
> instead of:
> 
> sofia/external-huawei_gw/XXXXXX
> 
> Regards,
> Fran?ois.

> 
> On Fri, 2012-12-14 at 12:20 +0100, Regis M wrote:
> > It works, we're doing it on a production system.
> > The message seems to be more a agent State problem instead of gateway
> > problem. Your agent doesn't seems to been "Waiting" or you don't

> > correctly affect it in a tier with you queue. There's 2 agents thing
> > to check Status and State. and Tier association
> > Check the configuration by make call working with a "normal" sip agent

> > and then, you could try by changing his contact parameter to gateway
> > outside the fs callcenter box.
> > 
> > Regards
> > 
> > 
> > 2012/12/14 Brian Foster <bdfoster at endigotech.com>

> >         +1 I don't really know if this is possible. We've tried to do
> >         it this way but we ended up using some LUA scripts and
> >         mod_fifo. Unfortunately I can't release the scripts because

> >         the client won't allow me. 
> >         
> >         Sent from my iPhone 
> >         
> >         On Dec 13, 2012, at 1:31 PM, Juan Pablo L.
> >         <jpablolorenzetti at hotmail.com> wrote:

> >         
> >         
> >         > Hi, i m trying to set up a callcenter but the lines for the
> >         > agents are mobile phones in a mobile network and are not
> >         > attached to freeswitch, i m trying to set it up with the

> >         > following: 
> >         > 
> >         > 
> >         > <agent name="1000 at default" type="callback"
> >         > contact="[call_timeout=10]sofia/external-huawei_gw/XXXXXX"

> >         > status="Available" max-no-answer="3" wrap-up-time="10"
> >         > reject-delay-time="10" busy-delay-time="60" /> 
> >         > 

> >         > 
> >         > XXXXX being the mobile number but i m getting the following
> >         > error: 
> >         > 
> >         > 
> >         > Member 2018 <2018> in queue 'callcenter at default' reached max

> >         > wait of 0 sec. with no agent plus join grace period of 5
> >         > sec. 
> >         > 
> >         > 
> >         > and i see that freeswitch does not even try to dial out to

> >         > the trunk.  
> >         > 
> >         > 
> >         > i m wondering if configuring it the way i need it is even
> >         > possible as i think the problem may reside in the fact that

> >         > freeswitch does not actually see the agent registered.
> >         > 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
> >         > 
> >         > FreeSWITCH-users mailing list
> >         > FreeSWITCH-users at lists.freeswitch.org

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

> >         > http://www.freeswitch.org
> >         > 
> >         
> >         _________________________________________________________________________

> >         Professional FreeSWITCH Consulting 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

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

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

> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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

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

_________________________________________________________________________

Professional FreeSWITCH Consulting 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

FreeSWITCH-users mailing 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  Mon Dec 17 20:05:41 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 17 Dec 2012 12:05:41 -0500
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>
	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>
Message-ID: <500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>

HI Anthony (et al),

 

I have finally captured the failure with SIPTRACE enabled.  See pastebin:
http://pastebin.freeswitch.org/20342  I have tried to clip this down to just
the one call.  It is from "anonymous" . <4104850152>->+14108825019 in
context from-trunk

 

The first error I see is "nta: received 406 Not Acceptable for INVITE
(37544793)"

 

I have no idea what that means though.  I can't figure out why these calls
work many times and fail others.  My best guess right now is that this is a
capacity limitation of the NAT/ROUTER at their site.  I think some of these
phones in this ring group may be double NAT'ed just to make my life fun.

 

Any diagnosis of these logs would be greatly appreciated.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Thursday, December 13, 2012 12:35 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

You can set tracelevel under global settings in sofia.conf.xml

 

<param name="tracelevel" value="debug"/>

 

or from cli

 

sofia tracelevel debug

 

that will make the traces in debug level instead of console so then if your
console level is less than debug you won't see them but they will still go
in the log.

 

 

 

On Thu, Dec 13, 2012 at 11:26 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

http://lists.freeswitch.org/pipermail/freeswitch-users/2010-February/053516.
html





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From mario_fs at mgtech.com  Mon Dec 17 20:08:19 2012
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 17 Dec 2012 09:08:19 -0800
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAHKQagH8uu1nOBvgZ+nr_o0dV4u+G=3wfoaEx9pfxZd30F7YWw@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<CAFiqYu=wL6+yk67AzzfF_HKrMBBreqeb5jybX7r8LjoXBHVs0A@mail.gmail.com>
	<CAHKQagFbEdS+wLt-CH7t_=B5UESRQkVVnFf3HSt8EB4=Gq=pOA@mail.gmail.com>
	<CABH8QX0WNTaMHXbFeJ8ip+D7ADn+b45nc2=fSHdf5F8OpWzMNQ@mail.gmail.com>
	<CAHKQagH8uu1nOBvgZ+nr_o0dV4u+G=3wfoaEx9pfxZd30F7YWw@mail.gmail.com>
Message-ID: <468A3B64-5121-4DC9-AC68-4B5050EAAD37@mgtech.com>

I am planning on putting up an example of how I have 2 WAN lines both available to FreeSwitch, both used at the same time (load balanced), both are failover for each other. I have tested pulling plugs, etc. There is nothing in FreeSwitch defined for the lines and it runs with "-nonat", the router does SIP ALG fine. I thought the setup example might be useful to someone since it took so long to figure it out. The only hold up has been that I have other wiki updates to do and had FreeSwitch issues this year, each time one is fixed another comes up (memory leak right now) so that has eaten up all the time I could put into the wiki. I have already started it and promise to post on the ML once it's up, should be in the next 2 months. It may help someone in the same situation and requirements.
Mario G

On Dec 17, 2012, at 6:17 AM, Cal Leeming [Simplicity Media Ltd] wrote:

> Any other experiences/thoughts from others on this?
> 
> Feedback so far has been great, lets keep it coming guys!
> 
> Cal
> 
> On Sun, Dec 16, 2012 at 9:52 PM, Jo?o Mesquita <jmesquita at freeswitch.org> wrote:
> My take on the subject is that we are trying to tackle 2 different problems at once.
> 
> On one hand we have how NAT works and what problems it create on a VoIP world. I really was not able to find any condensed documentation on the internet that would describe with VoIP in mind what a admin need to know about SIP packets and NAT handling. If there was one, we could just add a link to the wiki page. Since NAT has no "one configuration" fix, user NEEDS to know about it in order to fix his own scenario.
> 
> On the other hand, we have how FS particularly deals with NAT (client and server side). I think there is more documentation to be added/created on that end as well. NAT is a complicated matter indeed and understanding Sofia profiles alone is a challenge. Add NAT to the mix and the configuration can look like black magic.
> 
> So, question is, who can elaborate/contribute/find a definitive guide to NAT so we can lecture users and who knows all about the NAT handling internals so we can document each and every option available?
> 
> Jo?o Mesquita
> 
> 
> 
> On Sun, Dec 16, 2012 at 6:21 PM, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:
> It seems that most of us agree there is no single answer to fix NAT problems - the double-NAT is something I hadn't thought about.
> 
> Sean mentioned having a checklist of approaches, which would be good addition for the documentation fix.
> 
> I agree that enabling NAT HACK by default could break clients that are functioning normally, but not if it is only enabled automatically under certain conditions (described in the original email). 
> 
> However - the upside is that it gives another layer of "this just works".. the downside is that it gives users a reason to not bother looking at why their NAT is broken in the first place.
> 
> With that in mind, I'm thinking that just a documentation fix is the answer here, to avoid user lazyness.
> 
> Cal
> 
> On Sun, Dec 16, 2012 at 7:53 PM, Steven Ayre <steveayre at gmail.com> wrote:
> "There seems to be a large number of discussions surrounding NAT traversal, as well as lots of documentation, but with no concrete answers. "
> 
> Part of the problem is that:
> Not all NAT implementations function in the same way (eg some rewrite ports others do not)
> Not all SIP ALG implementations work the same/work
> Not all clients handle NAT in the same way
> You can encounter other odd situations such as double NAT that further complicate matters
> So what works in one case might not work in another, so it's hard to give a concrete 'this is how to do it' that'll work in all cases. And often that means you need to find a failing client first then put in a NDLB workaround for that specific client. You could enable them by default, but that then can cause problems in other cases where the clients handle NAT correctly.
> 
> Roll on NAT-less IPv6 for true end-to-end connectivity*. :o)
> 
> -Steve
> 
> 
> 
> On 16 December 2012 16:15, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:
> Any and all feedback on this thread would be much welcomed.
> 
> Hello,
> 
> There seems to be a large number of discussions surrounding NAT traversal, as well as lots of documentation, but with no concrete answers. 
> 
> The NAT related wiki documentation is tedious, and depending on the outcome of this thread, I'd like to spend some time cleaning it up.
> 
> The most common problem (the same as ours) was having a router with broken ALG and a softphone that does not seem to work with STUN.
> 
> The following REGISTER is sent from a phone.
> 
> REGISTER sip:1.2.3.4:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.102:57787;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport
> Max-Forwards: 70
> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>
> To: "foxx"<sip:2000 at 1.2.3.4:5060>
> From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448
> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.
> CSeq: 7 REGISTER
> Expires: 120
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
> Supported: replaces
> User-Agent: 3CXPhone 6.0.25732.0
> Content-Length: 0
> 
> As you can see, the client's public IP is not specified anywhere. FreeSWITCH offers several ways around this, the main ones being;
> 
> * NDLB-connectile-dysfunction
> * NDLB-force-rport
> * apply-nat-acl
> * sip-force-contact
> 
> The one that has worked in our case was "NDLB-connectile-dysfunction" (otherwise known as NAT HACK), however there seems to be a lot of negative comments about using this.
> 
> From what I can tell, the general argument is that NAT HACK is considered a non RFC compliant hack, and the SIP phones should be doing a better job of keeping to the RFCs.
> 
> In principle, this is a fair argument - but in practise, it's not a reasonable assumption that all phones are RFC compliant, and (imho) not a reasonable argument to have this functionality disabled by default.
> 
> So, I'd like to present the following arguments;
> 
> * Are there any other negative aspects about using NDLB-connectile-dysfunction, other than it is a non compliant RFC hack?
> 
> * Why is NDLB-connectile-dysfunction not enabled by default when certain conditions are met? In the event that FreeSWITCH receives a REGISTER from a phone specifying a Contact/Via as 192.168.0.0/16, but received on a public IP, then it should be obvious that NAT is broken and automatically try to circumvent it.
> 
> * People seem to get confused between server side and client side NAT problems, and that they both need to be resolved in a different way. The documentation doesn't seem to reflect this clearly.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
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From bdfoster at endigotech.com  Mon Dec 17 20:20:03 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 17 Dec 2012 12:20:03 -0500
Subject: [Freeswitch-users] G729 - [INCOMPATIBLE_DESTINATION]
In-Reply-To: <CAP2fbbeZaQG3ai6+GVyyY08=rJS=L1FJWmNk4wmb_j2J_Lhtbg@mail.gmail.com>
References: <CAP2fbbcixeTy4H+KPbxJQvAhCDWmco5a0jF0XQkGHBtE2Qkmdg@mail.gmail.com>
	<CAFiqYuk+OwUVPCmy8H1Yq5hW-+Hhy40DS1RdRe8T6oBTjeY2LA@mail.gmail.com>
	<CAMWiMy_0f44SwcXZR4CQgP7=RBqrTYW8fQQhAa8m1nR+kKv=oQ@mail.gmail.com>
	<CAP2fbbfczFPcBF5ZF0G_+FL_+n0XZPBcu6KCCSHU1ryny2hBKg@mail.gmail.com>
	<CAFiqYukVJW5BzbXeRtJmonAXxudRvuW9=h7SwCQjyyTHk3th6g@mail.gmail.com>
	<CAFiqYu=0ShYLYE+Y3rEtUw6eKp1EjSkxPaM2YkqSnYFksrfqEw@mail.gmail.com>
	<CAP2fbbeZaQG3ai6+GVyyY08=rJS=L1FJWmNk4wmb_j2J_Lhtbg@mail.gmail.com>
Message-ID: <CAKV-jFx+fff8nK4k7ccDpcCa7OjXHiM2uqvfnus5=DzLeeb=CQ@mail.gmail.com>

*Dear OP,*
*
*
*We've detected that your FreeSWITCH version *is seriously old. Because you
have an old version of FreeSWITCH, this particular issue you are facing
could have already been fixed, and is no longer relevant. Our suggesion:
update and reproduce (your issue, get your mind out of the gutter). By
doing your part FreeSWITCH, you can help save a FreeSWITCH developer from
the constant migraines they face when trying to fix a bug that's already
been fixed, and let him concentrate on bringing you the best FreeSWITCH
experience possible. *It's the only way.*
*

The current stable release* of FreeSWITCH is
1.2.5.3<http://files.freeswitch.org/freeswitch-1.2.5.3.tar.bz2>,
released on 07-December-2012.

*The stable 1.2 series* is available in Git branch
v1.2.stable<http://git.freeswitch.org/git/freeswitch/log/?h=v1.2.stable>.
This is the recommended version for most applications. Patches to the 1.2
series will be made here and periodically released as the next 1.2.x
version. Since not every patch will mean a newer version you will get the
very latest patches to the stable series by using this branch. The branch
is a moving target - the checkout date and revision will form part of the
version number.

*If you wish to try an even newer experimental version*, you can use the Master
Branch from Git <http://git.freeswitch.org/git/freeswitch/log/>. New
features will only be added to the Master branch. This will have the
newest, latest, and greatest features, but due to upgrading of the bundled
libraries it might be less stable than v1.2.stable for a short while. If
you use this branch you should test the installation before allowing users
to use it to confirm it is stable. You can then use 'git checkout' to
install the same tested revision on all your systems. NOTE that this is a
formal change of development practice. Master used to be the recommended
deployment target for all users, and now the tarballs and the Git stable
branch serve that purpose. Older documentation/tutorials may still exist
recommending using Git, but these will now be out-of-date.

*Thank you so much* for bringing this to our attention. Seriously. However
the developers can't really do anything without some help from you. *You
can make a difference.* Update FreeSWITCH today :)



On Mon, Dec 17, 2012 at 6:30 AM, Balamurugan Mahendran <b2m at a-cti.com>wrote:

> Yes - both are same.  Actually call is from Android SIP client. Trying to
> make outbound call from it - Getting incoming call with NO issues.
>
> Thanks,
> Bala
>
>
> On Mon, Dec 17, 2012 at 4:52 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28 18-32-35
>>> -0600
>>
>>
>> Your version is pretty old now - almost a year. Are you able to reproduce
>> this problem on the latest head of 1.2.stable and/or master?
>>
>> It could be this is a bug that's already been resolved in the past 12
>> months.
>>
>> -Steve
>>
>>
>>
>> On 17 December 2012 11:21, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>>> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>>>
>>> This looks odd, although I'm not sure why it's not working.
>>>
>>> The remote endpoint is using the static payload type 0 for G729.
>>> Normally that's reserved for G711 ulaw, while G729 uses 18. But the rtpmap
>>> should allow overriding that.
>>>
>>>> G729:0:8000:20:64000
>>>
>>>
>>> The 64000 rate here also looks odd - it's comparing 64khz to 8khz and I
>>> suspect that's also a reason why it doesn't recognise it. I don't know
>>> where that 64000 is coming from though.
>>>
>>> -Steve
>>>
>>>
>>>
>>>
>>>
>>> On 17 December 2012 07:31, Balamurugan Mahendran <b2m at a-cti.com> wrote:
>>>
>>>> Thanks for the help!!  Please let me know if I missed anything.
>>>>
>>>>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,SILK at 8000h
>>>> ,SILK at 12000h,SILK at 16000h,SILK at 24000h,G7221 at 32000h,G7221 at 16000h
>>>> ,G722,PCMU,PCMA,GSM"/>
>>>>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,SILK at 8000h
>>>> ,SILK at 12000h,SILK at 16000h,SILK at 24000h,G7221 at 32000h,G7221 at 16000h
>>>> ,G722,PCMU,PCMA,GSM"/>
>>>>
>>>>
>>>>  freeswitch at internal> g729_info
>>>> Permitted G729 channels: 10
>>>> Encoders in use: 4
>>>> Decoders in use: 3
>>>>
>>>> freeswitch at internal>
>>>>
>>>>
>>>>
>>>> recv 742 bytes from udp/[182.79.149.142]:16251 at 07:27:19.212195:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    INVITE sip:500 at 50.54.12.39 SIP/2.0
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport;branch=z9hG4bK68891
>>>>
>>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>>
>>>>    Max-Forwards: 70
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>>
>>>>    To: <sip:500 at 50.54.12.39>
>>>>
>>>>    Call-ID: 047707180743 at 10.4.2.7
>>>>
>>>>    CSeq: 1 INVITE
>>>>
>>>>    Contact: <sip:501 at 182.79.149.142:16251;transport=udp>
>>>>
>>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>>
>>>>    Expires: 3600
>>>>
>>>>    User-Agent: SyncPhone 1.0
>>>>
>>>>    Content-Length: 244
>>>>
>>>>    Content-Type: application/sdp
>>>>
>>>>
>>>>
>>>>    v=0
>>>>
>>>>    o=- 1111 1111 IN IP4 182.79.149.142
>>>>
>>>>    s=Session SIP/SDP
>>>>
>>>>    c=IN IP4 182.79.149.142
>>>>
>>>>    t=0 0
>>>>
>>>>    a=sendrecv
>>>>
>>>>    a=rtcp:57395 IN IP4 182.79.149.142
>>>>
>>>>    m=audio 26474 RTP/AVP 0 101
>>>>
>>>>    a=rtpmap:0 G729/8000
>>>>
>>>>    a=rtpmap:101 telephone-event/8000
>>>>
>>>>    a=fmtp:101 0-15
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> send 321 bytes to udp/[182.79.149.142]:16251 at 07:27:19.212520:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    SIP/2.0 100 Trying
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:16251
>>>> ;rport=16251;branch=z9hG4bK68891
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>>
>>>>    To: <sip:500 at 50.54.12.39>
>>>>
>>>>    Call-ID: 047707180743 at 10.4.2.7
>>>>
>>>>    CSeq: 1 INVITE
>>>>
>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>> 18-32-35 -0600
>>>>
>>>>    Content-Length: 0
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:7481 IP 182.79.149.142
>>>> Approved by acl "domains[]". Access Granted.
>>>>
>>>> 2012-12-17 07:27:19.206419 [NOTICE] switch_channel.c:930 New Channel
>>>> sofia/internal/501 at 50.54.12.39 [30a75cd4-481b-11e2-84b4-e5c28303057e]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_NEW
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:380
>>>> (sofia/internal/501 at 50.54.12.39) State NEW
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:5502 Channel sofia/internal/
>>>> 501 at 50.54.12.39 entering state [received][100]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:5513 Remote SDP:
>>>>
>>>> v=0
>>>>
>>>> o=- 1111 1111 IN IP4 182.79.149.142
>>>>
>>>> s=Session SIP/SDP
>>>>
>>>> c=IN IP4 182.79.149.142
>>>>
>>>> t=0 0
>>>>
>>>> a=sendrecv
>>>>
>>>> a=rtcp:57395 IN IP4 182.79.149.142
>>>>
>>>> m=audio 26474 RTP/AVP 0 101
>>>>
>>>> a=rtpmap:0 G729/8000
>>>>
>>>> a=rtpmap:101 telephone-event/8000
>>>>
>>>> a=fmtp:101 0-15
>>>>
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
>>>> send/recv payload to 101
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_channel.c:2852
>>>> (sofia/internal/501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
>>>>
>>>> 2012-12-17 07:27:19.206419 [NOTICE] sofia.c:5781 Hangup sofia/internal/
>>>> 501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_channel.c:2875 Send signal
>>>> sofia/internal/501 at 50.54.12.39 [KILL]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send
>>>> signal sofia/internal/501 at 50.54.12.39 [BREAK]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_HANGUP
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:602
>>>> (sofia/internal/501 at 50.54.12.39) State HANGUP
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:469 Channel
>>>> sofia/internal/501 at 50.54.12.39 hanging up, cause:
>>>> INCOMPATIBLE_DESTINATION
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:534 Responding to INVITE
>>>> with: 488
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:47
>>>> sofia/internal/501 at 50.54.12.39 Standard HANGUP, cause:
>>>> INCOMPATIBLE_DESTINATION
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:602
>>>> (sofia/internal/501 at 50.54.12.39) State HANGUP going to sleep
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:393
>>>> (sofia/internal/501 at 50.54.12.39) State Change CS_HANGUP -> CS_REPORTING
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send
>>>> signal sofia/internal/501 at 50.54.12.39 [BREAK]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_REPORTING
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:662
>>>> (sofia/internal/501 at 50.54.12.39) State REPORTING
>>>>
>>>> send 816 bytes to udp/[182.79.149.142]:16251 at 07:27:19.215295:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    SIP/2.0 488 Not Acceptable Here
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:16251
>>>> ;rport=16251;branch=z9hG4bK68891
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>>
>>>>    To: <sip:500 at 50.54.12.39>;tag=8K31Z9S2DymjN
>>>>
>>>>    Call-ID: 047707180743 at 10.4.2.7
>>>>
>>>>    CSeq: 1 INVITE
>>>>
>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>> 18-32-35 -0600
>>>>
>>>>    Accept: application/sdp
>>>>
>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>
>>>>    Supported: timer, precondition, path, replaces
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:79
>>>> sofia/internal/501 at 50.54.12.39 Standard REPORTING, cause:
>>>> INCOMPATIBLE_DESTINATION
>>>>
>>>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>>> sla, include-session-description, presence.winfo, message-summary, refer
>>>>
>>>>    Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"2012-12-17
>>>> 07:27:19.206419 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/
>>>> 501 at 50.54.12.39) State REPORTING going to sleep
>>>>
>>>>
>>>>    Content-Length: 0
>>>>
>>>>    Remote-Party-ID: "500" <sip:500 at 50.54.12.39
>>>> >;party=calling;privacy=off;screen=no
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:387
>>>> (sofia/internal/501 at 50.54.12.39) State Change CS_REPORTING ->
>>>> CS_DESTROY
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send
>>>> signal sofia/internal/501 at 50.54.12.39 [BREAK]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1380 Session
>>>> 204 (sofia/internal/501 at 50.54.12.39) Locked, Waiting on external
>>>> entities
>>>>
>>>> 2012-12-17 07:27:19.206419 [NOTICE] switch_core_session.c:1398 Session
>>>> 204 (sofia/internal/501 at 50.54.12.39) Ended
>>>>
>>>> 2012-12-17 07:27:19.206419 [NOTICE] switch_core_session.c:1400 Close
>>>> Channel sofia/internal/501 at 50.54.12.39 [CS_DESTROY]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:491
>>>> (sofia/internal/501 at 50.54.12.39) Callstate Change HANGUP -> DOWN
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:494
>>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_DESTROY
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:504
>>>> (sofia/internal/501 at 50.54.12.39) State DESTROY
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:374 sofia/internal/
>>>> 501 at 50.54.12.39 SOFIA DESTROY
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:86
>>>> sofia/internal/501 at 50.54.12.39 Standard DESTROY
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:504
>>>> (sofia/internal/501 at 50.54.12.39) State DESTROY going to sleep
>>>>
>>>> recv 407 bytes from udp/[182.79.149.142]:16251 at 07:27:19.581401:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    ACK sip:500 at 50.54.12.39 SIP/2.0
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport;branch=z9hG4bK68891
>>>>
>>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>>
>>>>    Max-Forwards: 70
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>>
>>>>    To: <sip:500 at 50.54.12.39>;tag=8K31Z9S2DymjN
>>>>
>>>>    Call-ID: 047707180743 at 10.4.2.7
>>>>
>>>>    CSeq: 1 ACK
>>>>
>>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>>
>>>>    User-Agent: SyncPhone 1.0
>>>>
>>>>    Content-Length: 0
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> recv 465 bytes from udp/[182.79.149.142]:36420 at 07:27:20.672450:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    REGISTER sip:50.54.12.39 SIP/2.0
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport;branch=z9hG4bK20451
>>>>
>>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>>
>>>>    Max-Forwards: 70
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>>
>>>>    To: <sip:501 at 50.54.12.39>
>>>>
>>>>    Call-ID: 790899967509 at 10.4.2.7
>>>>
>>>>    CSeq: 1 REGISTER
>>>>
>>>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>
>>>>
>>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>>
>>>>    Expires: 3600
>>>>
>>>>    User-Agent: SyncPhone 1.0
>>>>
>>>>    Content-Length: 0
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> 2012-12-17 07:27:20.666425 [DEBUG] sofia_reg.c:1417 Send challenge for [
>>>> 501 at 50.54.12.39]
>>>>
>>>> 2012-12-17 07:27:20.666425 [WARNING] sofia_reg.c:1421 SIP auth
>>>> challenge (REGISTER) on sofia profile 'internal' for [501 at 50.54.12.39]
>>>> from ip 182.79.149.142
>>>>
>>>> send 626 bytes to udp/[182.79.149.142]:36420 at 07:27:20.673200:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    SIP/2.0 401 Unauthorized
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:36420
>>>> ;rport=36420;branch=z9hG4bK20451
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>>
>>>>    To: <sip:501 at 50.54.12.39>;tag=9vvt14a6a7a5g
>>>>
>>>>    Call-ID: 790899967509 at 10.4.2.7
>>>>
>>>>    CSeq: 1 REGISTER
>>>>
>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>> 18-32-35 -0600
>>>>
>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>
>>>>    Supported: timer, precondition, path, replaces
>>>>
>>>>    WWW-Authenticate: Digest realm="50.54.12.39",
>>>> nonce="31862298-481b-11e2-84b6-e5c28303057e", algorithm=MD5, qop="auth"
>>>>
>>>>    Content-Length: 0
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> recv 707 bytes from udp/[182.79.149.142]:36420 at 07:27:21.094563:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    REGISTER sip:50.54.12.39 SIP/2.0
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport;branch=z9hG4bK84890
>>>>
>>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>>
>>>>    Max-Forwards: 70
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>>
>>>>    To: <sip:501 at 50.54.12.39>
>>>>
>>>>    Call-ID: 790899967509 at 10.4.2.7
>>>>
>>>>    CSeq: 2 REGISTER
>>>>
>>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>>
>>>>    Expires: 3600
>>>>
>>>>    User-Agent: SyncPhone 1.0
>>>>
>>>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>
>>>>
>>>>    Authorization: Digest username="501", realm="50.54.12.39",
>>>> nonce="31862298-481b-11e2-84b6-e5c28303057e", uri="sip:50.54.12.39",
>>>> algorithm=MD5, qop=auth, nc=00000001, cnonce="a5d0aab2288476b3",
>>>> response="eb8ccc768c7f9fee038f0d5dfc599931"
>>>>
>>>>    Content-Length: 0
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'record_stereo' = 'true'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'default_gateway' = 'example.com'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'default_areacode' = '918'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'transfer_fallback_extension' = 'operator'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'toll_allow' = 'domestic,international,local'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'accountcode' = '501'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'user_context' = 'default'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'effective_caller_id_name' = 'Extension 501'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'effective_caller_id_number' = '501'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'outbound_caller_id_name' = 'FS'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'outbound_caller_id_number' = '9732386040'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'callgroup' = 'techsupport'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:1575 Register:
>>>>
>>>> From:    [501 at 10.252.148.21]
>>>>
>>>> Contact: ["user" <sip:501 at 182.79.149.142:36420;transport=udp>]
>>>>
>>>> Expires: [3600]
>>>>
>>>> send 600 bytes to udp/[182.79.149.142]:36420 at 07:27:21.096965:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    SIP/2.0 200 OK
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:36420
>>>> ;rport=36420;branch=z9hG4bK84890
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>>
>>>>    To: <sip:501 at 50.54.12.39>;tag=a6NK3ZU97F1Qc
>>>>
>>>>    Call-ID: 790899967509 at 10.4.2.7
>>>>
>>>>    CSeq: 2 REGISTER
>>>>
>>>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>;expires=3600
>>>>
>>>>    Date: Mon, 17 Dec 2012 07:27:21 GMT
>>>>
>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>> 18-32-35 -0600
>>>>
>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>
>>>>    Supported: timer, precondition, path, replaces
>>>>
>>>>    Content-Length: 0
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> send 926 bytes to udp/[182.79.149.142]:36420 at 07:27:21.112029:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    NOTIFY sip:501 at 182.79.149.142:36420;transport=udp SIP/2.0
>>>>
>>>>    Via: SIP/2.0/UDP 50.54.12.39;rport;branch=z9hG4bKXXQBeBHUQ5yZp
>>>>
>>>>    Max-Forwards: 70
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=BFFc5tcD5rQar
>>>>
>>>>    To: <sip:501 at 50.54.12.39>
>>>>
>>>>    Call-ID: 09276738-c2be-1230-ec86-22000afc9415
>>>>
>>>>    CSeq: 37528556 NOTIFY
>>>>
>>>>    Contact: <sip:mod_sofia at 50.54.12.39:5060>
>>>>
>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>> 18-32-35 -0600
>>>>
>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>
>>>>    Supported: timer, precondition, path, replaces
>>>>
>>>>    Event: message-summary
>>>>
>>>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>>> sla, include-session-description, presence.winfo, message-summary, refer
>>>>
>>>>    Subscription-State: terminated;reason=noresource
>>>>
>>>>    Content-Type: application/simple-message-summary
>>>>
>>>>    Content-Length: 64
>>>>
>>>>
>>>>
>>>>    Messages-Waiting: no
>>>>
>>>>    Message-Account: sip:501 at 10.252.148.21
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> recv 295 bytes from udp/[182.79.149.142]:36420 at 07:27:21.531463:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    SIP/2.0 405 Method Not Allowed
>>>>
>>>>    Via: SIP/2.0/UDP 50.54.12.39;branch=z9hG4bKXXQBeBHUQ5yZp;rport=5060
>>>>
>>>>    To: <sip:501 at 50.54.12.39>
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=BFFc5tcD5rQar
>>>>
>>>>    Call-ID: 09276738-c2be-1230-ec86-22000afc9415
>>>>
>>>>    CSeq: 37528556 NOTIFY
>>>>
>>>>    Server: SyncPhone 1.0
>>>>
>>>>    Content-Length: 0
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>
>>>>
>>>>
>>>> On Mon, Dec 17, 2012 at 2:25 AM, curriegrad2004 <
>>>> curriegrad2004 at gmail.com> wrote:
>>>>
>>>>> If he is using mod_g729 then the log would throw an error saying that
>>>>> the codec only does passthrough mode, however he hasn't shown that
>>>>> part of the log to us yet
>>>>>
>>>>> On Sun, Dec 16, 2012 at 12:05 PM, Steven Ayre <steveayre at gmail.com>
>>>>> wrote:
>>>>> > What G729 module are you using, and what codec is in use on the
>>>>> other leg?
>>>>> >
>>>>> > G729 needs licenses due to patents.
>>>>> >
>>>>> > mod_g729 is passthrough only but requires no licenses. This is
>>>>> because it
>>>>> > merely forwards the data, but doesn't perform the
>>>>> encoding/transcoding step
>>>>> > that the patents cover. But that means it can't transcode between
>>>>> different
>>>>> > codecs.
>>>>> > mod_com_g729 is the licensed version:
>>>>> http://www.freeswitch.org/node/235
>>>>> >
>>>>> > With the passthrough mod_g729 codec, if the other leg has selected a
>>>>> codec
>>>>> > other than g729 you'll see an error in your logs and it'll hangup
>>>>> with that
>>>>> > reason.
>>>>> >
>>>>> > You can tweak your codec negotiation to avoid this (eg
>>>>> > late-negotiation=true), or you can use the licensed version.
>>>>> >
>>>>> >
>>>>> >
>>>>> >
>>>>> > On 16 December 2012 11:40, Balamurugan Mahendran <b2m at a-cti.com>
>>>>> wrote:
>>>>> >>
>>>>> >> Need help on Codec Negotiation
>>>>> >>
>>>>> >>
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
>>>>> >> send/recv payload to 101
>>>>> >>
>>>>> >>
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] switch_channel.c:2852
>>>>> >> (sofia/internal/501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
>>>>> >> 2012-12-16 11:35:52.246422 [NOTICE] sofia.c:5781 Hangup
>>>>> >> sofia/internal/501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>>>> >>
>>>>> >> 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
>>>>> >>
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >>
>>>>> >
>>>>> >
>>>>> >
>>>>> _________________________________________________________________________
>>>>> > Professional FreeSWITCH Consulting 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
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

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From cal.leeming at simplicitymedialtd.co.uk  Mon Dec 17 20:20:29 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Mon, 17 Dec 2012 17:20:29 +0000
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <468A3B64-5121-4DC9-AC68-4B5050EAAD37@mgtech.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<CAFiqYu=wL6+yk67AzzfF_HKrMBBreqeb5jybX7r8LjoXBHVs0A@mail.gmail.com>
	<CAHKQagFbEdS+wLt-CH7t_=B5UESRQkVVnFf3HSt8EB4=Gq=pOA@mail.gmail.com>
	<CABH8QX0WNTaMHXbFeJ8ip+D7ADn+b45nc2=fSHdf5F8OpWzMNQ@mail.gmail.com>
	<CAHKQagH8uu1nOBvgZ+nr_o0dV4u+G=3wfoaEx9pfxZd30F7YWw@mail.gmail.com>
	<468A3B64-5121-4DC9-AC68-4B5050EAAD37@mgtech.com>
Message-ID: <CAHKQagHff5fupd=92VoA2DLHwM-Yojs=zUbb3Uzk=RhPnFiGMw@mail.gmail.com>

Hi Mario,

If at all possible - feel free to just dump into the thread, and I can take
care of getting it into a nice format into the wiki (everyones credits will
be kept of course).

Same for everyone else!

Cal

On Mon, Dec 17, 2012 at 5:08 PM, Mario G <mario_fs at mgtech.com> wrote:

> I am planning on putting up an example of how I have 2 WAN lines both
> available to FreeSwitch, both used at the same time (load balanced), both
> are failover for each other. I have tested pulling plugs, etc. There is
> nothing in FreeSwitch defined for the lines and it runs with "-nonat", the
> router does SIP ALG fine. I thought the setup example might be useful to
> someone since it took so long to figure it out. The only hold up has been
> that I have other wiki updates to do and had FreeSwitch issues this year,
> each time one is fixed another comes up (memory leak right now) so that has
> eaten up all the time I could put into the wiki. I have already started it
> and promise to post on the ML once it's up, should be in the next 2 months.
> It may help someone in the same situation and requirements.
> Mario G
>
>
> On Dec 17, 2012, at 6:17 AM, Cal Leeming [Simplicity Media Ltd] wrote:
>
> Any other experiences/thoughts from others on this?
>
> Feedback so far has been great, lets keep it coming guys!
>
> Cal
>
> On Sun, Dec 16, 2012 at 9:52 PM, Jo?o Mesquita <jmesquita at freeswitch.org>wrote:
>
>> My take on the subject is that we are trying to tackle 2 different
>> problems at once.
>>
>> On one hand we have how NAT works and what problems it create on a VoIP
>> world. I really was not able to find any condensed documentation on the
>> internet that would describe with VoIP in mind what a admin need to know
>> about SIP packets and NAT handling. If there was one, we could just add a
>> link to the wiki page. Since NAT has no "one configuration" fix, user NEEDS
>> to know about it in order to fix his own scenario.
>>
>> On the other hand, we have how FS particularly deals with NAT (client and
>> server side). I think there is more documentation to be added/created on
>> that end as well. NAT is a complicated matter indeed and understanding
>> Sofia profiles alone is a challenge. Add NAT to the mix and the
>> configuration can look like black magic.
>>
>> So, question is, who can elaborate/contribute/find a definitive guide to
>> NAT so we can lecture users and who knows all about the NAT handling
>> internals so we can document each and every option available?
>>
>> Jo?o Mesquita
>>
>>
>>
>> On Sun, Dec 16, 2012 at 6:21 PM, Cal Leeming [Simplicity Media Ltd] <
>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>
>>> It seems that most of us agree there is no single answer to fix NAT
>>> problems - the double-NAT is something I hadn't thought about.
>>>
>>> Sean mentioned having a checklist of approaches, which would be good
>>> addition for the documentation fix.
>>>
>>> I agree that enabling NAT HACK by default could break clients that are
>>> functioning normally, but not if it is only enabled automatically under
>>> certain conditions (described in the original email).
>>>
>>> However - the upside is that it gives another layer of "this just
>>> works".. the downside is that it gives users a reason to not bother looking
>>> at why their NAT is broken in the first place.
>>>
>>> With that in mind, I'm thinking that just a documentation fix is the
>>> answer here, to avoid user lazyness.
>>>
>>> Cal
>>>
>>> On Sun, Dec 16, 2012 at 7:53 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> "There seems to be a large number of discussions surrounding NAT
>>>> traversal, as well as lots of documentation, but with no concrete answers. "
>>>>
>>>> Part of the problem is that:
>>>>
>>>>    - Not all NAT implementations function in the same way (eg some
>>>>    rewrite ports others do not)
>>>>    - Not all SIP ALG implementations work the same/work
>>>>    - Not all clients handle NAT in the same way
>>>>    - You can encounter other odd situations such as double NAT that
>>>>    further complicate matters
>>>>
>>>> So what works in one case might not work in another, so it's hard to
>>>> give a concrete 'this is how to do it' that'll work in all cases. And often
>>>> that means you need to find a failing client first then put in a NDLB
>>>> workaround for that specific client. You could enable them by default, but
>>>> that then can cause problems in other cases where the clients handle NAT
>>>> correctly.
>>>>
>>>> Roll on NAT-less IPv6 for true end-to-end connectivity*. :o)
>>>>
>>>> -Steve
>>>>
>>>>
>>>>
>>>> On 16 December 2012 16:15, Cal Leeming [Simplicity Media Ltd] <
>>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>
>>>>> *Any and all feedback on this thread would be much welcomed.*
>>>>>
>>>>> Hello,
>>>>> *
>>>>> *
>>>>> There seems to be a large number of discussions surrounding NAT
>>>>> traversal, as well as lots of documentation, but with no concrete answers.
>>>>>
>>>>> The NAT related wiki documentation is tedious, and depending on the
>>>>> outcome of this thread, I'd like to spend some time cleaning it up.
>>>>>
>>>>> The most common problem (the same as ours) was having a router with
>>>>> broken ALG and a softphone that does not seem to work with STUN.
>>>>>
>>>>> The following REGISTER is sent from a phone.
>>>>>
>>>>> REGISTER sip:1.2.3.4:5060 SIP/2.0
>>>>> Via: SIP/2.0/UDP 192.168.1.102:57787
>>>>> ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport
>>>>> Max-Forwards: 70
>>>>> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>
>>>>> To: "foxx"<sip:2000 at 1.2.3.4:5060>
>>>>> From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448
>>>>> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.
>>>>> CSeq: 7 REGISTER
>>>>> Expires: 120
>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
>>>>> REFER, INFO, MESSAGE
>>>>> Supported: replaces
>>>>> User-Agent: 3CXPhone 6.0.25732.0
>>>>> Content-Length: 0
>>>>>
>>>>> As you can see, the client's public IP is not specified
>>>>> anywhere. FreeSWITCH offers several ways around this, the main ones being;
>>>>>
>>>>> * NDLB-connectile-dysfunction
>>>>> * NDLB-force-rport
>>>>> * apply-nat-acl
>>>>> * sip-force-contact
>>>>>
>>>>> The one that has worked in our case was "NDLB-connectile-dysfunction"
>>>>> (otherwise known as NAT HACK), however there seems to be a lot of negative
>>>>> comments about using this.
>>>>>
>>>>> From what I can tell, the general argument is that NAT HACK is
>>>>> considered a non RFC compliant hack, and the SIP phones should be doing a
>>>>> better job of keeping to the RFCs.
>>>>>
>>>>> In principle, this is a fair argument - but in practise, it's not a
>>>>> reasonable assumption that all phones are RFC compliant, and (imho) not a
>>>>> reasonable argument to have this functionality disabled by default.
>>>>>
>>>>> So, I'd like to present the following arguments;
>>>>>
>>>>> * Are there any other negative aspects about
>>>>> using NDLB-connectile-dysfunction, other than it is a non compliant RFC
>>>>> hack?
>>>>>
>>>>> * Why is NDLB-connectile-dysfunction not enabled by default when
>>>>> certain conditions are met? In the event that FreeSWITCH receives a
>>>>> REGISTER from a phone specifying a Contact/Via as 192.168.0.0/16, but
>>>>> received on a public IP, then it should be obvious that NAT is broken and
>>>>> automatically try to circumvent it.
>>>>>
>>>>> * People seem to get confused between server side and client side NAT
>>>>> problems, and that they both need to be resolved in a different way. The
>>>>> documentation doesn't seem to reflect this clearly.
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting 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
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Dec 17 20:32:24 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 Dec 2012 11:32:24 -0600
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>
	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>
	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>
Message-ID: <CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>

The siptrace is not in here?  Did you enable it on the console and then
capture to the log file perhaps?
you should do "sofia tracelevel debug" too to route traces to the log file.

Its hard to say for sure with no sip trace but it seems like the far end is
rejecting the session timer re-invite causing the call to end.


On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> http://pastebin.freeswitch.org/20342




-- 
Anthony Minessale II

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
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From madan.mallikarjun at outlook.com  Mon Dec 17 15:37:09 2012
From: madan.mallikarjun at outlook.com (madan.mallikarjun)
Date: Mon, 17 Dec 2012 04:37:09 -0800 (PST)
Subject: [Freeswitch-users] FreeSwitch
Message-ID: <1355747829414-7585608.post@n2.nabble.com>

Hello All, 

           I m new to freeswitch and need all the help from where to start
and how to proceed . Is there any sort of documented way how FS can be
installed coz frankly freeswitch seems to be scattered.

thanks 
Madan 



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-tp7585608.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Mon Dec 17 20:41:58 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 Dec 2012 11:41:58 -0600
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAHKQagHff5fupd=92VoA2DLHwM-Yojs=zUbb3Uzk=RhPnFiGMw@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<CAFiqYu=wL6+yk67AzzfF_HKrMBBreqeb5jybX7r8LjoXBHVs0A@mail.gmail.com>
	<CAHKQagFbEdS+wLt-CH7t_=B5UESRQkVVnFf3HSt8EB4=Gq=pOA@mail.gmail.com>
	<CABH8QX0WNTaMHXbFeJ8ip+D7ADn+b45nc2=fSHdf5F8OpWzMNQ@mail.gmail.com>
	<CAHKQagH8uu1nOBvgZ+nr_o0dV4u+G=3wfoaEx9pfxZd30F7YWw@mail.gmail.com>
	<468A3B64-5121-4DC9-AC68-4B5050EAAD37@mgtech.com>
	<CAHKQagHff5fupd=92VoA2DLHwM-Yojs=zUbb3Uzk=RhPnFiGMw@mail.gmail.com>
Message-ID: <CAKbxfG-xL2h_5UuVoE1ib560AEhB6JmSu01tGecDet=Ev8A_pw@mail.gmail.com>

NDLB-connectile-dysfunction and sip-force-contact are both pretty
much deprecated but remain functional as a last resort.
Most problems can be solved with one of agressive-nat-detection,
NDLB-force-rport, or apply-nat-acl

agressive-nat-detection and apply-nat-acl are both geared at the 'how' NAT
is detected rather then the how to deal with it which is now centralized
into the core of sofia with fs_path support.

Stay tuned for the refresh of the FS Book where I just finished a large
chapter on NAT.  Kind of ironic that the following week everyone joins
forces to document NAT so that's great.






On Mon, Dec 17, 2012 at 11:20 AM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Hi Mario,
>
> If at all possible - feel free to just dump into the thread, and I can
> take care of getting it into a nice format into the wiki (everyones credits
> will be kept of course).
>
> Same for everyone else!
>
> Cal
>
>
> On Mon, Dec 17, 2012 at 5:08 PM, Mario G <mario_fs at mgtech.com> wrote:
>
>> I am planning on putting up an example of how I have 2 WAN lines both
>> available to FreeSwitch, both used at the same time (load balanced), both
>> are failover for each other. I have tested pulling plugs, etc. There is
>> nothing in FreeSwitch defined for the lines and it runs with "-nonat", the
>> router does SIP ALG fine. I thought the setup example might be useful to
>> someone since it took so long to figure it out. The only hold up has been
>> that I have other wiki updates to do and had FreeSwitch issues this year,
>> each time one is fixed another comes up (memory leak right now) so that has
>> eaten up all the time I could put into the wiki. I have already started it
>> and promise to post on the ML once it's up, should be in the next 2 months.
>> It may help someone in the same situation and requirements.
>> Mario G
>>
>>
>> On Dec 17, 2012, at 6:17 AM, Cal Leeming [Simplicity Media Ltd] wrote:
>>
>> Any other experiences/thoughts from others on this?
>>
>> Feedback so far has been great, lets keep it coming guys!
>>
>> Cal
>>
>> On Sun, Dec 16, 2012 at 9:52 PM, Jo?o Mesquita <jmesquita at freeswitch.org>wrote:
>>
>>> My take on the subject is that we are trying to tackle 2 different
>>> problems at once.
>>>
>>> On one hand we have how NAT works and what problems it create on a VoIP
>>> world. I really was not able to find any condensed documentation on the
>>> internet that would describe with VoIP in mind what a admin need to know
>>> about SIP packets and NAT handling. If there was one, we could just add a
>>> link to the wiki page. Since NAT has no "one configuration" fix, user NEEDS
>>> to know about it in order to fix his own scenario.
>>>
>>> On the other hand, we have how FS particularly deals with NAT (client
>>> and server side). I think there is more documentation to be added/created
>>> on that end as well. NAT is a complicated matter indeed and understanding
>>> Sofia profiles alone is a challenge. Add NAT to the mix and the
>>> configuration can look like black magic.
>>>
>>> So, question is, who can elaborate/contribute/find a definitive guide to
>>> NAT so we can lecture users and who knows all about the NAT handling
>>> internals so we can document each and every option available?
>>>
>>> Jo?o Mesquita
>>>
>>>
>>>
>>> On Sun, Dec 16, 2012 at 6:21 PM, Cal Leeming [Simplicity Media Ltd] <
>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>
>>>> It seems that most of us agree there is no single answer to fix NAT
>>>> problems - the double-NAT is something I hadn't thought about.
>>>>
>>>> Sean mentioned having a checklist of approaches, which would be good
>>>> addition for the documentation fix.
>>>>
>>>> I agree that enabling NAT HACK by default could break clients that are
>>>> functioning normally, but not if it is only enabled automatically under
>>>> certain conditions (described in the original email).
>>>>
>>>> However - the upside is that it gives another layer of "this just
>>>> works".. the downside is that it gives users a reason to not bother looking
>>>> at why their NAT is broken in the first place.
>>>>
>>>> With that in mind, I'm thinking that just a documentation fix is the
>>>> answer here, to avoid user lazyness.
>>>>
>>>> Cal
>>>>
>>>> On Sun, Dec 16, 2012 at 7:53 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>
>>>>> "There seems to be a large number of discussions surrounding NAT
>>>>> traversal, as well as lots of documentation, but with no concrete answers. "
>>>>>
>>>>> Part of the problem is that:
>>>>>
>>>>>    - Not all NAT implementations function in the same way (eg some
>>>>>    rewrite ports others do not)
>>>>>    - Not all SIP ALG implementations work the same/work
>>>>>    - Not all clients handle NAT in the same way
>>>>>    - You can encounter other odd situations such as double NAT that
>>>>>    further complicate matters
>>>>>
>>>>> So what works in one case might not work in another, so it's hard to
>>>>> give a concrete 'this is how to do it' that'll work in all cases. And often
>>>>> that means you need to find a failing client first then put in a NDLB
>>>>> workaround for that specific client. You could enable them by default, but
>>>>> that then can cause problems in other cases where the clients handle NAT
>>>>> correctly.
>>>>>
>>>>> Roll on NAT-less IPv6 for true end-to-end connectivity*. :o)
>>>>>
>>>>> -Steve
>>>>>
>>>>>
>>>>>
>>>>> On 16 December 2012 16:15, Cal Leeming [Simplicity Media Ltd] <
>>>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>>
>>>>>> *Any and all feedback on this thread would be much welcomed.*
>>>>>>
>>>>>> Hello,
>>>>>> *
>>>>>> *
>>>>>> There seems to be a large number of discussions surrounding NAT
>>>>>> traversal, as well as lots of documentation, but with no concrete answers.
>>>>>>
>>>>>> The NAT related wiki documentation is tedious, and depending on the
>>>>>> outcome of this thread, I'd like to spend some time cleaning it up.
>>>>>>
>>>>>> The most common problem (the same as ours) was having a router with
>>>>>> broken ALG and a softphone that does not seem to work with STUN.
>>>>>>
>>>>>> The following REGISTER is sent from a phone.
>>>>>>
>>>>>> REGISTER sip:1.2.3.4:5060 SIP/2.0
>>>>>> Via: SIP/2.0/UDP 192.168.1.102:57787
>>>>>> ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport
>>>>>> Max-Forwards: 70
>>>>>> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>
>>>>>> To: "foxx"<sip:2000 at 1.2.3.4:5060>
>>>>>> From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448
>>>>>> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.
>>>>>> CSeq: 7 REGISTER
>>>>>> Expires: 120
>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE,
>>>>>> NOTIFY, REFER, INFO, MESSAGE
>>>>>> Supported: replaces
>>>>>> User-Agent: 3CXPhone 6.0.25732.0
>>>>>> Content-Length: 0
>>>>>>
>>>>>> As you can see, the client's public IP is not specified
>>>>>> anywhere. FreeSWITCH offers several ways around this, the main ones being;
>>>>>>
>>>>>> * NDLB-connectile-dysfunction
>>>>>> * NDLB-force-rport
>>>>>> * apply-nat-acl
>>>>>> * sip-force-contact
>>>>>>
>>>>>> The one that has worked in our case was "NDLB-connectile-dysfunction"
>>>>>> (otherwise known as NAT HACK), however there seems to be a lot of negative
>>>>>> comments about using this.
>>>>>>
>>>>>> From what I can tell, the general argument is that NAT HACK is
>>>>>> considered a non RFC compliant hack, and the SIP phones should be doing a
>>>>>> better job of keeping to the RFCs.
>>>>>>
>>>>>> In principle, this is a fair argument - but in practise, it's not a
>>>>>> reasonable assumption that all phones are RFC compliant, and (imho) not a
>>>>>> reasonable argument to have this functionality disabled by default.
>>>>>>
>>>>>> So, I'd like to present the following arguments;
>>>>>>
>>>>>> * Are there any other negative aspects about
>>>>>> using NDLB-connectile-dysfunction, other than it is a non compliant RFC
>>>>>> hack?
>>>>>>
>>>>>> * Why is NDLB-connectile-dysfunction not enabled by default when
>>>>>> certain conditions are met? In the event that FreeSWITCH receives a
>>>>>> REGISTER from a phone specifying a Contact/Via as 192.168.0.0/16,
>>>>>> but received on a public IP, then it should be obvious that NAT is broken
>>>>>> and automatically try to circumvent it.
>>>>>>
>>>>>> * People seem to get confused between server side and client side NAT
>>>>>> problems, and that they both need to be resolved in a different way. The
>>>>>> documentation doesn't seem to reflect this clearly.
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting 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
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Anthony Minessale II

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
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From krice at freeswitch.org  Mon Dec 17 20:53:59 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 17 Dec 2012 11:53:59 -0600
Subject: [Freeswitch-users] FreeSwitch
In-Reply-To: <1355747829414-7585608.post@n2.nabble.com>
Message-ID: <CCF4B857.62370%krice@freeswitch.org>

Theres tons of information on the wiki, theres atleast 2 freeswitch books...
The real question is what are you trying to do? Theres dozens of dozens of
ways to deploy FreeSWITCH... As with any opensource switching software you
can pretty much do whatever you want, there in lies the problem, you can do
whatever you want...

But to start I would check out the FreeSWITCH book... Its available from all
the eBook places


On 12/17/12 6:37 AM, "madan.mallikarjun" <madan.mallikarjun at outlook.com>
wrote:

> Hello All, 
> 
>            I m new to freeswitch and need all the help from where to start
> and how to proceed . Is there any sort of documented way how FS can be
> installed coz frankly freeswitch seems to be scattered.
> 
> thanks 
> Madan 
> 
> 
> 
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-tp7585608.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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From cal.leeming at simplicitymedialtd.co.uk  Mon Dec 17 20:56:39 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Mon, 17 Dec 2012 17:56:39 +0000
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAKbxfG-xL2h_5UuVoE1ib560AEhB6JmSu01tGecDet=Ev8A_pw@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<CAFiqYu=wL6+yk67AzzfF_HKrMBBreqeb5jybX7r8LjoXBHVs0A@mail.gmail.com>
	<CAHKQagFbEdS+wLt-CH7t_=B5UESRQkVVnFf3HSt8EB4=Gq=pOA@mail.gmail.com>
	<CABH8QX0WNTaMHXbFeJ8ip+D7ADn+b45nc2=fSHdf5F8OpWzMNQ@mail.gmail.com>
	<CAHKQagH8uu1nOBvgZ+nr_o0dV4u+G=3wfoaEx9pfxZd30F7YWw@mail.gmail.com>
	<468A3B64-5121-4DC9-AC68-4B5050EAAD37@mgtech.com>
	<CAHKQagHff5fupd=92VoA2DLHwM-Yojs=zUbb3Uzk=RhPnFiGMw@mail.gmail.com>
	<CAKbxfG-xL2h_5UuVoE1ib560AEhB6JmSu01tGecDet=Ev8A_pw@mail.gmail.com>
Message-ID: <CAHKQagHZD62XYckx4YDNiJCFSv8uPYpiA92kfW7mR9JSgmq2hA@mail.gmail.com>

Nice, I'll be looking forward to seeing this..!

I had little success with using 'apply-nat-acl' on its own, so I'll take
another look at that (if anyone else has had success using this, please
take a minute to reply!)

Cal

On Mon, Dec 17, 2012 at 5:41 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> apply-nat-acl
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From cal.leeming at simplicitymedialtd.co.uk  Mon Dec 17 20:58:08 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Mon, 17 Dec 2012 17:58:08 +0000
Subject: [Freeswitch-users] FreeSwitch
In-Reply-To: <1355747829414-7585608.post@n2.nabble.com>
References: <1355747829414-7585608.post@n2.nabble.com>
Message-ID: <CAHKQagEgC4NCLHWCm6J1YD3UrEtN6r358=LLE3O6M2idH=+AxA@mail.gmail.com>

Welcome to FreeSWITCH!

You might want to start here;
http://wiki.freeswitch.org/wiki/Main_Page

There are a whole bunch of sections under the "New Users - Start Here".

Hope this helps

Cal

On Mon, Dec 17, 2012 at 12:37 PM, madan.mallikarjun <
madan.mallikarjun at outlook.com> wrote:

> Hello All,
>
>            I m new to freeswitch and need all the help from where to start
> and how to proceed . Is there any sort of documented way how FS can be
> installed coz frankly freeswitch seems to be scattered.
>
> thanks
> Madan
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-tp7585608.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From frederick at targointernet.com  Mon Dec 17 21:16:07 2012
From: frederick at targointernet.com (Frederick Pruneau)
Date: Mon, 17 Dec 2012 13:16:07 -0500
Subject: [Freeswitch-users] mod_fifo ringall
Message-ID: <50CF6167.1060702@targointernet.com>

Hello all!

I have set up a queue with mod_fifo. I am trying to ring agents' phones 
all at once. I have looked at the wiki page 
http://wiki.freeswitch.org/wiki/Mod_fifo. Everything is working except 
that the phones are ringing one by one. I don't find the information to 
setup ringall strategy. I have done a Google search. It seems that it is 
possible but I don't find any information how to set it up.

Can you help me with this?

Fred


From jmesquita at freeswitch.org  Mon Dec 17 21:20:36 2012
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 17 Dec 2012 15:20:36 -0300
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAHKQagHZD62XYckx4YDNiJCFSv8uPYpiA92kfW7mR9JSgmq2hA@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<CAFiqYu=wL6+yk67AzzfF_HKrMBBreqeb5jybX7r8LjoXBHVs0A@mail.gmail.com>
	<CAHKQagFbEdS+wLt-CH7t_=B5UESRQkVVnFf3HSt8EB4=Gq=pOA@mail.gmail.com>
	<CABH8QX0WNTaMHXbFeJ8ip+D7ADn+b45nc2=fSHdf5F8OpWzMNQ@mail.gmail.com>
	<CAHKQagH8uu1nOBvgZ+nr_o0dV4u+G=3wfoaEx9pfxZd30F7YWw@mail.gmail.com>
	<468A3B64-5121-4DC9-AC68-4B5050EAAD37@mgtech.com>
	<CAHKQagHff5fupd=92VoA2DLHwM-Yojs=zUbb3Uzk=RhPnFiGMw@mail.gmail.com>
	<CAKbxfG-xL2h_5UuVoE1ib560AEhB6JmSu01tGecDet=Ev8A_pw@mail.gmail.com>
	<CAHKQagHZD62XYckx4YDNiJCFSv8uPYpiA92kfW7mR9JSgmq2hA@mail.gmail.com>
Message-ID: <CABH8QX3dOjXxSBhRuhQEEZmQi1EFy5ZKuwW0xRWDqXnULAaqxQ@mail.gmail.com>

On the waiting list for the book then!! ;)

Jo?o Mesquita
FreeSWITCH? Solutions



On Mon, Dec 17, 2012 at 2:56 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Nice, I'll be looking forward to seeing this..!
>
> I had little success with using 'apply-nat-acl' on its own, so I'll take
> another look at that (if anyone else has had success using this, please
> take a minute to reply!)
>
> Cal
>
> On Mon, Dec 17, 2012 at 5:41 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> apply-nat-acl
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 bdfoster at endigotech.com  Mon Dec 17 21:23:59 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 17 Dec 2012 13:23:59 -0500
Subject: [Freeswitch-users] mod_fifo ringall
In-Reply-To: <50CF6167.1060702@targointernet.com>
References: <50CF6167.1060702@targointernet.com>
Message-ID: <CAKV-jFy-+RyVjLR1B8Xc0y1sDkhCmWh5bx7CiTqSuqVn1q9gUg@mail.gmail.com>

Depending on your situation,
mod_callcenter<http://wiki.freeswitch.org/wiki/Mod_callcenter>may be a
better fit for you.

If it makes you feel any better I can't find how to do a ringall with
mod_fifo either.


On Mon, Dec 17, 2012 at 1:16 PM, Frederick Pruneau <
frederick at targointernet.com> wrote:

> Hello all!
>
> I have set up a queue with mod_fifo. I am trying to ring agents' phones
> all at once. I have looked at the wiki page
> http://wiki.freeswitch.org/wiki/Mod_fifo. Everything is working except
> that the phones are ringing one by one. I don't find the information to
> setup ringall strategy. I have done a Google search. It seems that it is
> possible but I don't find any information how to set it up.
>
> Can you help me with this?
>
> Fred
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
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.
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From frederick at targointernet.com  Mon Dec 17 21:33:33 2012
From: frederick at targointernet.com (Frederick Pruneau)
Date: Mon, 17 Dec 2012 13:33:33 -0500
Subject: [Freeswitch-users] mod_fifo ringall
In-Reply-To: <CAKV-jFy-+RyVjLR1B8Xc0y1sDkhCmWh5bx7CiTqSuqVn1q9gUg@mail.gmail.com>
References: <50CF6167.1060702@targointernet.com>
	<CAKV-jFy-+RyVjLR1B8Xc0y1sDkhCmWh5bx7CiTqSuqVn1q9gUg@mail.gmail.com>
Message-ID: <50CF657D.1060408@targointernet.com>

mod_callcenter offers better options. But there is one option that I 
need and It does not seem to be available in mod_callcenter but it is in 
mod_fifo. The option is the exit key. For example, if the caller is 
impatient and needs to exit the queue (in my case, exit to voicemail), 
caller presses 2 to exit.

Is there any option like this in mod_callcenter?

On 2012-12-17 13:23, Brian Foster wrote:
> Depending on your situation, mod_callcenter 
> <http://wiki.freeswitch.org/wiki/Mod_callcenter> may be a better fit 
> for you.
>
> If it makes you feel any better I can't find how to do a ringall with 
> mod_fifo either.
>
>
> On Mon, Dec 17, 2012 at 1:16 PM, Frederick Pruneau 
> <frederick at targointernet.com <mailto:frederick at targointernet.com>> wrote:
>
>     Hello all!
>
>     I have set up a queue with mod_fifo. I am trying to ring agents'
>     phones
>     all at once. I have looked at the wiki page
>     http://wiki.freeswitch.org/wiki/Mod_fifo. Everything is working except
>     that the phones are ringing one by one. I don't find the
>     information to
>     setup ringall strategy. I have done a Google search. It seems that
>     it is
>     possible but I don't find any information how to set it up.
>
>     Can you help me with this?
>
>     Fred
>
>     _________________________________________________________________________
>     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
>
>
>
>
> -- 
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com <mailto: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 steveayre at gmail.com  Mon Dec 17 21:45:44 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 17 Dec 2012 18:45:44 +0000
Subject: [Freeswitch-users] FreeSwitch
In-Reply-To: <CAHKQagEgC4NCLHWCm6J1YD3UrEtN6r358=LLE3O6M2idH=+AxA@mail.gmail.com>
References: <1355747829414-7585608.post@n2.nabble.com>
	<CAHKQagEgC4NCLHWCm6J1YD3UrEtN6r358=LLE3O6M2idH=+AxA@mail.gmail.com>
Message-ID: <CAFiqYuk+jotVFK9jXQXUZJMMd77JoenWS=AWsQ+cLnpn4Bg5dQ@mail.gmail.com>

In partcular:
http://wiki.freeswitch.org/wiki/Installation_Guide
http://wiki.freeswitch.org/wiki/Getting_Started_Guide


On 17 December 2012 17:58, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Welcome to FreeSWITCH!
>
> You might want to start here;
> http://wiki.freeswitch.org/wiki/Main_Page
>
> There are a whole bunch of sections under the "New Users - Start Here".
>
> Hope this helps
>
> Cal
>
>
> On Mon, Dec 17, 2012 at 12:37 PM, madan.mallikarjun <
> madan.mallikarjun at outlook.com> wrote:
>
>> Hello All,
>>
>>            I m new to freeswitch and need all the help from where to start
>> and how to proceed . Is there any sort of documented way how FS can be
>> installed coz frankly freeswitch seems to be scattered.
>>
>> thanks
>> Madan
>>
>>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-tp7585608.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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 steveayre at gmail.com  Mon Dec 17 21:47:12 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 17 Dec 2012 18:47:12 +0000
Subject: [Freeswitch-users] G729 - [INCOMPATIBLE_DESTINATION]
In-Reply-To: <CAP2fbbeZaQG3ai6+GVyyY08=rJS=L1FJWmNk4wmb_j2J_Lhtbg@mail.gmail.com>
References: <CAP2fbbcixeTy4H+KPbxJQvAhCDWmco5a0jF0XQkGHBtE2Qkmdg@mail.gmail.com>
	<CAFiqYuk+OwUVPCmy8H1Yq5hW-+Hhy40DS1RdRe8T6oBTjeY2LA@mail.gmail.com>
	<CAMWiMy_0f44SwcXZR4CQgP7=RBqrTYW8fQQhAa8m1nR+kKv=oQ@mail.gmail.com>
	<CAP2fbbfczFPcBF5ZF0G_+FL_+n0XZPBcu6KCCSHU1ryny2hBKg@mail.gmail.com>
	<CAFiqYukVJW5BzbXeRtJmonAXxudRvuW9=h7SwCQjyyTHk3th6g@mail.gmail.com>
	<CAFiqYu=0ShYLYE+Y3rEtUw6eKp1EjSkxPaM2YkqSnYFksrfqEw@mail.gmail.com>
	<CAP2fbbeZaQG3ai6+GVyyY08=rJS=L1FJWmNk4wmb_j2J_Lhtbg@mail.gmail.com>
Message-ID: <CAFiqYunHqqSdMBVhANxduvi3pNU987EqL2qGBFB-U1bX-5KC-w@mail.gmail.com>

If you're reproducing it, then a full debug log on master head from today
would be much more useful than from the 1year old version.

-Steve



On 17 December 2012 11:30, Balamurugan Mahendran <b2m at a-cti.com> wrote:

> Yes - both are same.  Actually call is from Android SIP client. Trying to
> make outbound call from it - Getting incoming call with NO issues.
>
> Thanks,
> Bala
>
>
> On Mon, Dec 17, 2012 at 4:52 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28 18-32-35
>>> -0600
>>
>>
>> Your version is pretty old now - almost a year. Are you able to reproduce
>> this problem on the latest head of 1.2.stable and/or master?
>>
>> It could be this is a bug that's already been resolved in the past 12
>> months.
>>
>> -Steve
>>
>>
>>
>> On 17 December 2012 11:21, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
>>>> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>>>
>>> This looks odd, although I'm not sure why it's not working.
>>>
>>> The remote endpoint is using the static payload type 0 for G729.
>>> Normally that's reserved for G711 ulaw, while G729 uses 18. But the rtpmap
>>> should allow overriding that.
>>>
>>>> G729:0:8000:20:64000
>>>
>>>
>>> The 64000 rate here also looks odd - it's comparing 64khz to 8khz and I
>>> suspect that's also a reason why it doesn't recognise it. I don't know
>>> where that 64000 is coming from though.
>>>
>>> -Steve
>>>
>>>
>>>
>>>
>>>
>>> On 17 December 2012 07:31, Balamurugan Mahendran <b2m at a-cti.com> wrote:
>>>
>>>> Thanks for the help!!  Please let me know if I missed anything.
>>>>
>>>>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,SILK at 8000h
>>>> ,SILK at 12000h,SILK at 16000h,SILK at 24000h,G7221 at 32000h,G7221 at 16000h
>>>> ,G722,PCMU,PCMA,GSM"/>
>>>>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,SILK at 8000h
>>>> ,SILK at 12000h,SILK at 16000h,SILK at 24000h,G7221 at 32000h,G7221 at 16000h
>>>> ,G722,PCMU,PCMA,GSM"/>
>>>>
>>>>
>>>>  freeswitch at internal> g729_info
>>>> Permitted G729 channels: 10
>>>> Encoders in use: 4
>>>> Decoders in use: 3
>>>>
>>>> freeswitch at internal>
>>>>
>>>>
>>>>
>>>> recv 742 bytes from udp/[182.79.149.142]:16251 at 07:27:19.212195:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    INVITE sip:500 at 50.54.12.39 SIP/2.0
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport;branch=z9hG4bK68891
>>>>
>>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>>
>>>>    Max-Forwards: 70
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>>
>>>>    To: <sip:500 at 50.54.12.39>
>>>>
>>>>    Call-ID: 047707180743 at 10.4.2.7
>>>>
>>>>    CSeq: 1 INVITE
>>>>
>>>>    Contact: <sip:501 at 182.79.149.142:16251;transport=udp>
>>>>
>>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>>
>>>>    Expires: 3600
>>>>
>>>>    User-Agent: SyncPhone 1.0
>>>>
>>>>    Content-Length: 244
>>>>
>>>>    Content-Type: application/sdp
>>>>
>>>>
>>>>
>>>>    v=0
>>>>
>>>>    o=- 1111 1111 IN IP4 182.79.149.142
>>>>
>>>>    s=Session SIP/SDP
>>>>
>>>>    c=IN IP4 182.79.149.142
>>>>
>>>>    t=0 0
>>>>
>>>>    a=sendrecv
>>>>
>>>>    a=rtcp:57395 IN IP4 182.79.149.142
>>>>
>>>>    m=audio 26474 RTP/AVP 0 101
>>>>
>>>>    a=rtpmap:0 G729/8000
>>>>
>>>>    a=rtpmap:101 telephone-event/8000
>>>>
>>>>    a=fmtp:101 0-15
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> send 321 bytes to udp/[182.79.149.142]:16251 at 07:27:19.212520:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    SIP/2.0 100 Trying
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:16251
>>>> ;rport=16251;branch=z9hG4bK68891
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>>
>>>>    To: <sip:500 at 50.54.12.39>
>>>>
>>>>    Call-ID: 047707180743 at 10.4.2.7
>>>>
>>>>    CSeq: 1 INVITE
>>>>
>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>> 18-32-35 -0600
>>>>
>>>>    Content-Length: 0
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:7481 IP 182.79.149.142
>>>> Approved by acl "domains[]". Access Granted.
>>>>
>>>> 2012-12-17 07:27:19.206419 [NOTICE] switch_channel.c:930 New Channel
>>>> sofia/internal/501 at 50.54.12.39 [30a75cd4-481b-11e2-84b4-e5c28303057e]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_NEW
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:380
>>>> (sofia/internal/501 at 50.54.12.39) State NEW
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:5502 Channel sofia/internal/
>>>> 501 at 50.54.12.39 entering state [received][100]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:5513 Remote SDP:
>>>>
>>>> v=0
>>>>
>>>> o=- 1111 1111 IN IP4 182.79.149.142
>>>>
>>>> s=Session SIP/SDP
>>>>
>>>> c=IN IP4 182.79.149.142
>>>>
>>>> t=0 0
>>>>
>>>> a=sendrecv
>>>>
>>>> a=rtcp:57395 IN IP4 182.79.149.142
>>>>
>>>> m=audio 26474 RTP/AVP 0 101
>>>>
>>>> a=rtpmap:0 G729/8000
>>>>
>>>> a=rtpmap:101 telephone-event/8000
>>>>
>>>> a=fmtp:101 0-15
>>>>
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>> Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
>>>> send/recv payload to 101
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_channel.c:2852
>>>> (sofia/internal/501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
>>>>
>>>> 2012-12-17 07:27:19.206419 [NOTICE] sofia.c:5781 Hangup sofia/internal/
>>>> 501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_channel.c:2875 Send signal
>>>> sofia/internal/501 at 50.54.12.39 [KILL]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send
>>>> signal sofia/internal/501 at 50.54.12.39 [BREAK]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_HANGUP
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:602
>>>> (sofia/internal/501 at 50.54.12.39) State HANGUP
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:469 Channel
>>>> sofia/internal/501 at 50.54.12.39 hanging up, cause:
>>>> INCOMPATIBLE_DESTINATION
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:534 Responding to INVITE
>>>> with: 488
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:47
>>>> sofia/internal/501 at 50.54.12.39 Standard HANGUP, cause:
>>>> INCOMPATIBLE_DESTINATION
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:602
>>>> (sofia/internal/501 at 50.54.12.39) State HANGUP going to sleep
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:393
>>>> (sofia/internal/501 at 50.54.12.39) State Change CS_HANGUP -> CS_REPORTING
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send
>>>> signal sofia/internal/501 at 50.54.12.39 [BREAK]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_REPORTING
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:662
>>>> (sofia/internal/501 at 50.54.12.39) State REPORTING
>>>>
>>>> send 816 bytes to udp/[182.79.149.142]:16251 at 07:27:19.215295:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    SIP/2.0 488 Not Acceptable Here
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:16251
>>>> ;rport=16251;branch=z9hG4bK68891
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>>
>>>>    To: <sip:500 at 50.54.12.39>;tag=8K31Z9S2DymjN
>>>>
>>>>    Call-ID: 047707180743 at 10.4.2.7
>>>>
>>>>    CSeq: 1 INVITE
>>>>
>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>> 18-32-35 -0600
>>>>
>>>>    Accept: application/sdp
>>>>
>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>
>>>>    Supported: timer, precondition, path, replaces
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:79
>>>> sofia/internal/501 at 50.54.12.39 Standard REPORTING, cause:
>>>> INCOMPATIBLE_DESTINATION
>>>>
>>>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>>> sla, include-session-description, presence.winfo, message-summary, refer
>>>>
>>>>    Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"2012-12-17
>>>> 07:27:19.206419 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/
>>>> 501 at 50.54.12.39) State REPORTING going to sleep
>>>>
>>>>
>>>>    Content-Length: 0
>>>>
>>>>    Remote-Party-ID: "500" <sip:500 at 50.54.12.39
>>>> >;party=calling;privacy=off;screen=no
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:387
>>>> (sofia/internal/501 at 50.54.12.39) State Change CS_REPORTING ->
>>>> CS_DESTROY
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send
>>>> signal sofia/internal/501 at 50.54.12.39 [BREAK]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1380 Session
>>>> 204 (sofia/internal/501 at 50.54.12.39) Locked, Waiting on external
>>>> entities
>>>>
>>>> 2012-12-17 07:27:19.206419 [NOTICE] switch_core_session.c:1398 Session
>>>> 204 (sofia/internal/501 at 50.54.12.39) Ended
>>>>
>>>> 2012-12-17 07:27:19.206419 [NOTICE] switch_core_session.c:1400 Close
>>>> Channel sofia/internal/501 at 50.54.12.39 [CS_DESTROY]
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:491
>>>> (sofia/internal/501 at 50.54.12.39) Callstate Change HANGUP -> DOWN
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:494
>>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_DESTROY
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:504
>>>> (sofia/internal/501 at 50.54.12.39) State DESTROY
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:374 sofia/internal/
>>>> 501 at 50.54.12.39 SOFIA DESTROY
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:86
>>>> sofia/internal/501 at 50.54.12.39 Standard DESTROY
>>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:504
>>>> (sofia/internal/501 at 50.54.12.39) State DESTROY going to sleep
>>>>
>>>> recv 407 bytes from udp/[182.79.149.142]:16251 at 07:27:19.581401:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    ACK sip:500 at 50.54.12.39 SIP/2.0
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport;branch=z9hG4bK68891
>>>>
>>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>>
>>>>    Max-Forwards: 70
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>>
>>>>    To: <sip:500 at 50.54.12.39>;tag=8K31Z9S2DymjN
>>>>
>>>>    Call-ID: 047707180743 at 10.4.2.7
>>>>
>>>>    CSeq: 1 ACK
>>>>
>>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>>
>>>>    User-Agent: SyncPhone 1.0
>>>>
>>>>    Content-Length: 0
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> recv 465 bytes from udp/[182.79.149.142]:36420 at 07:27:20.672450:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    REGISTER sip:50.54.12.39 SIP/2.0
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport;branch=z9hG4bK20451
>>>>
>>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>>
>>>>    Max-Forwards: 70
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>>
>>>>    To: <sip:501 at 50.54.12.39>
>>>>
>>>>    Call-ID: 790899967509 at 10.4.2.7
>>>>
>>>>    CSeq: 1 REGISTER
>>>>
>>>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>
>>>>
>>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>>
>>>>    Expires: 3600
>>>>
>>>>    User-Agent: SyncPhone 1.0
>>>>
>>>>    Content-Length: 0
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> 2012-12-17 07:27:20.666425 [DEBUG] sofia_reg.c:1417 Send challenge for [
>>>> 501 at 50.54.12.39]
>>>>
>>>> 2012-12-17 07:27:20.666425 [WARNING] sofia_reg.c:1421 SIP auth
>>>> challenge (REGISTER) on sofia profile 'internal' for [501 at 50.54.12.39]
>>>> from ip 182.79.149.142
>>>>
>>>> send 626 bytes to udp/[182.79.149.142]:36420 at 07:27:20.673200:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    SIP/2.0 401 Unauthorized
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:36420
>>>> ;rport=36420;branch=z9hG4bK20451
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>>
>>>>    To: <sip:501 at 50.54.12.39>;tag=9vvt14a6a7a5g
>>>>
>>>>    Call-ID: 790899967509 at 10.4.2.7
>>>>
>>>>    CSeq: 1 REGISTER
>>>>
>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>> 18-32-35 -0600
>>>>
>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>
>>>>    Supported: timer, precondition, path, replaces
>>>>
>>>>    WWW-Authenticate: Digest realm="50.54.12.39",
>>>> nonce="31862298-481b-11e2-84b6-e5c28303057e", algorithm=MD5, qop="auth"
>>>>
>>>>    Content-Length: 0
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> recv 707 bytes from udp/[182.79.149.142]:36420 at 07:27:21.094563:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    REGISTER sip:50.54.12.39 SIP/2.0
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport;branch=z9hG4bK84890
>>>>
>>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>>
>>>>    Max-Forwards: 70
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>>
>>>>    To: <sip:501 at 50.54.12.39>
>>>>
>>>>    Call-ID: 790899967509 at 10.4.2.7
>>>>
>>>>    CSeq: 2 REGISTER
>>>>
>>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>>
>>>>    Expires: 3600
>>>>
>>>>    User-Agent: SyncPhone 1.0
>>>>
>>>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>
>>>>
>>>>    Authorization: Digest username="501", realm="50.54.12.39",
>>>> nonce="31862298-481b-11e2-84b6-e5c28303057e", uri="sip:50.54.12.39",
>>>> algorithm=MD5, qop=auth, nc=00000001, cnonce="a5d0aab2288476b3",
>>>> response="eb8ccc768c7f9fee038f0d5dfc599931"
>>>>
>>>>    Content-Length: 0
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'record_stereo' = 'true'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'default_gateway' = 'example.com'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'default_areacode' = '918'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'transfer_fallback_extension' = 'operator'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'toll_allow' = 'domestic,international,local'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'accountcode' = '501'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'user_context' = 'default'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'effective_caller_id_name' = 'Extension 501'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'effective_caller_id_number' = '501'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'outbound_caller_id_name' = 'FS'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'outbound_caller_id_number' = '9732386040'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header ->
>>>> 'callgroup' = 'techsupport'
>>>>
>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:1575 Register:
>>>>
>>>> From:    [501 at 10.252.148.21]
>>>>
>>>> Contact: ["user" <sip:501 at 182.79.149.142:36420;transport=udp>]
>>>>
>>>> Expires: [3600]
>>>>
>>>> send 600 bytes to udp/[182.79.149.142]:36420 at 07:27:21.096965:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    SIP/2.0 200 OK
>>>>
>>>>    Via: SIP/2.0/UDP 182.79.149.142:36420
>>>> ;rport=36420;branch=z9hG4bK84890
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>>
>>>>    To: <sip:501 at 50.54.12.39>;tag=a6NK3ZU97F1Qc
>>>>
>>>>    Call-ID: 790899967509 at 10.4.2.7
>>>>
>>>>    CSeq: 2 REGISTER
>>>>
>>>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>;expires=3600
>>>>
>>>>    Date: Mon, 17 Dec 2012 07:27:21 GMT
>>>>
>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>> 18-32-35 -0600
>>>>
>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>
>>>>    Supported: timer, precondition, path, replaces
>>>>
>>>>    Content-Length: 0
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> send 926 bytes to udp/[182.79.149.142]:36420 at 07:27:21.112029:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    NOTIFY sip:501 at 182.79.149.142:36420;transport=udp SIP/2.0
>>>>
>>>>    Via: SIP/2.0/UDP 50.54.12.39;rport;branch=z9hG4bKXXQBeBHUQ5yZp
>>>>
>>>>    Max-Forwards: 70
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=BFFc5tcD5rQar
>>>>
>>>>    To: <sip:501 at 50.54.12.39>
>>>>
>>>>    Call-ID: 09276738-c2be-1230-ec86-22000afc9415
>>>>
>>>>    CSeq: 37528556 NOTIFY
>>>>
>>>>    Contact: <sip:mod_sofia at 50.54.12.39:5060>
>>>>
>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>> 18-32-35 -0600
>>>>
>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>
>>>>    Supported: timer, precondition, path, replaces
>>>>
>>>>    Event: message-summary
>>>>
>>>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>>> sla, include-session-description, presence.winfo, message-summary, refer
>>>>
>>>>    Subscription-State: terminated;reason=noresource
>>>>
>>>>    Content-Type: application/simple-message-summary
>>>>
>>>>    Content-Length: 64
>>>>
>>>>
>>>>
>>>>    Messages-Waiting: no
>>>>
>>>>    Message-Account: sip:501 at 10.252.148.21
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> recv 295 bytes from udp/[182.79.149.142]:36420 at 07:27:21.531463:
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>    SIP/2.0 405 Method Not Allowed
>>>>
>>>>    Via: SIP/2.0/UDP 50.54.12.39;branch=z9hG4bKXXQBeBHUQ5yZp;rport=5060
>>>>
>>>>    To: <sip:501 at 50.54.12.39>
>>>>
>>>>    From: <sip:501 at 50.54.12.39>;tag=BFFc5tcD5rQar
>>>>
>>>>    Call-ID: 09276738-c2be-1230-ec86-22000afc9415
>>>>
>>>>    CSeq: 37528556 NOTIFY
>>>>
>>>>    Server: SyncPhone 1.0
>>>>
>>>>    Content-Length: 0
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>
>>>>
>>>>
>>>> On Mon, Dec 17, 2012 at 2:25 AM, curriegrad2004 <
>>>> curriegrad2004 at gmail.com> wrote:
>>>>
>>>>> If he is using mod_g729 then the log would throw an error saying that
>>>>> the codec only does passthrough mode, however he hasn't shown that
>>>>> part of the log to us yet
>>>>>
>>>>> On Sun, Dec 16, 2012 at 12:05 PM, Steven Ayre <steveayre at gmail.com>
>>>>> wrote:
>>>>> > What G729 module are you using, and what codec is in use on the
>>>>> other leg?
>>>>> >
>>>>> > G729 needs licenses due to patents.
>>>>> >
>>>>> > mod_g729 is passthrough only but requires no licenses. This is
>>>>> because it
>>>>> > merely forwards the data, but doesn't perform the
>>>>> encoding/transcoding step
>>>>> > that the patents cover. But that means it can't transcode between
>>>>> different
>>>>> > codecs.
>>>>> > mod_com_g729 is the licensed version:
>>>>> http://www.freeswitch.org/node/235
>>>>> >
>>>>> > With the passthrough mod_g729 codec, if the other leg has selected a
>>>>> codec
>>>>> > other than g729 you'll see an error in your logs and it'll hangup
>>>>> with that
>>>>> > reason.
>>>>> >
>>>>> > You can tweak your codec negotiation to avoid this (eg
>>>>> > late-negotiation=true), or you can use the licensed version.
>>>>> >
>>>>> >
>>>>> >
>>>>> >
>>>>> > On 16 December 2012 11:40, Balamurugan Mahendran <b2m at a-cti.com>
>>>>> wrote:
>>>>> >>
>>>>> >> Need help on Codec Negotiation
>>>>> >>
>>>>> >>
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare
>>>>> >> [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
>>>>> >> send/recv payload to 101
>>>>> >>
>>>>> >>
>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] switch_channel.c:2852
>>>>> >> (sofia/internal/501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
>>>>> >> 2012-12-16 11:35:52.246422 [NOTICE] sofia.c:5781 Hangup
>>>>> >> sofia/internal/501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>>>> >>
>>>>> >> 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
>>>>> >>
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >>
>>>>> >
>>>>> >
>>>>> >
>>>>> _________________________________________________________________________
>>>>> > Professional FreeSWITCH Consulting 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
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 marketing at cluecon.com  Mon Dec 17 22:11:12 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 17 Dec 2012 11:11:12 -0800
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxVxZ==+Q=sXAZo_HhJw2m2dxx5H_3wsC3-pXJLZahDF7w@mail.gmail.com>

A happy and cold (in the northern hemisphere) Monday to you all!

I'd like to start this week's news and notes by alerting everyone to the
fact that there's been a significant addition to the functionality of the
XML dialplan. If you've been around FreeSWITCH for any length of time
you've probably read the words, "Nested conditions not allowed!" in yours
or someone else's FreeSWITCH logs. As of last
Friday<http://fisheye.freeswitch.org/changelog/freeswitch.git?cs=9b7b46dbca9c4c458121dc80c3eeedf8e5ec9643>that
has changed! In response to Jira
FS-4935 <http://jira.freeswitch.org/browse/FS-4935> Anthony has added
provisional support for nested <condition> tags inside the dialplan.
(Thanks to IRC user vipkilla for adding this to the
wiki<http://wiki.freeswitch.org/wiki/Dialplan_XML#Nested_Conditions>already.)
As you may know we are working the second edition of the FreeSWITCH
"bridge" book<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book>.
I will be updating chapters 5 and 8 to reflect this new change.

It seems appropriate that with this new feature we should talk about it on this
week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_19>.
Ken Rice and I will work up some simple examples of how to use the nested
conditions and how they relate to the existing XML dialplan controls such
as the break attribute and the regex tag. If you have a dialplan example
that works well with nested conditions please email Ken and me off list.

One last reminder for our Windows users: Dave Kompel shared with us some useful
information <http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_12> for
gathering debug data when FreeSWITCH crashes under Windows. Our other main
Windows guru, Jeff Lenk, was also on the call and gave some helpful input.
If you are running under Windows you now have more tools at your disposal
with which to analyze crash data and open Jira tickets.

Have a great week and we'll talk to you on Wednesday.

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From msc at freeswitch.org  Mon Dec 17 22:22:57 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Dec 2012 11:22:57 -0800
Subject: [Freeswitch-users] Newbie question: Executing Lua scripts for
 incoming calls, how concurrency is to be handled?
In-Reply-To: <965759A53E43FE439E43565A7715E5F058F4156D32@oa-exchange1.oa.com.au>
References: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au>
	<50CE5D6E.2020401@quentustech.com>
	<965759A53E43FE439E43565A7715E5F058F4156D32@oa-exchange1.oa.com.au>
Message-ID: <CAKzWOxWFh3RNYg9NY4Dqa-=f9KZOaXTvM8NGzHpvEvzV16VUWA@mail.gmail.com>

You don't have to have actual human agents for mod_fifo. You could define a
user for each modem and then manually "log in" those "agents" on the
command line using the fifo_member API command. Something like this:

fifo_member add fifo_name {fifo_member_wait=nowait}user/1234

Where 1234 is the user id of one of the modems. You would need a SIP
registration from the PBX to FreeSWITCH for each of the modem extensions.

Having modems go through a VoIP system sounds a bit scary. What application
are you building?
-MC

On Sun, Dec 16, 2012 at 5:41 PM, Sirish Masur Mohan <
Sirish.MasurMohan at oa.com.au> wrote:

> Hi William,
>
> Thanks for the reply.
>
> My setup is as follows:
> Client(Caller)->dialup modem->PBX->FreeSWITCH->PBX-> 4 dialup
> modems->Server(Receiver)
>
> I basically need FreeSWITCH to bridge the incoming call to the best
> external destination (out of the 4 available), so that the modem training,
> connection etc can take place smoothly, before exchange of data. I am not
> sure if mod_fifo would help me in this scenario, as, I would require an
> agent to dial in and read the fifo. Could you please clarify?
>
> Thanks!
>
> With regards,
> Sirish
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William King
> Sent: Monday, 17 December 2012 10:47 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for
> incoming calls, how concurrency is to be handled?
>
> Sounds like you want to take a look into mod_fifo.
>
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From ahmed at netelsat.net  Tue Dec 18 00:15:03 2012
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Tue, 18 Dec 2012 02:15:03 +0500
Subject: [Freeswitch-users] show calls/channels b-leg false information
Message-ID: <CALFFxt1KYV67zv5D6xn5yekSbMkVyBtKn7+50Sma-ymVXPM1ig@mail.gmail.com>

Hi All,
i am trying to see information like show calls or show channels .
only confusion is wrong/missing information in *b_name and **b_ip_addr .*
these both are taken from a leg while here i was expecting to see call is
connected to which gateway.
*
*
*
*
i am having like :    *<b_name>sofia/external/919123123334</b_name>*

Means there is no "gateway name mentioned in b_name.

and in b_ip_addr its like :
*    <b_ip_addr>10.10.10.1</b_ip_addr>*

while this is ip call is coming from and b leg i was expecting as ip of
gateway call is going to.

Please let me know if i am missing something.

usring FreeSWITCH Version 1.2.5.3+git~20121206T171012Z~d74bef3f2f (git
d74bef3 2012-12-06 17:10:12Z)

Thanking you
Ahmed
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From msc at freeswitch.org  Tue Dec 18 01:43:07 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Dec 2012 14:43:07 -0800
Subject: [Freeswitch-users] show calls/channels b-leg false information
In-Reply-To: <CALFFxt1KYV67zv5D6xn5yekSbMkVyBtKn7+50Sma-ymVXPM1ig@mail.gmail.com>
References: <CALFFxt1KYV67zv5D6xn5yekSbMkVyBtKn7+50Sma-ymVXPM1ig@mail.gmail.com>
Message-ID: <CAKzWOxXj0vQv-1kQf0FFX9ZcOJ-Zdt+_X0S90+sKtgLkv0yo6Q@mail.gmail.com>

Get a call up and running and then do a uuid_dump of the a leg. I think
you'll find that there are some channel variables that get set. I don't
recall them off the top of my head but look for something like sip_gateway
or sip_gateway_name.

-MC

On Mon, Dec 17, 2012 at 1:15 PM, Ahmed Sboor <ahmed at netelsat.net> wrote:

> Hi All,
> i am trying to see information like show calls or show channels .
> only confusion is wrong/missing information in *b_name and **b_ip_addr .*
> these both are taken from a leg while here i was expecting to see call is
> connected to which gateway.
> *
> *
> *
> *
> i am having like :    *<b_name>sofia/external/919123123334</b_name>*
>
> Means there is no "gateway name mentioned in b_name.
>
> and in b_ip_addr its like :
> *    <b_ip_addr>10.10.10.1</b_ip_addr>*
>
> while this is ip call is coming from and b leg i was expecting as ip of
> gateway call is going to.
>
> Please let me know if i am missing something.
>
> usring FreeSWITCH Version 1.2.5.3+git~20121206T171012Z~d74bef3f2f (git
> d74bef3 2012-12-06 17:10:12Z)
>
> Thanking you
> Ahmed
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Tue Dec 18 01:54:31 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Dec 2012 14:54:31 -0800
Subject: [Freeswitch-users] mod_fifo ringall
In-Reply-To: <50CF6167.1060702@targointernet.com>
References: <50CF6167.1060702@targointernet.com>
Message-ID: <CAKzWOxVqXE2ZcbFbR2DnsT8wBT9FLb8LuOFt6gDQyJoORXqxVw@mail.gmail.com>

I don't think this ever got properly documented, but maybe you could test.
In conf/autoload_configs/fifo.conf.xml trying creating the fifo with an
attribute outbound_strategy="ringall". The other main strategy is
"enterprise". Would you mind doing some testing and letting us know what
shakes out? I'll help you get it documented on the wiki.

Thanks,
MC

On Mon, Dec 17, 2012 at 10:16 AM, Frederick Pruneau <
frederick at targointernet.com> wrote:

> Hello all!
>
> I have set up a queue with mod_fifo. I am trying to ring agents' phones
> all at once. I have looked at the wiki page
> http://wiki.freeswitch.org/wiki/Mod_fifo. Everything is working except
> that the phones are ringing one by one. I don't find the information to
> setup ringall strategy. I have done a Google search. It seems that it is
> possible but I don't find any information how to set it up.
>
> Can you help me with this?
>
> Fred
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Tue Dec 18 01:56:25 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Dec 2012 14:56:25 -0800
Subject: [Freeswitch-users] mod_fifo ringall
In-Reply-To: <50CF657D.1060408@targointernet.com>
References: <50CF6167.1060702@targointernet.com>
	<CAKV-jFy-+RyVjLR1B8Xc0y1sDkhCmWh5bx7CiTqSuqVn1q9gUg@mail.gmail.com>
	<50CF657D.1060408@targointernet.com>
Message-ID: <CAKzWOxXggFRpNL=P2NrvvNFXcHgw7kfvRiNvrkJ3vDQ1_WpX5w@mail.gmail.com>

They are two different philosophies for accomplishing the same basic
purpose: connect lots of inbound callers with waiting agents, making their
wait times as short as possible. Moc's mod_callcenter is very much a "true
ACD" whereas mod_fifo is, as you may have guessed, a true FIFO. However,
they both have enormous customizability. To each his own.

-MC

On Mon, Dec 17, 2012 at 10:33 AM, Frederick Pruneau <
frederick at targointernet.com> wrote:

>  mod_callcenter offers better options. But there is one option that I
> need and It does not seem to be available in mod_callcenter but it is in
> mod_fifo. The option is the exit key. For example, if the caller is
> impatient and needs to exit the queue (in my case, exit to voicemail),
> caller presses 2 to exit.
>
> Is there any option like this in mod_callcenter?
>
>
> On 2012-12-17 13:23, Brian Foster wrote:
>
> Depending on your situation, mod_callcenter<http://wiki.freeswitch.org/wiki/Mod_callcenter>may be a better fit for you.
>
>  If it makes you feel any better I can't find how to do a ringall with
> mod_fifo either.
>
>
> On Mon, Dec 17, 2012 at 1:16 PM, Frederick Pruneau <
> frederick at targointernet.com> wrote:
>
>> Hello all!
>>
>> I have set up a queue with mod_fifo. I am trying to ring agents' phones
>> all at once. I have looked at the wiki page
>> http://wiki.freeswitch.org/wiki/Mod_fifo. Everything is working except
>> that the phones are ringing one by one. I don't find the information to
>> setup ringall strategy. I have done a Google search. It seems that it is
>> possible but I don't find any information how to set it up.
>>
>> Can you help me with this?
>>
>> Fred
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>
>
>
>  --
> 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.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From nickolayr at gmail.com  Tue Dec 18 01:58:08 2012
From: nickolayr at gmail.com (Nikolay Rogoshchenkov)
Date: Mon, 17 Dec 2012 17:58:08 -0500
Subject: [Freeswitch-users] Build error with mod_spandsp
Message-ID: <CAAQCs32knLQBNJbghyCEXYDgw628kXFtPEAQo4aJ=i2okYKEvg@mail.gmail.com>

Hello,

Are anybody know how can we solve problem with building *mod_spandsp* for
FreeBSD 8.2.
I also also tried "*gmake spandsp-reconf*" (as recommended in
FS-4577<http://jira.freeswitch.org/browse/FS-4577>),
but results are the same:

[...]
*making all mod_spandsp*
*Making all in src*
*libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I.. -I/usr/local/include
-I/usr/local/include/libxml2 -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff
-DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden
-DHAVE_VISIBILITY=1 -g -O2 -MT t4_rx.lo -MD -MP -MF .deps/t4_rx.Tpo -c
t4_rx.c  -fPIC -DPIC -o t4_rx.o*
*t4_rx.c:85: error: expected '=', ',', ';', 'asm' or '__attribute__' before
'tiff_fx_field_array'*
*t4_rx.c: In function 'write_tiff_image':*
*t4_rx.c:320: warning: implicit declaration of function
'TIFFCreateCustomDirectory'*
*t4_rx.c:320: error: 'tiff_fx_field_array' undeclared (first use in this
function)*
*t4_rx.c:320: error: (Each undeclared identifier is reported only once*
*t4_rx.c:320: error: for each function it appears in.)*
*t4_rx.c:327: warning: implicit declaration of function
'TIFFWriteCustomDirectory'*
*gmake[7]: *** [t4_rx.lo] Error 1*
*gmake[6]: *** [all] Error 2*
*gmake[5]: *** [all-recursive] Error 1*
*gmake[4]: *** [/usr/src/freeswitch/libs/spandsp/src/libspandsp.la] Error 2*
*gmake[3]: *** [mod_spandsp-all] Error 1*
*gmake[2]: *** [all-recursive] Error 1*
*gmake[1]: *** [all-recursive] Error 1*
*gmake: *** [all] Error 2*
*#*

Thank you!

--
nikk
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From ahmed at netelsat.net  Tue Dec 18 02:20:07 2012
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Tue, 18 Dec 2012 04:20:07 +0500
Subject: [Freeswitch-users] show calls/channels b-leg false information
In-Reply-To: <CAKzWOxXj0vQv-1kQf0FFX9ZcOJ-Zdt+_X0S90+sKtgLkv0yo6Q@mail.gmail.com>
References: <CALFFxt1KYV67zv5D6xn5yekSbMkVyBtKn7+50Sma-ymVXPM1ig@mail.gmail.com>
	<CAKzWOxXj0vQv-1kQf0FFX9ZcOJ-Zdt+_X0S90+sKtgLkv0yo6Q@mail.gmail.com>
Message-ID: <CALFFxt0VqA6uUVnqkPwZAXgmQb9La4NcJYD0tg-9b0uWC8OYLg@mail.gmail.com>

Hi,
i tried dump and variable_sip_to_host is the variable which tells me what i
am looking for, And now question is how to get value of this variable in
b_ip_addr while doing show calls as xml ?


On Tue, Dec 18, 2012 at 3:43 AM, Michael Collins <msc at freeswitch.org> wrote:

> Get a call up and running and then do a uuid_dump of the a leg. I think
> you'll find that there are some channel variables that get set. I don't
> recall them off the top of my head but look for something like sip_gateway
> or sip_gateway_name.
>
> -MC
>
> On Mon, Dec 17, 2012 at 1:15 PM, Ahmed Sboor <ahmed at netelsat.net> wrote:
>
>> Hi All,
>> i am trying to see information like show calls or show channels .
>> only confusion is wrong/missing information in *b_name and **b_ip_addr .*
>> these both are taken from a leg while here i was expecting to see call is
>> connected to which gateway.
>> *
>> *
>> *
>> *
>> i am having like :    *<b_name>sofia/external/919123123334</b_name>*
>>
>> Means there is no "gateway name mentioned in b_name.
>>
>> and in b_ip_addr its like :
>> *    <b_ip_addr>10.10.10.1</b_ip_addr>*
>>
>> while this is ip call is coming from and b leg i was expecting as ip of
>> gateway call is going to.
>>
>> Please let me know if i am missing something.
>>
>> usring FreeSWITCH Version 1.2.5.3+git~20121206T171012Z~d74bef3f2f (git
>> d74bef3 2012-12-06 17:10:12Z)
>>
>> Thanking you
>>  Ahmed
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 sdevoy at bizfocused.com  Tue Dec 18 02:51:34 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 17 Dec 2012 18:51:34 -0500
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>
	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>
Message-ID: <541b01cddcb1$72330200$56990600$@bizfocused.com>

Waiting for another failure with siptrace REALLY on this time.

 

If the user has clicked DND on these cisco phones, could that cause these
messages?

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 12:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

The siptrace is not in here?  Did you enable it on the console and then
capture to the log file perhaps?

you should do "sofia tracelevel debug" too to route traces to the log file.

 

Its hard to say for sure with no sip trace but it seems like the far end is
rejecting the session timer re-invite causing the call to end.

 

 

On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

http://pastebin.freeswitch.org/20342





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From anthony.minessale at gmail.com  Tue Dec 18 03:00:11 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 Dec 2012 18:00:11 -0600
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <541b01cddcb1$72330200$56990600$@bizfocused.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>
	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>
	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>
	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>
	<541b01cddcb1$72330200$56990600$@bizfocused.com>
Message-ID: <CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>

usually the 406 is done in an established call to refuse a codec change
during re-invite.
Its possible the other end thinks we want to change the codec when we do
the session-timer re-invite but I'm sure we don't but the sip trace can
help shed some light.  You can run a pcap too at the same time so when we
find the bad call in the logs we can filter it out of the pcap too.  To
avoid it getting too big you can just restart it every so often or use
sippcapdump and delete calls that are not affected.




On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Waiting for another failure with siptrace REALLY on this time.****
>
> ** **
>
> If the user has clicked DND on these cisco phones, could that cause these
> messages?****
>
> ** **
>
> Sean****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 17, 2012 12:32 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
> ** **
>
> The siptrace is not in here?  Did you enable it on the console and then
> capture to the log file perhaps?****
>
> you should do "sofia tracelevel debug" too to route traces to the log file.
> ****
>
> ** **
>
> Its hard to say for sure with no sip trace but it seems like the far end
> is rejecting the session timer re-invite causing the call to end.****
>
> ** **
>
> ** **
>
> On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com>
> wrote:****
>
> http://pastebin.freeswitch.org/20342****
>
>
>
> ****
>
> ** **
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
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From steveayre at gmail.com  Tue Dec 18 03:16:44 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 18 Dec 2012 00:16:44 +0000
Subject: [Freeswitch-users] show calls/channels b-leg false information
In-Reply-To: <CALFFxt0VqA6uUVnqkPwZAXgmQb9La4NcJYD0tg-9b0uWC8OYLg@mail.gmail.com>
References: <CALFFxt1KYV67zv5D6xn5yekSbMkVyBtKn7+50Sma-ymVXPM1ig@mail.gmail.com>
	<CAKzWOxXj0vQv-1kQf0FFX9ZcOJ-Zdt+_X0S90+sKtgLkv0yo6Q@mail.gmail.com>
	<CALFFxt0VqA6uUVnqkPwZAXgmQb9La4NcJYD0tg-9b0uWC8OYLg@mail.gmail.com>
Message-ID: <CAFiqYum2zQEkLEP3Ot7-Jzjq4odJL0QU4bO0+gexZh1F7Uw7yA@mail.gmail.com>

You can't... the database doesn't store every channel variable. Try
watching ESL events instead.



On 17 December 2012 23:20, Ahmed Sboor <ahmed at netelsat.net> wrote:

> Hi,
> i tried dump and variable_sip_to_host is the variable which tells me what
> i am looking for, And now question is how to get value of this variable in
> b_ip_addr while doing show calls as xml ?
>
>
> On Tue, Dec 18, 2012 at 3:43 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Get a call up and running and then do a uuid_dump of the a leg. I think
>> you'll find that there are some channel variables that get set. I don't
>> recall them off the top of my head but look for something like sip_gateway
>> or sip_gateway_name.
>>
>> -MC
>>
>> On Mon, Dec 17, 2012 at 1:15 PM, Ahmed Sboor <ahmed at netelsat.net> wrote:
>>
>>> Hi All,
>>> i am trying to see information like show calls or show channels .
>>> only confusion is wrong/missing information in *b_name and **b_ip_addr .
>>> *
>>> these both are taken from a leg while here i was expecting to see call
>>> is connected to which gateway.
>>> *
>>> *
>>> *
>>> *
>>> i am having like :    *<b_name>sofia/external/919123123334</b_name>*
>>>
>>> Means there is no "gateway name mentioned in b_name.
>>>
>>> and in b_ip_addr its like :
>>> *    <b_ip_addr>10.10.10.1</b_ip_addr>*
>>>
>>> while this is ip call is coming from and b leg i was expecting as ip of
>>> gateway call is going to.
>>>
>>> Please let me know if i am missing something.
>>>
>>> usring FreeSWITCH Version 1.2.5.3+git~20121206T171012Z~d74bef3f2f (git
>>> d74bef3 2012-12-06 17:10:12Z)
>>>
>>> Thanking you
>>>  Ahmed
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 18 03:18:18 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 18 Dec 2012 00:18:18 +0000
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>
	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>
	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>
	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>
	<541b01cddcb1$72330200$56990600$@bizfocused.com>
	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>
Message-ID: <CAFiqYumFAPae7GbVxCtKzqSG6H86pgGUnW4Hbosp0_gNbr-yNQ@mail.gmail.com>

Getting the sip_call_id variable (Call-ID header) for each leg (eg from xml
cdrs) will give you a field that makes it very easy to pull the call out of
a pcap.

-Steve



On 18 December 2012 00:00, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> usually the 406 is done in an established call to refuse a codec change
> during re-invite.
> Its possible the other end thinks we want to change the codec when we do
> the session-timer re-invite but I'm sure we don't but the sip trace can
> help shed some light.  You can run a pcap too at the same time so when we
> find the bad call in the logs we can filter it out of the pcap too.  To
> avoid it getting too big you can just restart it every so often or use
> sippcapdump and delete calls that are not affected.
>
>
>
>
> On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>
>> Waiting for another failure with siptrace REALLY on this time.****
>>
>> ** **
>>
>> If the user has clicked DND on these cisco phones, could that cause these
>> messages?****
>>
>> ** **
>>
>> Sean****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
>> Minessale
>> *Sent:* Monday, December 17, 2012 12:32 PM
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
>> [SERVICE_NOT_IMPLEMENTED]****
>>
>> ** **
>>
>> The siptrace is not in here?  Did you enable it on the console and then
>> capture to the log file perhaps?****
>>
>> you should do "sofia tracelevel debug" too to route traces to the log
>> file.****
>>
>> ** **
>>
>> Its hard to say for sure with no sip trace but it seems like the far end
>> is rejecting the session timer re-invite causing the call to end.****
>>
>> ** **
>>
>> ** **
>>
>> On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com>
>> wrote:****
>>
>> http://pastebin.freeswitch.org/20342****
>>
>>
>>
>> ****
>>
>> ** **
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From robert.hadley at teotech.com  Tue Dec 18 03:28:14 2012
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 18 Dec 2012 00:28:14 +0000
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKzWOxVxZ==+Q=sXAZo_HhJw2m2dxx5H_3wsC3-pXJLZahDF7w@mail.gmail.com>
References: <CAKzWOxVxZ==+Q=sXAZo_HhJw2m2dxx5H_3wsC3-pXJLZahDF7w@mail.gmail.com>
Message-ID: <71943DD5C22943448A24B7C5CDC23807307223F7@CH1PRD0411MB430.namprd04.prod.outlook.com>

Hi Michael,

Thanks for the update about nested conditions (and to Anthony for doing the work :-)   Question, will this be added to the 1.2.x stable branch, and if so, when?

Regards,
Robert

From: Michael Collins [mailto:marketing at cluecon.com]
Sent: Monday, December 17, 2012 11:11 AM
To: freeswitch-users at lists.freeswitch.org; freeswitch-dev at lists.freeswitch.org; freeswitch-cluecon at lists.freeswitch.org
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes

A happy and cold (in the northern hemisphere) Monday to you all!

I'd like to start this week's news and notes by alerting everyone to the fact that there's been a significant addition to the functionality of the XML dialplan. If you've been around FreeSWITCH for any length of time you've probably read the words, "Nested conditions not allowed!" in yours or someone else's FreeSWITCH logs. As of last Friday<http://fisheye.freeswitch.org/changelog/freeswitch.git?cs=9b7b46dbca9c4c458121dc80c3eeedf8e5ec9643> that has changed! In response to Jira FS-4935<http://jira.freeswitch.org/browse/FS-4935> Anthony has added provisional support for nested <condition> tags inside the dialplan. (Thanks to IRC user vipkilla for adding this to the wiki<http://wiki.freeswitch.org/wiki/Dialplan_XML#Nested_Conditions> already.) As you may know we are working the second edition of the FreeSWITCH "bridge" book<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book>. I will be updating chapters 5 and 8 to reflect this new change.

It seems appropriate that with this new feature we should talk about it on this week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_19>. Ken Rice and I will work up some simple examples of how to use the nested conditions and how they relate to the existing XML dialplan controls such as the break attribute and the regex tag. If you have a dialplan example that works well with nested conditions please email Ken and me off list.

One last reminder for our Windows users: Dave Kompel shared with us some useful information<http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_12> for gathering debug data when FreeSWITCH crashes under Windows. Our other main Windows guru, Jeff Lenk, was also on the call and gave some helpful input. If you are running under Windows you now have more tools at your disposal with which to analyze crash data and open Jira tickets.

Have a great week and we'll talk to you on Wednesday.

--
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE

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From ahmed at netelsat.net  Tue Dec 18 03:32:28 2012
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Tue, 18 Dec 2012 05:32:28 +0500
Subject: [Freeswitch-users] show calls/channels b-leg false information
In-Reply-To: <CAFiqYum2zQEkLEP3Ot7-Jzjq4odJL0QU4bO0+gexZh1F7Uw7yA@mail.gmail.com>
References: <CALFFxt1KYV67zv5D6xn5yekSbMkVyBtKn7+50Sma-ymVXPM1ig@mail.gmail.com>
	<CAKzWOxXj0vQv-1kQf0FFX9ZcOJ-Zdt+_X0S90+sKtgLkv0yo6Q@mail.gmail.com>
	<CALFFxt0VqA6uUVnqkPwZAXgmQb9La4NcJYD0tg-9b0uWC8OYLg@mail.gmail.com>
	<CAFiqYum2zQEkLEP3Ot7-Jzjq4odJL0QU4bO0+gexZh1F7Uw7yA@mail.gmail.com>
Message-ID: <CALFFxt3KPoeN+-RBe4QVQKn+t7yrK+_WW8E8v+DhJtPqr7urXA@mail.gmail.com>

Actually my purpose is just to monitor live calls . All is going fine but
if database is not storing the channel variable how its coming as in field
" ip_addr" ? b_ip_addr is just getting the same value while it should be
having Other leg . there is another field b_dest that is also showing
correct value. Why not for b_ip_addr then ?
yesterday i was searching old threads and somewhere Anthony replied to
similar Question that it has to be updated in db . But how to do that
nowhere mentioned .



On Tue, Dec 18, 2012 at 5:16 AM, Steven Ayre <steveayre at gmail.com> wrote:

>
>
>
> On 17 December 2012 23:20, Ahmed Sboor <ahmed at netelsat.net> wrote:
>
>> Hi,
>> i tried dump and variable_sip_to_host is the variable which tells me what
>> i am looking for, And now question is how to get value of this variable in
>> b_ip_addr while doing show calls as xml ?
>>
>>
>> On Tue, Dec 18, 2012 at 3:43 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Get a call up and running and then do a uuid_dump of the a leg. I think
>>> you'll find that there are some channel variables that get set. I don't
>>> recall them off the top of my head but look for something like sip_gateway
>>> or sip_gateway_name.
>>>
>>> -MC
>>>
>>> On Mon, Dec 17, 2012 at 1:15 PM, Ahmed Sboor <ahmed at netelsat.net> wrote:
>>>
>>>> Hi All,
>>>> i am trying to see information like show calls or show channels .
>>>> only confusion is wrong/missing information in *b_name and **b_ip_addr
>>>> .*
>>>> these both are taken from a leg while here i was expecting to see call
>>>> is connected to which gateway.
>>>> *
>>>> *
>>>> *
>>>> *
>>>> i am having like :    *<b_name>sofia/external/919123123334</b_name>*
>>>>
>>>> Means there is no "gateway name mentioned in b_name.
>>>>
>>>> and in b_ip_addr its like :
>>>> *    <b_ip_addr>10.10.10.1</b_ip_addr>*
>>>>
>>>> while this is ip call is coming from and b leg i was expecting as ip of
>>>> gateway call is going to.
>>>>
>>>> Please let me know if i am missing something.
>>>>
>>>> usring FreeSWITCH Version 1.2.5.3+git~20121206T171012Z~d74bef3f2f (git
>>>> d74bef3 2012-12-06 17:10:12Z)
>>>>
>>>> Thanking you
>>>>  Ahmed
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>>>
>>>>
>>>
>>>
>>> --
>>> Michael S Collins
>>> Twitter: @mercutioviz
>>> http://www.FreeSWITCH.org
>>> http://www.ClueCon.com
>>> http://www.OSTAG.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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 Dec 18 03:45:04 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 Dec 2012 18:45:04 -0600
Subject: [Freeswitch-users] Build error with mod_spandsp
In-Reply-To: <CAAQCs32knLQBNJbghyCEXYDgw628kXFtPEAQo4aJ=i2okYKEvg@mail.gmail.com>
References: <CAAQCs32knLQBNJbghyCEXYDgw628kXFtPEAQo4aJ=i2okYKEvg@mail.gmail.com>
Message-ID: <CAKbxfG_FBDNYY_v5oX1HXr1SzgF4bMxoEe3ooB53f16vGh-dsg@mail.gmail.com>

try

make spandsp-reconf
make install

or

git pull
make sure





On Mon, Dec 17, 2012 at 4:58 PM, Nikolay Rogoshchenkov
<nickolayr at gmail.com>wrote:

> Hello,
>
> Are anybody know how can we solve problem with building *mod_spandsp* for
> FreeBSD 8.2.
> I also also tried "*gmake spandsp-reconf*" (as recommended in FS-4577<http://jira.freeswitch.org/browse/FS-4577>),
> but results are the same:
>
> [...]
> *making all mod_spandsp*
> *Making all in src*
> *libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I.. -I/usr/local/include
> -I/usr/local/include/libxml2 -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff
> -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden
> -DHAVE_VISIBILITY=1 -g -O2 -MT t4_rx.lo -MD -MP -MF .deps/t4_rx.Tpo -c
> t4_rx.c  -fPIC -DPIC -o t4_rx.o*
> *t4_rx.c:85: error: expected '=', ',', ';', 'asm' or '__attribute__'
> before 'tiff_fx_field_array'*
> *t4_rx.c: In function 'write_tiff_image':*
> *t4_rx.c:320: warning: implicit declaration of function
> 'TIFFCreateCustomDirectory'*
> *t4_rx.c:320: error: 'tiff_fx_field_array' undeclared (first use in this
> function)*
> *t4_rx.c:320: error: (Each undeclared identifier is reported only once*
> *t4_rx.c:320: error: for each function it appears in.)*
> *t4_rx.c:327: warning: implicit declaration of function
> 'TIFFWriteCustomDirectory'*
> *gmake[7]: *** [t4_rx.lo] Error 1*
> *gmake[6]: *** [all] Error 2*
> *gmake[5]: *** [all-recursive] Error 1*
> *gmake[4]: *** [/usr/src/freeswitch/libs/spandsp/src/libspandsp.la] Error
> 2*
> *gmake[3]: *** [mod_spandsp-all] Error 1*
> *gmake[2]: *** [all-recursive] Error 1*
> *gmake[1]: *** [all-recursive] Error 1*
> *gmake: *** [all] Error 2*
> *#*
>
> Thank you!
>
> --
> nikk
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Anthony Minessale II

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
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From dujinfang at gmail.com  Tue Dec 18 04:11:19 2012
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 18 Dec 2012 09:11:19 +0800
Subject: [Freeswitch-users] Video call
In-Reply-To: <A7B83A68-4703-438A-8A0F-74E669B1DB84@kavun.ch>
References: <A7B83A68-4703-438A-8A0F-74E669B1DB84@kavun.ch>
Message-ID: <70B3867611F64EFA9DAB536F87B82E33@gmail.com>

reset everything to default 

and do sth. like this in vars.xml

  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,H264"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,H264"/>


restart FS 

-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Monday, December 17, 2012 at 11:03 PM, Emrah wrote:

> Hi there,
> 
> I am having a hard time getting video working.
> 
> I have mod_h26x loaded.
> I tried setting proxy_media=true before bridging.
> I tried forcing codecs with absolute_codec_string.
> I tried inherit_codec=true.
> 
> I get no video no matter what I do.
> 
> The codec I need is H264.
> 
> What is the procedure to get video calling working on FS?
> 
> Cheers and thanks,
> Emrah
> 
> _________________________________________________________________________
> 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
> 
> 


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From sdevoy at bizfocused.com  Tue Dec 18 04:44:00 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 17 Dec 2012 20:44:00 -0500
Subject: [Freeswitch-users] Linksys Registration disappears after 5 minutes
Message-ID: <54bb01cddcc1$2752f780$75f8e680$@bizfocused.com>

Hi,

 

I have a Linksys ATA (RPT300).  I have it successfully Registering through a
NAT connection.  But, 5 minutes later FS no longer shows it registered.  The
RTP300 still says it is registered and can of course dial out, but not in.

 

For what it is worth, this RTP300 has been working for months and months.
Since we now have Polycom phones on this LAN, we had to enable SIP ALG and
NDLB.  Cisco phones and Polycom phones are all working great now, but I
cannot figure out what setting to change in the Linksys device.

 

There are no words to express how much I hate NAT and SIP.

 

Thanks,

Sean

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From nickolayr at gmail.com  Tue Dec 18 04:50:11 2012
From: nickolayr at gmail.com (Nikolay Rogoshchenkov)
Date: Mon, 17 Dec 2012 20:50:11 -0500
Subject: [Freeswitch-users] Build error with mod_spandsp
In-Reply-To: <CAKbxfG_FBDNYY_v5oX1HXr1SzgF4bMxoEe3ooB53f16vGh-dsg@mail.gmail.com>
References: <CAAQCs32knLQBNJbghyCEXYDgw628kXFtPEAQo4aJ=i2okYKEvg@mail.gmail.com>
	<CAKbxfG_FBDNYY_v5oX1HXr1SzgF4bMxoEe3ooB53f16vGh-dsg@mail.gmail.com>
Message-ID: <CAAQCs307Gr4BXL0m5pK1=n_kk=NxEzcMCBcFbGDx+4mEHBPO7A@mail.gmail.com>

Thank you Anthony,

Did you mean *g*make or make?
PS: wiki told me to use gmake for FreeBSD

--
nikk



On Mon, Dec 17, 2012 at 7:45 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try
>
> make spandsp-reconf
> make install
>
> or
>
> git pull
> make sure
>
>
>
>
>
> On Mon, Dec 17, 2012 at 4:58 PM, Nikolay Rogoshchenkov <
> nickolayr at gmail.com> wrote:
>
>> Hello,
>>
>> Are anybody know how can we solve problem with building *mod_spandsp*for FreeBSD 8.2.
>> I also also tried "*gmake spandsp-reconf*" (as recommended in FS-4577<http://jira.freeswitch.org/browse/FS-4577>),
>> but results are the same:
>>
>> [...]
>> *making all mod_spandsp*
>> *Making all in src*
>> *libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I.. -I/usr/local/include
>> -I/usr/local/include/libxml2 -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff
>> -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden
>> -DHAVE_VISIBILITY=1 -g -O2 -MT t4_rx.lo -MD -MP -MF .deps/t4_rx.Tpo -c
>> t4_rx.c  -fPIC -DPIC -o t4_rx.o*
>> *t4_rx.c:85: error: expected '=', ',', ';', 'asm' or '__attribute__'
>> before 'tiff_fx_field_array'*
>> *t4_rx.c: In function 'write_tiff_image':*
>> *t4_rx.c:320: warning: implicit declaration of function
>> 'TIFFCreateCustomDirectory'*
>> *t4_rx.c:320: error: 'tiff_fx_field_array' undeclared (first use in this
>> function)*
>> *t4_rx.c:320: error: (Each undeclared identifier is reported only once*
>> *t4_rx.c:320: error: for each function it appears in.)*
>> *t4_rx.c:327: warning: implicit declaration of function
>> 'TIFFWriteCustomDirectory'*
>> *gmake[7]: *** [t4_rx.lo] Error 1*
>> *gmake[6]: *** [all] Error 2*
>> *gmake[5]: *** [all-recursive] Error 1*
>> *gmake[4]: *** [/usr/src/freeswitch/libs/spandsp/src/libspandsp.la]
>> Error 2*
>> *gmake[3]: *** [mod_spandsp-all] Error 1*
>> *gmake[2]: *** [all-recursive] Error 1*
>> *gmake[1]: *** [all-recursive] Error 1*
>> *gmake: *** [all] Error 2*
>> *#*
>>
>> Thank you!
>>
>> --
>> nikk
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 18 04:54:33 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 Dec 2012 19:54:33 -0600
Subject: [Freeswitch-users] Linksys Registration disappears after 5
	minutes
In-Reply-To: <54bb01cddcc1$2752f780$75f8e680$@bizfocused.com>
References: <54bb01cddcc1$2752f780$75f8e680$@bizfocused.com>
Message-ID: <CAKbxfG8aL4teQ5SVxXc4p+MMV8cVRSyrTR4-5M8_S5gwdcbJ6A@mail.gmail.com>

Try going back to what you had that was working and instead of evil ALG etc:

Set sofia profile param  NDLB-force-rport to "safe"



On Mon, Dec 17, 2012 at 7:44 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Hi,****
>
> ** **
>
> I have a Linksys ATA (RPT300).  I have it successfully Registering through
> a NAT connection.  But, 5 minutes later FS no longer shows it registered.
> The RTP300 still says it is registered and can of course dial out, but not
> in.****
>
> ** **
>
> For what it is worth, this RTP300 has been working for months and months.
> Since we now have Polycom phones on this LAN, we had to enable SIP ALG and
> NDLB.  Cisco phones and Polycom phones are all working great now, but I
> cannot figure out what setting to change in the Linksys device.****
>
> ** **
>
> There are no words to express how much I hate NAT and SIP.****
>
> ** **
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
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From anthony.minessale at gmail.com  Tue Dec 18 04:55:10 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 Dec 2012 19:55:10 -0600
Subject: [Freeswitch-users] Build error with mod_spandsp
In-Reply-To: <CAAQCs307Gr4BXL0m5pK1=n_kk=NxEzcMCBcFbGDx+4mEHBPO7A@mail.gmail.com>
References: <CAAQCs32knLQBNJbghyCEXYDgw628kXFtPEAQo4aJ=i2okYKEvg@mail.gmail.com>
	<CAKbxfG_FBDNYY_v5oX1HXr1SzgF4bMxoEe3ooB53f16vGh-dsg@mail.gmail.com>
	<CAAQCs307Gr4BXL0m5pK1=n_kk=NxEzcMCBcFbGDx+4mEHBPO7A@mail.gmail.com>
Message-ID: <CAKbxfG_b5A1HVkYUuYRJjWaTRPQppEpymv=aHXL=CgQDTr3adg@mail.gmail.com>

Yes gmake


On Mon, Dec 17, 2012 at 7:50 PM, Nikolay Rogoshchenkov
<nickolayr at gmail.com>wrote:

> Thank you Anthony,
>
> Did you mean *g*make or make?
> PS: wiki told me to use gmake for FreeBSD
>
> --
> nikk
>
>
>
> On Mon, Dec 17, 2012 at 7:45 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> try
>>
>> make spandsp-reconf
>> make install
>>
>> or
>>
>> git pull
>> make sure
>>
>>
>>
>>
>>
>> On Mon, Dec 17, 2012 at 4:58 PM, Nikolay Rogoshchenkov <
>> nickolayr at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> Are anybody know how can we solve problem with building *mod_spandsp*for FreeBSD 8.2.
>>> I also also tried "*gmake spandsp-reconf*" (as recommended in FS-4577<http://jira.freeswitch.org/browse/FS-4577>),
>>> but results are the same:
>>>
>>> [...]
>>> *making all mod_spandsp*
>>> *Making all in src*
>>> *libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I.. -I/usr/local/include
>>> -I/usr/local/include/libxml2 -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff
>>> -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden
>>> -DHAVE_VISIBILITY=1 -g -O2 -MT t4_rx.lo -MD -MP -MF .deps/t4_rx.Tpo -c
>>> t4_rx.c  -fPIC -DPIC -o t4_rx.o*
>>> *t4_rx.c:85: error: expected '=', ',', ';', 'asm' or '__attribute__'
>>> before 'tiff_fx_field_array'*
>>> *t4_rx.c: In function 'write_tiff_image':*
>>> *t4_rx.c:320: warning: implicit declaration of function
>>> 'TIFFCreateCustomDirectory'*
>>> *t4_rx.c:320: error: 'tiff_fx_field_array' undeclared (first use in
>>> this function)*
>>> *t4_rx.c:320: error: (Each undeclared identifier is reported only once*
>>> *t4_rx.c:320: error: for each function it appears in.)*
>>> *t4_rx.c:327: warning: implicit declaration of function
>>> 'TIFFWriteCustomDirectory'*
>>> *gmake[7]: *** [t4_rx.lo] Error 1*
>>> *gmake[6]: *** [all] Error 2*
>>> *gmake[5]: *** [all-recursive] Error 1*
>>> *gmake[4]: *** [/usr/src/freeswitch/libs/spandsp/src/libspandsp.la]
>>> Error 2*
>>> *gmake[3]: *** [mod_spandsp-all] Error 1*
>>> *gmake[2]: *** [all-recursive] Error 1*
>>> *gmake[1]: *** [all-recursive] Error 1*
>>> *gmake: *** [all] Error 2*
>>> *#*
>>>
>>> Thank you!
>>>
>>> --
>>> nikk
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Anthony Minessale II

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
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From Sirish.MasurMohan at oa.com.au  Tue Dec 18 05:02:20 2012
From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan)
Date: Tue, 18 Dec 2012 13:02:20 +1100
Subject: [Freeswitch-users] Newbie question: Executing Lua scripts for
 incoming calls, how concurrency is to be handled?
In-Reply-To: <CAKzWOxWFh3RNYg9NY4Dqa-=f9KZOaXTvM8NGzHpvEvzV16VUWA@mail.gmail.com>
References: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au>
	<50CE5D6E.2020401@quentustech.com>
	<965759A53E43FE439E43565A7715E5F058F4156D32@oa-exchange1.oa.com.au>
	<CAKzWOxWFh3RNYg9NY4Dqa-=f9KZOaXTvM8NGzHpvEvzV16VUWA@mail.gmail.com>
Message-ID: <965759A53E43FE439E43565A7715E5F058F4156D9E@oa-exchange1.oa.com.au>

Hi Michael,

Thanks for the reply.

>> You would need a SIP registration from the PBX to FreeSWITCH for each of the modem extensions
I am seen SIP clients such as X-Lite sending out the SIP registrations, but could you please clarify as to how this can be achieved in the PBX? The final production environment would be out in the customer's PBX, which I may not have complete control of..

>> What application are you building?
I may not be able to provide the details because of the NDA with customer, but what I am trying to achieve is, to replace an existing IVR with FreeSWITCH in an old existing setup of the customer - that's the reason why we continue working with dialup modems!

With regards,
Sirish

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, 18 December 2012 6:23 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

You don't have to have actual human agents for mod_fifo. You could define a user for each modem and then manually "log in" those "agents" on the command line using the fifo_member API command. Something like this:

fifo_member add fifo_name {fifo_member_wait=nowait}user/1234

Where 1234 is the user id of one of the modems. You would need a SIP registration from the PBX to FreeSWITCH for each of the modem extensions.

Having modems go through a VoIP system sounds a bit scary. What application are you building?
-MC
On Sun, Dec 16, 2012 at 5:41 PM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au<mailto:Sirish.MasurMohan at oa.com.au>> wrote:
Hi William,

Thanks for the reply.

My setup is as follows:
Client(Caller)->dialup modem->PBX->FreeSWITCH->PBX-> 4 dialup modems->Server(Receiver)

I basically need FreeSWITCH to bridge the incoming call to the best external destination (out of the 4 available), so that the modem training, connection etc can take place smoothly, before exchange of data. I am not sure if mod_fifo would help me in this scenario, as, I would require an agent to dial in and read the fifo. Could you please clarify?

Thanks!

With regards,
Sirish

-----Original Message-----
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 William King
Sent: Monday, 17 December 2012 10:47 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

Sounds like you want to take a look into mod_fifo.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337<tel:%28877%29%20211-9337>
Office: (206) 388-4772<tel:%28206%29%20388-4772>
Cell:   (253) 686-5518<tel:%28253%29%20686-5518>
william.king at quentustech.com<mailto:william.king at quentustech.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

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From sdevoy at bizfocused.com  Tue Dec 18 05:18:48 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 17 Dec 2012 21:18:48 -0500
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>	<541b01cddcb1$72330200$56990600$@bizfocused.com>
	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>
Message-ID: <550801cddcc6$040145c0$0c03d140$@bizfocused.com>

Ah, this is interesting.  I have pointed out to the customer that this error
has ONLY happened when the call is from THIS SPECIFIC caller.  They said it
is likely just because this is a current customer site where they have many
calls.  That may point to a codec issue, but why only occasionally?

 

After hours today, I tired an identical enterprise bridge statement  from my
office something like 50 times and could not reproduce the error.

I am waiting for a recurrence.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 7:00 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

usually the 406 is done in an established call to refuse a codec change
during re-invite.

Its possible the other end thinks we want to change the codec when we do the
session-timer re-invite but I'm sure we don't but the sip trace can help
shed some light.  You can run a pcap too at the same time so when we find
the bad call in the logs we can filter it out of the pcap too.  To avoid it
getting too big you can just restart it every so often or use sippcapdump
and delete calls that are not affected.

 

 

 

On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Waiting for another failure with siptrace REALLY on this time.

 

If the user has clicked DND on these cisco phones, could that cause these
messages?

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 12:32 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

The siptrace is not in here?  Did you enable it on the console and then
capture to the log file perhaps?

you should do "sofia tracelevel debug" too to route traces to the log file.

 

Its hard to say for sure with no sip trace but it seems like the far end is
rejecting the session timer re-invite causing the call to end.

 

 

On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

http://pastebin.freeswitch.org/20342





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900> 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From sdevoy at bizfocused.com  Tue Dec 18 05:40:01 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 17 Dec 2012 21:40:01 -0500
Subject: [Freeswitch-users] Linksys Registration disappears after
	5	minutes
In-Reply-To: <CAKbxfG8aL4teQ5SVxXc4p+MMV8cVRSyrTR4-5M8_S5gwdcbJ6A@mail.gmail.com>
References: <54bb01cddcc1$2752f780$75f8e680$@bizfocused.com>
	<CAKbxfG8aL4teQ5SVxXc4p+MMV8cVRSyrTR4-5M8_S5gwdcbJ6A@mail.gmail.com>
Message-ID: <553001cddcc8$fac448b0$f04cda10$@bizfocused.com>

Anthony,

Your killing me man!  Without SIP ALG, I could not find any configuration to
get 2 Polycom 335s Registered.

 

NDLB-force-rport to "safe"      -  made no difference.

 

I left rport to safe and tried removing :

<variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>
from this user.    -  WORKED!!!

 

So on my FIOS router with SIP ALG on,  Polycom and Cisco phones REQUIRE NDLB
variable and LinkSys RPT300 CANNOT HAVE IT in the user profile.  This is
starting to feel a lot like witchcraft! 

 

Have you ever read about BF Skinner's pigeons that were given random
reinforcement rewards?  

http://vidallena.org/skinpal.htm

 

Thanks again,

Sean

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 8:55 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Linksys Registration disappears after 5
minutes

 

Try going back to what you had that was working and instead of evil ALG etc:

 

Set sofia profile param  NDLB-force-rport to "safe"

 

 

On Mon, Dec 17, 2012 at 7:44 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Hi,

 

I have a Linksys ATA (RPT300).  I have it successfully Registering through a
NAT connection.  But, 5 minutes later FS no longer shows it registered.  The
RTP300 still says it is registered and can of course dial out, but not in.

 

For what it is worth, this RTP300 has been working for months and months.
Since we now have Polycom phones on this LAN, we had to enable SIP ALG and
NDLB.  Cisco phones and Polycom phones are all working great now, but I
cannot figure out what setting to change in the Linksys device.

 

There are no words to express how much I hate NAT and SIP.

 

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

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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From krice at freeswitch.org  Tue Dec 18 05:47:47 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 17 Dec 2012 20:47:47 -0600
Subject: [Freeswitch-users] Linksys Registration disappears after 5
 minutes
In-Reply-To: <553001cddcc8$fac448b0$f04cda10$@bizfocused.com>
Message-ID: <CCF53573.623BB%krice@freeswitch.org>

Setting NDLB-force-rport to safe was specifically done for Polycoms


On 12/17/12 8:40 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> Anthony,
> Your killing me man!  Without SIP ALG, I could not find any configuration to
> get 2 Polycom 335s Registered.
>  
> NDLB-force-rport to "safe?      -  made no difference.
>  
> I left rport to safe and tried removing :
> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> from
> this user.    -  WORKED!!!
>  
> So on my FIOS router with SIP ALG on,  Polycom and Cisco phones REQUIRE NDLB
> variable and LinkSys RPT300 CANNOT HAVE IT in the user profile.  This is
> starting to feel a lot like witchcraft!
>  
> Have you ever read about BF Skinner?s pigeons that were given random
> reinforcement rewards?
> http://vidallena.org/skinpal.htm
>  
> Thanks again,
> Sean
>  
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
> Minessale
> Sent: Monday, December 17, 2012 8:55 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Linksys Registration disappears after 5
> minutes
>  
> 
> Try going back to what you had that was working and instead of evil ALG etc:
> 
>  
> 
> Set sofia profile param  NDLB-force-rport to "safe"
> 
>  
> 
>  
> 
> On Mon, Dec 17, 2012 at 7:44 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> 
> Hi,
>  
> I have a Linksys ATA (RPT300).  I have it successfully Registering through a
> NAT connection.  But, 5 minutes later FS no longer shows it registered.  The
> RTP300 still says it is registered and can of course dial out, but not in.
>  
> For what it is worth, this RTP300 has been working for months and months.
> Since we now have Polycom phones on this LAN, we had to enable SIP ALG and
> NDLB.  Cisco phones and Polycom phones are all working great now, but I cannot
> figure out what setting to change in the Linksys device.
>  
> There are no words to express how much I hate NAT and SIP.
>  
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
>  

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From msc at freeswitch.org  Tue Dec 18 09:21:46 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Dec 2012 22:21:46 -0800
Subject: [Freeswitch-users] Linksys Registration disappears after 5
	minutes
In-Reply-To: <553001cddcc8$fac448b0$f04cda10$@bizfocused.com>
References: <54bb01cddcc1$2752f780$75f8e680$@bizfocused.com>
	<CAKbxfG8aL4teQ5SVxXc4p+MMV8cVRSyrTR4-5M8_S5gwdcbJ6A@mail.gmail.com>
	<553001cddcc8$fac448b0$f04cda10$@bizfocused.com>
Message-ID: <CAKzWOxWCaVnv0vB2caDzqnBqyBdHbenOHAPsbfb3dcqY8Kbo-Q@mail.gmail.com>

Unless you have a REALLY compelling reason there's nothing preventing you
from making a two different SIP profiles: one for the Poly/Cisco and one
for the Linksys. As long as you use the "user/xxxx" dialstring then FS
won't care which profile the device is registered on.

-MC

On Mon, Dec 17, 2012 at 6:40 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Anthony,****
>
> Your killing me man!  Without SIP ALG, I could not find any configuration
> to get 2 Polycom 335s Registered.****
>
> ** **
>
> NDLB-force-rport to "safe?      -  made no difference.****
>
> ** **
>
> I left rport to safe and tried *removing* :****
>
> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> from
> this user.    -  WORKED!!!****
>
> ** **
>
> So on my FIOS router with SIP ALG on,  Polycom and Cisco phones REQUIRE
> NDLB variable and LinkSys RPT300 CANNOT HAVE IT in the user profile.  This
> is starting to feel a lot like witchcraft! ****
>
> ** **
>
> Have you ever read about BF Skinner?s pigeons that were given random
> reinforcement rewards?  ****
>
> http://vidallena.org/skinpal.htm****
>
> ** **
>
> Thanks again,****
>
> Sean****
>
> ** **
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 17, 2012 8:55 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Linksys Registration disappears after 5
> minutes****
>
> ** **
>
> Try going back to what you had that was working and instead of evil ALG
> etc:****
>
> ** **
>
> Set sofia profile param  NDLB-force-rport to "safe"****
>
> ** **
>
> ** **
>
> On Mon, Dec 17, 2012 at 7:44 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> Hi,****
>
>  ****
>
> I have a Linksys ATA (RPT300).  I have it successfully Registering through
> a NAT connection.  But, 5 minutes later FS no longer shows it registered.
> The RTP300 still says it is registered and can of course dial out, but not
> in.****
>
>  ****
>
> For what it is worth, this RTP300 has been working for months and months.
> Since we now have Polycom phones on this LAN, we had to enable SIP ALG and
> NDLB.  Cisco phones and Polycom phones are all working great now, but I
> cannot figure out what setting to change in the Linksys device.****
>
>  ****
>
> There are no words to express how much I hate NAT and SIP.****
>
>  ****
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
> ** **
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Tue Dec 18 09:24:09 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Dec 2012 22:24:09 -0800
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <71943DD5C22943448A24B7C5CDC23807307223F7@CH1PRD0411MB430.namprd04.prod.outlook.com>
References: <CAKzWOxVxZ==+Q=sXAZo_HhJw2m2dxx5H_3wsC3-pXJLZahDF7w@mail.gmail.com>
	<71943DD5C22943448A24B7C5CDC23807307223F7@CH1PRD0411MB430.namprd04.prod.outlook.com>
Message-ID: <CAKzWOxX7=KSc9V2tpf_N5S966ZDT-06wkLjD1=0982geCdVTyQ@mail.gmail.com>

Yes, it will go into the stable branch when we decide to do the next
release, which I believe is 1.2.6. Ken Rice could give you more information
but I suspect it won't be more than a week or two depending on feedback
from early adopters...

-MC

On Mon, Dec 17, 2012 at 4:28 PM, Robert Hadley <robert.hadley at teotech.com>wrote:

>  Hi Michael,****
>
> ** **
>
> Thanks for the update about nested conditions (and to Anthony for doing
> the work :-)   Question, will this be added to the 1.2.x stable branch, and
> if so, when?****
>
> ** **
>
> Regards,****
>
> Robert****
>
> ** **
>
> *From:* Michael Collins [mailto:marketing at cluecon.com]
> *Sent:* Monday, December 17, 2012 11:11 AM
> *To:* freeswitch-users at lists.freeswitch.org;
> freeswitch-dev at lists.freeswitch.org;
> freeswitch-cluecon at lists.freeswitch.org
> *Subject:* [Freeswitch-users] FreeSWITCH Weekly News and Notes****
>
> ** **
>
> A happy and cold (in the northern hemisphere) Monday to you all!
>
> I'd like to start this week's news and notes by alerting everyone to the
> fact that there's been a significant addition to the functionality of the
> XML dialplan. If you've been around FreeSWITCH for any length of time
> you've probably read the words, "Nested conditions not allowed!" in yours
> or someone else's FreeSWITCH logs. As of last Friday<http://fisheye.freeswitch.org/changelog/freeswitch.git?cs=9b7b46dbca9c4c458121dc80c3eeedf8e5ec9643>that has changed! In response to Jira
> FS-4935 <http://jira.freeswitch.org/browse/FS-4935> Anthony has added
> provisional support for nested <condition> tags inside the dialplan.
> (Thanks to IRC user vipkilla for adding this to the wiki<http://wiki.freeswitch.org/wiki/Dialplan_XML#Nested_Conditions>already.) As you may know we are working the second edition of the FreeSWITCH
> "bridge" book<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book>.
> I will be updating chapters 5 and 8 to reflect this new change.
>
> It seems appropriate that with this new feature we should talk about it on this
> week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_19>.
> Ken Rice and I will work up some simple examples of how to use the nested
> conditions and how they relate to the existing XML dialplan controls such
> as the break attribute and the regex tag. If you have a dialplan example
> that works well with nested conditions please email Ken and me off list.
>
> One last reminder for our Windows users: Dave Kompel shared with us some useful
> information <http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_12> for
> gathering debug data when FreeSWITCH crashes under Windows. Our other main
> Windows guru, Jeff Lenk, was also on the call and gave some helpful input.
> If you are running under Windows you now have more tools at your disposal
> with which to analyze crash data and open Jira tickets.
>
> Have a great week and we'll talk to you on Wednesday.
>
> --
> Michael S Collins****
>
> ClueCon Team****
>
> http://www.cluecon.com****
>
> 877-7-4ACLUE****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Tue Dec 18 09:31:57 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Dec 2012 22:31:57 -0800
Subject: [Freeswitch-users] show calls/channels b-leg false information
In-Reply-To: <CALFFxt3KPoeN+-RBe4QVQKn+t7yrK+_WW8E8v+DhJtPqr7urXA@mail.gmail.com>
References: <CALFFxt1KYV67zv5D6xn5yekSbMkVyBtKn7+50Sma-ymVXPM1ig@mail.gmail.com>
	<CAKzWOxXj0vQv-1kQf0FFX9ZcOJ-Zdt+_X0S90+sKtgLkv0yo6Q@mail.gmail.com>
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Message-ID: <CAKzWOxV92K9R4ZOSxxs5=9CJkjbFK1oCowdGni1ia_qjO9ZW6g@mail.gmail.com>

Ah - be careful. The "database" is not the entire picture. When you do show
channels or show calls it pulls right from some sqlite3 tables that are
frequently updated by FreeSWITCH. You have a few choices depending upon
just how intense your monitoring will be:

#1 - Follow Steven's advice and monitor CHANNEL events on the event socket
#2 - Use some form of polling and "uuid_getvar <uuid> sip_gateway_name" to
get the name for each channel
#3 - Modify the source code to store the sip_gateway_name in the same table
that stores the rest of the show channels/show calls information. (Most
people don't realize that you can do this. :)

Personally I like event socket/ESL so I would go with #1 first.

-MC

On Mon, Dec 17, 2012 at 4:32 PM, Ahmed Sboor <ahmed at netelsat.net> wrote:

> Actually my purpose is just to monitor live calls . All is going fine but
> if database is not storing the channel variable how its coming as in field
> " ip_addr" ? b_ip_addr is just getting the same value while it should be
> having Other leg . there is another field b_dest that is also showing
> correct value. Why not for b_ip_addr then ?
> yesterday i was searching old threads and somewhere Anthony replied to
> similar Question that it has to be updated in db . But how to do that
> nowhere mentioned .
>
>
>
> On Tue, Dec 18, 2012 at 5:16 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>>
>>
>>
>> On 17 December 2012 23:20, Ahmed Sboor <ahmed at netelsat.net> wrote:
>>
>>> Hi,
>>> i tried dump and variable_sip_to_host is the variable which tells me
>>> what i am looking for, And now question is how to get value of this
>>> variable in b_ip_addr while doing show calls as xml ?
>>>
>>>
>>> On Tue, Dec 18, 2012 at 3:43 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Get a call up and running and then do a uuid_dump of the a leg. I think
>>>> you'll find that there are some channel variables that get set. I don't
>>>> recall them off the top of my head but look for something like sip_gateway
>>>> or sip_gateway_name.
>>>>
>>>> -MC
>>>>
>>>> On Mon, Dec 17, 2012 at 1:15 PM, Ahmed Sboor <ahmed at netelsat.net>wrote:
>>>>
>>>>> Hi All,
>>>>> i am trying to see information like show calls or show channels .
>>>>> only confusion is wrong/missing information in *b_name and **b_ip_addr
>>>>> .*
>>>>> these both are taken from a leg while here i was expecting to see call
>>>>> is connected to which gateway.
>>>>> *
>>>>> *
>>>>> *
>>>>> *
>>>>> i am having like :    *<b_name>sofia/external/919123123334</b_name>*
>>>>>
>>>>> Means there is no "gateway name mentioned in b_name.
>>>>>
>>>>> and in b_ip_addr its like :
>>>>> *    <b_ip_addr>10.10.10.1</b_ip_addr>*
>>>>>
>>>>> while this is ip call is coming from and b leg i was expecting as ip
>>>>> of gateway call is going to.
>>>>>
>>>>> Please let me know if i am missing something.
>>>>>
>>>>> usring FreeSWITCH Version 1.2.5.3+git~20121206T171012Z~d74bef3f2f (git
>>>>> d74bef3 2012-12-06 17:10:12Z)
>>>>>
>>>>> Thanking you
>>>>>  Ahmed
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting 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
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Michael S Collins
>>>> Twitter: @mercutioviz
>>>> http://www.FreeSWITCH.org
>>>> http://www.ClueCon.com
>>>> http://www.OSTAG.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From steveayre at gmail.com  Tue Dec 18 10:51:26 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 18 Dec 2012 07:51:26 +0000
Subject: [Freeswitch-users] show calls/channels b-leg false information
In-Reply-To: <CAKzWOxV92K9R4ZOSxxs5=9CJkjbFK1oCowdGni1ia_qjO9ZW6g@mail.gmail.com>
References: <CALFFxt1KYV67zv5D6xn5yekSbMkVyBtKn7+50Sma-ymVXPM1ig@mail.gmail.com>
	<CAKzWOxXj0vQv-1kQf0FFX9ZcOJ-Zdt+_X0S90+sKtgLkv0yo6Q@mail.gmail.com>
	<CALFFxt0VqA6uUVnqkPwZAXgmQb9La4NcJYD0tg-9b0uWC8OYLg@mail.gmail.com>
	<CAFiqYum2zQEkLEP3Ot7-Jzjq4odJL0QU4bO0+gexZh1F7Uw7yA@mail.gmail.com>
	<CALFFxt3KPoeN+-RBe4QVQKn+t7yrK+_WW8E8v+DhJtPqr7urXA@mail.gmail.com>
	<CAKzWOxV92K9R4ZOSxxs5=9CJkjbFK1oCowdGni1ia_qjO9ZW6g@mail.gmail.com>
Message-ID: <CAFiqYumJzso8FZ3PEJqLYO-kFUrJLOHO38LjXMRzrXaucKTnQg@mail.gmail.com>

It's worth noting that show channels/calls just map to a SQL query on the
sqlite/ODBC tables used by the core. They're a partial overview of the
call, as Michael notes not the full picture.

In particular it's the core that's writing to that database - so it's not
going to contain anything endpoint-specific. So all the sip_ fields set by
mod_sofia won't be stored there.

They do at least give you the UUID of the call - you could get the other
variables from the channels using uuid_getvar using the same ESL connection
you already have open to run the show api in.

Michael,
#3 is possible yes, but won't that give you a diff against upstream that'll
never be merged and as upstream changes won't apply cleanly any more,
making it a headache to maintain?

-Steve


On 18 December 2012 06:31, Michael Collins <msc at freeswitch.org> wrote:

> Ah - be careful. The "database" is not the entire picture. When you do
> show channels or show calls it pulls right from some sqlite3 tables that
> are frequently updated by FreeSWITCH. You have a few choices depending upon
> just how intense your monitoring will be:
>
> #1 - Follow Steven's advice and monitor CHANNEL events on the event socket
> #2 - Use some form of polling and "uuid_getvar <uuid> sip_gateway_name" to
> get the name for each channel
> #3 - Modify the source code to store the sip_gateway_name in the same
> table that stores the rest of the show channels/show calls information.
> (Most people don't realize that you can do this. :)
>
> Personally I like event socket/ESL so I would go with #1 first.
>
> -MC
>
>
> On Mon, Dec 17, 2012 at 4:32 PM, Ahmed Sboor <ahmed at netelsat.net> wrote:
>
>> Actually my purpose is just to monitor live calls . All is going fine but
>> if database is not storing the channel variable how its coming as in field
>> " ip_addr" ? b_ip_addr is just getting the same value while it should be
>> having Other leg . there is another field b_dest that is also showing
>> correct value. Why not for b_ip_addr then ?
>> yesterday i was searching old threads and somewhere Anthony replied to
>> similar Question that it has to be updated in db . But how to do that
>> nowhere mentioned .
>>
>>
>>
>> On Tue, Dec 18, 2012 at 5:16 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>>
>>>
>>>
>>> On 17 December 2012 23:20, Ahmed Sboor <ahmed at netelsat.net> wrote:
>>>
>>>> Hi,
>>>> i tried dump and variable_sip_to_host is the variable which tells me
>>>> what i am looking for, And now question is how to get value of this
>>>> variable in b_ip_addr while doing show calls as xml ?
>>>>
>>>>
>>>> On Tue, Dec 18, 2012 at 3:43 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> Get a call up and running and then do a uuid_dump of the a leg. I
>>>>> think you'll find that there are some channel variables that get set. I
>>>>> don't recall them off the top of my head but look for something like
>>>>> sip_gateway or sip_gateway_name.
>>>>>
>>>>> -MC
>>>>>
>>>>> On Mon, Dec 17, 2012 at 1:15 PM, Ahmed Sboor <ahmed at netelsat.net>wrote:
>>>>>
>>>>>> Hi All,
>>>>>> i am trying to see information like show calls or show channels .
>>>>>> only confusion is wrong/missing information in *b_name and **b_ip_addr
>>>>>> .*
>>>>>> these both are taken from a leg while here i was expecting to see
>>>>>> call is connected to which gateway.
>>>>>> *
>>>>>> *
>>>>>> *
>>>>>> *
>>>>>> i am having like :    *<b_name>sofia/external/919123123334</b_name>*
>>>>>>
>>>>>> Means there is no "gateway name mentioned in b_name.
>>>>>>
>>>>>> and in b_ip_addr its like :
>>>>>> *    <b_ip_addr>10.10.10.1</b_ip_addr>*
>>>>>>
>>>>>> while this is ip call is coming from and b leg i was expecting as ip
>>>>>> of gateway call is going to.
>>>>>>
>>>>>> Please let me know if i am missing something.
>>>>>>
>>>>>> usring FreeSWITCH Version 1.2.5.3+git~20121206T171012Z~d74bef3f2f
>>>>>> (git d74bef3 2012-12-06 17:10:12Z)
>>>>>>
>>>>>> Thanking you
>>>>>>  Ahmed
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting 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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Michael S Collins
>>>>> Twitter: @mercutioviz
>>>>> http://www.FreeSWITCH.org
>>>>> http://www.ClueCon.com
>>>>> http://www.OSTAG.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting 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
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 b2m at a-cti.com  Tue Dec 18 11:27:44 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Tue, 18 Dec 2012 13:57:44 +0530
Subject: [Freeswitch-users] G729 - [INCOMPATIBLE_DESTINATION]
In-Reply-To: <CAFiqYunHqqSdMBVhANxduvi3pNU987EqL2qGBFB-U1bX-5KC-w@mail.gmail.com>
References: <CAP2fbbcixeTy4H+KPbxJQvAhCDWmco5a0jF0XQkGHBtE2Qkmdg@mail.gmail.com>
	<CAFiqYuk+OwUVPCmy8H1Yq5hW-+Hhy40DS1RdRe8T6oBTjeY2LA@mail.gmail.com>
	<CAMWiMy_0f44SwcXZR4CQgP7=RBqrTYW8fQQhAa8m1nR+kKv=oQ@mail.gmail.com>
	<CAP2fbbfczFPcBF5ZF0G_+FL_+n0XZPBcu6KCCSHU1ryny2hBKg@mail.gmail.com>
	<CAFiqYukVJW5BzbXeRtJmonAXxudRvuW9=h7SwCQjyyTHk3th6g@mail.gmail.com>
	<CAFiqYu=0ShYLYE+Y3rEtUw6eKp1EjSkxPaM2YkqSnYFksrfqEw@mail.gmail.com>
	<CAP2fbbeZaQG3ai6+GVyyY08=rJS=L1FJWmNk4wmb_j2J_Lhtbg@mail.gmail.com>
	<CAFiqYunHqqSdMBVhANxduvi3pNU987EqL2qGBFB-U1bX-5KC-w@mail.gmail.com>
Message-ID: <CAP2fbbcnssdM=C+PMi9QZOXAMfXWKFCQkt9RCxKp7CoWbOskHA@mail.gmail.com>

Thank You! will do update ASAP, will update my findings here.

Thanks,
Bala

On Tue, Dec 18, 2012 at 12:17 AM, Steven Ayre <steveayre at gmail.com> wrote:

> If you're reproducing it, then a full debug log on master head from today
> would be much more useful than from the 1year old version.
>
> -Steve
>
>
>
> On 17 December 2012 11:30, Balamurugan Mahendran <b2m at a-cti.com> wrote:
>
>> Yes - both are same.  Actually call is from Android SIP client. Trying to
>> make outbound call from it - Getting incoming call with NO issues.
>>
>> Thanks,
>> Bala
>>
>>
>> On Mon, Dec 17, 2012 at 4:52 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>> 18-32-35 -0600
>>>
>>>
>>> Your version is pretty old now - almost a year. Are you able to
>>> reproduce this problem on the latest head of 1.2.stable and/or master?
>>>
>>> It could be this is a bug that's already been resolved in the past 12
>>> months.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 17 December 2012 11:21, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>>>>
>>>> This looks odd, although I'm not sure why it's not working.
>>>>
>>>> The remote endpoint is using the static payload type 0 for G729.
>>>> Normally that's reserved for G711 ulaw, while G729 uses 18. But the rtpmap
>>>> should allow overriding that.
>>>>
>>>>> G729:0:8000:20:64000
>>>>
>>>>
>>>> The 64000 rate here also looks odd - it's comparing 64khz to 8khz and I
>>>> suspect that's also a reason why it doesn't recognise it. I don't know
>>>> where that 64000 is coming from though.
>>>>
>>>> -Steve
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On 17 December 2012 07:31, Balamurugan Mahendran <b2m at a-cti.com> wrote:
>>>>
>>>>> Thanks for the help!!  Please let me know if I missed anything.
>>>>>
>>>>>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,SILK at 8000h
>>>>> ,SILK at 12000h,SILK at 16000h,SILK at 24000h,G7221 at 32000h,G7221 at 16000h
>>>>> ,G722,PCMU,PCMA,GSM"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,SILK at 8000h
>>>>> ,SILK at 12000h,SILK at 16000h,SILK at 24000h,G7221 at 32000h,G7221 at 16000h
>>>>> ,G722,PCMU,PCMA,GSM"/>
>>>>>
>>>>>
>>>>>  freeswitch at internal> g729_info
>>>>> Permitted G729 channels: 10
>>>>> Encoders in use: 4
>>>>> Decoders in use: 3
>>>>>
>>>>> freeswitch at internal>
>>>>>
>>>>>
>>>>>
>>>>> recv 742 bytes from udp/[182.79.149.142]:16251 at 07:27:19.212195:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>>    INVITE sip:500 at 50.54.12.39 SIP/2.0
>>>>>
>>>>>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport;branch=z9hG4bK68891
>>>>>
>>>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>>>
>>>>>    Max-Forwards: 70
>>>>>
>>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>>>
>>>>>    To: <sip:500 at 50.54.12.39>
>>>>>
>>>>>    Call-ID: 047707180743 at 10.4.2.7
>>>>>
>>>>>    CSeq: 1 INVITE
>>>>>
>>>>>    Contact: <sip:501 at 182.79.149.142:16251;transport=udp>
>>>>>
>>>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>>>
>>>>>    Expires: 3600
>>>>>
>>>>>    User-Agent: SyncPhone 1.0
>>>>>
>>>>>    Content-Length: 244
>>>>>
>>>>>    Content-Type: application/sdp
>>>>>
>>>>>
>>>>>
>>>>>    v=0
>>>>>
>>>>>    o=- 1111 1111 IN IP4 182.79.149.142
>>>>>
>>>>>    s=Session SIP/SDP
>>>>>
>>>>>    c=IN IP4 182.79.149.142
>>>>>
>>>>>    t=0 0
>>>>>
>>>>>    a=sendrecv
>>>>>
>>>>>    a=rtcp:57395 IN IP4 182.79.149.142
>>>>>
>>>>>    m=audio 26474 RTP/AVP 0 101
>>>>>
>>>>>    a=rtpmap:0 G729/8000
>>>>>
>>>>>    a=rtpmap:101 telephone-event/8000
>>>>>
>>>>>    a=fmtp:101 0-15
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> send 321 bytes to udp/[182.79.149.142]:16251 at 07:27:19.212520:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>>    SIP/2.0 100 Trying
>>>>>
>>>>>    Via: SIP/2.0/UDP 182.79.149.142:16251
>>>>> ;rport=16251;branch=z9hG4bK68891
>>>>>
>>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>>>
>>>>>    To: <sip:500 at 50.54.12.39>
>>>>>
>>>>>    Call-ID: 047707180743 at 10.4.2.7
>>>>>
>>>>>    CSeq: 1 INVITE
>>>>>
>>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>>> 18-32-35 -0600
>>>>>
>>>>>    Content-Length: 0
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:7481 IP 182.79.149.142
>>>>> Approved by acl "domains[]". Access Granted.
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [NOTICE] switch_channel.c:930 New Channel
>>>>> sofia/internal/501 at 50.54.12.39 [30a75cd4-481b-11e2-84b4-e5c28303057e]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>>>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_NEW
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:380
>>>>> (sofia/internal/501 at 50.54.12.39) State NEW
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:5502 Channel sofia/internal/
>>>>> 501 at 50.54.12.39 entering state [received][100]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia.c:5513 Remote SDP:
>>>>>
>>>>> v=0
>>>>>
>>>>> o=- 1111 1111 IN IP4 182.79.149.142
>>>>>
>>>>> s=Session SIP/SDP
>>>>>
>>>>> c=IN IP4 182.79.149.142
>>>>>
>>>>> t=0 0
>>>>>
>>>>> a=sendrecv
>>>>>
>>>>> a=rtcp:57395 IN IP4 182.79.149.142
>>>>>
>>>>> m=audio 26474 RTP/AVP 0 101
>>>>>
>>>>> a=rtpmap:0 G729/8000
>>>>>
>>>>> a=rtpmap:101 telephone-event/8000
>>>>>
>>>>> a=fmtp:101 0-15
>>>>>
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>> Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
>>>>> send/recv payload to 101
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_channel.c:2852
>>>>> (sofia/internal/501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [NOTICE] sofia.c:5781 Hangup sofia/internal/
>>>>> 501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_channel.c:2875 Send signal
>>>>> sofia/internal/501 at 50.54.12.39 [KILL]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send
>>>>> signal sofia/internal/501 at 50.54.12.39 [BREAK]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>>>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_HANGUP
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:602
>>>>> (sofia/internal/501 at 50.54.12.39) State HANGUP
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:469 Channel
>>>>> sofia/internal/501 at 50.54.12.39 hanging up, cause:
>>>>> INCOMPATIBLE_DESTINATION
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:534 Responding to
>>>>> INVITE with: 488
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:47
>>>>> sofia/internal/501 at 50.54.12.39 Standard HANGUP, cause:
>>>>> INCOMPATIBLE_DESTINATION
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:602
>>>>> (sofia/internal/501 at 50.54.12.39) State HANGUP going to sleep
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:393
>>>>> (sofia/internal/501 at 50.54.12.39) State Change CS_HANGUP ->
>>>>> CS_REPORTING
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send
>>>>> signal sofia/internal/501 at 50.54.12.39 [BREAK]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:362
>>>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_REPORTING
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:662
>>>>> (sofia/internal/501 at 50.54.12.39) State REPORTING
>>>>>
>>>>> send 816 bytes to udp/[182.79.149.142]:16251 at 07:27:19.215295:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>>    SIP/2.0 488 Not Acceptable Here
>>>>>
>>>>>    Via: SIP/2.0/UDP 182.79.149.142:16251
>>>>> ;rport=16251;branch=z9hG4bK68891
>>>>>
>>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>>>
>>>>>    To: <sip:500 at 50.54.12.39>;tag=8K31Z9S2DymjN
>>>>>
>>>>>    Call-ID: 047707180743 at 10.4.2.7
>>>>>
>>>>>    CSeq: 1 INVITE
>>>>>
>>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>>> 18-32-35 -0600
>>>>>
>>>>>    Accept: application/sdp
>>>>>
>>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>
>>>>>    Supported: timer, precondition, path, replaces
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:79
>>>>> sofia/internal/501 at 50.54.12.39 Standard REPORTING, cause:
>>>>> INCOMPATIBLE_DESTINATION
>>>>>
>>>>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>>>> sla, include-session-description, presence.winfo, message-summary, refer
>>>>>
>>>>>    Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"2012-12-17
>>>>> 07:27:19.206419 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/
>>>>> 501 at 50.54.12.39) State REPORTING going to sleep
>>>>>
>>>>>
>>>>>    Content-Length: 0
>>>>>
>>>>>    Remote-Party-ID: "500" <sip:500 at 50.54.12.39
>>>>> >;party=calling;privacy=off;screen=no
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:387
>>>>> (sofia/internal/501 at 50.54.12.39) State Change CS_REPORTING ->
>>>>> CS_DESTROY
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1180 Send
>>>>> signal sofia/internal/501 at 50.54.12.39 [BREAK]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_session.c:1380 Session
>>>>> 204 (sofia/internal/501 at 50.54.12.39) Locked, Waiting on external
>>>>> entities
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [NOTICE] switch_core_session.c:1398 Session
>>>>> 204 (sofia/internal/501 at 50.54.12.39) Ended
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [NOTICE] switch_core_session.c:1400 Close
>>>>> Channel sofia/internal/501 at 50.54.12.39 [CS_DESTROY]
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:491
>>>>> (sofia/internal/501 at 50.54.12.39) Callstate Change HANGUP -> DOWN
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:494
>>>>> (sofia/internal/501 at 50.54.12.39) Running State Change CS_DESTROY
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:504
>>>>> (sofia/internal/501 at 50.54.12.39) State DESTROY
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] mod_sofia.c:374 sofia/internal/
>>>>> 501 at 50.54.12.39 SOFIA DESTROY
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:86
>>>>> sofia/internal/501 at 50.54.12.39 Standard DESTROY
>>>>>
>>>>> 2012-12-17 07:27:19.206419 [DEBUG] switch_core_state_machine.c:504
>>>>> (sofia/internal/501 at 50.54.12.39) State DESTROY going to sleep
>>>>>
>>>>> recv 407 bytes from udp/[182.79.149.142]:16251 at 07:27:19.581401:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>>    ACK sip:500 at 50.54.12.39 SIP/2.0
>>>>>
>>>>>    Via: SIP/2.0/UDP 182.79.149.142:16251;rport;branch=z9hG4bK68891
>>>>>
>>>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>>>
>>>>>    Max-Forwards: 70
>>>>>
>>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK03044060
>>>>>
>>>>>    To: <sip:500 at 50.54.12.39>;tag=8K31Z9S2DymjN
>>>>>
>>>>>    Call-ID: 047707180743 at 10.4.2.7
>>>>>
>>>>>    CSeq: 1 ACK
>>>>>
>>>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>>>
>>>>>    User-Agent: SyncPhone 1.0
>>>>>
>>>>>    Content-Length: 0
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> recv 465 bytes from udp/[182.79.149.142]:36420 at 07:27:20.672450:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>>    REGISTER sip:50.54.12.39 SIP/2.0
>>>>>
>>>>>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport;branch=z9hG4bK20451
>>>>>
>>>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>>>
>>>>>    Max-Forwards: 70
>>>>>
>>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>>>
>>>>>    To: <sip:501 at 50.54.12.39>
>>>>>
>>>>>    Call-ID: 790899967509 at 10.4.2.7
>>>>>
>>>>>    CSeq: 1 REGISTER
>>>>>
>>>>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>
>>>>>
>>>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>>>
>>>>>    Expires: 3600
>>>>>
>>>>>    User-Agent: SyncPhone 1.0
>>>>>
>>>>>    Content-Length: 0
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> 2012-12-17 07:27:20.666425 [DEBUG] sofia_reg.c:1417 Send challenge for
>>>>> [501 at 50.54.12.39]
>>>>>
>>>>> 2012-12-17 07:27:20.666425 [WARNING] sofia_reg.c:1421 SIP auth
>>>>> challenge (REGISTER) on sofia profile 'internal' for [501 at 50.54.12.39]
>>>>> from ip 182.79.149.142
>>>>>
>>>>> send 626 bytes to udp/[182.79.149.142]:36420 at 07:27:20.673200:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>>    SIP/2.0 401 Unauthorized
>>>>>
>>>>>    Via: SIP/2.0/UDP 182.79.149.142:36420
>>>>> ;rport=36420;branch=z9hG4bK20451
>>>>>
>>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>>>
>>>>>    To: <sip:501 at 50.54.12.39>;tag=9vvt14a6a7a5g
>>>>>
>>>>>    Call-ID: 790899967509 at 10.4.2.7
>>>>>
>>>>>    CSeq: 1 REGISTER
>>>>>
>>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>>> 18-32-35 -0600
>>>>>
>>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>
>>>>>    Supported: timer, precondition, path, replaces
>>>>>
>>>>>    WWW-Authenticate: Digest realm="50.54.12.39",
>>>>> nonce="31862298-481b-11e2-84b6-e5c28303057e", algorithm=MD5, qop="auth"
>>>>>
>>>>>    Content-Length: 0
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> recv 707 bytes from udp/[182.79.149.142]:36420 at 07:27:21.094563:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>>    REGISTER sip:50.54.12.39 SIP/2.0
>>>>>
>>>>>    Via: SIP/2.0/UDP 182.79.149.142:36420;rport;branch=z9hG4bK84890
>>>>>
>>>>>    Route: <sip:50.54.12.39;transport=udp;lr>
>>>>>
>>>>>    Max-Forwards: 70
>>>>>
>>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>>>
>>>>>    To: <sip:501 at 50.54.12.39>
>>>>>
>>>>>    Call-ID: 790899967509 at 10.4.2.7
>>>>>
>>>>>    CSeq: 2 REGISTER
>>>>>
>>>>>    Allow:  INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS
>>>>>
>>>>>    Expires: 3600
>>>>>
>>>>>    User-Agent: SyncPhone 1.0
>>>>>
>>>>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>
>>>>>
>>>>>    Authorization: Digest username="501", realm="50.54.12.39",
>>>>> nonce="31862298-481b-11e2-84b6-e5c28303057e", uri="sip:50.54.12.39",
>>>>> algorithm=MD5, qop=auth, nc=00000001, cnonce="a5d0aab2288476b3",
>>>>> response="eb8ccc768c7f9fee038f0d5dfc599931"
>>>>>
>>>>>    Content-Length: 0
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header
>>>>> -> 'record_stereo' = 'true'
>>>>>
>>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header
>>>>> -> 'default_gateway' = 'example.com'
>>>>>
>>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header
>>>>> -> 'default_areacode' = '918'
>>>>>
>>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header
>>>>> -> 'transfer_fallback_extension' = 'operator'
>>>>>
>>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header
>>>>> -> 'toll_allow' = 'domestic,international,local'
>>>>>
>>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header
>>>>> -> 'accountcode' = '501'
>>>>>
>>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header
>>>>> -> 'user_context' = 'default'
>>>>>
>>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header
>>>>> -> 'effective_caller_id_name' = 'Extension 501'
>>>>>
>>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header
>>>>> -> 'effective_caller_id_number' = '501'
>>>>>
>>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header
>>>>> -> 'outbound_caller_id_name' = 'FS'
>>>>>
>>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header
>>>>> -> 'outbound_caller_id_number' = '9732386040'
>>>>>
>>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:2639 event_add_header
>>>>> -> 'callgroup' = 'techsupport'
>>>>>
>>>>> 2012-12-17 07:27:21.086425 [DEBUG] sofia_reg.c:1575 Register:
>>>>>
>>>>> From:    [501 at 10.252.148.21]
>>>>>
>>>>> Contact: ["user" <sip:501 at 182.79.149.142:36420;transport=udp>]
>>>>>
>>>>> Expires: [3600]
>>>>>
>>>>> send 600 bytes to udp/[182.79.149.142]:36420 at 07:27:21.096965:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>>    SIP/2.0 200 OK
>>>>>
>>>>>    Via: SIP/2.0/UDP 182.79.149.142:36420
>>>>> ;rport=36420;branch=z9hG4bK84890
>>>>>
>>>>>    From: <sip:501 at 50.54.12.39>;tag=z9hG4bK35760121
>>>>>
>>>>>    To: <sip:501 at 50.54.12.39>;tag=a6NK3ZU97F1Qc
>>>>>
>>>>>    Call-ID: 790899967509 at 10.4.2.7
>>>>>
>>>>>    CSeq: 2 REGISTER
>>>>>
>>>>>    Contact: <sip:501 at 182.79.149.142:36420;transport=udp>;expires=3600
>>>>>
>>>>>    Date: Mon, 17 Dec 2012 07:27:21 GMT
>>>>>
>>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>>> 18-32-35 -0600
>>>>>
>>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>
>>>>>    Supported: timer, precondition, path, replaces
>>>>>
>>>>>    Content-Length: 0
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> send 926 bytes to udp/[182.79.149.142]:36420 at 07:27:21.112029:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>>    NOTIFY sip:501 at 182.79.149.142:36420;transport=udp SIP/2.0
>>>>>
>>>>>    Via: SIP/2.0/UDP 50.54.12.39;rport;branch=z9hG4bKXXQBeBHUQ5yZp
>>>>>
>>>>>    Max-Forwards: 70
>>>>>
>>>>>    From: <sip:501 at 50.54.12.39>;tag=BFFc5tcD5rQar
>>>>>
>>>>>    To: <sip:501 at 50.54.12.39>
>>>>>
>>>>>    Call-ID: 09276738-c2be-1230-ec86-22000afc9415
>>>>>
>>>>>    CSeq: 37528556 NOTIFY
>>>>>
>>>>>    Contact: <sip:mod_sofia at 50.54.12.39:5060>
>>>>>
>>>>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9f8d37d 2012-01-28
>>>>> 18-32-35 -0600
>>>>>
>>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>
>>>>>    Supported: timer, precondition, path, replaces
>>>>>
>>>>>    Event: message-summary
>>>>>
>>>>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>>>> sla, include-session-description, presence.winfo, message-summary, refer
>>>>>
>>>>>    Subscription-State: terminated;reason=noresource
>>>>>
>>>>>    Content-Type: application/simple-message-summary
>>>>>
>>>>>    Content-Length: 64
>>>>>
>>>>>
>>>>>
>>>>>    Messages-Waiting: no
>>>>>
>>>>>    Message-Account: sip:501 at 10.252.148.21
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> recv 295 bytes from udp/[182.79.149.142]:36420 at 07:27:21.531463:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>>    SIP/2.0 405 Method Not Allowed
>>>>>
>>>>>    Via: SIP/2.0/UDP 50.54.12.39;branch=z9hG4bKXXQBeBHUQ5yZp;rport=5060
>>>>>
>>>>>    To: <sip:501 at 50.54.12.39>
>>>>>
>>>>>    From: <sip:501 at 50.54.12.39>;tag=BFFc5tcD5rQar
>>>>>
>>>>>    Call-ID: 09276738-c2be-1230-ec86-22000afc9415
>>>>>
>>>>>    CSeq: 37528556 NOTIFY
>>>>>
>>>>>    Server: SyncPhone 1.0
>>>>>
>>>>>    Content-Length: 0
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Mon, Dec 17, 2012 at 2:25 AM, curriegrad2004 <
>>>>> curriegrad2004 at gmail.com> wrote:
>>>>>
>>>>>> If he is using mod_g729 then the log would throw an error saying that
>>>>>> the codec only does passthrough mode, however he hasn't shown that
>>>>>> part of the log to us yet
>>>>>>
>>>>>> On Sun, Dec 16, 2012 at 12:05 PM, Steven Ayre <steveayre at gmail.com>
>>>>>> wrote:
>>>>>> > What G729 module are you using, and what codec is in use on the
>>>>>> other leg?
>>>>>> >
>>>>>> > G729 needs licenses due to patents.
>>>>>> >
>>>>>> > mod_g729 is passthrough only but requires no licenses. This is
>>>>>> because it
>>>>>> > merely forwards the data, but doesn't perform the
>>>>>> encoding/transcoding step
>>>>>> > that the patents cover. But that means it can't transcode between
>>>>>> different
>>>>>> > codecs.
>>>>>> > mod_com_g729 is the licensed version:
>>>>>> http://www.freeswitch.org/node/235
>>>>>> >
>>>>>> > With the passthrough mod_g729 codec, if the other leg has selected
>>>>>> a codec
>>>>>> > other than g729 you'll see an error in your logs and it'll hangup
>>>>>> with that
>>>>>> > reason.
>>>>>> >
>>>>>> > You can tweak your codec negotiation to avoid this (eg
>>>>>> > late-negotiation=true), or you can use the licensed version.
>>>>>> >
>>>>>> >
>>>>>> >
>>>>>> >
>>>>>> > On 16 December 2012 11:40, Balamurugan Mahendran <b2m at a-cti.com>
>>>>>> wrote:
>>>>>> >>
>>>>>> >> Need help on Codec Negotiation
>>>>>> >>
>>>>>> >>
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [G729:0:8000:20:64000]/[G729:18:8000:20:8000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [G729:0:8000:20:64000]/[SILK:117:8000:20:20000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [G729:0:8000:20:64000]/[SILK:118:12000:20:25000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [G729:0:8000:20:64000]/[SILK:119:16000:20:30000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [G729:0:8000:20:64000]/[SILK:120:24000:20:40000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [G729:0:8000:20:64000]/[G7221:115:32000:20:48000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [G729:0:8000:20:64000]/[G7221:107:16000:20:32000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [G729:0:8000:20:64000]/[G722:9:8000:20:64000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [G729:0:8000:20:64000]/[GSM:3:8000:20:13200]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec
>>>>>> Compare
>>>>>> >> [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf
>>>>>> >> send/recv payload to 101
>>>>>> >>
>>>>>> >>
>>>>>> >> 2012-12-16 11:35:52.246422 [DEBUG] switch_channel.c:2852
>>>>>> >> (sofia/internal/501 at 50.54.12.39) Callstate Change DOWN -> HANGUP
>>>>>> >> 2012-12-16 11:35:52.246422 [NOTICE] sofia.c:5781 Hangup
>>>>>> >> sofia/internal/501 at 50.54.12.39 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>>>>> >>
>>>>>> >> 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
>>>>>> >>
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >> http://www.freeswitch.org
>>>>>> >>
>>>>>> >
>>>>>> >
>>>>>> >
>>>>>> _________________________________________________________________________
>>>>>> > Professional FreeSWITCH Consulting 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
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From madan.mallikarjun at outlook.com  Tue Dec 18 13:17:27 2012
From: madan.mallikarjun at outlook.com (madan.mallikarjun)
Date: Tue, 18 Dec 2012 02:17:27 -0800 (PST)
Subject: [Freeswitch-users] FreeSwitch
In-Reply-To: <1355747829414-7585608.post@n2.nabble.com>
References: <1355747829414-7585608.post@n2.nabble.com>
Message-ID: <1355825847949-7585662.post@n2.nabble.com>

hello all,
 
           thanks for the updates still there is a lot of confusion when it
comes to the path's that has to be followed when the set up has to be
installed . kindly guide me where i m wrong and let me know.

1. cd /usr/local/src
2. mkdir freeswitch : where i wish do the set up installation
3. cd /usr/local/src/freeswitch
4. wget the stable version path 
5. now that i have the file in freeswitch folder freeswitch-1.2.5.3.tar.bz2
i try to tar it i get the following error for the command that i write 
6.  tar -xvfz freeswitch-1.2.5.3.tar.bz2

thanks in advance 
Madan Mallikarjun



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-tp7585608p7585662.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From babak.freeswitch at gmail.com  Tue Dec 18 13:56:30 2012
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Tue, 18 Dec 2012 14:26:30 +0330
Subject: [Freeswitch-users] attended transfer to rxfax
Message-ID: <CAEVfcaZukd+gs3ReiS=ro-K3rTGKgEoHmNn_QWqwgN+Uks2iOw@mail.gmail.com>

Hi
How can I implement this:
user calls to a number which starts sending fax after user presses 1
(channel A)
after user hears fax signaling starts to attended transfer to a local
extension which executes rxfax (Channel B)
after user hears local extension rxfax signaling presses transfer to
complete the transfer
but no fax is received!!
I trace logs and see that channel B ends after user presses transfer. it
seems I need to create a C channel which is running txfax but I donno how
to do it
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From ssoni at lifesize.com  Tue Dec 18 14:21:19 2012
From: ssoni at lifesize.com (Sanjay Soni)
Date: Tue, 18 Dec 2012 05:21:19 -0600
Subject: [Freeswitch-users] FreeSwitch
In-Reply-To: <1355825847949-7585662.post@n2.nabble.com>
References: <1355747829414-7585608.post@n2.nabble.com>
	<1355825847949-7585662.post@n2.nabble.com>
Message-ID: <F3C02C5E8650B74FA0C199FA1A01D26E169E2FE2EF@ausmsex00.austin.kmvtechnologies.com>

This should help you http://how-to.wikia.com/wiki/How_to_untar_a_tar_file_or_gzip-bz2_tar_file ?? ! .gz is different from .bz2 !


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of madan.mallikarjun
Sent: 18 December 2012 15:47
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSwitch

hello all,
 
           thanks for the updates still there is a lot of confusion when it
comes to the path's that has to be followed when the set up has to be
installed . kindly guide me where i m wrong and let me know.

1. cd /usr/local/src
2. mkdir freeswitch : where i wish do the set up installation
3. cd /usr/local/src/freeswitch
4. wget the stable version path 
5. now that i have the file in freeswitch folder freeswitch-1.2.5.3.tar.bz2
i try to tar it i get the following error for the command that i write 
6.  tar -xvfz freeswitch-1.2.5.3.tar.bz2

thanks in advance 
Madan Mallikarjun



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-tp7585608p7585662.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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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 Dec 18 14:24:53 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 18 Dec 2012 11:24:53 +0000
Subject: [Freeswitch-users] FreeSwitch
In-Reply-To: <1355825847949-7585662.post@n2.nabble.com>
References: <1355747829414-7585608.post@n2.nabble.com>
	<1355825847949-7585662.post@n2.nabble.com>
Message-ID: <CAFiqYu=Lp0ivYOMTdnDe++zi3vyBy7ai82F4EYGMuGUSyixZtQ@mail.gmail.com>

>
> 1. cd /usr/local/src
> 2. mkdir freeswitch : where i wish do the set up installation


This is where you will compile FreeSWITCH, but not where it'll be
installed. It will install to the standardised FHS paths (/usr/local/bin
etc),

i try to tar it i get the following error for the command that i write

6.  tar -xvfz freeswitch-1.2.5.3.tar.bz2


The correct command for a .tar.bz2 is:
tar -jxvf freeswitch-1.2.5.3.tar.bz2

-j = .bz2, -z is for .gz. They're different compression algorithm. bzip2
generally gives better compression, but takes longer to compress.
-x = extract
-v = verbose (optional), will tell you each file extracted
-f filename = what file to extract
-f *must* appear just before the filename, in the one you were trying you
had xvfz, the fz means it would've been looking for a tarball called 'z'


-Steve




On 18 December 2012 10:17, madan.mallikarjun
<madan.mallikarjun at outlook.com>wrote:

> hello all,
>
>            thanks for the updates still there is a lot of confusion when it
> comes to the path's that has to be followed when the set up has to be
> installed . kindly guide me where i m wrong and let me know.
>
> 1. cd /usr/local/src
> 2. mkdir freeswitch : where i wish do the set up installation
> 3. cd /usr/local/src/freeswitch
> 4. wget the stable version path
> 5. now that i have the file in freeswitch folder freeswitch-1.2.5.3.tar.bz2
> i try to tar it i get the following error for the command that i write
> 6.  tar -xvfz freeswitch-1.2.5.3.tar.bz2
>
> thanks in advance
> Madan Mallikarjun
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-tp7585608p7585662.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 18 14:38:56 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 18 Dec 2012 12:38:56 +0100
Subject: [Freeswitch-users] FreeSwitch
In-Reply-To: <CAFiqYu=Lp0ivYOMTdnDe++zi3vyBy7ai82F4EYGMuGUSyixZtQ@mail.gmail.com>
References: <1355747829414-7585608.post@n2.nabble.com>
	<1355825847949-7585662.post@n2.nabble.com>
	<CAFiqYu=Lp0ivYOMTdnDe++zi3vyBy7ai82F4EYGMuGUSyixZtQ@mail.gmail.com>
Message-ID: <CALXCt0oRyKB7H+vPbOWxSq9g-Dt2h1Mj_m87gQAN-aj7Q8Z-hA@mail.gmail.com>

I don't want to be the bad boy telling unpleasant things, but if you
are not able to use tar, or to follow the wiki, you better first get
yourself acquainted and familiar with Linux, and only then try to
install complex softwatre like whatever VOIP. If you do not have
adequate background in Linux, you'll end up with lot of frustration,
trash your time, and nothing will work. Maybe you wanna ask a friend
that's familiar with Linux administration to be on your side for a
while.

-giovanni

On Tue, Dec 18, 2012 at 12:24 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> 1. cd /usr/local/src
>> 2. mkdir freeswitch : where i wish do the set up installation
>
>
> This is where you will compile FreeSWITCH, but not where it'll be installed.
> It will install to the standardised FHS paths (/usr/local/bin etc),
>
>> i try to tar it i get the following error for the command that i write
>>
>> 6.  tar -xvfz freeswitch-1.2.5.3.tar.bz2
>
>
> The correct command for a .tar.bz2 is:
> tar -jxvf freeswitch-1.2.5.3.tar.bz2
>
> -j = .bz2, -z is for .gz. They're different compression algorithm. bzip2
> generally gives better compression, but takes longer to compress.
> -x = extract
> -v = verbose (optional), will tell you each file extracted
> -f filename = what file to extract
> -f must appear just before the filename, in the one you were trying you had
> xvfz, the fz means it would've been looking for a tarball called 'z'
>
>
> -Steve
>
>
>
>
> On 18 December 2012 10:17, madan.mallikarjun <madan.mallikarjun at outlook.com>
> wrote:
>>
>> hello all,
>>
>>            thanks for the updates still there is a lot of confusion when
>> it
>> comes to the path's that has to be followed when the set up has to be
>> installed . kindly guide me where i m wrong and let me know.
>>
>> 1. cd /usr/local/src
>> 2. mkdir freeswitch : where i wish do the set up installation
>> 3. cd /usr/local/src/freeswitch
>> 4. wget the stable version path
>> 5. now that i have the file in freeswitch folder
>> freeswitch-1.2.5.3.tar.bz2
>> i try to tar it i get the following error for the command that i write
>> 6.  tar -xvfz freeswitch-1.2.5.3.tar.bz2
>>
>> thanks in advance
>> Madan Mallikarjun
>>
>>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-tp7585608p7585662.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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at gmail.com  Tue Dec 18 14:42:46 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 18 Dec 2012 12:42:46 +0100
Subject: [Freeswitch-users] Mutiple GSMopen USB modem setup
In-Reply-To: <CAHKqMb69d22pUBqDEw16MwEh_EDR-wMG=j78zYKGFq1ubGZ0DA@mail.gmail.com>
References: <CAHKqMb6OBRVtOkEepF-a5fVXz_x-NbvhDijvU+dgCBsGcQ5sKg@mail.gmail.com>
	<CALXCt0pT2UoMWZ0dUPC2zKTbm_7zd2Z_2UAr1e8i78JO24uXTw@mail.gmail.com>
	<CAHKqMb69d22pUBqDEw16MwEh_EDR-wMG=j78zYKGFq1ubGZ0DA@mail.gmail.com>
Message-ID: <CALXCt0pradfFFKxvderhXAmrcMaaSP-uajJxLUoopg1_u787bg@mail.gmail.com>

Hi Nandy,

how it was? It works with that udev thing? Can you describe for me, so
I can put it in the wiki?

-giovanni

On Sun, Dec 16, 2012 at 3:08 AM, Nandy Dagondon <nandy1925 at gmail.com> wrote:
> Thanks for the feedback Giovanni.  Re fixing the ttyUSB port assignment, I
> have searched a guy made a script to scan USB serial numbers using udev,
> then created symbolic links. Will try it out later.
> /Nandy
>
>
> On Thu, Dec 13, 2012 at 7:02 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>>
>> On Thu, Dec 13, 2012 at 4:44 AM, Nandy Dagondon <nandy1925 at gmail.com>
>> wrote:
>> > I have run GSMOpen on CentOS 5.7 defying the Wiki advisory. Questions:
>>
>> I would invite you to go for the OS distros detailed in the wiki page.
>> You'll probably encounter problems with different distros, and you're
>> on your own to solve it.
>>
>> > 1. What is the maximum number of USB modems tested? Can we get the
>> > numbers
>> > and the CPU used?
>>
>> I've heard about 48 concurrent, and 64. Me personally have tested with
>> 5. No CPU consumption. The critical part is the USB BUS. So use
>> cascading and POWERED good usb 2.0 hubs
>>
>> > 2. I'll be installing multiple modems each connected to a different
>> > mobile
>> > network. Is the /dev/ttyUSB assignments  constant for every modem?
>> > Meaning
>> > it doesn't change if I plug it on different USB jacks.
>>
>> it will change not only if you change USB port, but also randomly if
>> you stay on the same USB port and reboot (and sometimes also without
>> rebooting). That's a "feature" of Linux distros (a demented one,
>> cannot understand why they choose this behavior).
>>
>> Soon or later I'll look into this, and come out with a solution (I've
>> made some preliminary research and reasoning about in the past).
>>
>> If you have a commercial interest in that, and a real budget for it,
>> contact me in private as consultant, or put a public bounty on it.
>>
>> -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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From Alexander.Haugg at c4b.de  Tue Dec 18 15:05:59 2012
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Tue, 18 Dec 2012 12:05:59 +0000
Subject: [Freeswitch-users] build error with new Freeswitch 'nametab.h': No
 such file or directory
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC837D409@vc4bex2.corp.c4b.de>

Hi,

i hope the question is not reapplied

today i checkout the newest version of freeswqitch and get following error:

"E:\TFS_neu\C4B UC\Main\3rd Party\FreeSWITCH\tmp_3\freeswitch\Freeswitch.2008.sln" (rebuild target) (1) ->
(_Libraries\xmlrpc-c\xmltok:Rebuild target) ->
  ..\..\xmlrpc-c\lib\expat\xmltok\xmltok.c(10): fatal error C1083: Cannot open include file: 'nametab.h': No such file or directory

Exist a workaround or bugfix for this problem?

Thanks for your help
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From madan.mallikarjun at outlook.com  Tue Dec 18 15:11:39 2012
From: madan.mallikarjun at outlook.com (madan.mallikarjun)
Date: Tue, 18 Dec 2012 04:11:39 -0800 (PST)
Subject: [Freeswitch-users] FreeSwitch
In-Reply-To: <CALXCt0oRyKB7H+vPbOWxSq9g-Dt2h1Mj_m87gQAN-aj7Q8Z-hA@mail.gmail.com>
References: <1355747829414-7585608.post@n2.nabble.com>
	<1355825847949-7585662.post@n2.nabble.com>
	<CAFiqYu=Lp0ivYOMTdnDe++zi3vyBy7ai82F4EYGMuGUSyixZtQ@mail.gmail.com>
	<CALXCt0oRyKB7H+vPbOWxSq9g-Dt2h1Mj_m87gQAN-aj7Q8Z-hA@mail.gmail.com>
Message-ID: <BAY002-W15576BFBCB8F01C2C21A0F8F9310@phx.gbl>

thanks and will make sure that i m well acquainted before posting  

Date: Tue, 18 Dec 2012 03:45:17 -0800
From: ml-node+s2379917n7585666h59 at n2.nabble.com
To: madan.mallikarjun at outlook.com
Subject: Re: FreeSwitch



	I don't want to be the bad boy telling unpleasant things, but if you

are not able to use tar, or to follow the wiki, you better first get

yourself acquainted and familiar with Linux, and only then try to

install complex softwatre like whatever VOIP. If you do not have

adequate background in Linux, you'll end up with lot of frustration,

trash your time, and nothing will work. Maybe you wanna ask a friend

that's familiar with Linux administration to be on your side for a

while.


-giovanni


On Tue, Dec 18, 2012 at 12:24 PM, Steven Ayre <[hidden email]> wrote:

>> 1. cd /usr/local/src

>> 2. mkdir freeswitch : where i wish do the set up installation

>

>

> This is where you will compile FreeSWITCH, but not where it'll be installed.

> It will install to the standardised FHS paths (/usr/local/bin etc),

>

>> i try to tar it i get the following error for the command that i write

>>

>> 6.  tar -xvfz freeswitch-1.2.5.3.tar.bz2

>

>

> The correct command for a .tar.bz2 is:

> tar -jxvf freeswitch-1.2.5.3.tar.bz2

>

> -j = .bz2, -z is for .gz. They're different compression algorithm. bzip2

> generally gives better compression, but takes longer to compress.

> -x = extract

> -v = verbose (optional), will tell you each file extracted

> -f filename = what file to extract

> -f must appear just before the filename, in the one you were trying you had

> xvfz, the fz means it would've been looking for a tarball called 'z'

>

>

> -Steve

>

>

>

>

> On 18 December 2012 10:17, madan.mallikarjun <[hidden email]>

> wrote:

>>

>> hello all,

>>

>>            thanks for the updates still there is a lot of confusion when

>> it

>> comes to the path's that has to be followed when the set up has to be

>> installed . kindly guide me where i m wrong and let me know.

>>

>> 1. cd /usr/local/src

>> 2. mkdir freeswitch : where i wish do the set up installation

>> 3. cd /usr/local/src/freeswitch

>> 4. wget the stable version path

>> 5. now that i have the file in freeswitch folder

>> freeswitch-1.2.5.3.tar.bz2

>> i try to tar it i get the following error for the command that i write

>> 6.  tar -xvfz freeswitch-1.2.5.3.tar.bz2

>>

>> thanks in advance

>> Madan Mallikarjun

>>

>>

>>

>> --

>> View this message in context:

>> http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-tp7585608p7585662.html
>> 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
>>

>> 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
>

> 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
>



-- 

Sincerely,


Giovanni Maruzzelli

Cell : +39-347-2665618


_________________________________________________________________________

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

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/FreeSwitch-tp7585608p7585666.html
	
	
		
		To unsubscribe from FreeSwitch, click here.

		NAML
	 		 	   		  



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-tp7585608p7585668.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From dvl36.ripe.nick at gmail.com  Tue Dec 18 16:03:34 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Tue, 18 Dec 2012 15:03:34 +0200
Subject: [Freeswitch-users] Mutiple GSMopen USB modem setup
In-Reply-To: <CAHKqMb69d22pUBqDEw16MwEh_EDR-wMG=j78zYKGFq1ubGZ0DA@mail.gmail.com>
References: <CAHKqMb6OBRVtOkEepF-a5fVXz_x-NbvhDijvU+dgCBsGcQ5sKg@mail.gmail.com>
	<CALXCt0pT2UoMWZ0dUPC2zKTbm_7zd2Z_2UAr1e8i78JO24uXTw@mail.gmail.com>
	<CAHKqMb69d22pUBqDEw16MwEh_EDR-wMG=j78zYKGFq1ubGZ0DA@mail.gmail.com>
Message-ID: <CAMjaK3D0Xw4cnXeO1gQr-9+MD1JCuq1m6Xk3qKS0YFQDm6=z5g@mail.gmail.com>

Hi!
Huawei USB modems does not have accessible via USB protocol serial numbers.
At least E1550, E153 and E171.
Best wishes,
 Dmitry.

2012/12/16 Nandy Dagondon <nandy1925 at gmail.com>

> Thanks for the feedback Giovanni.  Re fixing the ttyUSB port assignment, I
> have searched a guy made a script to scan USB serial numbers using udev,
> then created symbolic links. Will try it out later.
> /Nandy
>
>
>
> On Thu, Dec 13, 2012 at 7:02 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>
>> On Thu, Dec 13, 2012 at 4:44 AM, Nandy Dagondon <nandy1925 at gmail.com>
>> wrote:
>> > I have run GSMOpen on CentOS 5.7 defying the Wiki advisory. Questions:
>>
>> I would invite you to go for the OS distros detailed in the wiki page.
>> You'll probably encounter problems with different distros, and you're
>> on your own to solve it.
>>
>> > 1. What is the maximum number of USB modems tested? Can we get the
>> numbers
>> > and the CPU used?
>>
>> I've heard about 48 concurrent, and 64. Me personally have tested with
>> 5. No CPU consumption. The critical part is the USB BUS. So use
>> cascading and POWERED good usb 2.0 hubs
>>
>> > 2. I'll be installing multiple modems each connected to a different
>> mobile
>> > network. Is the /dev/ttyUSB assignments  constant for every modem?
>> Meaning
>> > it doesn't change if I plug it on different USB jacks.
>>
>> it will change not only if you change USB port, but also randomly if
>> you stay on the same USB port and reboot (and sometimes also without
>> rebooting). That's a "feature" of Linux distros (a demented one,
>> cannot understand why they choose this behavior).
>>
>> Soon or later I'll look into this, and come out with a solution (I've
>> made some preliminary research and reasoning about in the past).
>>
>> If you have a commercial interest in that, and a real budget for it,
>> contact me in private as consultant, or put a public bounty on it.
>>
>> -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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 peter.olsson at visionutveckling.se  Tue Dec 18 16:16:20 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 18 Dec 2012 13:16:20 +0000
Subject: [Freeswitch-users] build error with new Freeswitch 'nametab.h':
 No such file or directory
Message-ID: <1FFF97C269757C458224B7C895F35F151ED6B1@cantor.std.visionutv.se>

I suspect you're using an old version?

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Alexander Haugg
Skickat: den 18 december 2012 13:06
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] build error with new Freeswitch 'nametab.h': No such file or directory

Hi,

i hope the question is not reapplied

today i checkout the newest version of freeswqitch and get following error:

"E:\TFS_neu\C4B UC\Main\3rd Party\FreeSWITCH\tmp_3\freeswitch\Freeswitch.2008.sln" (rebuild target) (1) ->
(_Libraries\xmlrpc-c\xmltok:Rebuild target) ->
  ..\..\xmlrpc-c\lib\expat\xmltok\xmltok.c(10): fatal error C1083: Cannot open include file: 'nametab.h': No such file or directory

Exist a workaround or bugfix for this problem?

Thanks for your help
!DSPAM:50d059ae32761979713419!
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From frederick at targointernet.com  Tue Dec 18 16:33:31 2012
From: frederick at targointernet.com (Frederick Pruneau)
Date: Tue, 18 Dec 2012 08:33:31 -0500
Subject: [Freeswitch-users] mod_fifo ringall
In-Reply-To: <CAKzWOxVqXE2ZcbFbR2DnsT8wBT9FLb8LuOFt6gDQyJoORXqxVw@mail.gmail.com>
References: <50CF6167.1060702@targointernet.com>
	<CAKzWOxVqXE2ZcbFbR2DnsT8wBT9FLb8LuOFt6gDQyJoORXqxVw@mail.gmail.com>
Message-ID: <50D070AB.7050805@targointernet.com>

I tested the two strategies and it still ring one by one. Is there any 
other option that I need to add or modify? Here is my fifo config:

     <fifo name="support_fifo_1@$${domain}" importance="0" 
outbound_strategy="ringall">
     <member timeout="15" simo="1" 
lag="5">{call_timeout=30,fifo_member_wait=nowait}user/223@$${domain}</member>
     <member timeout="15" simo="1" 
lag="5">{call_timeout=30,fifo_member_wait=nowait}user/224@$${domain}</member> 

     </fifo>

On 2012-12-17 17:54, Michael Collins wrote:
> I don't think this ever got properly documented, but maybe you could 
> test. In conf/autoload_configs/fifo.conf.xml trying creating the fifo 
> with an attribute outbound_strategy="ringall". The other main strategy 
> is "enterprise". Would you mind doing some testing and letting us know 
> what shakes out? I'll help you get it documented on the wiki.
>
> Thanks,
> MC
>
> On Mon, Dec 17, 2012 at 10:16 AM, Frederick Pruneau 
> <frederick at targointernet.com <mailto:frederick at targointernet.com>> wrote:
>
>     Hello all!
>
>     I have set up a queue with mod_fifo. I am trying to ring agents'
>     phones
>     all at once. I have looked at the wiki page
>     http://wiki.freeswitch.org/wiki/Mod_fifo. Everything is working except
>     that the phones are ringing one by one. I don't find the
>     information to
>     setup ringall strategy. I have done a Google search. It seems that
>     it is
>     possible but I don't find any information how to set it up.
>
>     Can you help me with this?
>
>     Fred
>
>     _________________________________________________________________________
>     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
>
>
>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 jeff at jefflenk.com  Tue Dec 18 18:31:43 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 18 Dec 2012 07:31:43 -0800 (PST)
Subject: [Freeswitch-users] build error with new Freeswitch 'nametab.h':
 No such file or directory
In-Reply-To: <B60DA91F5EA0E24A96E73CDE43CC9EC837D409@vc4bex2.corp.c4b.de>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC837D409@vc4bex2.corp.c4b.de>
Message-ID: <1355844703594-7585673.post@n2.nabble.com>

Was an older build present before updating. You may need to do a git clean
-fdx.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/build-error-with-new-Freeswitch-nametab-h-No-such-file-or-directory-tp7585669p7585673.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rnbrady at gmail.com  Tue Dec 18 20:00:09 2012
From: rnbrady at gmail.com (Richard Brady)
Date: Tue, 18 Dec 2012 17:00:09 +0000
Subject: [Freeswitch-users] SIP re-INVITEs and DTMF renegotiation
Message-ID: <CAN5esQLE8OOjouM8zFRK7csDp6p7-B6OpeNxPsjjQNkVYOew0g@mail.gmail.com>

Hi folks

I have the joy of talking to a Cisco device which behaves in weird and
wonderful ways.

When transferring a caller (in this case FreeSWITCH) to MOH it negotiates
RFC2833 out of the SDP.

When reconnecting the caller to a more meaningful endpoint, it renegotiates
the media again, but using delayed offer/answer, which means FreeSWITCH
must perform the offer.

Since FreeSWITCH has seen RFC2833 removed previously:

    [DEBUG] sofia_glue.c:5277 No 2833 in SDP. Disable 2833 dtmf and switch
to INFO.

It does not offer RFC2833 in the new SDP (if it did, the Cisco would now
accept it and answer with it).

Any suggestion for how I can force it to re-offer RFC2833?

Regards,
Richard
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From rnbrady at gmail.com  Tue Dec 18 20:43:07 2012
From: rnbrady at gmail.com (Richard Brady)
Date: Tue, 18 Dec 2012 17:43:07 +0000
Subject: [Freeswitch-users] SIP re-INVITEs and DTMF renegotiation
In-Reply-To: <CAN5esQLE8OOjouM8zFRK7csDp6p7-B6OpeNxPsjjQNkVYOew0g@mail.gmail.com>
References: <CAN5esQLE8OOjouM8zFRK7csDp6p7-B6OpeNxPsjjQNkVYOew0g@mail.gmail.com>
Message-ID: <CAN5esQLNBubvtbD=XVMPr3jOdYHCjxPWq791dpdt7X-3p=5=BQ@mail.gmail.com>

Looks like <param name="liberal-dtmf" value="true"/> in the profile does
the trick!


On 18 December 2012 17:00, Richard Brady <rnbrady at gmail.com> wrote:

> Hi folks
>
> I have the joy of talking to a Cisco device which behaves in weird and
> wonderful ways.
>
> When transferring a caller (in this case FreeSWITCH) to MOH it negotiates
> RFC2833 out of the SDP.
>
> When reconnecting the caller to a more meaningful endpoint, it
> renegotiates the media again, but using delayed offer/answer, which means
> FreeSWITCH must perform the offer.
>
> Since FreeSWITCH has seen RFC2833 removed previously:
>
>     [DEBUG] sofia_glue.c:5277 No 2833 in SDP. Disable 2833 dtmf and switch
> to INFO.
>
> It does not offer RFC2833 in the new SDP (if it did, the Cisco would now
> accept it and answer with it).
>
> Any suggestion for how I can force it to re-offer RFC2833?
>
> Regards,
> Richard
>
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From steveayre at gmail.com  Tue Dec 18 21:16:44 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 18 Dec 2012 18:16:44 +0000
Subject: [Freeswitch-users] SIP re-INVITEs and DTMF renegotiation
In-Reply-To: <CAN5esQLNBubvtbD=XVMPr3jOdYHCjxPWq791dpdt7X-3p=5=BQ@mail.gmail.com>
References: <CAN5esQLE8OOjouM8zFRK7csDp6p7-B6OpeNxPsjjQNkVYOew0g@mail.gmail.com>
	<CAN5esQLNBubvtbD=XVMPr3jOdYHCjxPWq791dpdt7X-3p=5=BQ@mail.gmail.com>
Message-ID: <CAFiqYum_YyEciw54UYYffD0hO3Q5uN_GEYPx2dLrTuuRG8B2sA@mail.gmail.com>

Thanks for letting us know what worked, it'll be a great help to anyone
Googling this problem in future. :o)



On 18 December 2012 17:43, Richard Brady <rnbrady at gmail.com> wrote:

> Looks like <param name="liberal-dtmf" value="true"/> in the profile does
> the trick!
>
>
>
> On 18 December 2012 17:00, Richard Brady <rnbrady at gmail.com> wrote:
>
>> Hi folks
>>
>> I have the joy of talking to a Cisco device which behaves in weird and
>> wonderful ways.
>>
>> When transferring a caller (in this case FreeSWITCH) to MOH it negotiates
>> RFC2833 out of the SDP.
>>
>> When reconnecting the caller to a more meaningful endpoint, it
>> renegotiates the media again, but using delayed offer/answer, which means
>> FreeSWITCH must perform the offer.
>>
>> Since FreeSWITCH has seen RFC2833 removed previously:
>>
>>     [DEBUG] sofia_glue.c:5277 No 2833 in SDP. Disable 2833 dtmf and
>> switch to INFO.
>>
>> It does not offer RFC2833 in the new SDP (if it did, the Cisco would now
>> accept it and answer with it).
>>
>> Any suggestion for how I can force it to re-offer RFC2833?
>>
>> Regards,
>> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 19 00:37:20 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 18 Dec 2012 16:37:20 -0500
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>	<541b01cddcb1$72330200$56990600$@bizfocused.com>
	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>
Message-ID: <5da601cddd67$dc76b6f0$956424d0$@bizfocused.com>

OK, I got one the logs!  I have trimmed the file down, but it is still over
5 MB and pastebin just chokes on it.

 

It is on my server:  http://www.bizfocused.com/service_not_found.log

 

Thanks for looking.

 

If you really need pcap, I am afraid I need detailed instructions on
installing/running it.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 7:00 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

usually the 406 is done in an established call to refuse a codec change
during re-invite.

Its possible the other end thinks we want to change the codec when we do the
session-timer re-invite but I'm sure we don't but the sip trace can help
shed some light.  You can run a pcap too at the same time so when we find
the bad call in the logs we can filter it out of the pcap too.  To avoid it
getting too big you can just restart it every so often or use sippcapdump
and delete calls that are not affected.

 

 

 

On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Waiting for another failure with siptrace REALLY on this time.

 

If the user has clicked DND on these cisco phones, could that cause these
messages?

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 12:32 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

The siptrace is not in here?  Did you enable it on the console and then
capture to the log file perhaps?

you should do "sofia tracelevel debug" too to route traces to the log file.

 

Its hard to say for sure with no sip trace but it seems like the far end is
rejecting the session timer re-invite causing the call to end.

 

 

On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

http://pastebin.freeswitch.org/20342





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900> 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From sdevoy at bizfocused.com  Wed Dec 19 00:52:01 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 18 Dec 2012 16:52:01 -0500
Subject: [Freeswitch-users] New install 1.2.Stable blows up compiling
	mod_perl.so
Message-ID: <5dc601cddd69$e97a1890$bc6e49b0$@bizfocused.com>

I just started a clean install of the stable tree on a brand new install of
Centos 5.3.

Here is EXACTLY what I have done so far:

Logged in to root

Cd /usr/local/src

yum install gcc

yum -y install zlib-devel openssl-devel cpio expat-devel gettext-devel

cd /usr/local/src

wget http://git-core.googlecode.com/files/git-1.7.9.tar.gz

tar xvzf git-1.7.9.tar.gz

cd git-1.7.9

./configure

make

make install

yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig

 

cd /usr/local/src

git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git

./bootstrap.sh    

./configure

tftp modules.conf from existing centos/fs server   (did this kill it)?  If
so how can I get it back?

make & make install  

 

after about 20 minutes the output is:

gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c  -fPIC -DPIC
-o .libs/mod_perl.o

gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c -o
mod_perl.o >/dev/null 2>&1

Creating mod_perl.so...

/usr/bin/ld: cannot find -lgdbm

collect2: ld returned 1 exit status

gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E -Wl,-rpath
-Wl,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE freeswitch_perl.o
mod_perl_wrap.o perlxsi.o  /usr/local/src/freeswitch/.libs/libfreeswitch.so
-L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib
/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid
-L/usr/local/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lrt -lssl
-lcrypto -lz -lncurses -ljpeg -lodbc -L/usr/local/lib
/usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a
-L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -lperl -lresolv -lnsl
-lgdbm -ldb -ldl -lm -lcrypt -lutil -lpthread -lc   -Wl,--rpath
-Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod

make[5]: *** [mod_perl.so] Error 1

make[4]: *** [all] Error 1

make[3]: *** [mod_perl-all] Error 1

make[2]: *** [all-recursive] Error 1

make[1]: *** [all-recursive] Error 1

make: *** [all] Error 2

 

 

The modules.conf I upload is:

#applications/mod_abstraction

#applications/mod_avmd

applications/mod_blacklist

applications/mod_callcenter

applications/mod_cidlookup

applications/mod_cluechoo

applications/mod_commands

applications/mod_conference

applications/mod_curl

applications/mod_db

applications/mod_directory

#applications/mod_distributor

applications/mod_dptools

#applications/mod_easyroute

applications/mod_enum

applications/mod_esf

#applications/mod_esl

applications/mod_expr

applications/mod_fifo

#applications/mod_fsk

applications/mod_fsv

applications/mod_hash

applications/mod_httapi

#applications/mod_http_cache

#applications/mod_ladspa

applications/mod_lcr

#applications/mod_memcache

#applications/mod_mongo

#applications/mod_nibblebill

#applications/mod_osp

#applications/mod_redis

#applications/mod_rss

applications/mod_sms

#applications/mod_snapshot

#applications/mod_snipe_hunt

#applications/mod_snom

#applications/mod_soundtouch

applications/mod_spandsp

#applications/mod_spy

#applications/mod_stress

applications/mod_valet_parking

#applications/mod_vmd

applications/mod_voicemail

applications/mod_voicemail_ivr

#applications/mod_random

#asr_tts/mod_cepstral

asr_tts/mod_flite

asr_tts/mod_pocketsphinx

asr_tts/mod_tts_commandline

#asr_tts/mod_unimrcp

codecs/mod_amr

#codecs/mod_amrwb

codecs/mod_bv

#codecs/mod_celt

#codecs/mod_codec2

#codecs/mod_com_g729

#codecs/mod_dahdi_codec

codecs/mod_g723_1

codecs/mod_g729

codecs/mod_h26x

#codecs/mod_ilbc

#codecs/mod_isac

#codecs/mod_opus

#codecs/mod_sangoma_codec

#codecs/mod_silk

#codecs/mod_siren

codecs/mod_speex

dialplans/mod_dialplan_asterisk

#dialplans/mod_dialplan_directory

dialplans/mod_dialplan_xml

#directories/mod_ldap

#endpoints/mod_alsa

endpoints/mod_dingaling

#endpoints/mod_h323

#endpoints/mod_khomp

endpoints/mod_loopback

#endpoints/mod_opal

#endpoints/mod_portaudio

endpoints/mod_rtmp

#endpoints/mod_skinny

#endpoints/mod_skypopen

endpoints/mod_sofia

event_handlers/mod_cdr_csv

#event_handlers/mod_cdr_mongodb

#event_handlers/mod_cdr_pg_csv

event_handlers/mod_cdr_sqlite

#event_handlers/mod_erlang_event

#event_handlers/mod_event_multicast

event_handlers/mod_event_socket

#event_handlers/mod_event_zmq

#event_handlers/mod_radius_cdr

event_handlers/mod_snmp

formats/mod_local_stream

formats/mod_native_file

#formats/mod_portaudio_stream

#formats/mod_shell_stream

formats/mod_shout

formats/mod_sndfile

formats/mod_tone_stream

#formats/mod_vlc

#languages/mod_java

languages/mod_lua

#languages/mod_managed

languages/mod_perl

languages/mod_python

languages/mod_spidermonkey

#languages/mod_yaml

loggers/mod_console

loggers/mod_logfile

loggers/mod_syslog

#say/mod_say_de

say/mod_say_en

#say/mod_say_es

#say/mod_say_fr

#say/mod_say_he

#say/mod_say_hu

#say/mod_say_it

#say/mod_say_nl

#say/mod_say_pt

#say/mod_say_ru

#say/mod_say_th

#say/mod_say_zh

#timers/mod_posix_timer

#timers/mod_timerfd

xml_int/mod_xml_cdr

xml_int/mod_xml_curl

#xml_int/mod_xml_ldap

xml_int/mod_xml_rpc

xml_int/mod_xml_scgi

 

#../../libs/freetdm/mod_freetdm

#../../libs/openzap/mod_openzap

 

## Experimental Modules (don't cry if they're broken)

#../../contrib/mod/xml_int/mod_xml_odbc

 

 

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From yungwei at resolvity.com  Wed Dec 19 00:56:08 2012
From: yungwei at resolvity.com (Yungwei Chen)
Date: Tue, 18 Dec 2012 16:56:08 -0500
Subject: [Freeswitch-users] voices in the recordings are out of sync
Message-ID: <33095823FD21DF429B481B5163264B799F6D8752B6@VMBX102.ihostexchange.net>

Hi,

I found one issue that voices are always out of sync in the recordings.
I am running freeswitch-1.2.5.1-1 on CentOS 5 (i386), which was installed from yum.
I am having trouble installing the latest version from source due to an error: Autoconf version 2.62 or higher is required.
It would be nice if someone can reproduce this issue against HEAD. Thanks.

Here're the steps to reproduce it. The idea is to call a phone number and then bridge to another phone number while the entire session is being recorded.
1. In dialplan/public.xml, make sure you have a dialplan to handle any 10 digit phone numbers.
    <extension name="public_extensions">
      <condition field="destination_number" expression="^\d{10}$">
        <action application="transfer" data="main XML default"/>
      </condition>
    </extension>

2. In dialplan/default/main.xml, make sure you have an extension to handle the call in the default context.
  <extension name="test">
      <condition field="${destination_number}" expression="^main$">
        <action application="set" data="RECORD_STEREO=false"/>
        <action application="answer" />
        <action application="record_session" data="/tmp/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
        <action application="bridge" data="{ignore_early_media=false}[leg_timeout=60]sofia/gateway/gw1/1234567890"/>
      </condition>
  </extension>

3. In sip_profiles/external/gateways.xml, make sure you have a gateway that allows you to make an outbound call.
    <include>
        <gateway name="gw1">
          <param name="username" value=""/>
          <param name="password" value=""/>
          <param name="realm" value=""/>
          <param name="from-domain" value=""/>
          <param name="extension" value=""/>
          <param name="expire-seconds" value="60"/>
          <param name="register" value="false"/>
          <param name="retry-seconds" value="60"/>
        </gateway>
    </include>

4. make a call to one of the allowed 10-digit phone numbers in your environment.
5. Once the call is answered, the caller shall start to count from 1 to 60 with some pause after each number.
6. The callee shall repeat each number he/she heard from the caller.
7. You should be able to hear that 2 voices in the recoridng (/tmp/rec.wav) are out of sync.


From abaci64 at gmail.com  Wed Dec 19 01:00:53 2012
From: abaci64 at gmail.com (Abaci)
Date: Tue, 18 Dec 2012 17:00:53 -0500
Subject: [Freeswitch-users] New install 1.2.Stable blows up compiling
 mod_perl.so
In-Reply-To: <5dc601cddd69$e97a1890$bc6e49b0$@bizfocused.com>
References: <5dc601cddd69$e97a1890$bc6e49b0$@bizfocused.com>
Message-ID: <50D0E795.1090303@gmail.com>

yum install gdbm-devel db4-devel

On 12/18/2012 4:52 PM, Sean Devoy wrote:
>
> I just started a clean install of the stable tree on a brand new 
> install of Centos 5.3.
>
> Here is EXACTLY what I have done so far:
>
> Logged in to root
>
> Cd /usr/local/src
>
> yum install gcc
>
> yum -y install zlib-devel openssl-devel cpio expat-devel gettext-devel
>
> cd /usr/local/src
>
> wget http://git-core.googlecode.com/files/git-1.7.9.tar.gz
>
> tar xvzf git-1.7.9.tar.gz
>
> cd git-1.7.9
>
> ./configure
>
> make
>
> make install
>
> yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool 
> make ncurses-devel pkgconfig
>
> cd /usr/local/src
>
> git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
>
> ./bootstrap.sh
>
> ./configure
>
> tftp modules.conf from existing centos/fs server   (did this kill 
> it)?  If so how can I get it back?
>
> make & make install
>
> after about 20 minutes the output is:
>
> gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing 
> -pipe -Wdeclaration-after-statement -I/usr/local/include 
> -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm 
> -I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL 
> -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 
> -I/usr/local/src/freeswitch/libs/stfu -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 -DHAVE_CONFIG_H -c 
> /usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c -fPIC 
> -DPIC -o .libs/mod_perl.o
>
> gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing 
> -pipe -Wdeclaration-after-statement -I/usr/local/include 
> -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm 
> -I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL 
> -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 
> -I/usr/local/src/freeswitch/libs/stfu -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 -DHAVE_CONFIG_H -c 
> /usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c -o 
> mod_perl.o >/dev/null 2>&1
>
> Creating mod_perl.so...
>
> /usr/bin/ld: cannot find -lgdbm
>
> collect2: ld returned 1 exit status
>
> gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing 
> -pipe -Wdeclaration-after-statement -I/usr/local/include 
> -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm 
> -I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL 
> -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 
> -I/usr/local/src/freeswitch/libs/stfu -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_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E -Wl,-rpath 
> -Wl,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE 
> freeswitch_perl.o mod_perl_wrap.o perlxsi.o 
> /usr/local/src/freeswitch/.libs/libfreeswitch.so 
> -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib 
> /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a 
> /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid 
> -L/usr/local/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lrt -lssl 
> -lcrypto -lz -lncurses -ljpeg -lodbc -L/usr/local/lib 
> /usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a 
> -L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -lperl -lresolv 
> -lnsl -lgdbm -ldb -ldl -lm -lcrypt -lutil -lpthread -lc   -Wl,--rpath 
> -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod
>
> make[5]: *** [mod_perl.so] Error 1
>
> make[4]: *** [all] Error 1
>
> make[3]: *** [mod_perl-all] Error 1
>
> make[2]: *** [all-recursive] Error 1
>
> make[1]: *** [all-recursive] Error 1
>
> make: *** [all] Error 2
>
> The modules.conf I upload is:
>
> #applications/mod_abstraction
>
> #applications/mod_avmd
>
> applications/mod_blacklist
>
> applications/mod_callcenter
>
> applications/mod_cidlookup
>
> applications/mod_cluechoo
>
> applications/mod_commands
>
> applications/mod_conference
>
> applications/mod_curl
>
> applications/mod_db
>
> applications/mod_directory
>
> #applications/mod_distributor
>
> applications/mod_dptools
>
> #applications/mod_easyroute
>
> applications/mod_enum
>
> applications/mod_esf
>
> #applications/mod_esl
>
> applications/mod_expr
>
> applications/mod_fifo
>
> #applications/mod_fsk
>
> applications/mod_fsv
>
> applications/mod_hash
>
> applications/mod_httapi
>
> #applications/mod_http_cache
>
> #applications/mod_ladspa
>
> applications/mod_lcr
>
> #applications/mod_memcache
>
> #applications/mod_mongo
>
> #applications/mod_nibblebill
>
> #applications/mod_osp
>
> #applications/mod_redis
>
> #applications/mod_rss
>
> applications/mod_sms
>
> #applications/mod_snapshot
>
> #applications/mod_snipe_hunt
>
> #applications/mod_snom
>
> #applications/mod_soundtouch
>
> applications/mod_spandsp
>
> #applications/mod_spy
>
> #applications/mod_stress
>
> applications/mod_valet_parking
>
> #applications/mod_vmd
>
> applications/mod_voicemail
>
> applications/mod_voicemail_ivr
>
> #applications/mod_random
>
> #asr_tts/mod_cepstral
>
> asr_tts/mod_flite
>
> asr_tts/mod_pocketsphinx
>
> asr_tts/mod_tts_commandline
>
> #asr_tts/mod_unimrcp
>
> codecs/mod_amr
>
> #codecs/mod_amrwb
>
> codecs/mod_bv
>
> #codecs/mod_celt
>
> #codecs/mod_codec2
>
> #codecs/mod_com_g729
>
> #codecs/mod_dahdi_codec
>
> codecs/mod_g723_1
>
> codecs/mod_g729
>
> codecs/mod_h26x
>
> #codecs/mod_ilbc
>
> #codecs/mod_isac
>
> #codecs/mod_opus
>
> #codecs/mod_sangoma_codec
>
> #codecs/mod_silk
>
> #codecs/mod_siren
>
> codecs/mod_speex
>
> dialplans/mod_dialplan_asterisk
>
> #dialplans/mod_dialplan_directory
>
> dialplans/mod_dialplan_xml
>
> #directories/mod_ldap
>
> #endpoints/mod_alsa
>
> endpoints/mod_dingaling
>
> #endpoints/mod_h323
>
> #endpoints/mod_khomp
>
> endpoints/mod_loopback
>
> #endpoints/mod_opal
>
> #endpoints/mod_portaudio
>
> endpoints/mod_rtmp
>
> #endpoints/mod_skinny
>
> #endpoints/mod_skypopen
>
> endpoints/mod_sofia
>
> event_handlers/mod_cdr_csv
>
> #event_handlers/mod_cdr_mongodb
>
> #event_handlers/mod_cdr_pg_csv
>
> event_handlers/mod_cdr_sqlite
>
> #event_handlers/mod_erlang_event
>
> #event_handlers/mod_event_multicast
>
> event_handlers/mod_event_socket
>
> #event_handlers/mod_event_zmq
>
> #event_handlers/mod_radius_cdr
>
> event_handlers/mod_snmp
>
> formats/mod_local_stream
>
> formats/mod_native_file
>
> #formats/mod_portaudio_stream
>
> #formats/mod_shell_stream
>
> formats/mod_shout
>
> formats/mod_sndfile
>
> formats/mod_tone_stream
>
> #formats/mod_vlc
>
> #languages/mod_java
>
> languages/mod_lua
>
> #languages/mod_managed
>
> languages/mod_perl
>
> languages/mod_python
>
> languages/mod_spidermonkey
>
> #languages/mod_yaml
>
> loggers/mod_console
>
> loggers/mod_logfile
>
> loggers/mod_syslog
>
> #say/mod_say_de
>
> say/mod_say_en
>
> #say/mod_say_es
>
> #say/mod_say_fr
>
> #say/mod_say_he
>
> #say/mod_say_hu
>
> #say/mod_say_it
>
> #say/mod_say_nl
>
> #say/mod_say_pt
>
> #say/mod_say_ru
>
> #say/mod_say_th
>
> #say/mod_say_zh
>
> #timers/mod_posix_timer
>
> #timers/mod_timerfd
>
> xml_int/mod_xml_cdr
>
> xml_int/mod_xml_curl
>
> #xml_int/mod_xml_ldap
>
> xml_int/mod_xml_rpc
>
> xml_int/mod_xml_scgi
>
> #../../libs/freetdm/mod_freetdm
>
> #../../libs/openzap/mod_openzap
>
> ## Experimental Modules (don't cry if they're broken)
>
> #../../contrib/mod/xml_int/mod_xml_odbc
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 sdevoy at bizfocused.com  Wed Dec 19 01:23:44 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 18 Dec 2012 17:23:44 -0500
Subject: [Freeswitch-users] New install 1.2.Stable blows up compiling
	mod_perl.so
In-Reply-To: <50D0E795.1090303@gmail.com>
References: <5dc601cddd69$e97a1890$bc6e49b0$@bizfocused.com>
	<50D0E795.1090303@gmail.com>
Message-ID: <5e0401cddd6e$57c742b0$0755c810$@bizfocused.com>

Thank you!

That resulted in a message saying I needed to install bison, so I ran yum
install bison.

 

That resulted in shit though:

gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o

gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o

libtool: link: `VbrTag.lo' is not a valid libtool object

make[9]: *** [libmp3lame.la] Error 1

make[8]: *** [all-recursive] Error 1

Making all in frontend

libtool: link: `VbrTag.lo' is not a valid libtool object

make[9]: *** [libmp3lame.la] Error 1

make[8]: *** [all-recursive] Error 1

Making all in frontend

make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.

Making all in Dll

Making all in debian

Making all in doc

Making all in html

Making all in man

Making all in include

Making all in misc

Making all in dshow

Making all in ACM

Making all in ADbg

Making all in ddk

make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.

Making all in Dll

Making all in tinyxml

Making all in debian

Making all in doc

Making all in mac

Making all in macosx

Making all in html

Making all in man

Making all in English.lproj

Making all in include

Making all in LAME.xcodeproj

Making all in misc

Making all in dshow

Making all in vc_solution

Making all in ACM

make[7]: *** [all-recursive] Error 1

make[6]: *** [all] Error 2

make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2

make[4]: *** [install] Error 1

make[3]: *** [mod_shout-install] Error 1

Making all in ADbg

Making all in ddk

Making all in tinyxml

Making all in mac

Making all in macosx

Making all in English.lproj

Making all in LAME.xcodeproj

Making all in vc_solution

make[7]: *** [all-recursive] Error 1

make[6]: *** [all] Error 2

make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2

make[4]: *** [all] Error 1

make[3]: *** [mod_shout-all] Error 1

make[2]: *** [all-recursive] Error 1

make[1]: *** [all-recursive] Error 1

make: *** [all] Error 2

make[2]: *** [install-recursive] Error 1

make[1]: *** [install-recursive] Error 1

make: *** [install] Error 2

[1]+  Exit 2                  make

 

Can't we get the INSTALL page of the wiki right?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abaci
Sent: Tuesday, December 18, 2012 5:01 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

 

yum install gdbm-devel db4-devel
 

On 12/18/2012 4:52 PM, Sean Devoy wrote:

I just started a clean install of the stable tree on a brand new install of
Centos 5.3.

Here is EXACTLY what I have done so far:

Logged in to root

Cd /usr/local/src

yum install gcc

yum -y install zlib-devel openssl-devel cpio expat-devel gettext-devel

cd /usr/local/src

wget http://git-core.googlecode.com/files/git-1.7.9.tar.gz

tar xvzf git-1.7.9.tar.gz

cd git-1.7.9

./configure

make

make install

yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig

 

cd /usr/local/src

git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git

./bootstrap.sh    

./configure

tftp modules.conf from existing centos/fs server   (did this kill it)?  If
so how can I get it back?

make & make install  

 

after about 20 minutes the output is:

gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c  -fPIC -DPIC
-o .libs/mod_perl.o

gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c -o
mod_perl.o >/dev/null 2>&1

Creating mod_perl.so...

/usr/bin/ld: cannot find -lgdbm

collect2: ld returned 1 exit status

gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E -Wl,-rpath
-Wl,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE freeswitch_perl.o
mod_perl_wrap.o perlxsi.o  /usr/local/src/freeswitch/.libs/libfreeswitch.so
-L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib
/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid
-L/usr/local/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lrt -lssl
-lcrypto -lz -lncurses -ljpeg -lodbc -L/usr/local/lib
/usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a
-L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -lperl -lresolv -lnsl
-lgdbm -ldb -ldl -lm -lcrypt -lutil -lpthread -lc   -Wl,--rpath
-Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod

make[5]: *** [mod_perl.so] Error 1

make[4]: *** [all] Error 1

make[3]: *** [mod_perl-all] Error 1

make[2]: *** [all-recursive] Error 1

make[1]: *** [all-recursive] Error 1

make: *** [all] Error 2

 

 

The modules.conf I upload is:

#applications/mod_abstraction

#applications/mod_avmd

applications/mod_blacklist

applications/mod_callcenter

applications/mod_cidlookup

applications/mod_cluechoo

applications/mod_commands

applications/mod_conference

applications/mod_curl

applications/mod_db

applications/mod_directory

#applications/mod_distributor

applications/mod_dptools

#applications/mod_easyroute

applications/mod_enum

applications/mod_esf

#applications/mod_esl

applications/mod_expr

applications/mod_fifo

#applications/mod_fsk

applications/mod_fsv

applications/mod_hash

applications/mod_httapi

#applications/mod_http_cache

#applications/mod_ladspa

applications/mod_lcr

#applications/mod_memcache

#applications/mod_mongo

#applications/mod_nibblebill

#applications/mod_osp

#applications/mod_redis

#applications/mod_rss

applications/mod_sms

#applications/mod_snapshot

#applications/mod_snipe_hunt

#applications/mod_snom

#applications/mod_soundtouch

applications/mod_spandsp

#applications/mod_spy

#applications/mod_stress

applications/mod_valet_parking

#applications/mod_vmd

applications/mod_voicemail

applications/mod_voicemail_ivr

#applications/mod_random

#asr_tts/mod_cepstral

asr_tts/mod_flite

asr_tts/mod_pocketsphinx

asr_tts/mod_tts_commandline

#asr_tts/mod_unimrcp

codecs/mod_amr

#codecs/mod_amrwb

codecs/mod_bv

#codecs/mod_celt

#codecs/mod_codec2

#codecs/mod_com_g729

#codecs/mod_dahdi_codec

codecs/mod_g723_1

codecs/mod_g729

codecs/mod_h26x

#codecs/mod_ilbc

#codecs/mod_isac

#codecs/mod_opus

#codecs/mod_sangoma_codec

#codecs/mod_silk

#codecs/mod_siren

codecs/mod_speex

dialplans/mod_dialplan_asterisk

#dialplans/mod_dialplan_directory

dialplans/mod_dialplan_xml

#directories/mod_ldap

#endpoints/mod_alsa

endpoints/mod_dingaling

#endpoints/mod_h323

#endpoints/mod_khomp

endpoints/mod_loopback

#endpoints/mod_opal

#endpoints/mod_portaudio

endpoints/mod_rtmp

#endpoints/mod_skinny

#endpoints/mod_skypopen

endpoints/mod_sofia

event_handlers/mod_cdr_csv

#event_handlers/mod_cdr_mongodb

#event_handlers/mod_cdr_pg_csv

event_handlers/mod_cdr_sqlite

#event_handlers/mod_erlang_event

#event_handlers/mod_event_multicast

event_handlers/mod_event_socket

#event_handlers/mod_event_zmq

#event_handlers/mod_radius_cdr

event_handlers/mod_snmp

formats/mod_local_stream

formats/mod_native_file

#formats/mod_portaudio_stream

#formats/mod_shell_stream

formats/mod_shout

formats/mod_sndfile

formats/mod_tone_stream

#formats/mod_vlc

#languages/mod_java

languages/mod_lua

#languages/mod_managed

languages/mod_perl

languages/mod_python

languages/mod_spidermonkey

#languages/mod_yaml

loggers/mod_console

loggers/mod_logfile

loggers/mod_syslog

#say/mod_say_de

say/mod_say_en

#say/mod_say_es

#say/mod_say_fr

#say/mod_say_he

#say/mod_say_hu

#say/mod_say_it

#say/mod_say_nl

#say/mod_say_pt

#say/mod_say_ru

#say/mod_say_th

#say/mod_say_zh

#timers/mod_posix_timer

#timers/mod_timerfd

xml_int/mod_xml_cdr

xml_int/mod_xml_curl

#xml_int/mod_xml_ldap

xml_int/mod_xml_rpc

xml_int/mod_xml_scgi

 

#../../libs/freetdm/mod_freetdm

#../../libs/openzap/mod_openzap

 

## Experimental Modules (don't cry if they're broken)

#../../contrib/mod/xml_int/mod_xml_odbc

 

 






_________________________________________________________________________
Professional FreeSWITCH Consulting 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 Sirish.MasurMohan at oa.com.au  Wed Dec 19 01:36:55 2012
From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan)
Date: Wed, 19 Dec 2012 09:36:55 +1100
Subject: [Freeswitch-users] Newbie question: Executing Lua scripts for
 incoming calls, how concurrency is to be handled?
References: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au>
	<50CE5D6E.2020401@quentustech.com>
	<965759A53E43FE439E43565A7715E5F058F4156D32@oa-exchange1.oa.com.au>
	<CAKzWOxWFh3RNYg9NY4Dqa-=f9KZOaXTvM8NGzHpvEvzV16VUWA@mail.gmail.com> 
Message-ID: <965759A53E43FE439E43565A7715E5F058F4156DD8@oa-exchange1.oa.com.au>

Hey Guys,

Would really appreciate if you could help me out here - isn't there a way to handle concurrent calls in the dial plan, especially when Lua scripts, accessing global variables, are executed on receiving calls?

Is mod_fifo the closest I could get to handle concurrency (as Michael has explained)? If yes, how do I trigger SIP registrations, especially working with a PBX which I don't have full control of?

With regards,
Sirish

From: Sirish Masur Mohan
Sent: Tuesday, 18 December 2012 1:02 PM
To: FreeSWITCH Users Help
Subject: RE: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

Hi Michael,

Thanks for the reply.

>> You would need a SIP registration from the PBX to FreeSWITCH for each of the modem extensions
I am seen SIP clients such as X-Lite sending out the SIP registrations, but could you please clarify as to how this can be achieved in the PBX? The final production environment would be out in the customer's PBX, which I may not have complete control of..

>> What application are you building?
I may not be able to provide the details because of the NDA with customer, but what I am trying to achieve is, to replace an existing IVR with FreeSWITCH in an old existing setup of the customer - that's the reason why we continue working with dialup modems!

With regards,
Sirish

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: Tuesday, 18 December 2012 6:23 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

You don't have to have actual human agents for mod_fifo. You could define a user for each modem and then manually "log in" those "agents" on the command line using the fifo_member API command. Something like this:

fifo_member add fifo_name {fifo_member_wait=nowait}user/1234

Where 1234 is the user id of one of the modems. You would need a SIP registration from the PBX to FreeSWITCH for each of the modem extensions.

Having modems go through a VoIP system sounds a bit scary. What application are you building?
-MC
On Sun, Dec 16, 2012 at 5:41 PM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au<mailto:Sirish.MasurMohan at oa.com.au>> wrote:
Hi William,

Thanks for the reply.

My setup is as follows:
Client(Caller)->dialup modem->PBX->FreeSWITCH->PBX-> 4 dialup modems->Server(Receiver)

I basically need FreeSWITCH to bridge the incoming call to the best external destination (out of the 4 available), so that the modem training, connection etc can take place smoothly, before exchange of data. I am not sure if mod_fifo would help me in this scenario, as, I would require an agent to dial in and read the fifo. Could you please clarify?

Thanks!

With regards,
Sirish

-----Original Message-----
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 William King
Sent: Monday, 17 December 2012 10:47 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

Sounds like you want to take a look into mod_fifo.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337<tel:%28877%29%20211-9337>
Office: (206) 388-4772<tel:%28206%29%20388-4772>
Cell:   (253) 686-5518<tel:%28253%29%20686-5518>
william.king at quentustech.com<mailto:william.king at quentustech.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From krice at freeswitch.org  Wed Dec 19 01:37:13 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 18 Dec 2012 16:37:13 -0600
Subject: [Freeswitch-users] New install 1.2.Stable blows up compiling
 mod_perl.so
In-Reply-To: <5e0401cddd6e$57c742b0$0755c810$@bizfocused.com>
Message-ID: <CCF64C39.62413%krice@freeswitch.org>

If you installed on a centos base install then follow the directions on the
wiki at 
http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS

That usually avoids all these errors...  I usually do something like

 yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig unixODBC-devel openssl-devel libogg-devel
libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel
expat-devel zlib zlib-devel bzip2 which

You may of course need to add a few extra RPMs if you are compiling things
that you have to uncomment in modules.conf

 



On 12/18/12 4:23 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> Thank you!
> That resulted in a message saying I needed to install bison, so I ran yum
> install bison.
>  
> That resulted in shit though:
> gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe -g
> -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
> mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
>  gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe -g
> -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
> mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
> libtool: link: `VbrTag.lo' is not a valid libtool object
> make[9]: *** [libmp3lame.la] Error 1
> make[8]: *** [all-recursive] Error 1
> Making all in frontend
> libtool: link: `VbrTag.lo' is not a valid libtool object
> make[9]: *** [libmp3lame.la] Error 1
> make[8]: *** [all-recursive] Error 1
> Making all in frontend
> make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
> `lame'.  Stop.
> Making all in Dll
> Making all in debian
> Making all in doc
> Making all in html
> Making all in man
> Making all in include
> Making all in misc
> Making all in dshow
> Making all in ACM
> Making all in ADbg
> Making all in ddk
> make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
> `lame'.  Stop.
> Making all in Dll
> Making all in tinyxml
> Making all in debian
> Making all in doc
> Making all in mac
> Making all in macosx
> Making all in html
> Making all in man
> Making all in English.lproj
> Making all in include
> Making all in LAME.xcodeproj
> Making all in misc
> Making all in dshow
> Making all in vc_solution
> Making all in ACM
> make[7]: *** [all-recursive] Error 1
> make[6]: *** [all] Error 2
> make[5]: *** 
> [/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error 2
> make[4]: *** [install] Error 1
> make[3]: *** [mod_shout-install] Error 1
> Making all in ADbg
> Making all in ddk
> Making all in tinyxml
> Making all in mac
> Making all in macosx
> Making all in English.lproj
> Making all in LAME.xcodeproj
> Making all in vc_solution
> make[7]: *** [all-recursive] Error 1
> make[6]: *** [all] Error 2
> make[5]: *** 
> [/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error 2
> make[4]: *** [all] Error 1
> make[3]: *** [mod_shout-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
> make[2]: *** [install-recursive] Error 1
> make[1]: *** [install-recursive] Error 1
> make: *** [install] Error 2
> [1]+  Exit 2                  make
>  
> Can?t we get the INSTALL page of the wiki right?
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abaci
> Sent: Tuesday, December 18, 2012 5:01 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
> mod_perl.so
>  
> yum install gdbm-devel db4-devel
>  
> On 12/18/2012 4:52 PM, Sean Devoy wrote:
>> I just started a clean install of the stable tree on a brand new install of
>> Centos 5.3.
>> Here is EXACTLY what I have done so far:
>> Logged in to root
>> Cd /usr/local/src
>> yum install gcc
>> yum -y install zlib-devel openssl-devel cpio expat-devel gettext-devel
>> cd /usr/local/src
>> wget http://git-core.googlecode.com/files/git-1.7.9.tar.gz
>> tar xvzf git-1.7.9.tar.gz
>> cd git-1.7.9
>> ./configure
>> make
>> make install
>> yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
>> ncurses-devel pkgconfig
>>  
>> cd /usr/local/src
>> git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
>> ./bootstrap.sh  
>> ./configure
>> tftp modules.conf from existing centos/fs server   (did this kill it)?  If so
>> how can I get it back?
>> make & make install
>>  
>> after about 20 minutes the output is:
>> gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
>> -Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
>> -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
>> -I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
>> -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
>> -I/usr/local/src/freeswitch/libs/stfu -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
>> -DHAVE_CONFIG_H -c
>> /usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c  -fPIC -DPIC
>> -o .libs/mod_perl.o
>> gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
>> -Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
>> -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
>> -I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
>> -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
>> -I/usr/local/src/freeswitch/libs/stfu -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
>> -DHAVE_CONFIG_H -c
>> /usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c -o mod_perl.o
>> >/dev/null 2>&1
>> Creating mod_perl.so...
>> /usr/bin/ld: cannot find -lgdbm
>> collect2: ld returned 1 exit status
>> gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
>> -Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
>> -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
>> -I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
>> -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
>> -I/usr/local/src/freeswitch/libs/stfu -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_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E -Wl,-rpath
>> -Wl,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE freeswitch_perl.o
>> mod_perl_wrap.o perlxsi.o  /usr/local/src/freeswitch/.libs/libfreeswitch.so
>> -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib
>> /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
>> /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid
>> -L/usr/local/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lrt -lssl -lcrypto
>> -lz -lncurses -ljpeg -lodbc -L/usr/local/lib
>> /usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a
>> -L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -lperl -lresolv -lnsl
>> -lgdbm -ldb -ldl -lm -lcrypt -lutil -lpthread -lc   -Wl,--rpath
>> -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod
>> make[5]: *** [mod_perl.so] Error 1
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_perl-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>> make[1]: *** [all-recursive] Error 1
>> make: *** [all] Error 2
>>  
>>  
>> The modules.conf I upload is:
>> #applications/mod_abstraction
>> #applications/mod_avmd
>> applications/mod_blacklist
>> applications/mod_callcenter
>> applications/mod_cidlookup
>> applications/mod_cluechoo
>> applications/mod_commands
>> applications/mod_conference
>> applications/mod_curl
>> applications/mod_db
>> applications/mod_directory
>> #applications/mod_distributor
>> applications/mod_dptools
>> #applications/mod_easyroute
>> applications/mod_enum
>> applications/mod_esf
>> #applications/mod_esl
>> applications/mod_expr
>> applications/mod_fifo
>> #applications/mod_fsk
>> applications/mod_fsv
>> applications/mod_hash
>> applications/mod_httapi
>> #applications/mod_http_cache
>> #applications/mod_ladspa
>> applications/mod_lcr
>> #applications/mod_memcache
>> #applications/mod_mongo
>> #applications/mod_nibblebill
>> #applications/mod_osp
>> #applications/mod_redis
>> #applications/mod_rss
>> applications/mod_sms
>> #applications/mod_snapshot
>> #applications/mod_snipe_hunt
>> #applications/mod_snom
>> #applications/mod_soundtouch
>> applications/mod_spandsp
>> #applications/mod_spy
>> #applications/mod_stress
>> applications/mod_valet_parking
>> #applications/mod_vmd
>> applications/mod_voicemail
>> applications/mod_voicemail_ivr
>> #applications/mod_random
>> #asr_tts/mod_cepstral
>> asr_tts/mod_flite
>> asr_tts/mod_pocketsphinx
>> asr_tts/mod_tts_commandline
>> #asr_tts/mod_unimrcp
>> codecs/mod_amr
>> #codecs/mod_amrwb
>> codecs/mod_bv
>> #codecs/mod_celt
>> #codecs/mod_codec2
>> #codecs/mod_com_g729
>> #codecs/mod_dahdi_codec
>> codecs/mod_g723_1
>> codecs/mod_g729
>> codecs/mod_h26x
>> #codecs/mod_ilbc
>> #codecs/mod_isac
>> #codecs/mod_opus
>> #codecs/mod_sangoma_codec
>> #codecs/mod_silk
>> #codecs/mod_siren
>> codecs/mod_speex
>> dialplans/mod_dialplan_asterisk
>> #dialplans/mod_dialplan_directory
>> dialplans/mod_dialplan_xml
>> #directories/mod_ldap
>> #endpoints/mod_alsa
>> endpoints/mod_dingaling
>> #endpoints/mod_h323
>> #endpoints/mod_khomp
>> endpoints/mod_loopback
>> #endpoints/mod_opal
>> #endpoints/mod_portaudio
>> endpoints/mod_rtmp
>> #endpoints/mod_skinny
>> #endpoints/mod_skypopen
>> endpoints/mod_sofia
>> event_handlers/mod_cdr_csv
>> #event_handlers/mod_cdr_mongodb
>> #event_handlers/mod_cdr_pg_csv
>> event_handlers/mod_cdr_sqlite
>> #event_handlers/mod_erlang_event
>> #event_handlers/mod_event_multicast
>> event_handlers/mod_event_socket
>> #event_handlers/mod_event_zmq
>> #event_handlers/mod_radius_cdr
>> event_handlers/mod_snmp
>> formats/mod_local_stream
>> formats/mod_native_file
>> #formats/mod_portaudio_stream
>> #formats/mod_shell_stream
>> formats/mod_shout
>> formats/mod_sndfile
>> formats/mod_tone_stream
>> #formats/mod_vlc
>> #languages/mod_java
>> languages/mod_lua
>> #languages/mod_managed
>> languages/mod_perl
>> languages/mod_python
>> languages/mod_spidermonkey
>> #languages/mod_yaml
>> loggers/mod_console
>> loggers/mod_logfile
>> loggers/mod_syslog
>> #say/mod_say_de
>> say/mod_say_en
>> #say/mod_say_es
>> #say/mod_say_fr
>> #say/mod_say_he
>> #say/mod_say_hu
>> #say/mod_say_it
>> #say/mod_say_nl
>> #say/mod_say_pt
>> #say/mod_say_ru
>> #say/mod_say_th
>> #say/mod_say_zh
>> #timers/mod_posix_timer
>> #timers/mod_timerfd
>> xml_int/mod_xml_cdr
>> xml_int/mod_xml_curl
>> #xml_int/mod_xml_ldap
>> xml_int/mod_xml_rpc
>> xml_int/mod_xml_scgi
>>  
>> #../../libs/freetdm/mod_freetdm
>> #../../libs/openzap/mod_openzap
>>  
>> ## Experimental Modules (don't cry if they're broken)
>> #../../contrib/mod/xml_int/mod_xml_odbc
>>  
>>  
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From sdevoy at bizfocused.com  Wed Dec 19 01:57:45 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 18 Dec 2012 17:57:45 -0500
Subject: [Freeswitch-users] New install 1.2.Stable blows up compiling
	mod_perl.so
In-Reply-To: <CCF64C39.62413%krice@freeswitch.org>
References: <5e0401cddd6e$57c742b0$0755c810$@bizfocused.com>
	<CCF64C39.62413%krice@freeswitch.org>
Message-ID: <5e4e01cddd73$1855b4e0$49011ea0$@bizfocused.com>

Thanks.  That is EXACTLY what I am trying to follow.  I am thrown by "There
were no issues during installation and testing other than EPEL
requirements."

And "Installation of Git required adding the EPEL repository to obtain Git.
For EPEL information visit the EPEL <http://fedoraproject.org/wiki/EPEL>
site, The EPEL repository is added with 1 click."

 

I cannot understand how to install EPEL or how to click with my SSH window.

 

Anyone that can help me with that?

 

I did not install the "optional packages" so I will now. And try again.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Tuesday, December 18, 2012 5:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

 

If you installed on a centos base install then follow the directions on the
wiki at 
http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS

That usually avoids all these errors...  I usually do something like 

 yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig unixODBC-devel openssl-devel libogg-devel
libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel
expat-devel zlib zlib-devel bzip2 which

You may of course need to add a few extra RPMs if you are compiling things
that you have to uncomment in modules.conf

 



On 12/18/12 4:23 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

Thank you!
That resulted in a message saying I needed to install bison, so I ran yum
install bison.
 
That resulted in shit though:
gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
 gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
libtool: link: `VbrTag.lo' is not a valid libtool object
make[9]: *** [libmp3lame.la] Error 1
make[8]: *** [all-recursive] Error 1
Making all in frontend
libtool: link: `VbrTag.lo' is not a valid libtool object
make[9]: *** [libmp3lame.la] Error 1
make[8]: *** [all-recursive] Error 1
Making all in frontend
make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.
Making all in Dll
Making all in debian
Making all in doc
Making all in html
Making all in man
Making all in include
Making all in misc
Making all in dshow
Making all in ACM
Making all in ADbg
Making all in ddk
make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.
Making all in Dll
Making all in tinyxml
Making all in debian
Making all in doc
Making all in mac
Making all in macosx
Making all in html
Making all in man
Making all in English.lproj
Making all in include
Making all in LAME.xcodeproj
Making all in misc
Making all in dshow
Making all in vc_solution
Making all in ACM
make[7]: *** [all-recursive] Error 1
make[6]: *** [all] Error 2
make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2
make[4]: *** [install] Error 1
make[3]: *** [mod_shout-install] Error 1
Making all in ADbg
Making all in ddk
Making all in tinyxml
Making all in mac
Making all in macosx
Making all in English.lproj
Making all in LAME.xcodeproj
Making all in vc_solution
make[7]: *** [all-recursive] Error 1
make[6]: *** [all] Error 2
make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2
make[4]: *** [all] Error 1
make[3]: *** [mod_shout-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
make[2]: *** [install-recursive] Error 1
make[1]: *** [install-recursive] Error 1
make: *** [install] Error 2
[1]+  Exit 2                  make
 
Can't we get the INSTALL page of the wiki right?
 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abaci
Sent: Tuesday, December 18, 2012 5:01 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

yum install gdbm-devel db4-devel
 
On 12/18/2012 4:52 PM, Sean Devoy wrote:

I just started a clean install of the stable tree on a brand new install of
Centos 5.3.
Here is EXACTLY what I have done so far:
Logged in to root
Cd /usr/local/src
yum install gcc
yum -y install zlib-devel openssl-devel cpio expat-devel gettext-devel
cd /usr/local/src
wget http://git-core.googlecode.com/files/git-1.7.9.tar.gz
tar xvzf git-1.7.9.tar.gz
cd git-1.7.9
./configure
make
make install
yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig
 
cd /usr/local/src
git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
./bootstrap.sh    
./configure
tftp modules.conf from existing centos/fs server   (did this kill it)?  If
so how can I get it back?
make & make install  
 
after about 20 minutes the output is:
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c  -fPIC -DPIC
-o .libs/mod_perl.o
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c -o
mod_perl.o >/dev/null 2>&1
Creating mod_perl.so...
/usr/bin/ld: cannot find -lgdbm
collect2: ld returned 1 exit status
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E -Wl,-rpath
-Wl,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE freeswitch_perl.o
mod_perl_wrap.o perlxsi.o  /usr/local/src/freeswitch/.libs/libfreeswitch.so
-L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib
/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid
-L/usr/local/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lrt -lssl
-lcrypto -lz -lncurses -ljpeg -lodbc -L/usr/local/lib
/usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a
-L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -lperl -lresolv -lnsl
-lgdbm -ldb -ldl -lm -lcrypt -lutil -lpthread -lc   -Wl,--rpath
-Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_perl.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_perl-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
 
 
The modules.conf I upload is:
#applications/mod_abstraction
#applications/mod_avmd
applications/mod_blacklist
applications/mod_callcenter
applications/mod_cidlookup
applications/mod_cluechoo
applications/mod_commands
applications/mod_conference
applications/mod_curl
applications/mod_db
applications/mod_directory
#applications/mod_distributor
applications/mod_dptools
#applications/mod_easyroute
applications/mod_enum
applications/mod_esf
#applications/mod_esl
applications/mod_expr
applications/mod_fifo
#applications/mod_fsk
applications/mod_fsv
applications/mod_hash
applications/mod_httapi
#applications/mod_http_cache
#applications/mod_ladspa
applications/mod_lcr
#applications/mod_memcache
#applications/mod_mongo
#applications/mod_nibblebill
#applications/mod_osp
#applications/mod_redis
#applications/mod_rss
applications/mod_sms
#applications/mod_snapshot
#applications/mod_snipe_hunt
#applications/mod_snom
#applications/mod_soundtouch
applications/mod_spandsp
#applications/mod_spy
#applications/mod_stress
applications/mod_valet_parking
#applications/mod_vmd
applications/mod_voicemail
applications/mod_voicemail_ivr
#applications/mod_random
#asr_tts/mod_cepstral
asr_tts/mod_flite
asr_tts/mod_pocketsphinx
asr_tts/mod_tts_commandline
#asr_tts/mod_unimrcp
codecs/mod_amr
#codecs/mod_amrwb
codecs/mod_bv
#codecs/mod_celt
#codecs/mod_codec2
#codecs/mod_com_g729
#codecs/mod_dahdi_codec
codecs/mod_g723_1
codecs/mod_g729
codecs/mod_h26x
#codecs/mod_ilbc
#codecs/mod_isac
#codecs/mod_opus
#codecs/mod_sangoma_codec
#codecs/mod_silk
#codecs/mod_siren
codecs/mod_speex
dialplans/mod_dialplan_asterisk
#dialplans/mod_dialplan_directory
dialplans/mod_dialplan_xml
#directories/mod_ldap
#endpoints/mod_alsa
endpoints/mod_dingaling
#endpoints/mod_h323
#endpoints/mod_khomp
endpoints/mod_loopback
#endpoints/mod_opal
#endpoints/mod_portaudio
endpoints/mod_rtmp
#endpoints/mod_skinny
#endpoints/mod_skypopen
endpoints/mod_sofia
event_handlers/mod_cdr_csv
#event_handlers/mod_cdr_mongodb
#event_handlers/mod_cdr_pg_csv
event_handlers/mod_cdr_sqlite
#event_handlers/mod_erlang_event
#event_handlers/mod_event_multicast
event_handlers/mod_event_socket
#event_handlers/mod_event_zmq
#event_handlers/mod_radius_cdr
event_handlers/mod_snmp
formats/mod_local_stream
formats/mod_native_file
#formats/mod_portaudio_stream
#formats/mod_shell_stream
formats/mod_shout
formats/mod_sndfile
formats/mod_tone_stream
#formats/mod_vlc
#languages/mod_java
languages/mod_lua
#languages/mod_managed
languages/mod_perl
languages/mod_python
languages/mod_spidermonkey
#languages/mod_yaml
loggers/mod_console
loggers/mod_logfile
loggers/mod_syslog
#say/mod_say_de
say/mod_say_en
#say/mod_say_es
#say/mod_say_fr
#say/mod_say_he
#say/mod_say_hu
#say/mod_say_it
#say/mod_say_nl
#say/mod_say_pt
#say/mod_say_ru
#say/mod_say_th
#say/mod_say_zh
#timers/mod_posix_timer
#timers/mod_timerfd
xml_int/mod_xml_cdr
xml_int/mod_xml_curl
#xml_int/mod_xml_ldap
xml_int/mod_xml_rpc
xml_int/mod_xml_scgi
 
#../../libs/freetdm/mod_freetdm
#../../libs/openzap/mod_openzap
 
## Experimental Modules (don't cry if they're broken)
#../../contrib/mod/xml_int/mod_xml_odbc
 




_________________________________________________________________________
Professional FreeSWITCH Consulting 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

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


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From sdevoy at bizfocused.com  Wed Dec 19 02:00:55 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 18 Dec 2012 18:00:55 -0500
Subject: [Freeswitch-users] New install 1.2.Stable blows up compiling
	mod_perl.so
In-Reply-To: <CCF64C39.62413%krice@freeswitch.org>
References: <5e0401cddd6e$57c742b0$0755c810$@bizfocused.com>
	<CCF64C39.62413%krice@freeswitch.org>
Message-ID: <5e5301cddd73$8989cf20$9c9d6d60$@bizfocused.com>

Running your yum install . resulted in "Nothing to do".

 

Yet the make ends with:

/bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=compile
gcc -g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT
-D_GNU_SOURCE -D_LARGEFILE64_SOURCE   -I./include
-I/usr/local/src/freeswitch/libs/apr/include/arch/unix -I./include/arch/unix
-I/usr/local/src/freeswitch/libs/apr/include  -o user/unix/groupinfo.lo -c
user/unix/groupinfo.c && touch user/unix/groupinfo.lo

/bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=compile
gcc -g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT
-D_GNU_SOURCE -D_LARGEFILE64_SOURCE   -I./include
-I/usr/local/src/freeswitch/libs/apr/include/arch/unix -I./include/arch/unix
-I/usr/local/src/freeswitch/libs/apr/include  -o user/unix/groupinfo.lo -c
user/unix/groupinfo.c && touch user/unix/groupinfo.lo

/bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link  gcc
-g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
-D_LARGEFILE64_SOURCE   -I./include
-I/usr/local/src/freeswitch/libs/apr/include/arch/unix -I./include/arch/unix
-I/usr/local/src/freeswitch/libs/apr/include  -version-info 2:8:2    -o
libapr-1.la -rpath /usr/local/freeswitch/lib passwd/apr_getpass.lo
strings/apr_cpystrn.lo strings/apr_strnatcmp.lo strings/apr_strings.lo
strings/apr_strtok.lo strings/apr_fnmatch.lo strings/apr_snprintf.lo
tables/apr_tables.lo tables/apr_hash.lo atomic/unix/apr_atomic.lo
dso/unix/dso.lo file_io/unix/flock.lo file_io/unix/readwrite.lo
file_io/unix/filepath_util.lo file_io/unix/seek.lo file_io/unix/dir.lo
file_io/unix/mktemp.lo file_io/unix/filedup.lo file_io/unix/tempdir.lo
file_io/unix/filepath.lo file_io/unix/pipe.lo file_io/unix/open.lo
file_io/unix/filestat.lo file_io/unix/copy.lo file_io/unix/fileacc.lo
file_io/unix/fullrw.lo locks/unix/thread_rwlock.lo
locks/unix/thread_mutex.lo locks/unix/thread_cond.lo
locks/unix/proc_mutex.lo locks/unix/global_mutex.lo memory/unix/apr_pools.lo
misc/unix/charset.lo misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo
misc/unix/start.lo misc/unix/errorcodes.lo misc/unix/getopt.lo
misc/unix/otherchild.lo mmap/unix/mmap.lo mmap/unix/common.lo
network_io/unix/sockaddr.lo network_io/unix/sockopt.lo
network_io/unix/sendrecv.lo network_io/unix/multicast.lo
network_io/unix/sockets.lo network_io/unix/inet_ntop.lo
network_io/unix/inet_pton.lo poll/unix/epoll.lo poll/unix/select.lo
poll/unix/poll.lo poll/unix/port.lo poll/unix/kqueue.lo random/unix/sha2.lo
random/unix/apr_random.lo random/unix/sha2_glue.lo shmem/unix/shm.lo
support/unix/waitio.lo threadproc/unix/procsup.lo threadproc/unix/thread.lo
threadproc/unix/signals.lo threadproc/unix/proc.lo
threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
-lpthread

libtool: link: `strings/apr_strnatcmp.lo' is not a valid libtool object

make[2]: *** [libapr-1.la] Error 1

make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make[1]: *** [all-recursive] Error 1

make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make: *** [libs/apr/libapr-1.la] Error 2

/bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link  gcc
-g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
-D_LARGEFILE64_SOURCE   -I./include
-I/usr/local/src/freeswitch/libs/apr/include/arch/unix -I./include/arch/unix
-I/usr/local/src/freeswitch/libs/apr/include  -version-info 2:8:2    -o
libapr-1.la -rpath /usr/local/freeswitch/lib passwd/apr_getpass.lo
strings/apr_cpystrn.lo strings/apr_strnatcmp.lo strings/apr_strings.lo
strings/apr_strtok.lo strings/apr_fnmatch.lo strings/apr_snprintf.lo
tables/apr_tables.lo tables/apr_hash.lo atomic/unix/apr_atomic.lo
dso/unix/dso.lo file_io/unix/flock.lo file_io/unix/readwrite.lo
file_io/unix/filepath_util.lo file_io/unix/seek.lo file_io/unix/dir.lo
file_io/unix/mktemp.lo file_io/unix/filedup.lo file_io/unix/tempdir.lo
file_io/unix/filepath.lo file_io/unix/pipe.lo file_io/unix/open.lo
file_io/unix/filestat.lo file_io/unix/copy.lo file_io/unix/fileacc.lo
file_io/unix/fullrw.lo locks/unix/thread_rwlock.lo
locks/unix/thread_mutex.lo locks/unix/thread_cond.lo
locks/unix/proc_mutex.lo locks/unix/global_mutex.lo memory/unix/apr_pools.lo
misc/unix/charset.lo misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo
misc/unix/start.lo misc/unix/errorcodes.lo misc/unix/getopt.lo
misc/unix/otherchild.lo mmap/unix/mmap.lo mmap/unix/common.lo
network_io/unix/sockaddr.lo network_io/unix/sockopt.lo
network_io/unix/sendrecv.lo network_io/unix/multicast.lo
network_io/unix/sockets.lo network_io/unix/inet_ntop.lo
network_io/unix/inet_pton.lo poll/unix/epoll.lo poll/unix/select.lo
poll/unix/poll.lo poll/unix/port.lo poll/unix/kqueue.lo random/unix/sha2.lo
random/unix/apr_random.lo random/unix/sha2_glue.lo shmem/unix/shm.lo
support/unix/waitio.lo threadproc/unix/procsup.lo threadproc/unix/thread.lo
threadproc/unix/signals.lo threadproc/unix/proc.lo
threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
-lpthread

libtool: link: `strings/apr_strnatcmp.lo' is not a valid libtool object

make[2]: *** [libapr-1.la] Error 1

make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make[1]: *** [all-recursive] Error 1

make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make: *** [libs/apr/libapr-1.la] Error 2

[1]+  Exit 2                  make

 

I am missing some prereq or the EPEL is critical and I don't know how to add
it.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Tuesday, December 18, 2012 5:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

 

If you installed on a centos base install then follow the directions on the
wiki at 
http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS

That usually avoids all these errors...  I usually do something like 

 yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig unixODBC-devel openssl-devel libogg-devel
libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel
expat-devel zlib zlib-devel bzip2 which

You may of course need to add a few extra RPMs if you are compiling things
that you have to uncomment in modules.conf

 



On 12/18/12 4:23 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

Thank you!
That resulted in a message saying I needed to install bison, so I ran yum
install bison.
 
That resulted in shit though:
gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
 gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
libtool: link: `VbrTag.lo' is not a valid libtool object
make[9]: *** [libmp3lame.la] Error 1
make[8]: *** [all-recursive] Error 1
Making all in frontend
libtool: link: `VbrTag.lo' is not a valid libtool object
make[9]: *** [libmp3lame.la] Error 1
make[8]: *** [all-recursive] Error 1
Making all in frontend
make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.
Making all in Dll
Making all in debian
Making all in doc
Making all in html
Making all in man
Making all in include
Making all in misc
Making all in dshow
Making all in ACM
Making all in ADbg
Making all in ddk
make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.
Making all in Dll
Making all in tinyxml
Making all in debian
Making all in doc
Making all in mac
Making all in macosx
Making all in html
Making all in man
Making all in English.lproj
Making all in include
Making all in LAME.xcodeproj
Making all in misc
Making all in dshow
Making all in vc_solution
Making all in ACM
make[7]: *** [all-recursive] Error 1
make[6]: *** [all] Error 2
make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2
make[4]: *** [install] Error 1
make[3]: *** [mod_shout-install] Error 1
Making all in ADbg
Making all in ddk
Making all in tinyxml
Making all in mac
Making all in macosx
Making all in English.lproj
Making all in LAME.xcodeproj
Making all in vc_solution
make[7]: *** [all-recursive] Error 1
make[6]: *** [all] Error 2
make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2
make[4]: *** [all] Error 1
make[3]: *** [mod_shout-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
make[2]: *** [install-recursive] Error 1
make[1]: *** [install-recursive] Error 1
make: *** [install] Error 2
[1]+  Exit 2                  make
 
Can't we get the INSTALL page of the wiki right?
 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abaci
Sent: Tuesday, December 18, 2012 5:01 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

yum install gdbm-devel db4-devel
 
On 12/18/2012 4:52 PM, Sean Devoy wrote:

I just started a clean install of the stable tree on a brand new install of
Centos 5.3.
Here is EXACTLY what I have done so far:
Logged in to root
Cd /usr/local/src
yum install gcc
yum -y install zlib-devel openssl-devel cpio expat-devel gettext-devel
cd /usr/local/src
wget http://git-core.googlecode.com/files/git-1.7.9.tar.gz
tar xvzf git-1.7.9.tar.gz
cd git-1.7.9
./configure
make
make install
yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig
 
cd /usr/local/src
git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
./bootstrap.sh    
./configure
tftp modules.conf from existing centos/fs server   (did this kill it)?  If
so how can I get it back?
make & make install  
 
after about 20 minutes the output is:
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c  -fPIC -DPIC
-o .libs/mod_perl.o
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c -o
mod_perl.o >/dev/null 2>&1
Creating mod_perl.so...
/usr/bin/ld: cannot find -lgdbm
collect2: ld returned 1 exit status
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E -Wl,-rpath
-Wl,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE freeswitch_perl.o
mod_perl_wrap.o perlxsi.o  /usr/local/src/freeswitch/.libs/libfreeswitch.so
-L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib
/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid
-L/usr/local/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lrt -lssl
-lcrypto -lz -lncurses -ljpeg -lodbc -L/usr/local/lib
/usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a
-L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -lperl -lresolv -lnsl
-lgdbm -ldb -ldl -lm -lcrypt -lutil -lpthread -lc   -Wl,--rpath
-Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_perl.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_perl-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
 
 
The modules.conf I upload is:
#applications/mod_abstraction
#applications/mod_avmd
applications/mod_blacklist
applications/mod_callcenter
applications/mod_cidlookup
applications/mod_cluechoo
applications/mod_commands
applications/mod_conference
applications/mod_curl
applications/mod_db
applications/mod_directory
#applications/mod_distributor
applications/mod_dptools
#applications/mod_easyroute
applications/mod_enum
applications/mod_esf
#applications/mod_esl
applications/mod_expr
applications/mod_fifo
#applications/mod_fsk
applications/mod_fsv
applications/mod_hash
applications/mod_httapi
#applications/mod_http_cache
#applications/mod_ladspa
applications/mod_lcr
#applications/mod_memcache
#applications/mod_mongo
#applications/mod_nibblebill
#applications/mod_osp
#applications/mod_redis
#applications/mod_rss
applications/mod_sms
#applications/mod_snapshot
#applications/mod_snipe_hunt
#applications/mod_snom
#applications/mod_soundtouch
applications/mod_spandsp
#applications/mod_spy
#applications/mod_stress
applications/mod_valet_parking
#applications/mod_vmd
applications/mod_voicemail
applications/mod_voicemail_ivr
#applications/mod_random
#asr_tts/mod_cepstral
asr_tts/mod_flite
asr_tts/mod_pocketsphinx
asr_tts/mod_tts_commandline
#asr_tts/mod_unimrcp
codecs/mod_amr
#codecs/mod_amrwb
codecs/mod_bv
#codecs/mod_celt
#codecs/mod_codec2
#codecs/mod_com_g729
#codecs/mod_dahdi_codec
codecs/mod_g723_1
codecs/mod_g729
codecs/mod_h26x
#codecs/mod_ilbc
#codecs/mod_isac
#codecs/mod_opus
#codecs/mod_sangoma_codec
#codecs/mod_silk
#codecs/mod_siren
codecs/mod_speex
dialplans/mod_dialplan_asterisk
#dialplans/mod_dialplan_directory
dialplans/mod_dialplan_xml
#directories/mod_ldap
#endpoints/mod_alsa
endpoints/mod_dingaling
#endpoints/mod_h323
#endpoints/mod_khomp
endpoints/mod_loopback
#endpoints/mod_opal
#endpoints/mod_portaudio
endpoints/mod_rtmp
#endpoints/mod_skinny
#endpoints/mod_skypopen
endpoints/mod_sofia
event_handlers/mod_cdr_csv
#event_handlers/mod_cdr_mongodb
#event_handlers/mod_cdr_pg_csv
event_handlers/mod_cdr_sqlite
#event_handlers/mod_erlang_event
#event_handlers/mod_event_multicast
event_handlers/mod_event_socket
#event_handlers/mod_event_zmq
#event_handlers/mod_radius_cdr
event_handlers/mod_snmp
formats/mod_local_stream
formats/mod_native_file
#formats/mod_portaudio_stream
#formats/mod_shell_stream
formats/mod_shout
formats/mod_sndfile
formats/mod_tone_stream
#formats/mod_vlc
#languages/mod_java
languages/mod_lua
#languages/mod_managed
languages/mod_perl
languages/mod_python
languages/mod_spidermonkey
#languages/mod_yaml
loggers/mod_console
loggers/mod_logfile
loggers/mod_syslog
#say/mod_say_de
say/mod_say_en
#say/mod_say_es
#say/mod_say_fr
#say/mod_say_he
#say/mod_say_hu
#say/mod_say_it
#say/mod_say_nl
#say/mod_say_pt
#say/mod_say_ru
#say/mod_say_th
#say/mod_say_zh
#timers/mod_posix_timer
#timers/mod_timerfd
xml_int/mod_xml_cdr
xml_int/mod_xml_curl
#xml_int/mod_xml_ldap
xml_int/mod_xml_rpc
xml_int/mod_xml_scgi
 
#../../libs/freetdm/mod_freetdm
#../../libs/openzap/mod_openzap
 
## Experimental Modules (don't cry if they're broken)
#../../contrib/mod/xml_int/mod_xml_odbc
 




_________________________________________________________________________
Professional FreeSWITCH Consulting 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

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


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From sdevoy at bizfocused.com  Wed Dec 19 02:02:34 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 18 Dec 2012 18:02:34 -0500
Subject: [Freeswitch-users] New install 1.2.Stable blows up
	compiling	mod_perl.so
In-Reply-To: <5e4e01cddd73$1855b4e0$49011ea0$@bizfocused.com>
References: <5e0401cddd6e$57c742b0$0755c810$@bizfocused.com>	<CCF64C39.62413%krice@freeswitch.org>
	<5e4e01cddd73$1855b4e0$49011ea0$@bizfocused.com>
Message-ID: <5e6d01cddd73$c42fb1d0$4c8f1570$@bizfocused.com>

Sorry, I should have mentioned I deleted the /usr/local/src/freswitch
treeand started over with:

 

cd /usr/local/src
git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git



 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Devoy
Sent: Tuesday, December 18, 2012 5:58 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

 

Thanks.  That is EXACTLY what I am trying to follow.  I am thrown by "There
were no issues during installation and testing other than EPEL
requirements."

And "Installation of Git required adding the EPEL repository to obtain Git.
For EPEL information visit the EPEL <http://fedoraproject.org/wiki/EPEL>
site, The EPEL repository is added with 1 click."

 

I cannot understand how to install EPEL or how to click with my SSH window.

 

Anyone that can help me with that?

 

I did not install the "optional packages" so I will now. And try again.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Tuesday, December 18, 2012 5:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

 

If you installed on a centos base install then follow the directions on the
wiki at 
http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS

That usually avoids all these errors...  I usually do something like 

 yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig unixODBC-devel openssl-devel libogg-devel
libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel
expat-devel zlib zlib-devel bzip2 which

You may of course need to add a few extra RPMs if you are compiling things
that you have to uncomment in modules.conf

 



On 12/18/12 4:23 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

Thank you!
That resulted in a message saying I needed to install bison, so I ran yum
install bison.
 
That resulted in shit though:
gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
 gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
libtool: link: `VbrTag.lo' is not a valid libtool object
make[9]: *** [libmp3lame.la] Error 1
make[8]: *** [all-recursive] Error 1
Making all in frontend
libtool: link: `VbrTag.lo' is not a valid libtool object
make[9]: *** [libmp3lame.la] Error 1
make[8]: *** [all-recursive] Error 1
Making all in frontend
make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.
Making all in Dll
Making all in debian
Making all in doc
Making all in html
Making all in man
Making all in include
Making all in misc
Making all in dshow
Making all in ACM
Making all in ADbg
Making all in ddk
make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.
Making all in Dll
Making all in tinyxml
Making all in debian
Making all in doc
Making all in mac
Making all in macosx
Making all in html
Making all in man
Making all in English.lproj
Making all in include
Making all in LAME.xcodeproj
Making all in misc
Making all in dshow
Making all in vc_solution
Making all in ACM
make[7]: *** [all-recursive] Error 1
make[6]: *** [all] Error 2
make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2
make[4]: *** [install] Error 1
make[3]: *** [mod_shout-install] Error 1
Making all in ADbg
Making all in ddk
Making all in tinyxml
Making all in mac
Making all in macosx
Making all in English.lproj
Making all in LAME.xcodeproj
Making all in vc_solution
make[7]: *** [all-recursive] Error 1
make[6]: *** [all] Error 2
make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2
make[4]: *** [all] Error 1
make[3]: *** [mod_shout-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
make[2]: *** [install-recursive] Error 1
make[1]: *** [install-recursive] Error 1
make: *** [install] Error 2
[1]+  Exit 2                  make
 
Can't we get the INSTALL page of the wiki right?
 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abaci
Sent: Tuesday, December 18, 2012 5:01 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

yum install gdbm-devel db4-devel
 
On 12/18/2012 4:52 PM, Sean Devoy wrote:

I just started a clean install of the stable tree on a brand new install of
Centos 5.3.
Here is EXACTLY what I have done so far:
Logged in to root
Cd /usr/local/src
yum install gcc
yum -y install zlib-devel openssl-devel cpio expat-devel gettext-devel
cd /usr/local/src
wget http://git-core.googlecode.com/files/git-1.7.9.tar.gz
tar xvzf git-1.7.9.tar.gz
cd git-1.7.9
./configure
make
make install
yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig
 
cd /usr/local/src
git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
./bootstrap.sh    
./configure
tftp modules.conf from existing centos/fs server   (did this kill it)?  If
so how can I get it back?
make & make install  
 
after about 20 minutes the output is:
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c  -fPIC -DPIC
-o .libs/mod_perl.o
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c -o
mod_perl.o >/dev/null 2>&1
Creating mod_perl.so...
/usr/bin/ld: cannot find -lgdbm
collect2: ld returned 1 exit status
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E -Wl,-rpath
-Wl,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE freeswitch_perl.o
mod_perl_wrap.o perlxsi.o  /usr/local/src/freeswitch/.libs/libfreeswitch.so
-L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib
/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid
-L/usr/local/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lrt -lssl
-lcrypto -lz -lncurses -ljpeg -lodbc -L/usr/local/lib
/usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a
-L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -lperl -lresolv -lnsl
-lgdbm -ldb -ldl -lm -lcrypt -lutil -lpthread -lc   -Wl,--rpath
-Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_perl.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_perl-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
 
 
The modules.conf I upload is:
#applications/mod_abstraction
#applications/mod_avmd
applications/mod_blacklist
applications/mod_callcenter
applications/mod_cidlookup
applications/mod_cluechoo
applications/mod_commands
applications/mod_conference
applications/mod_curl
applications/mod_db
applications/mod_directory
#applications/mod_distributor
applications/mod_dptools
#applications/mod_easyroute
applications/mod_enum
applications/mod_esf
#applications/mod_esl
applications/mod_expr
applications/mod_fifo
#applications/mod_fsk
applications/mod_fsv
applications/mod_hash
applications/mod_httapi
#applications/mod_http_cache
#applications/mod_ladspa
applications/mod_lcr
#applications/mod_memcache
#applications/mod_mongo
#applications/mod_nibblebill
#applications/mod_osp
#applications/mod_redis
#applications/mod_rss
applications/mod_sms
#applications/mod_snapshot
#applications/mod_snipe_hunt
#applications/mod_snom
#applications/mod_soundtouch
applications/mod_spandsp
#applications/mod_spy
#applications/mod_stress
applications/mod_valet_parking
#applications/mod_vmd
applications/mod_voicemail
applications/mod_voicemail_ivr
#applications/mod_random
#asr_tts/mod_cepstral
asr_tts/mod_flite
asr_tts/mod_pocketsphinx
asr_tts/mod_tts_commandline
#asr_tts/mod_unimrcp
codecs/mod_amr
#codecs/mod_amrwb
codecs/mod_bv
#codecs/mod_celt
#codecs/mod_codec2
#codecs/mod_com_g729
#codecs/mod_dahdi_codec
codecs/mod_g723_1
codecs/mod_g729
codecs/mod_h26x
#codecs/mod_ilbc
#codecs/mod_isac
#codecs/mod_opus
#codecs/mod_sangoma_codec
#codecs/mod_silk
#codecs/mod_siren
codecs/mod_speex
dialplans/mod_dialplan_asterisk
#dialplans/mod_dialplan_directory
dialplans/mod_dialplan_xml
#directories/mod_ldap
#endpoints/mod_alsa
endpoints/mod_dingaling
#endpoints/mod_h323
#endpoints/mod_khomp
endpoints/mod_loopback
#endpoints/mod_opal
#endpoints/mod_portaudio
endpoints/mod_rtmp
#endpoints/mod_skinny
#endpoints/mod_skypopen
endpoints/mod_sofia
event_handlers/mod_cdr_csv
#event_handlers/mod_cdr_mongodb
#event_handlers/mod_cdr_pg_csv
event_handlers/mod_cdr_sqlite
#event_handlers/mod_erlang_event
#event_handlers/mod_event_multicast
event_handlers/mod_event_socket
#event_handlers/mod_event_zmq
#event_handlers/mod_radius_cdr
event_handlers/mod_snmp
formats/mod_local_stream
formats/mod_native_file
#formats/mod_portaudio_stream
#formats/mod_shell_stream
formats/mod_shout
formats/mod_sndfile
formats/mod_tone_stream
#formats/mod_vlc
#languages/mod_java
languages/mod_lua
#languages/mod_managed
languages/mod_perl
languages/mod_python
languages/mod_spidermonkey
#languages/mod_yaml
loggers/mod_console
loggers/mod_logfile
loggers/mod_syslog
#say/mod_say_de
say/mod_say_en
#say/mod_say_es
#say/mod_say_fr
#say/mod_say_he
#say/mod_say_hu
#say/mod_say_it
#say/mod_say_nl
#say/mod_say_pt
#say/mod_say_ru
#say/mod_say_th
#say/mod_say_zh
#timers/mod_posix_timer
#timers/mod_timerfd
xml_int/mod_xml_cdr
xml_int/mod_xml_curl
#xml_int/mod_xml_ldap
xml_int/mod_xml_rpc
xml_int/mod_xml_scgi
 
#../../libs/freetdm/mod_freetdm
#../../libs/openzap/mod_openzap
 
## Experimental Modules (don't cry if they're broken)
#../../contrib/mod/xml_int/mod_xml_odbc
 




_________________________________________________________________________
Professional FreeSWITCH Consulting 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

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


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From krice at freeswitch.org  Wed Dec 19 02:41:33 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 18 Dec 2012 17:41:33 -0600
Subject: [Freeswitch-users] New install 1.2.Stable blows up compiling
 mod_perl.so
In-Reply-To: <5e4e01cddd73$1855b4e0$49011ea0$@bizfocused.com>
Message-ID: <CCF65B4D.6241D%krice@freeswitch.org>

You probably don?t need epel unless you are installing on centos5... EPEL
just adds another yum REPO for getting more upto date packages...

If you look at the EPEL site, it tells you how to install it... Its usually
just add this RPM via yum install http://some.website/some/path/some.rpm



On 12/18/12 4:57 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> Thanks.  That is EXACTLY what I am trying to follow.  I am thrown by ?There
> were no issues during installation and testing other than EPEL requirements.?
> And ?Installation of Git required adding the EPEL repository to obtain Git.
> For EPEL information visit the EPEL <http://fedoraproject.org/wiki/EPEL>
> site, The EPEL repository is added with 1 click.?
>  
> I cannot understand how to install EPEL or how to click with my SSH window.
>  
> Anyone that can help me with that?
>  
> I did not install the ?optional packages? so I will now. And try again.
>  
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
> Sent: Tuesday, December 18, 2012 5:37 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
> mod_perl.so
>  
> If you installed on a centos base install then follow the directions on the
> wiki at 
> http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS
> 
> That usually avoids all these errors...  I usually do something like
> 
>  yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
> ncurses-devel pkgconfig unixODBC-devel openssl-devel libogg-devel
> libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel
> expat-devel zlib zlib-devel bzip2 which
> 
> You may of course need to add a few extra RPMs if you are compiling things
> that you have to uncomment in modules.conf
> 
>  
> 
> 
> 
> On 12/18/12 4:23 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:
> Thank you!
> That resulted in a message saying I needed to install bison, so I ran yum
> install bison.
>  
> That resulted in shit though:
> gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe -g
> -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
> mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
>  gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe -g
> -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
> mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
> libtool: link: `VbrTag.lo' is not a valid libtool object
> make[9]: *** [libmp3lame.la] Error 1
> make[8]: *** [all-recursive] Error 1
> Making all in frontend
> libtool: link: `VbrTag.lo' is not a valid libtool object
> make[9]: *** [libmp3lame.la] Error 1
> make[8]: *** [all-recursive] Error 1
> Making all in frontend
> make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
> `lame'.  Stop.
> Making all in Dll
> Making all in debian
> Making all in doc
> Making all in html
> Making all in man
> Making all in include
> Making all in misc
> Making all in dshow
> Making all in ACM
> Making all in ADbg
> Making all in ddk
> make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
> `lame'.  Stop.
> Making all in Dll
> Making all in tinyxml
> Making all in debian
> Making all in doc
> Making all in mac
> Making all in macosx
> Making all in html
> Making all in man
> Making all in English.lproj
> Making all in include
> Making all in LAME.xcodeproj
> Making all in misc
> Making all in dshow
> Making all in vc_solution
> Making all in ACM
> make[7]: *** [all-recursive] Error 1
> make[6]: *** [all] Error 2
> make[5]: *** 
> [/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error 2
> make[4]: *** [install] Error 1
> make[3]: *** [mod_shout-install] Error 1
> Making all in ADbg
> Making all in ddk
> Making all in tinyxml
> Making all in mac
> Making all in macosx
> Making all in English.lproj
> Making all in LAME.xcodeproj
> Making all in vc_solution
> make[7]: *** [all-recursive] Error 1
> make[6]: *** [all] Error 2
> make[5]: *** 
> [/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error 2
> make[4]: *** [all] Error 1
> make[3]: *** [mod_shout-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
> make[2]: *** [install-recursive] Error 1
> make[1]: *** [install-recursive] Error 1
> make: *** [install] Error 2
> [1]+  Exit 2                  make
>  
> Can?t we get the INSTALL page of the wiki right?
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abaci
> Sent: Tuesday, December 18, 2012 5:01 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
> mod_perl.so
> 
> yum install gdbm-devel db4-devel
>  
> On 12/18/2012 4:52 PM, Sean Devoy wrote:
> I just started a clean install of the stable tree on a brand new install of
> Centos 5.3.
> Here is EXACTLY what I have done so far:
> Logged in to root
> Cd /usr/local/src
> yum install gcc
> yum -y install zlib-devel openssl-devel cpio expat-devel gettext-devel
> cd /usr/local/src
> wget http://git-core.googlecode.com/files/git-1.7.9.tar.gz
> tar xvzf git-1.7.9.tar.gz
> cd git-1.7.9
> ./configure
> make
> make install
> yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
> ncurses-devel pkgconfig
>  
> cd /usr/local/src
> git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
> ./bootstrap.sh   
> ./configure
> tftp modules.conf from existing centos/fs server   (did this kill it)?  If so
> how can I get it back?
> make & make install
>  
> after about 20 minutes the output is:
> gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
> -Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
> -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
> -I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
> -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
> -I/usr/local/src/freeswitch/libs/stfu -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
> -DHAVE_CONFIG_H -c
> /usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c  -fPIC -DPIC
> -o .libs/mod_perl.o
> gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
> -Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
> -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
> -I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
> -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
> -I/usr/local/src/freeswitch/libs/stfu -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
> -DHAVE_CONFIG_H -c
> /usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c -o mod_perl.o
> >/dev/null 2>&1
> Creating mod_perl.so...
> /usr/bin/ld: cannot find -lgdbm
> collect2: ld returned 1 exit status
> gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
> -Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
> -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
> -I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
> -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
> -I/usr/local/src/freeswitch/libs/stfu -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_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E -Wl,-rpath
> -Wl,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE freeswitch_perl.o
> mod_perl_wrap.o perlxsi.o  /usr/local/src/freeswitch/.libs/libfreeswitch.so
> -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib
> /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
> /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid
> -L/usr/local/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lrt -lssl -lcrypto
> -lz -lncurses -ljpeg -lodbc -L/usr/local/lib
> /usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a
> -L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -lperl -lresolv -lnsl
> -lgdbm -ldb -ldl -lm -lcrypt -lutil -lpthread -lc   -Wl,--rpath
> -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod
> make[5]: *** [mod_perl.so] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_perl-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
>  
>  
> The modules.conf I upload is:
> #applications/mod_abstraction
> #applications/mod_avmd
> applications/mod_blacklist
> applications/mod_callcenter
> applications/mod_cidlookup
> applications/mod_cluechoo
> applications/mod_commands
> applications/mod_conference
> applications/mod_curl
> applications/mod_db
> applications/mod_directory
> #applications/mod_distributor
> applications/mod_dptools
> #applications/mod_easyroute
> applications/mod_enum
> applications/mod_esf
> #applications/mod_esl
> applications/mod_expr
> applications/mod_fifo
> #applications/mod_fsk
> applications/mod_fsv
> applications/mod_hash
> applications/mod_httapi
> #applications/mod_http_cache
> #applications/mod_ladspa
> applications/mod_lcr
> #applications/mod_memcache
> #applications/mod_mongo
> #applications/mod_nibblebill
> #applications/mod_osp
> #applications/mod_redis
> #applications/mod_rss
> applications/mod_sms
> #applications/mod_snapshot
> #applications/mod_snipe_hunt
> #applications/mod_snom
> #applications/mod_soundtouch
> applications/mod_spandsp
> #applications/mod_spy
> #applications/mod_stress
> applications/mod_valet_parking
> #applications/mod_vmd
> applications/mod_voicemail
> applications/mod_voicemail_ivr
> #applications/mod_random
> #asr_tts/mod_cepstral
> asr_tts/mod_flite
> asr_tts/mod_pocketsphinx
> asr_tts/mod_tts_commandline
> #asr_tts/mod_unimrcp
> codecs/mod_amr
> #codecs/mod_amrwb
> codecs/mod_bv
> #codecs/mod_celt
> #codecs/mod_codec2
> #codecs/mod_com_g729
> #codecs/mod_dahdi_codec
> codecs/mod_g723_1
> codecs/mod_g729
> codecs/mod_h26x
> #codecs/mod_ilbc
> #codecs/mod_isac
> #codecs/mod_opus
> #codecs/mod_sangoma_codec
> #codecs/mod_silk
> #codecs/mod_siren
> codecs/mod_speex
> dialplans/mod_dialplan_asterisk
> #dialplans/mod_dialplan_directory
> dialplans/mod_dialplan_xml
> #directories/mod_ldap
> #endpoints/mod_alsa
> endpoints/mod_dingaling
> #endpoints/mod_h323
> #endpoints/mod_khomp
> endpoints/mod_loopback
> #endpoints/mod_opal
> #endpoints/mod_portaudio
> endpoints/mod_rtmp
> #endpoints/mod_skinny
> #endpoints/mod_skypopen
> endpoints/mod_sofia
> event_handlers/mod_cdr_csv
> #event_handlers/mod_cdr_mongodb
> #event_handlers/mod_cdr_pg_csv
> event_handlers/mod_cdr_sqlite
> #event_handlers/mod_erlang_event
> #event_handlers/mod_event_multicast
> event_handlers/mod_event_socket
> #event_handlers/mod_event_zmq
> #event_handlers/mod_radius_cdr
> event_handlers/mod_snmp
> formats/mod_local_stream
> formats/mod_native_file
> #formats/mod_portaudio_stream
> #formats/mod_shell_stream
> formats/mod_shout
> formats/mod_sndfile
> formats/mod_tone_stream
> #formats/mod_vlc
> #languages/mod_java
> languages/mod_lua
> #languages/mod_managed
> languages/mod_perl
> languages/mod_python
> languages/mod_spidermonkey
> #languages/mod_yaml
> loggers/mod_console
> loggers/mod_logfile
> loggers/mod_syslog
> #say/mod_say_de
> say/mod_say_en
> #say/mod_say_es
> #say/mod_say_fr
> #say/mod_say_he
> #say/mod_say_hu
> #say/mod_say_it
> #say/mod_say_nl
> #say/mod_say_pt
> #say/mod_say_ru
> #say/mod_say_th
> #say/mod_say_zh
> #timers/mod_posix_timer
> #timers/mod_timerfd
> xml_int/mod_xml_cdr
> xml_int/mod_xml_curl
> #xml_int/mod_xml_ldap
> xml_int/mod_xml_rpc
> xml_int/mod_xml_scgi
>  
> #../../libs/freetdm/mod_freetdm
> #../../libs/openzap/mod_openzap
>  
> ## Experimental Modules (don't cry if they're broken)
> #../../contrib/mod/xml_int/mod_xml_odbc
>  
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch

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From krice at freeswitch.org  Wed Dec 19 02:43:19 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 18 Dec 2012 17:43:19 -0600
Subject: [Freeswitch-users] Do you have a Wishlist of someone you would like
 to see Do a presentation on the weekly conf call?
Message-ID: <CCF65BB7.6241D%krice@freeswitch.org>

Let us know who they are...

Email me here or off list so we can start getting the schedule nailed down..

K



-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From jrichey at itltd.net  Wed Dec 19 02:49:38 2012
From: jrichey at itltd.net (JRichey)
Date: Tue, 18 Dec 2012 15:49:38 -0800
Subject: [Freeswitch-users] New install 1.2.Stable blows up compiling
	mod_perl.so
Message-ID: <6ECAF1527329364583AB525CF34ABF950B31A743@ms.kallback.com>

The commands below will add the EPEL repo, just choose the right one for
your architecture.  If you have git already you shouldn't need EPEL, but you
may want it for ngrep, fail2ban, etc.
 
rpm -Uvh
http://dl.fedoraproject.org/pub/epel/5/x86_64/epel-release-5-4.noarch.rpm
<http://dl.fedoraproject.org/pub/epel/5/x86_64/epel-release-5-4.noarch.rpm> 
rpm -Uvh
http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm
<http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm> 
 

-----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, December 18, 2012 3:03 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so



Sorry, I should have mentioned I deleted the /usr/local/src/freswitch
treeand started over with:

 

cd /usr/local/src
git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git



 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Devoy
Sent: Tuesday, December 18, 2012 5:58 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

 

Thanks.  That is EXACTLY what I am trying to follow.  I am thrown by "There
were no issues during installation and testing other than EPEL
requirements."

And "Installation of Git required adding the EPEL repository to obtain Git.
For EPEL information visit the EPEL <http://fedoraproject.org/wiki/EPEL>
site, The EPEL repository is added with 1 click."

 

I cannot understand how to install EPEL or how to click with my SSH window.

 

Anyone that can help me with that?

 

I did not install the "optional packages" so I will now. And try again.

 

 

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 Ken
Rice
Sent: Tuesday, December 18, 2012 5:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

 

If you installed on a centos base install then follow the directions on the
wiki at 
http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS
<http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS> 

That usually avoids all these errors...  I usually do something like 

 yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig unixODBC-devel openssl-devel libogg-devel
libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel
expat-devel zlib zlib-devel bzip2 which

You may of course need to add a few extra RPMs if you are compiling things
that you have to uncomment in modules.conf

 



On 12/18/12 4:23 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

Thank you!
That resulted in a message saying I needed to install bison, so I ran yum
install bison.
 
That resulted in shit though:
gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
 gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
libtool: link: `VbrTag.lo' is not a valid libtool object
make[9]: *** [libmp3lame.la] Error 1
make[8]: *** [all-recursive] Error 1
Making all in frontend
libtool: link: `VbrTag.lo' is not a valid libtool object
make[9]: *** [libmp3lame.la] Error 1
make[8]: *** [all-recursive] Error 1
Making all in frontend
make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.
Making all in Dll
Making all in debian
Making all in doc
Making all in html
Making all in man
Making all in include
Making all in misc
Making all in dshow
Making all in ACM
Making all in ADbg
Making all in ddk
make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.
Making all in Dll
Making all in tinyxml
Making all in debian
Making all in doc
Making all in mac
Making all in macosx
Making all in html
Making all in man
Making all in English.lproj
Making all in include
Making all in LAME.xcodeproj
Making all in misc
Making all in dshow
Making all in vc_solution
Making all in ACM
make[7]: *** [all-recursive] Error 1
make[6]: *** [all] Error 2
make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2
make[4]: *** [install] Error 1
make[3]: *** [mod_shout-install] Error 1
Making all in ADbg
Making all in ddk
Making all in tinyxml
Making all in mac
Making all in macosx
Making all in English.lproj
Making all in LAME.xcodeproj
Making all in vc_solution
make[7]: *** [all-recursive] Error 1
make[6]: *** [all] Error 2
make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2
make[4]: *** [all] Error 1
make[3]: *** [mod_shout-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
make[2]: *** [install-recursive] Error 1
make[1]: *** [install-recursive] Error 1
make: *** [install] Error 2
[1]+  Exit 2                  make
 
Can't we get the INSTALL page of the wiki right?
 

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 Abaci
Sent: Tuesday, December 18, 2012 5:01 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

yum install gdbm-devel db4-devel
 
On 12/18/2012 4:52 PM, Sean Devoy wrote:

I just started a clean install of the stable tree on a brand new install of
Centos 5.3.
Here is EXACTLY what I have done so far:
Logged in to root
Cd /usr/local/src
yum install gcc
yum -y install zlib-devel openssl-devel cpio expat-devel gettext-devel
cd /usr/local/src
wget http://git-core.googlecode.com/files/git-1.7.9.tar.gz
<http://git-core.googlecode.com/files/git-1.7.9.tar.gz> 
tar xvzf git-1.7.9.tar.gz
cd git-1.7.9
./configure
make
make install
yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig
 
cd /usr/local/src
git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
./bootstrap.sh    
./configure
tftp modules.conf from existing centos/fs server   (did this kill it)?  If
so how can I get it back?
make & make install  
 
after about 20 minutes the output is:
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c  -fPIC -DPIC
-o .libs/mod_perl.o
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c -o
mod_perl.o >/dev/null 2>&1
Creating mod_perl.so...
/usr/bin/ld: cannot find -lgdbm
collect2: ld returned 1 exit status
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E -Wl,-rpath
-Wl,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE freeswitch_perl.o
mod_perl_wrap.o perlxsi.o  /usr/local/src/freeswitch/.libs/libfreeswitch.so
-L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib
/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid
-L/usr/local/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lrt -lssl
-lcrypto -lz -lncurses -ljpeg -lodbc -L/usr/local/lib
/usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a
-L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -lperl -lresolv -lnsl
-lgdbm -ldb -ldl -lm -lcrypt -lutil -lpthread -lc   -Wl,--rpath
-Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_perl.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_perl-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
 
 
The modules.conf I upload is:
#applications/mod_abstraction
#applications/mod_avmd
applications/mod_blacklist
applications/mod_callcenter
applications/mod_cidlookup
applications/mod_cluechoo
applications/mod_commands
applications/mod_conference
applications/mod_curl
applications/mod_db
applications/mod_directory
#applications/mod_distributor
applications/mod_dptools
#applications/mod_easyroute
applications/mod_enum
applications/mod_esf
#applications/mod_esl
applications/mod_expr
applications/mod_fifo
#applications/mod_fsk
applications/mod_fsv
applications/mod_hash
applications/mod_httapi
#applications/mod_http_cache
#applications/mod_ladspa
applications/mod_lcr
#applications/mod_memcache
#applications/mod_mongo
#applications/mod_nibblebill
#applications/mod_osp
#applications/mod_redis
#applications/mod_rss
applications/mod_sms
#applications/mod_snapshot
#applications/mod_snipe_hunt
#applications/mod_snom
#applications/mod_soundtouch
applications/mod_spandsp
#applications/mod_spy
#applications/mod_stress
applications/mod_valet_parking
#applications/mod_vmd
applications/mod_voicemail
applications/mod_voicemail_ivr
#applications/mod_random
#asr_tts/mod_cepstral
asr_tts/mod_flite
asr_tts/mod_pocketsphinx
asr_tts/mod_tts_commandline
#asr_tts/mod_unimrcp
codecs/mod_amr
#codecs/mod_amrwb
codecs/mod_bv
#codecs/mod_celt
#codecs/mod_codec2
#codecs/mod_com_g729
#codecs/mod_dahdi_codec
codecs/mod_g723_1
codecs/mod_g729
codecs/mod_h26x
#codecs/mod_ilbc
#codecs/mod_isac
#codecs/mod_opus
#codecs/mod_sangoma_codec
#codecs/mod_silk
#codecs/mod_siren
codecs/mod_speex
dialplans/mod_dialplan_asterisk
#dialplans/mod_dialplan_directory
dialplans/mod_dialplan_xml
#directories/mod_ldap
#endpoints/mod_alsa
endpoints/mod_dingaling
#endpoints/mod_h323
#endpoints/mod_khomp
endpoints/mod_loopback
#endpoints/mod_opal
#endpoints/mod_portaudio
endpoints/mod_rtmp
#endpoints/mod_skinny
#endpoints/mod_skypopen
endpoints/mod_sofia
event_handlers/mod_cdr_csv
#event_handlers/mod_cdr_mongodb
#event_handlers/mod_cdr_pg_csv
event_handlers/mod_cdr_sqlite
#event_handlers/mod_erlang_event
#event_handlers/mod_event_multicast
event_handlers/mod_event_socket
#event_handlers/mod_event_zmq
#event_handlers/mod_radius_cdr
event_handlers/mod_snmp
formats/mod_local_stream
formats/mod_native_file
#formats/mod_portaudio_stream
#formats/mod_shell_stream
formats/mod_shout
formats/mod_sndfile
formats/mod_tone_stream
#formats/mod_vlc
#languages/mod_java
languages/mod_lua
#languages/mod_managed
languages/mod_perl
languages/mod_python
languages/mod_spidermonkey
#languages/mod_yaml
loggers/mod_console
loggers/mod_logfile
loggers/mod_syslog
#say/mod_say_de
say/mod_say_en
#say/mod_say_es
#say/mod_say_fr
#say/mod_say_he
#say/mod_say_hu
#say/mod_say_it
#say/mod_say_nl
#say/mod_say_pt
#say/mod_say_ru
#say/mod_say_th
#say/mod_say_zh
#timers/mod_posix_timer
#timers/mod_timerfd
xml_int/mod_xml_cdr
xml_int/mod_xml_curl
#xml_int/mod_xml_ldap
xml_int/mod_xml_rpc
xml_int/mod_xml_scgi
 
#../../libs/freetdm/mod_freetdm
#../../libs/openzap/mod_openzap
 
## Experimental Modules (don't cry if they're broken)
#../../contrib/mod/xml_int/mod_xml_odbc
 




_________________________________________________________________________
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> 
 
FreeSWITCH-users mailing list
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:
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> 

FreeSWITCH-users mailing list
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> 


-- 
Ken
http://www.FreeSWITCH.org <http://www.FreeSWITCH.org> 
http://www.ClueCon.com <http://www.ClueCon.com> 
http://www.OSTAG.org <http://www.OSTAG.org> 
irc.freenode.net #freeswitch

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From mario_fs at mgtech.com  Wed Dec 19 02:51:35 2012
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 18 Dec 2012 15:51:35 -0800
Subject: [Freeswitch-users] New install 1.2.Stable blows up compiling
	mod_perl.so
In-Reply-To: <CCF65B4D.6241D%krice@freeswitch.org>
References: <CCF65B4D.6241D%krice@freeswitch.org>
Message-ID: <AEE3CADD-AB93-44A6-9D18-9DD2C883EBBF@mgtech.com>

FYI, I was the one that tested FS on CentOS (among about 5 other distros) since I was redoing the install wiki, and I remember EPEL was required on CentOS 5.5 at the time.
Mario G

On Dec 18, 2012, at 3:41 PM, Ken Rice wrote:

> You probably don?t need epel unless you are installing on centos5... EPEL just adds another yum REPO for getting more upto date packages...
> 
> If you look at the EPEL site, it tells you how to install it... Its usually just add this RPM via yum install http://some.website/some/path/some.rpm
> 
> 
> 
> On 12/18/12 4:57 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:
> 
>> Thanks.  That is EXACTLY what I am trying to follow.  I am thrown by ?There were no issues during installation and testing other than EPEL requirements.?
>> And ?Installation of Git required adding the EPEL repository to obtain Git. For EPEL information visit the EPEL <http://fedoraproject.org/wiki/EPEL>  site, The EPEL repository is added with 1 click.?
>>  
>> I cannot understand how to install EPEL or how to click with my SSH window.
>>  
>> Anyone that can help me with that?
>>  
>> I did not install the ?optional packages? so I will now. And try again.
>> 
>>  
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
>> Sent: Tuesday, December 18, 2012 5:37 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling mod_perl.so
>> 
>> If you installed on a centos base install then follow the directions on the wiki at 
>> http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS
>> 
>> That usually avoids all these errors...  I usually do something like 
>> 
>>  yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make ncurses-devel pkgconfig unixODBC-devel openssl-devel libogg-devel libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel expat-devel zlib zlib-devel bzip2 which
>> 
>> You may of course need to add a few extra RPMs if you are compiling things that you have to uncomment in modules.conf
>> 
>>  
>> 
>> 
>> 
>> On 12/18/12 4:23 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:
>> Thank you!
>> That resulted in a message saying I needed to install bison, so I ran yum install bison.
>>  
>> That resulted in shit though:
>> gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe -g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
>>  gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe -g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
>> libtool: link: `VbrTag.lo' is not a valid libtool object
>> make[9]: *** [libmp3lame.la] Error 1
>> make[8]: *** [all-recursive] Error 1
>> Making all in frontend
>> libtool: link: `VbrTag.lo' is not a valid libtool object
>> make[9]: *** [libmp3lame.la] Error 1
>> make[8]: *** [all-recursive] Error 1
>> Making all in frontend
>> make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by `lame'.  Stop.
>> Making all in Dll
>> Making all in debian
>> Making all in doc
>> Making all in html
>> Making all in man
>> Making all in include
>> Making all in misc
>> Making all in dshow
>> Making all in ACM
>> Making all in ADbg
>> Making all in ddk
>> make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by `lame'.  Stop.
>> Making all in Dll
>> Making all in tinyxml
>> Making all in debian
>> Making all in doc
>> Making all in mac
>> Making all in macosx
>> Making all in html
>> Making all in man
>> Making all in English.lproj
>> Making all in include
>> Making all in LAME.xcodeproj
>> Making all in misc
>> Making all in dshow
>> Making all in vc_solution
>> Making all in ACM
>> make[7]: *** [all-recursive] Error 1
>> make[6]: *** [all] Error 2
>> make[5]: *** [/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error 2
>> make[4]: *** [install] Error 1
>> make[3]: *** [mod_shout-install] Error 1
>> Making all in ADbg
>> Making all in ddk
>> Making all in tinyxml
>> Making all in mac
>> Making all in macosx
>> Making all in English.lproj
>> Making all in LAME.xcodeproj
>> Making all in vc_solution
>> make[7]: *** [all-recursive] Error 1
>> make[6]: *** [all] Error 2
>> make[5]: *** [/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error 2
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_shout-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>> make[1]: *** [all-recursive] Error 1
>> make: *** [all] Error 2
>> make[2]: *** [install-recursive] Error 1
>> make[1]: *** [install-recursive] Error 1
>> make: *** [install] Error 2
>> [1]+  Exit 2                  make
>>  
>> Can?t we get the INSTALL page of the wiki right?
>>  
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abaci
>> Sent: Tuesday, December 18, 2012 5:01 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling mod_perl.so
>> 
>> yum install gdbm-devel db4-devel
>>  
>> On 12/18/2012 4:52 PM, Sean Devoy wrote:
>> I just started a clean install of the stable tree on a brand new install of Centos 5.3.
>> Here is EXACTLY what I have done so far:
>> Logged in to root
>> Cd /usr/local/src
>> yum install gcc
>> yum -y install zlib-devel openssl-devel cpio expat-devel gettext-devel
>> cd /usr/local/src
>> wget http://git-core.googlecode.com/files/git-1.7.9.tar.gz
>> tar xvzf git-1.7.9.tar.gz
>> cd git-1.7.9
>> ./configure
>> make
>> make install
>> yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make ncurses-devel pkgconfig
>>  
>> cd /usr/local/src
>> git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
>> ./bootstrap.sh    
>> ./configure
>> tftp modules.conf from existing centos/fs server   (did this kill it)?  If so how can I get it back?
>> make & make install  
>>  
>> after about 20 minutes the output is:
>> gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe -Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm -I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL -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 -I/usr/local/src/freeswitch/libs/stfu -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 -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c  -fPIC -DPIC -o .libs/mod_perl.o
>> gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe -Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm -I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL -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 -I/usr/local/src/freeswitch/libs/stfu -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 -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c -o mod_perl.o >/dev/null 2>&1
>> Creating mod_perl.so...
>> /usr/bin/ld: cannot find -lgdbm
>> collect2: ld returned 1 exit status
>> gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe -Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm -I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL -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 -I/usr/local/src/freeswitch/libs/stfu -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_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E -Wl,-rpath -Wl,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE freeswitch_perl.o mod_perl_wrap.o perlxsi.o  /usr/local/src/freeswitch/.libs/libfreeswitch.so -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -L/usr/local/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lrt -lssl -lcrypto -lz -lncurses -ljpeg -lodbc -L/usr/local/lib /usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a -L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -lperl -lresolv -lnsl -lgdbm -ldb -ldl -lm -lcrypt -lutil -lpthread -lc   -Wl,--rpath -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod
>> make[5]: *** [mod_perl.so] Error 1
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_perl-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>> make[1]: *** [all-recursive] Error 1
>> make: *** [all] Error 2
>>  
>>  
>> The modules.conf I upload is:
>> #applications/mod_abstraction
>> #applications/mod_avmd
>> applications/mod_blacklist
>> applications/mod_callcenter
>> applications/mod_cidlookup
>> applications/mod_cluechoo
>> applications/mod_commands
>> applications/mod_conference
>> applications/mod_curl
>> applications/mod_db
>> applications/mod_directory
>> #applications/mod_distributor
>> applications/mod_dptools
>> #applications/mod_easyroute
>> applications/mod_enum
>> applications/mod_esf
>> #applications/mod_esl
>> applications/mod_expr
>> applications/mod_fifo
>> #applications/mod_fsk
>> applications/mod_fsv
>> applications/mod_hash
>> applications/mod_httapi
>> #applications/mod_http_cache
>> #applications/mod_ladspa
>> applications/mod_lcr
>> #applications/mod_memcache
>> #applications/mod_mongo
>> #applications/mod_nibblebill
>> #applications/mod_osp
>> #applications/mod_redis
>> #applications/mod_rss
>> applications/mod_sms
>> #applications/mod_snapshot
>> #applications/mod_snipe_hunt
>> #applications/mod_snom
>> #applications/mod_soundtouch
>> applications/mod_spandsp
>> #applications/mod_spy
>> #applications/mod_stress
>> applications/mod_valet_parking
>> #applications/mod_vmd
>> applications/mod_voicemail
>> applications/mod_voicemail_ivr
>> #applications/mod_random
>> #asr_tts/mod_cepstral
>> asr_tts/mod_flite
>> asr_tts/mod_pocketsphinx
>> asr_tts/mod_tts_commandline
>> #asr_tts/mod_unimrcp
>> codecs/mod_amr
>> #codecs/mod_amrwb
>> codecs/mod_bv
>> #codecs/mod_celt
>> #codecs/mod_codec2
>> #codecs/mod_com_g729
>> #codecs/mod_dahdi_codec
>> codecs/mod_g723_1
>> codecs/mod_g729
>> codecs/mod_h26x
>> #codecs/mod_ilbc
>> #codecs/mod_isac
>> #codecs/mod_opus
>> #codecs/mod_sangoma_codec
>> #codecs/mod_silk
>> #codecs/mod_siren
>> codecs/mod_speex
>> dialplans/mod_dialplan_asterisk
>> #dialplans/mod_dialplan_directory
>> dialplans/mod_dialplan_xml
>> #directories/mod_ldap
>> #endpoints/mod_alsa
>> endpoints/mod_dingaling
>> #endpoints/mod_h323
>> #endpoints/mod_khomp
>> endpoints/mod_loopback
>> #endpoints/mod_opal
>> #endpoints/mod_portaudio
>> endpoints/mod_rtmp
>> #endpoints/mod_skinny
>> #endpoints/mod_skypopen
>> endpoints/mod_sofia
>> event_handlers/mod_cdr_csv
>> #event_handlers/mod_cdr_mongodb
>> #event_handlers/mod_cdr_pg_csv
>> event_handlers/mod_cdr_sqlite
>> #event_handlers/mod_erlang_event
>> #event_handlers/mod_event_multicast
>> event_handlers/mod_event_socket
>> #event_handlers/mod_event_zmq
>> #event_handlers/mod_radius_cdr
>> event_handlers/mod_snmp
>> formats/mod_local_stream
>> formats/mod_native_file
>> #formats/mod_portaudio_stream
>> #formats/mod_shell_stream
>> formats/mod_shout
>> formats/mod_sndfile
>> formats/mod_tone_stream
>> #formats/mod_vlc
>> #languages/mod_java
>> languages/mod_lua
>> #languages/mod_managed
>> languages/mod_perl
>> languages/mod_python
>> languages/mod_spidermonkey
>> #languages/mod_yaml
>> loggers/mod_console
>> loggers/mod_logfile
>> loggers/mod_syslog
>> #say/mod_say_de
>> say/mod_say_en
>> #say/mod_say_es
>> #say/mod_say_fr
>> #say/mod_say_he
>> #say/mod_say_hu
>> #say/mod_say_it
>> #say/mod_say_nl
>> #say/mod_say_pt
>> #say/mod_say_ru
>> #say/mod_say_th
>> #say/mod_say_zh
>> #timers/mod_posix_timer
>> #timers/mod_timerfd
>> xml_int/mod_xml_cdr
>> xml_int/mod_xml_curl
>> #xml_int/mod_xml_ldap
>> xml_int/mod_xml_rpc
>> xml_int/mod_xml_scgi
>>  
>> #../../libs/freetdm/mod_freetdm
>> #../../libs/openzap/mod_openzap
>>  
>> ## Experimental Modules (don't cry if they're broken)
>> #../../contrib/mod/xml_int/mod_xml_odbc
>>  
>> 
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 19 03:32:31 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 18 Dec 2012 19:32:31 -0500
Subject: [Freeswitch-users] New install 1.2.Stable blows up
	compiling	mod_perl.so
In-Reply-To: <AEE3CADD-AB93-44A6-9D18-9DD2C883EBBF@mgtech.com>
References: <CCF65B4D.6241D%krice@freeswitch.org>
	<AEE3CADD-AB93-44A6-9D18-9DD2C883EBBF@mgtech.com>
Message-ID: <5eef01cddd80$551563f0$ff402bd0$@bizfocused.com>

What a total disaster!

 

I added the EPEL per JRichie - thank you.

rpm -Uvh
http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm

 

 

Then I decided to remove all the pre-reqs and reinstall.

I ran:

 yum remove autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig unixODBC-devel openssl-devel libogg-devel
libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel
expat-devel zlib zlib-devel bzip2 which

 

The results are of course totally disastrous.  I now have no yum or rpm.  I
looked up how to install them.  Yum says use rpm and rpm says use yum!

 

I can't even do a "cat /etc/redhat-release"  or "ls" => /bin/ls  command not
found.

 

Short of waiting for the ISP to get up tomorrow and completely
reprovisioning my Centos5, does anyone have any suggestions.

 

Is there a command to reinstall centos 5 w/o yum or rpm?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mario G
Sent: Tuesday, December 18, 2012 6:52 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

 

FYI, I was the one that tested FS on CentOS (among about 5 other distros)
since I was redoing the install wiki, and I remember EPEL was required on
CentOS 5.5 at the time.

Mario G

 

On Dec 18, 2012, at 3:41 PM, Ken Rice wrote:





You probably don't need epel unless you are installing on centos5... EPEL
just adds another yum REPO for getting more upto date packages...

If you look at the EPEL site, it tells you how to install it... Its usually
just add this RPM via yum install http://some.website/some/path/some.rpm



On 12/18/12 4:57 PM, "Sean Devoy" <sdevoy at bizfocused.com
<x-msg://19/sdevoy at bizfocused.com> > wrote:




Thanks.  That is EXACTLY what I am trying to follow.  I am thrown by "There
were no issues during installation and testing other than EPEL
requirements."
And "Installation of Git required adding the EPEL repository to obtain Git.
For EPEL information visit the EPEL <http://fedoraproject.org/wiki/EPEL>
site, The EPEL repository is added with 1 click."
 
I cannot understand how to install EPEL or how to click with my SSH window.
 
Anyone that can help me with that?
 
I did not install the "optional packages" so I will now. And try again.

 

From: freeswitch-users-bounces at lists.freeswitch.org
<x-msg://19/freeswitch-users-bounces at lists.freeswitch.org>
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Tuesday, December 18, 2012 5:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

If you installed on a centos base install then follow the directions on the
wiki at 
http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS

That usually avoids all these errors...  I usually do something like 

 yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig unixODBC-devel openssl-devel libogg-devel
libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel
expat-devel zlib zlib-devel bzip2 which

You may of course need to add a few extra RPMs if you are compiling things
that you have to uncomment in modules.conf

 



On 12/18/12 4:23 PM, "Sean Devoy" <sdevoy at bizfocused.com
<x-msg://19/sdevoy at bizfocused.com> > wrote:
Thank you!
That resulted in a message saying I needed to install bison, so I ran yum
install bison.
 
That resulted in shit though:
gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
 gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
libtool: link: `VbrTag.lo' is not a valid libtool object
make[9]: *** [libmp3lame.la] Error 1
make[8]: *** [all-recursive] Error 1
Making all in frontend
libtool: link: `VbrTag.lo' is not a valid libtool object
make[9]: *** [libmp3lame.la] Error 1
make[8]: *** [all-recursive] Error 1
Making all in frontend
make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.
Making all in Dll
Making all in debian
Making all in doc
Making all in html
Making all in man
Making all in include
Making all in misc
Making all in dshow
Making all in ACM
Making all in ADbg
Making all in ddk
make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.
Making all in Dll
Making all in tinyxml
Making all in debian
Making all in doc
Making all in mac
Making all in macosx
Making all in html
Making all in man
Making all in English.lproj
Making all in include
Making all in LAME.xcodeproj
Making all in misc
Making all in dshow
Making all in vc_solution
Making all in ACM
make[7]: *** [all-recursive] Error 1
make[6]: *** [all] Error 2
make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2
make[4]: *** [install] Error 1
make[3]: *** [mod_shout-install] Error 1
Making all in ADbg
Making all in ddk
Making all in tinyxml
Making all in mac
Making all in macosx
Making all in English.lproj
Making all in LAME.xcodeproj
Making all in vc_solution
make[7]: *** [all-recursive] Error 1
make[6]: *** [all] Error 2
make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2
make[4]: *** [all] Error 1
make[3]: *** [mod_shout-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
make[2]: *** [install-recursive] Error 1
make[1]: *** [install-recursive] Error 1
make: *** [install] Error 2
[1]+  Exit 2                  make
 
Can't we get the INSTALL page of the wiki right?
 

From: freeswitch-users-bounces at lists.freeswitch.org
<x-msg://19/freeswitch-users-bounces at lists.freeswitch.org>
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abaci
Sent: Tuesday, December 18, 2012 5:01 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

yum install gdbm-devel db4-devel
 
On 12/18/2012 4:52 PM, Sean Devoy wrote:
I just started a clean install of the stable tree on a brand new install of
Centos 5.3.
Here is EXACTLY what I have done so far:
Logged in to root
Cd /usr/local/src
yum install gcc
yum -y install zlib-devel openssl-devel cpio expat-devel gettext-devel
cd /usr/local/src
wget http://git-core.googlecode.com/files/git-1.7.9.tar.gz
tar xvzf git-1.7.9.tar.gz
cd git-1.7.9
./configure
make
make install
yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig
 
cd /usr/local/src
git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
./bootstrap.sh    
./configure
tftp modules.conf from existing centos/fs server   (did this kill it)?  If
so how can I get it back?
make & make install  
 
after about 20 minutes the output is:
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c  -fPIC -DPIC
-o .libs/mod_perl.o
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c -o
mod_perl.o >/dev/null 2>&1
Creating mod_perl.so...
/usr/bin/ld: cannot find -lgdbm
collect2: ld returned 1 exit status
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E -Wl,-rpath
-Wl,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE freeswitch_perl.o
mod_perl_wrap.o perlxsi.o  /usr/local/src/freeswitch/.libs/libfreeswitch.so
-L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib
/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid
-L/usr/local/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lrt -lssl
-lcrypto -lz -lncurses -ljpeg -lodbc -L/usr/local/lib
/usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a
-L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -lperl -lresolv -lnsl
-lgdbm -ldb -ldl -lm -lcrypt -lutil -lpthread -lc   -Wl,--rpath
-Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_perl.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_perl-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
 
 
The modules.conf I upload is:
#applications/mod_abstraction
#applications/mod_avmd
applications/mod_blacklist
applications/mod_callcenter
applications/mod_cidlookup
applications/mod_cluechoo
applications/mod_commands
applications/mod_conference
applications/mod_curl
applications/mod_db
applications/mod_directory
#applications/mod_distributor
applications/mod_dptools
#applications/mod_easyroute
applications/mod_enum
applications/mod_esf
#applications/mod_esl
applications/mod_expr
applications/mod_fifo
#applications/mod_fsk
applications/mod_fsv
applications/mod_hash
applications/mod_httapi
#applications/mod_http_cache
#applications/mod_ladspa
applications/mod_lcr
#applications/mod_memcache
#applications/mod_mongo
#applications/mod_nibblebill
#applications/mod_osp
#applications/mod_redis
#applications/mod_rss
applications/mod_sms
#applications/mod_snapshot
#applications/mod_snipe_hunt
#applications/mod_snom
#applications/mod_soundtouch
applications/mod_spandsp
#applications/mod_spy
#applications/mod_stress
applications/mod_valet_parking
#applications/mod_vmd
applications/mod_voicemail
applications/mod_voicemail_ivr
#applications/mod_random
#asr_tts/mod_cepstral
asr_tts/mod_flite
asr_tts/mod_pocketsphinx
asr_tts/mod_tts_commandline
#asr_tts/mod_unimrcp
codecs/mod_amr
#codecs/mod_amrwb
codecs/mod_bv
#codecs/mod_celt
#codecs/mod_codec2
#codecs/mod_com_g729
#codecs/mod_dahdi_codec
codecs/mod_g723_1
codecs/mod_g729
codecs/mod_h26x
#codecs/mod_ilbc
#codecs/mod_isac
#codecs/mod_opus
#codecs/mod_sangoma_codec
#codecs/mod_silk
#codecs/mod_siren
codecs/mod_speex
dialplans/mod_dialplan_asterisk
#dialplans/mod_dialplan_directory
dialplans/mod_dialplan_xml
#directories/mod_ldap
#endpoints/mod_alsa
endpoints/mod_dingaling
#endpoints/mod_h323
#endpoints/mod_khomp
endpoints/mod_loopback
#endpoints/mod_opal
#endpoints/mod_portaudio
endpoints/mod_rtmp
#endpoints/mod_skinny
#endpoints/mod_skypopen
endpoints/mod_sofia
event_handlers/mod_cdr_csv
#event_handlers/mod_cdr_mongodb
#event_handlers/mod_cdr_pg_csv
event_handlers/mod_cdr_sqlite
#event_handlers/mod_erlang_event
#event_handlers/mod_event_multicast
event_handlers/mod_event_socket
#event_handlers/mod_event_zmq
#event_handlers/mod_radius_cdr
event_handlers/mod_snmp
formats/mod_local_stream
formats/mod_native_file
#formats/mod_portaudio_stream
#formats/mod_shell_stream
formats/mod_shout
formats/mod_sndfile
formats/mod_tone_stream
#formats/mod_vlc
#languages/mod_java
languages/mod_lua
#languages/mod_managed
languages/mod_perl
languages/mod_python
languages/mod_spidermonkey
#languages/mod_yaml
loggers/mod_console
loggers/mod_logfile
loggers/mod_syslog
#say/mod_say_de
say/mod_say_en
#say/mod_say_es
#say/mod_say_fr
#say/mod_say_he
#say/mod_say_hu
#say/mod_say_it
#say/mod_say_nl
#say/mod_say_pt
#say/mod_say_ru
#say/mod_say_th
#say/mod_say_zh
#timers/mod_posix_timer
#timers/mod_timerfd
xml_int/mod_xml_cdr
xml_int/mod_xml_curl
#xml_int/mod_xml_ldap
xml_int/mod_xml_rpc
xml_int/mod_xml_scgi
 
#../../libs/freetdm/mod_freetdm
#../../libs/openzap/mod_openzap
 
## Experimental Modules (don't cry if they're broken)
#../../contrib/mod/xml_int/mod_xml_odbc
 




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <x-msg://19/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/> 
 
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
<x-msg://19/FreeSWITCH-users at 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 <x-msg://19/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/> 

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
<x-msg://19/FreeSWITCH-users at 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/> 


-- 
Ken
http://www.FreeSWITCH.org <http://www.FreeSWITCH.org/> 
http://www.ClueCon.com <http://www.ClueCon.com/> 
http://www.OSTAG.org <http://www.OSTAG.org/> 
irc.freenode.net #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

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

 

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From gerrylist at drouillard.ca  Wed Dec 19 03:13:38 2012
From: gerrylist at drouillard.ca (Gerald Drouillard)
Date: Tue, 18 Dec 2012 19:13:38 -0500
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
Message-ID: <50D106B2.90904@drouillard.ca>

On 12/16/2012 11:15 AM, Cal Leeming [Simplicity Media Ltd] wrote:
> *Any and all feedback on this thread would be much welcomed.*
>
>
We always disable ALG on the client side where possible.  It always 
seems to break the connection.  We also have the offsite phones register 
via tcp instead of udp and that helps.   You may also want to look at 
your keep alives and set them to around 20-40secs or so to keep the 
connection open with the server.

We have had good success with the opensource softphone jitsi.

-- 
Regards
--------------------------------------
Gerald Drouillard
Technology Architect
Drouillard & Associates, Inc.
http://www.Drouillard.biz

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From msc at freeswitch.org  Wed Dec 19 04:06:09 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Dec 2012 17:06:09 -0800
Subject: [Freeswitch-users] show calls/channels b-leg false information
In-Reply-To: <CAFiqYumJzso8FZ3PEJqLYO-kFUrJLOHO38LjXMRzrXaucKTnQg@mail.gmail.com>
References: <CALFFxt1KYV67zv5D6xn5yekSbMkVyBtKn7+50Sma-ymVXPM1ig@mail.gmail.com>
	<CAKzWOxXj0vQv-1kQf0FFX9ZcOJ-Zdt+_X0S90+sKtgLkv0yo6Q@mail.gmail.com>
	<CALFFxt0VqA6uUVnqkPwZAXgmQb9La4NcJYD0tg-9b0uWC8OYLg@mail.gmail.com>
	<CAFiqYum2zQEkLEP3Ot7-Jzjq4odJL0QU4bO0+gexZh1F7Uw7yA@mail.gmail.com>
	<CALFFxt3KPoeN+-RBe4QVQKn+t7yrK+_WW8E8v+DhJtPqr7urXA@mail.gmail.com>
	<CAKzWOxV92K9R4ZOSxxs5=9CJkjbFK1oCowdGni1ia_qjO9ZW6g@mail.gmail.com>
	<CAFiqYumJzso8FZ3PEJqLYO-kFUrJLOHO38LjXMRzrXaucKTnQg@mail.gmail.com>
Message-ID: <CAKzWOxXc5x7wm=Hbbx4=65Et3CXZS8HS8Ufjg9YNAw_W1BidKg@mail.gmail.com>

On Mon, Dec 17, 2012 at 11:51 PM, Steven Ayre <steveayre at gmail.com> wrote:

> It's worth noting that show channels/calls just map to a SQL query on the
> sqlite/ODBC tables used by the core. They're a partial overview of the
> call, as Michael notes not the full picture.
>
> In particular it's the core that's writing to that database - so it's not
> going to contain anything endpoint-specific. So all the sip_ fields set by
> mod_sofia won't be stored there.
>
> They do at least give you the UUID of the call - you could get the other
> variables from the channels using uuid_getvar using the same ESL connection
> you already have open to run the show api in.
>
> Michael,
> #3 is possible yes, but won't that give you a diff against upstream
> that'll never be merged and as upstream changes won't apply cleanly any
> more, making it a headache to maintain?
>
Yep, so it's not for the faint of heart.
-MC


>
> -Steve
>
>
> On 18 December 2012 06:31, Michael Collins <msc at freeswitch.org> wrote:
>
>> Ah - be careful. The "database" is not the entire picture. When you do
>> show channels or show calls it pulls right from some sqlite3 tables that
>> are frequently updated by FreeSWITCH. You have a few choices depending upon
>> just how intense your monitoring will be:
>>
>> #1 - Follow Steven's advice and monitor CHANNEL events on the event socket
>> #2 - Use some form of polling and "uuid_getvar <uuid> sip_gateway_name"
>> to get the name for each channel
>> #3 - Modify the source code to store the sip_gateway_name in the same
>> table that stores the rest of the show channels/show calls information.
>> (Most people don't realize that you can do this. :)
>>
>> Personally I like event socket/ESL so I would go with #1 first.
>>
>> -MC
>>
>>
>> On Mon, Dec 17, 2012 at 4:32 PM, Ahmed Sboor <ahmed at netelsat.net> wrote:
>>
>>> Actually my purpose is just to monitor live calls . All is going fine
>>> but if database is not storing the channel variable how its coming as in
>>> field " ip_addr" ? b_ip_addr is just getting the same value while it should
>>> be having Other leg . there is another field b_dest that is also showing
>>> correct value. Why not for b_ip_addr then ?
>>> yesterday i was searching old threads and somewhere Anthony replied to
>>> similar Question that it has to be updated in db . But how to do that
>>> nowhere mentioned .
>>>
>>>
>>>
>>> On Tue, Dec 18, 2012 at 5:16 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>>
>>>>
>>>>
>>>> On 17 December 2012 23:20, Ahmed Sboor <ahmed at netelsat.net> wrote:
>>>>
>>>>> Hi,
>>>>> i tried dump and variable_sip_to_host is the variable which tells me
>>>>> what i am looking for, And now question is how to get value of this
>>>>> variable in b_ip_addr while doing show calls as xml ?
>>>>>
>>>>>
>>>>> On Tue, Dec 18, 2012 at 3:43 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> Get a call up and running and then do a uuid_dump of the a leg. I
>>>>>> think you'll find that there are some channel variables that get set. I
>>>>>> don't recall them off the top of my head but look for something like
>>>>>> sip_gateway or sip_gateway_name.
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>> On Mon, Dec 17, 2012 at 1:15 PM, Ahmed Sboor <ahmed at netelsat.net>wrote:
>>>>>>
>>>>>>> Hi All,
>>>>>>> i am trying to see information like show calls or show channels .
>>>>>>> only confusion is wrong/missing information in *b_name and **b_ip_addr
>>>>>>> .*
>>>>>>> these both are taken from a leg while here i was expecting to see
>>>>>>> call is connected to which gateway.
>>>>>>> *
>>>>>>> *
>>>>>>> *
>>>>>>> *
>>>>>>> i am having like :    *<b_name>sofia/external/919123123334</b_name>*
>>>>>>>
>>>>>>> Means there is no "gateway name mentioned in b_name.
>>>>>>>
>>>>>>> and in b_ip_addr its like :
>>>>>>> *    <b_ip_addr>10.10.10.1</b_ip_addr>*
>>>>>>>
>>>>>>> while this is ip call is coming from and b leg i was expecting as ip
>>>>>>> of gateway call is going to.
>>>>>>>
>>>>>>> Please let me know if i am missing something.
>>>>>>>
>>>>>>> usring FreeSWITCH Version 1.2.5.3+git~20121206T171012Z~d74bef3f2f
>>>>>>> (git d74bef3 2012-12-06 17:10:12Z)
>>>>>>>
>>>>>>> Thanking you
>>>>>>>  Ahmed
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting 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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Michael S Collins
>>>>>> Twitter: @mercutioviz
>>>>>> http://www.FreeSWITCH.org
>>>>>> http://www.ClueCon.com
>>>>>> http://www.OSTAG.org
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting 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
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
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>
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-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jrichey at itltd.net  Wed Dec 19 04:22:56 2012
From: jrichey at itltd.net (JRichey)
Date: Tue, 18 Dec 2012 17:22:56 -0800
Subject: [Freeswitch-users] New install 1.2.Stable blows up compiling
	mod_perl.so
Message-ID: <6ECAF1527329364583AB525CF34ABF950B31A744@ms.kallback.com>

"yum remove zlib" will do a pretty good job of wiping out your system due to
dependencies.

-----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, December 18, 2012 4:33 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so



What a total disaster!

 

I added the EPEL per JRichie - thank you.

rpm -Uvh
http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm
<http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm> 

 

 

Then I decided to remove all the pre-reqs and reinstall.

I ran:

 yum remove autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig unixODBC-devel openssl-devel libogg-devel
libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel
expat-devel zlib zlib-devel bzip2 which

 

The results are of course totally disastrous.  I now have no yum or rpm.  I
looked up how to install them.  Yum says use rpm and rpm says use yum!

 

I can't even do a "cat /etc/redhat-release"  or "ls" => /bin/ls  command not
found.

 

Short of waiting for the ISP to get up tomorrow and completely
reprovisioning my Centos5, does anyone have any suggestions.

 

Is there a command to reinstall centos 5 w/o yum or rpm?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mario G
Sent: Tuesday, December 18, 2012 6:52 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

 

FYI, I was the one that tested FS on CentOS (among about 5 other distros)
since I was redoing the install wiki, and I remember EPEL was required on
CentOS 5.5 at the time.

Mario G

 

On Dec 18, 2012, at 3:41 PM, Ken Rice wrote:





You probably don't need epel unless you are installing on centos5... EPEL
just adds another yum REPO for getting more upto date packages...

If you look at the EPEL site, it tells you how to install it... Its usually
just add this RPM via yum install http://some.website/some/path/some.rpm
<http://some.website/some/path/some.rpm> 



On 12/18/12 4:57 PM, "Sean Devoy" < sdevoy at bizfocused.com
<x-msg://19/sdevoy at bizfocused.com> > wrote:




Thanks.  That is EXACTLY what I am trying to follow.  I am thrown by "There
were no issues during installation and testing other than EPEL
requirements."
And "Installation of Git required adding the EPEL repository to obtain Git.
For EPEL information visit the EPEL < http://fedoraproject.org/wiki/EPEL
<http://fedoraproject.org/wiki/EPEL> >  site, The EPEL repository is added
with 1 click."
 
I cannot understand how to install EPEL or how to click with my SSH window.
 
Anyone that can help me with that?
 
I did not install the "optional packages" so I will now. And try again.

 

From: freeswitch-users-bounces at lists.freeswitch.org
<x-msg://19/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 Ken
Rice
Sent: Tuesday, December 18, 2012 5:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

If you installed on a centos base install then follow the directions on the
wiki at 
http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS
<http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS> 

That usually avoids all these errors...  I usually do something like 

 yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig unixODBC-devel openssl-devel libogg-devel
libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel
expat-devel zlib zlib-devel bzip2 which

You may of course need to add a few extra RPMs if you are compiling things
that you have to uncomment in modules.conf

 



On 12/18/12 4:23 PM, "Sean Devoy" < sdevoy at bizfocused.com
<x-msg://19/sdevoy at bizfocused.com> > wrote:
Thank you!
That resulted in a message saying I needed to install bison, so I ran yum
install bison.
 
That resulted in shit though:
gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
 gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
libtool: link: `VbrTag.lo' is not a valid libtool object
make[9]: *** [libmp3lame.la] Error 1
make[8]: *** [all-recursive] Error 1
Making all in frontend
libtool: link: `VbrTag.lo' is not a valid libtool object
make[9]: *** [libmp3lame.la] Error 1
make[8]: *** [all-recursive] Error 1
Making all in frontend
make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.
Making all in Dll
Making all in debian
Making all in doc
Making all in html
Making all in man
Making all in include
Making all in misc
Making all in dshow
Making all in ACM
Making all in ADbg
Making all in ddk
make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.
Making all in Dll
Making all in tinyxml
Making all in debian
Making all in doc
Making all in mac
Making all in macosx
Making all in html
Making all in man
Making all in English.lproj
Making all in include
Making all in LAME.xcodeproj
Making all in misc
Making all in dshow
Making all in vc_solution
Making all in ACM
make[7]: *** [all-recursive] Error 1
make[6]: *** [all] Error 2
make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2
make[4]: *** [install] Error 1
make[3]: *** [mod_shout-install] Error 1
Making all in ADbg
Making all in ddk
Making all in tinyxml
Making all in mac
Making all in macosx
Making all in English.lproj
Making all in LAME.xcodeproj
Making all in vc_solution
make[7]: *** [all-recursive] Error 1
make[6]: *** [all] Error 2
make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2
make[4]: *** [all] Error 1
make[3]: *** [mod_shout-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
make[2]: *** [install-recursive] Error 1
make[1]: *** [install-recursive] Error 1
make: *** [install] Error 2
[1]+  Exit 2                  make
 
Can't we get the INSTALL page of the wiki right?
 

From: freeswitch-users-bounces at lists.freeswitch.org
<x-msg://19/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 Abaci
Sent: Tuesday, December 18, 2012 5:01 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

yum install gdbm-devel db4-devel
 
On 12/18/2012 4:52 PM, Sean Devoy wrote:
I just started a clean install of the stable tree on a brand new install of
Centos 5.3.
Here is EXACTLY what I have done so far:
Logged in to root
Cd /usr/local/src
yum install gcc
yum -y install zlib-devel openssl-devel cpio expat-devel gettext-devel
cd /usr/local/src
wget http://git-core.googlecode.com/files/git-1.7.9.tar.gz
<http://git-core.googlecode.com/files/git-1.7.9.tar.gz> 
tar xvzf git-1.7.9.tar.gz
cd git-1.7.9
./configure
make
make install
yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig
 
cd /usr/local/src
git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
<git://git.freeswitch.org/freeswitch.git> 
./bootstrap.sh    
./configure
tftp modules.conf from existing centos/fs server   (did this kill it)?  If
so how can I get it back?
make & make install  
 
after about 20 minutes the output is:
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c  -fPIC -DPIC
-o .libs/mod_perl.o
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c -o
mod_perl.o >/dev/null 2>&1
Creating mod_perl.so...
/usr/bin/ld: cannot find -lgdbm
collect2: ld returned 1 exit status
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E -Wl,-rpath
-Wl,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE freeswitch_perl.o
mod_perl_wrap.o perlxsi.o  /usr/local/src/freeswitch/.libs/libfreeswitch.so
-L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib
/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid
-L/usr/local/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lrt -lssl
-lcrypto -lz -lncurses -ljpeg -lodbc -L/usr/local/lib
/usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a
-L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -lperl -lresolv -lnsl
-lgdbm -ldb -ldl -lm -lcrypt -lutil -lpthread -lc   -Wl,--rpath
-Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_perl.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_perl-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
 
 
The modules.conf I upload is:
#applications/mod_abstraction
#applications/mod_avmd
applications/mod_blacklist
applications/mod_callcenter
applications/mod_cidlookup
applications/mod_cluechoo
applications/mod_commands
applications/mod_conference
applications/mod_curl
applications/mod_db
applications/mod_directory
#applications/mod_distributor
applications/mod_dptools
#applications/mod_easyroute
applications/mod_enum
applications/mod_esf
#applications/mod_esl
applications/mod_expr
applications/mod_fifo
#applications/mod_fsk
applications/mod_fsv
applications/mod_hash
applications/mod_httapi
#applications/mod_http_cache
#applications/mod_ladspa
applications/mod_lcr
#applications/mod_memcache
#applications/mod_mongo
#applications/mod_nibblebill
#applications/mod_osp
#applications/mod_redis
#applications/mod_rss
applications/mod_sms
#applications/mod_snapshot
#applications/mod_snipe_hunt
#applications/mod_snom
#applications/mod_soundtouch
applications/mod_spandsp
#applications/mod_spy
#applications/mod_stress
applications/mod_valet_parking
#applications/mod_vmd
applications/mod_voicemail
applications/mod_voicemail_ivr
#applications/mod_random
#asr_tts/mod_cepstral
asr_tts/mod_flite
asr_tts/mod_pocketsphinx
asr_tts/mod_tts_commandline
#asr_tts/mod_unimrcp
codecs/mod_amr
#codecs/mod_amrwb
codecs/mod_bv
#codecs/mod_celt
#codecs/mod_codec2
#codecs/mod_com_g729
#codecs/mod_dahdi_codec
codecs/mod_g723_1
codecs/mod_g729
codecs/mod_h26x
#codecs/mod_ilbc
#codecs/mod_isac
#codecs/mod_opus
#codecs/mod_sangoma_codec
#codecs/mod_silk
#codecs/mod_siren
codecs/mod_speex
dialplans/mod_dialplan_asterisk
#dialplans/mod_dialplan_directory
dialplans/mod_dialplan_xml
#directories/mod_ldap
#endpoints/mod_alsa
endpoints/mod_dingaling
#endpoints/mod_h323
#endpoints/mod_khomp
endpoints/mod_loopback
#endpoints/mod_opal
#endpoints/mod_portaudio
endpoints/mod_rtmp
#endpoints/mod_skinny
#endpoints/mod_skypopen
endpoints/mod_sofia
event_handlers/mod_cdr_csv
#event_handlers/mod_cdr_mongodb
#event_handlers/mod_cdr_pg_csv
event_handlers/mod_cdr_sqlite
#event_handlers/mod_erlang_event
#event_handlers/mod_event_multicast
event_handlers/mod_event_socket
#event_handlers/mod_event_zmq
#event_handlers/mod_radius_cdr
event_handlers/mod_snmp
formats/mod_local_stream
formats/mod_native_file
#formats/mod_portaudio_stream
#formats/mod_shell_stream
formats/mod_shout
formats/mod_sndfile
formats/mod_tone_stream
#formats/mod_vlc
#languages/mod_java
languages/mod_lua
#languages/mod_managed
languages/mod_perl
languages/mod_python
languages/mod_spidermonkey
#languages/mod_yaml
loggers/mod_console
loggers/mod_logfile
loggers/mod_syslog
#say/mod_say_de
say/mod_say_en
#say/mod_say_es
#say/mod_say_fr
#say/mod_say_he
#say/mod_say_hu
#say/mod_say_it
#say/mod_say_nl
#say/mod_say_pt
#say/mod_say_ru
#say/mod_say_th
#say/mod_say_zh
#timers/mod_posix_timer
#timers/mod_timerfd
xml_int/mod_xml_cdr
xml_int/mod_xml_curl
#xml_int/mod_xml_ldap
xml_int/mod_xml_rpc
xml_int/mod_xml_scgi
 
#../../libs/freetdm/mod_freetdm
#../../libs/openzap/mod_openzap
 
## Experimental Modules (don't cry if they're broken)
#../../contrib/mod/xml_int/mod_xml_odbc
 




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <x-msg://19/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/> 
 
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
<x-msg://19/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:
consulting at freeswitch.org <x-msg://19/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/> 

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
<x-msg://19/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/> 


-- 
Ken
http://www.FreeSWITCH.org <http://www.FreeSWITCH.org/> 
http://www.ClueCon.com <http://www.ClueCon.com/> 
http://www.OSTAG.org <http://www.OSTAG.org/> 
irc.freenode.net <http://irc.freenode.net>  #freeswitch

_________________________________________________________________________
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> 

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

 

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From sdevoy at bizfocused.com  Wed Dec 19 04:32:43 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 18 Dec 2012 20:32:43 -0500
Subject: [Freeswitch-users] New install 1.2.Stable blows up
	compiling	mod_perl.so
In-Reply-To: <6ECAF1527329364583AB525CF34ABF950B31A744@ms.kallback.com>
References: <6ECAF1527329364583AB525CF34ABF950B31A744@ms.kallback.com>
Message-ID: <5f6c01cddd88$be0287a0$3a0796e0$@bizfocused.com>

LOL  live and learn.  They are rebuilding my Centos 5.8 now.

Thanks.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of JRichey
Sent: Tuesday, December 18, 2012 8:23 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

 

"yum remove zlib" will do a pretty good job of wiping out your system due to
dependencies.

-----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, December 18, 2012 4:33 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

What a total disaster!

 

I added the EPEL per JRichie - thank you.

rpm -Uvh
http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm

 

 

Then I decided to remove all the pre-reqs and reinstall.

I ran:

 yum remove autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig unixODBC-devel openssl-devel libogg-devel
libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel
expat-devel zlib zlib-devel bzip2 which

 

The results are of course totally disastrous.  I now have no yum or rpm.  I
looked up how to install them.  Yum says use rpm and rpm says use yum!

 

I can't even do a "cat /etc/redhat-release"  or "ls" => /bin/ls  command not
found.

 

Short of waiting for the ISP to get up tomorrow and completely
reprovisioning my Centos5, does anyone have any suggestions.

 

Is there a command to reinstall centos 5 w/o yum or rpm?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mario G
Sent: Tuesday, December 18, 2012 6:52 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

 

FYI, I was the one that tested FS on CentOS (among about 5 other distros)
since I was redoing the install wiki, and I remember EPEL was required on
CentOS 5.5 at the time.

Mario G

 

On Dec 18, 2012, at 3:41 PM, Ken Rice wrote:

 

You probably don't need epel unless you are installing on centos5... EPEL
just adds another yum REPO for getting more upto date packages...

If you look at the EPEL site, it tells you how to install it... Its usually
just add this RPM via yum install http://some.website/some/path/some.rpm



On 12/18/12 4:57 PM, "Sean Devoy" <sdevoy at bizfocused.com
<x-msg://19/sdevoy at bizfocused.com> > wrote:



Thanks.  That is EXACTLY what I am trying to follow.  I am thrown by "There
were no issues during installation and testing other than EPEL
requirements."
And "Installation of Git required adding the EPEL repository to obtain Git.
For EPEL information visit the EPEL <http://fedoraproject.org/wiki/EPEL>
site, The EPEL repository is added with 1 click."
 
I cannot understand how to install EPEL or how to click with my SSH window.
 
Anyone that can help me with that?
 
I did not install the "optional packages" so I will now. And try again.

 

From: freeswitch-users-bounces at lists.freeswitch.org
<x-msg://19/freeswitch-users-bounces at lists.freeswitch.org>
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Tuesday, December 18, 2012 5:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

If you installed on a centos base install then follow the directions on the
wiki at 
http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS

That usually avoids all these errors...  I usually do something like 

 yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig unixODBC-devel openssl-devel libogg-devel
libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel
expat-devel zlib zlib-devel bzip2 which

You may of course need to add a few extra RPMs if you are compiling things
that you have to uncomment in modules.conf

 



On 12/18/12 4:23 PM, "Sean Devoy" <sdevoy at bizfocused.com
<x-msg://19/sdevoy at bizfocused.com> > wrote:
Thank you!
That resulted in a message saying I needed to install bison, so I ran yum
install bison.
 
That resulted in shit though:
gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
 gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I. -I../mpglib -I.. -Wall -pipe
-g -O2 -MT mpglib_interface.lo -MD -MP -MF .deps/mpglib_interface.Tpo -c
mpglib_interface.c  -fPIC -DPIC -o mpglib_interface.o
libtool: link: `VbrTag.lo' is not a valid libtool object
make[9]: *** [libmp3lame.la] Error 1
make[8]: *** [all-recursive] Error 1
Making all in frontend
libtool: link: `VbrTag.lo' is not a valid libtool object
make[9]: *** [libmp3lame.la] Error 1
make[8]: *** [all-recursive] Error 1
Making all in frontend
make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.
Making all in Dll
Making all in debian
Making all in doc
Making all in html
Making all in man
Making all in include
Making all in misc
Making all in dshow
Making all in ACM
Making all in ADbg
Making all in ddk
make[8]: *** No rule to make target `../libmp3lame/libmp3lame.la', needed by
`lame'.  Stop.
Making all in Dll
Making all in tinyxml
Making all in debian
Making all in doc
Making all in mac
Making all in macosx
Making all in html
Making all in man
Making all in English.lproj
Making all in include
Making all in LAME.xcodeproj
Making all in misc
Making all in dshow
Making all in vc_solution
Making all in ACM
make[7]: *** [all-recursive] Error 1
make[6]: *** [all] Error 2
make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2
make[4]: *** [install] Error 1
make[3]: *** [mod_shout-install] Error 1
Making all in ADbg
Making all in ddk
Making all in tinyxml
Making all in mac
Making all in macosx
Making all in English.lproj
Making all in LAME.xcodeproj
Making all in vc_solution
make[7]: *** [all-recursive] Error 1
make[6]: *** [all] Error 2
make[5]: ***
[/usr/local/src/freeswitch/libs/lame-3.98.4/libmp3lame/libmp3lame.la] Error
2
make[4]: *** [all] Error 1
make[3]: *** [mod_shout-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
make[2]: *** [install-recursive] Error 1
make[1]: *** [install-recursive] Error 1
make: *** [install] Error 2
[1]+  Exit 2                  make
 
Can't we get the INSTALL page of the wiki right?
 

From: freeswitch-users-bounces at lists.freeswitch.org
<x-msg://19/freeswitch-users-bounces at lists.freeswitch.org>
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abaci
Sent: Tuesday, December 18, 2012 5:01 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New install 1.2.Stable blows up compiling
mod_perl.so

yum install gdbm-devel db4-devel
 
On 12/18/2012 4:52 PM, Sean Devoy wrote:
I just started a clean install of the stable tree on a brand new install of
Centos 5.3.
Here is EXACTLY what I have done so far:
Logged in to root
Cd /usr/local/src
yum install gcc
yum -y install zlib-devel openssl-devel cpio expat-devel gettext-devel
cd /usr/local/src
wget http://git-core.googlecode.com/files/git-1.7.9.tar.gz
tar xvzf git-1.7.9.tar.gz
cd git-1.7.9
./configure
make
make install
yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make
ncurses-devel pkgconfig
 
cd /usr/local/src
git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
./bootstrap.sh    
./configure
tftp modules.conf from existing centos/fs server   (did this kill it)?  If
so how can I get it back?
make & make install  
 
after about 20 minutes the output is:
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c  -fPIC -DPIC
-o .libs/mod_perl.o
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/languages/mod_perl/mod_perl.c -o
mod_perl.o >/dev/null 2>&1
Creating mod_perl.so...
/usr/bin/ld: cannot find -lgdbm
collect2: ld returned 1 exit status
gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe
-Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm
-I/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -DEMBED_PERL
-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
-I/usr/local/src/freeswitch/libs/stfu -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_perl.so -shared -Wl,-x .libs/mod_perl.o -Wl,-E -Wl,-rpath
-Wl,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE freeswitch_perl.o
mod_perl_wrap.o perlxsi.o  /usr/local/src/freeswitch/.libs/libfreeswitch.so
-L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib
/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid
-L/usr/local/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lrt -lssl
-lcrypto -lz -lncurses -ljpeg -lodbc -L/usr/local/lib
/usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a
-L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE -lperl -lresolv -lnsl
-lgdbm -ldb -ldl -lm -lcrypt -lutil -lpthread -lc   -Wl,--rpath
-Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_perl.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_perl-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
 
 
The modules.conf I upload is:
#applications/mod_abstraction
#applications/mod_avmd
applications/mod_blacklist
applications/mod_callcenter
applications/mod_cidlookup
applications/mod_cluechoo
applications/mod_commands
applications/mod_conference
applications/mod_curl
applications/mod_db
applications/mod_directory
#applications/mod_distributor
applications/mod_dptools
#applications/mod_easyroute
applications/mod_enum
applications/mod_esf
#applications/mod_esl
applications/mod_expr
applications/mod_fifo
#applications/mod_fsk
applications/mod_fsv
applications/mod_hash
applications/mod_httapi
#applications/mod_http_cache
#applications/mod_ladspa
applications/mod_lcr
#applications/mod_memcache
#applications/mod_mongo
#applications/mod_nibblebill
#applications/mod_osp
#applications/mod_redis
#applications/mod_rss
applications/mod_sms
#applications/mod_snapshot
#applications/mod_snipe_hunt
#applications/mod_snom
#applications/mod_soundtouch
applications/mod_spandsp
#applications/mod_spy
#applications/mod_stress
applications/mod_valet_parking
#applications/mod_vmd
applications/mod_voicemail
applications/mod_voicemail_ivr
#applications/mod_random
#asr_tts/mod_cepstral
asr_tts/mod_flite
asr_tts/mod_pocketsphinx
asr_tts/mod_tts_commandline
#asr_tts/mod_unimrcp
codecs/mod_amr
#codecs/mod_amrwb
codecs/mod_bv
#codecs/mod_celt
#codecs/mod_codec2
#codecs/mod_com_g729
#codecs/mod_dahdi_codec
codecs/mod_g723_1
codecs/mod_g729
codecs/mod_h26x
#codecs/mod_ilbc
#codecs/mod_isac
#codecs/mod_opus
#codecs/mod_sangoma_codec
#codecs/mod_silk
#codecs/mod_siren
codecs/mod_speex
dialplans/mod_dialplan_asterisk
#dialplans/mod_dialplan_directory
dialplans/mod_dialplan_xml
#directories/mod_ldap
#endpoints/mod_alsa
endpoints/mod_dingaling
#endpoints/mod_h323
#endpoints/mod_khomp
endpoints/mod_loopback
#endpoints/mod_opal
#endpoints/mod_portaudio
endpoints/mod_rtmp
#endpoints/mod_skinny
#endpoints/mod_skypopen
endpoints/mod_sofia
event_handlers/mod_cdr_csv
#event_handlers/mod_cdr_mongodb
#event_handlers/mod_cdr_pg_csv
event_handlers/mod_cdr_sqlite
#event_handlers/mod_erlang_event
#event_handlers/mod_event_multicast
event_handlers/mod_event_socket
#event_handlers/mod_event_zmq
#event_handlers/mod_radius_cdr
event_handlers/mod_snmp
formats/mod_local_stream
formats/mod_native_file
#formats/mod_portaudio_stream
#formats/mod_shell_stream
formats/mod_shout
formats/mod_sndfile
formats/mod_tone_stream
#formats/mod_vlc
#languages/mod_java
languages/mod_lua
#languages/mod_managed
languages/mod_perl
languages/mod_python
languages/mod_spidermonkey
#languages/mod_yaml
loggers/mod_console
loggers/mod_logfile
loggers/mod_syslog
#say/mod_say_de
say/mod_say_en
#say/mod_say_es
#say/mod_say_fr
#say/mod_say_he
#say/mod_say_hu
#say/mod_say_it
#say/mod_say_nl
#say/mod_say_pt
#say/mod_say_ru
#say/mod_say_th
#say/mod_say_zh
#timers/mod_posix_timer
#timers/mod_timerfd
xml_int/mod_xml_cdr
xml_int/mod_xml_curl
#xml_int/mod_xml_ldap
xml_int/mod_xml_rpc
xml_int/mod_xml_scgi
 
#../../libs/freetdm/mod_freetdm
#../../libs/openzap/mod_openzap
 
## Experimental Modules (don't cry if they're broken)
#../../contrib/mod/xml_int/mod_xml_odbc
 




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <x-msg://19/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/> 
 
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
<x-msg://19/FreeSWITCH-users at 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 <x-msg://19/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/> 

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
<x-msg://19/FreeSWITCH-users at 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/> 


-- 
Ken
http://www.FreeSWITCH.org <http://www.FreeSWITCH.org/> 
http://www.ClueCon.com <http://www.ClueCon.com/> 
http://www.OSTAG.org <http://www.OSTAG.org/> 
irc.freenode.net #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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 Dec 19 04:47:09 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Dec 2012 17:47:09 -0800
Subject: [Freeswitch-users] New install 1.2.Stable blows up compiling
	mod_perl.so
In-Reply-To: <5f6c01cddd88$be0287a0$3a0796e0$@bizfocused.com>
References: <6ECAF1527329364583AB525CF34ABF950B31A744@ms.kallback.com>
	<5f6c01cddd88$be0287a0$3a0796e0$@bizfocused.com>
Message-ID: <CAKzWOxV6-8tu3a5A6n47hG=upWBGimJT+cL3mXSO+caFercZbQ@mail.gmail.com>

On Tue, Dec 18, 2012 at 5:32 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> LOL  live and learn.  They are rebuilding my Centos 5.8 now.****
>
> Thanks.
>

Just remember we're laughing *with* you, not at you, because pretty much
everyone one of us has a story to tell. "One time I had this critical
server in place and I typed rm -fr * when I meant to type cd /tmp && rm -fr
* ..."

In fact, on tomorrow's conference call we should have a contest to see who
has done the biggest CLI woopsie daisy of them all. My money's on Ken Rice
- that guy has some mad skillz. :)

-MC
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From krice at freeswitch.org  Wed Dec 19 04:58:33 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 18 Dec 2012 19:58:33 -0600
Subject: [Freeswitch-users] New install 1.2.Stable blows up compiling
 mod_perl.so
In-Reply-To: <CAKzWOxV6-8tu3a5A6n47hG=upWBGimJT+cL3mXSO+caFercZbQ@mail.gmail.com>
Message-ID: <CCF67B69.62430%krice@freeswitch.org>

Back in the day I ran a large multi-line pay BBS/ISP.... We used this
software called MajorBBS to run the whole thing... It used these loadable
modules, not much different then freeswitch modules we use today, the only
problem was they needed to be loaded in a specific order, the only way to
get that order correct was moving the modules out of and back into the
directory they lived in until you got them to load right (they loaded based
on unsorted directory order)... So as with most hard drives back in the day,
I had to run a defrag regularly which always screwed up the load order...
After a few pints one night I decide its a good time to do the routine
defrag, etc...

Needless to say after about an hour of screwing with the system, I?ve
decided I had enough, I would just restore the backup I made of the FAT
tables and let it sit til later... Only problem I had defragged since the
last FAT back up... ie: the files were no longer in the same spot on hdd...
Needless to say there were a lotta people angry they couldn?t get onto the
system for about 3 days while I unscrewed things and tried to work my day
job also

K

On 12/18/12 7:47 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> 
> 
> On Tue, Dec 18, 2012 at 5:32 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>> LOL? live and learn.? They are rebuilding my Centos 5.8 now.
>> Thanks.
> 
> Just remember we're laughing *with* you, not at you, because pretty much
> everyone one of us has a story to tell. "One time I had this critical server
> in place and I typed rm -fr * when I meant to type cd /tmp && rm -fr * ..."
> 
> In fact, on tomorrow's conference call we should have a contest to see who has
> done the biggest CLI woopsie daisy of them all. My money's on Ken Rice - that
> guy has some mad skillz. :)
> 
> -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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From msc at freeswitch.org  Wed Dec 19 05:25:58 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Dec 2012 18:25:58 -0800
Subject: [Freeswitch-users] voices in the recordings are out of sync
In-Reply-To: <33095823FD21DF429B481B5163264B799F6D8752B6@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B799F6D8752B6@VMBX102.ihostexchange.net>
Message-ID: <CAKzWOxVMYHSv5yOEiRdrAUGWOniwLfM-VEjzdtGZDyW2oTCHNw@mail.gmail.com>

Latest version of FreeSWITCH has some updates that may fix this issue. I
would update to 1.2.5.3 ASAP.
-MC

On Tue, Dec 18, 2012 at 1:56 PM, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> I found one issue that voices are always out of sync in the recordings.
> I am running freeswitch-1.2.5.1-1 on CentOS 5 (i386), which was installed
> from yum.
> I am having trouble installing the latest version from source due to an
> error: Autoconf version 2.62 or higher is required.
> It would be nice if someone can reproduce this issue against HEAD. Thanks.
>
> Here're the steps to reproduce it. The idea is to call a phone number and
> then bridge to another phone number while the entire session is being
> recorded.
> 1. In dialplan/public.xml, make sure you have a dialplan to handle any 10
> digit phone numbers.
>     <extension name="public_extensions">
>       <condition field="destination_number" expression="^\d{10}$">
>         <action application="transfer" data="main XML default"/>
>       </condition>
>     </extension>
>
> 2. In dialplan/default/main.xml, make sure you have an extension to handle
> the call in the default context.
>   <extension name="test">
>       <condition field="${destination_number}" expression="^main$">
>         <action application="set" data="RECORD_STEREO=false"/>
>         <action application="answer" />
>         <action application="record_session"
> data="/tmp/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
>         <action application="bridge"
> data="{ignore_early_media=false}[leg_timeout=60]sofia/gateway/gw1/1234567890"/>
>       </condition>
>   </extension>
>
> 3. In sip_profiles/external/gateways.xml, make sure you have a gateway
> that allows you to make an outbound call.
>     <include>
>         <gateway name="gw1">
>           <param name="username" value=""/>
>           <param name="password" value=""/>
>           <param name="realm" value=""/>
>           <param name="from-domain" value=""/>
>           <param name="extension" value=""/>
>           <param name="expire-seconds" value="60"/>
>           <param name="register" value="false"/>
>           <param name="retry-seconds" value="60"/>
>         </gateway>
>     </include>
>
> 4. make a call to one of the allowed 10-digit phone numbers in your
> environment.
> 5. Once the call is answered, the caller shall start to count from 1 to 60
> with some pause after each number.
> 6. The callee shall repeat each number he/she heard from the caller.
> 7. You should be able to hear that 2 voices in the recoridng
> (/tmp/rec.wav) are out of sync.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec 19 05:31:13 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Dec 2012 18:31:13 -0800
Subject: [Freeswitch-users] Newbie question: Executing Lua scripts for
 incoming calls, how concurrency is to be handled?
In-Reply-To: <965759A53E43FE439E43565A7715E5F058F4156DD8@oa-exchange1.oa.com.au>
References: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au>
	<50CE5D6E.2020401@quentustech.com>
	<965759A53E43FE439E43565A7715E5F058F4156D32@oa-exchange1.oa.com.au>
	<CAKzWOxWFh3RNYg9NY4Dqa-=f9KZOaXTvM8NGzHpvEvzV16VUWA@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156DD8@oa-exchange1.oa.com.au>
Message-ID: <CAKzWOxUYyfZ_8dLgy_t-Ov2a+YXEDj04dDkdLSn14Q-tZk-8pQ@mail.gmail.com>

To trigger SIP registrations you'd need the PBX to have a SIP client. I'm
assuming this is possible, but maybe that's a false assumption. How are you
physically connecting from FreeSWITCH to the PBX?

-MC

On Tue, Dec 18, 2012 at 2:36 PM, Sirish Masur Mohan <
Sirish.MasurMohan at oa.com.au> wrote:

> Hey Guys,****
>
> ** **
>
> Would really appreciate if you could help me out here ? isn?t there a way
> to handle concurrent calls in the dial plan, especially when Lua scripts,
> accessing global variables, are executed on receiving calls? ****
>
> ** **
>
> Is mod_fifo the closest I could get to handle concurrency (as Michael has
> explained)? If yes, how do I trigger SIP registrations, especially working
> with a PBX which I don?t have full control of?****
>
> ** **
>
> With regards,****
>
> Sirish****
>
> ** **
>
> *From:* Sirish Masur Mohan
> *Sent:* Tuesday, 18 December 2012 1:02 PM
> *To:* FreeSWITCH Users Help
> *Subject:* RE: [Freeswitch-users] Newbie question: Executing Lua scripts
> for incoming calls, how concurrency is to be handled?****
>
> ** **
>
> Hi Michael,****
>
> ** **
>
> Thanks for the reply. ****
>
> ** **
>
> >> You would need a SIP registration from the PBX to FreeSWITCH for each
> of the modem extensions****
>
> I am seen SIP clients such as X-Lite sending out the SIP registrations,
> but could you please clarify as to how this can be achieved in the PBX? The
> final production environment would be out in the customer?s PBX, which I
> may not have complete control of.. ****
>
> ** **
>
> >> What application are you building?****
>
> I may not be able to provide the details because of the NDA with customer,
> but what I am trying to achieve is, to replace an existing IVR with
> FreeSWITCH in an old existing setup of the customer ? that?s the reason why
> we continue working with dialup modems!****
>
> ** **
>
> With regards,****
>
> Sirish****
>
> ** **
>
> *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:* Tuesday, 18 December 2012 6:23 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Newbie question: Executing Lua scripts
> for incoming calls, how concurrency is to be handled?****
>
> ** **
>
> You don't have to have actual human agents for mod_fifo. You could define
> a user for each modem and then manually "log in" those "agents" on the
> command line using the fifo_member API command. Something like this:
>
> fifo_member add fifo_name {fifo_member_wait=nowait}user/1234
>
> Where 1234 is the user id of one of the modems. You would need a SIP
> registration from the PBX to FreeSWITCH for each of the modem extensions.
>
> Having modems go through a VoIP system sounds a bit scary. What
> application are you building?
> -MC****
>
> On Sun, Dec 16, 2012 at 5:41 PM, Sirish Masur Mohan <
> Sirish.MasurMohan at oa.com.au> wrote:****
>
> Hi William,
>
> Thanks for the reply.
>
> My setup is as follows:
> Client(Caller)->dialup modem->PBX->FreeSWITCH->PBX-> 4 dialup
> modems->Server(Receiver)
>
> I basically need FreeSWITCH to bridge the incoming call to the best
> external destination (out of the 4 available), so that the modem training,
> connection etc can take place smoothly, before exchange of data. I am not
> sure if mod_fifo would help me in this scenario, as, I would require an
> agent to dial in and read the fifo. Could you please clarify?
>
> Thanks!
>
> With regards,
> Sirish****
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William King
> Sent: Monday, 17 December 2012 10:47 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for
> incoming calls, how concurrency is to be handled?
>
> Sounds like you want to take a look into mod_fifo.
>
> 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
>
> ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec 19 05:39:23 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Dec 2012 18:39:23 -0800
Subject: [Freeswitch-users] attended transfer to rxfax
In-Reply-To: <CAEVfcaZukd+gs3ReiS=ro-K3rTGKgEoHmNn_QWqwgN+Uks2iOw@mail.gmail.com>
References: <CAEVfcaZukd+gs3ReiS=ro-K3rTGKgEoHmNn_QWqwgN+Uks2iOw@mail.gmail.com>
Message-ID: <CAKzWOxVxXcmfY+y-3iE89HDFS7qXX66pycSv12z4SkhvPni0zg@mail.gmail.com>

Why do you need to do an attended transfer? If the fax extension is just
going to do rxfax then perform a blind transfer and be done with it.
-MC

On Tue, Dec 18, 2012 at 2:56 AM, Babak Yakhchali <babak.freeswitch at gmail.com
> wrote:

> Hi
> How can I implement this:
> user calls to a number which starts sending fax after user presses 1
> (channel A)
> after user hears fax signaling starts to attended transfer to a local
> extension which executes rxfax (Channel B)
> after user hears local extension rxfax signaling presses transfer to
> complete the transfer
> but no fax is received!!
> I trace logs and see that channel B ends after user presses transfer. it
> seems I need to create a C channel which is running txfax but I donno how
> to do it
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec 19 05:42:14 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Dec 2012 18:42:14 -0800
Subject: [Freeswitch-users] fifo questions
In-Reply-To: <E75D158C1EE546CEBF617224942B26FF@gmail.com>
References: <E75D158C1EE546CEBF617224942B26FF@gmail.com>
Message-ID: <CAKzWOxXQpU5Lr7BYK2WUM_2kmSF-=z2cMVEWDk4MQ3_4iNJ+Kw@mail.gmail.com>

Seven,

Could you share with us your fifo config? Others have asked about how to
ring all agents and it seems our documentation is lacking.

Thanks!
-MC

On Fri, Dec 14, 2012 at 8:43 PM, Seven Du <dujinfang at gmail.com> wrote:

>  Hi,
>
>
> I'm using mod_fifo with onhook agents, and when a caller in it will ring
> all agents. The problem is that if an agent is placing an outbound call and
> the fifo still ring it regardless it's "busy". I manually set on the phone
> to accept only one channel solved the problem. But mod_fifo still try to
> ring it, is it possible to not ring the "busy" agent? I found
> fifo_track_calls, might work with this?
>
> also, what's the purpose of fifo_add_outbound? should it have a difference
> with fifo_member add ?
>
>
> Thanks.
>
>
> --
> Seven Du
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec 19 05:48:46 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Dec 2012 18:48:46 -0800
Subject: [Freeswitch-users] Conference Calls 404
In-Reply-To: <CA+NjZ2vwfxyDiG753NPABA9ZKzb7LhqKs03iw+bHvYeoAudAAQ@mail.gmail.com>
References: <CA+NjZ2vwfxyDiG753NPABA9ZKzb7LhqKs03iw+bHvYeoAudAAQ@mail.gmail.com>
Message-ID: <CAKzWOxVdynYUKcyqBAFpEp_Xxo3VWtAGOSqUuLKXwVTmd2Z-Og@mail.gmail.com>

I'll get with Ken and we'll figure it out.
Thanks,
MC

On Mon, Dec 17, 2012 at 5:12 AM, Nick Vines <jnvines at gmail.com> wrote:

> The two most recent calls, Nov 28 and Dec 5, torrent links are broken.
> Does anyone have the torrent file or the recordings?
>
> Thanks,
> Nick
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From sdevoy at bizfocused.com  Wed Dec 19 06:11:49 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 18 Dec 2012 22:11:49 -0500
Subject: [Freeswitch-users] Install 1.2.Stable still fails.  Ver 2
Message-ID: <5fdf01cddd96$9670f330$c352d990$@bizfocused.com>

OK, I had my Centos restored to base install 5.8.

 

The install fails.  Here is EXACTLY what I did:

 

Login to root
cd /usr/local/src

rpm -Uvh
http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm


git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git

cd /usr/local/src/freeswitch

./bootstrap.sh

./configure

make & make install

 

 

It fails with:

/bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link  gcc
-g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
-I./include -I/usr/local/src/freeswitch/libs/apr/include/arch/unix
-I./include/arch/unix -I/usr/local/src/freeswitch/libs/apr/include
-version-info 2:8:2    -o libapr-1.la -rpath /usr/local/freeswitch/lib
passwd/apr_getpass.lo strings/apr_cpystrn.lo strings/apr_strnatcmp.lo
strings/apr_strings.lo strings/apr_strtok.lo strings/apr_fnmatch.lo
strings/apr_snprintf.lo tables/apr_tables.lo tables/apr_hash.lo
atomic/unix/apr_atomic.lo dso/unix/dso.lo file_io/unix/flock.lo
file_io/unix/readwrite.lo file_io/unix/filepath_util.lo file_io/unix/seek.lo
file_io/unix/dir.lo file_io/unix/mktemp.lo file_io/unix/filedup.lo
file_io/unix/tempdir.lo file_io/unix/filepath.lo file_io/unix/pipe.lo
file_io/unix/open.lo file_io/unix/filestat.lo file_io/unix/copy.lo
file_io/unix/fileacc.lo file_io/unix/fullrw.lo locks/unix/thread_rwlock.lo
locks/unix/thread_mutex.lo locks/unix/thread_cond.lo
locks/unix/proc_mutex.lo locks/unix/global_mutex.lo memory/unix/apr_pools.lo
misc/unix/charset.lo misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo
misc/unix/start.lo misc/unix/errorcodes.lo misc/unix/getopt.lo
misc/unix/otherchild.lo mmap/unix/mmap.lo mmap/unix/common.lo
network_io/unix/sockaddr.lo network_io/unix/sockopt.lo
network_io/unix/sendrecv.lo network_io/unix/multicast.lo
network_io/unix/sockets.lo network_io/unix/inet_ntop.lo
network_io/unix/inet_pton.lo poll/unix/epoll.lo poll/unix/select.lo
poll/unix/poll.lo poll/unix/port.lo poll/unix/kqueue.lo random/unix/sha2.lo
random/unix/apr_random.lo random/unix/sha2_glue.lo shmem/unix/shm.lo
support/unix/waitio.lo threadproc/unix/procsup.lo threadproc/unix/thread.lo
threadproc/unix/signals.lo threadproc/unix/proc.lo
threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
-lpthread

libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object

make[2]: *** [libapr-1.la] Error 1

make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make[1]: *** [all-recursive] Error 1

make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make: *** [libs/apr/libapr-1.la] Error 2

/bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link  gcc
-g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
-I./include -I/usr/local/src/freeswitch/libs/apr/include/arch/unix
-I./include/arch/unix -I/usr/local/src/freeswitch/libs/apr/include
-version-info 2:8:2    -o libapr-1.la -rpath /usr/local/freeswitch/lib
passwd/apr_getpass.lo strings/apr_cpystrn.lo strings/apr_strnatcmp.lo
strings/apr_strings.lo strings/apr_strtok.lo strings/apr_fnmatch.lo
strings/apr_snprintf.lo tables/apr_tables.lo tables/apr_hash.lo
atomic/unix/apr_atomic.lo dso/unix/dso.lo file_io/unix/flock.lo
file_io/unix/readwrite.lo file_io/unix/filepath_util.lo file_io/unix/seek.lo
file_io/unix/dir.lo file_io/unix/mktemp.lo file_io/unix/filedup.lo
file_io/unix/tempdir.lo file_io/unix/filepath.lo file_io/unix/pipe.lo
file_io/unix/open.lo file_io/unix/filestat.lo file_io/unix/copy.lo
file_io/unix/fileacc.lo file_io/unix/fullrw.lo locks/unix/thread_rwlock.lo
locks/unix/thread_mutex.lo locks/unix/thread_cond.lo
locks/unix/proc_mutex.lo locks/unix/global_mutex.lo memory/unix/apr_pools.lo
misc/unix/charset.lo misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo
misc/unix/start.lo misc/unix/errorcodes.lo misc/unix/getopt.lo
misc/unix/otherchild.lo mmap/unix/mmap.lo mmap/unix/common.lo
network_io/unix/sockaddr.lo network_io/unix/sockopt.lo
network_io/unix/sendrecv.lo network_io/unix/multicast.lo
network_io/unix/sockets.lo network_io/unix/inet_ntop.lo
network_io/unix/inet_pton.lo poll/unix/epoll.lo poll/unix/select.lo
poll/unix/poll.lo poll/unix/port.lo poll/unix/kqueue.lo random/unix/sha2.lo
random/unix/apr_random.lo random/unix/sha2_glue.lo shmem/unix/shm.lo
support/unix/waitio.lo threadproc/unix/procsup.lo threadproc/unix/thread.lo
threadproc/unix/signals.lo threadproc/unix/proc.lo
threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
-lpthread

libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object

make[2]: *** [libapr-1.la] Error 1

make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make[1]: *** [all-recursive] Error 1

make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make: *** [libs/apr/libapr-1.la] Error 2

[1]+  Exit 2                  make

 

What next guys?

 

Sean

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From Sirish.MasurMohan at oa.com.au  Wed Dec 19 06:39:58 2012
From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan)
Date: Wed, 19 Dec 2012 14:39:58 +1100
Subject: [Freeswitch-users] Newbie question: Executing Lua scripts for
 incoming calls, how concurrency is to be handled?
In-Reply-To: <CAKzWOxUYyfZ_8dLgy_t-Ov2a+YXEDj04dDkdLSn14Q-tZk-8pQ@mail.gmail.com>
References: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au>
	<50CE5D6E.2020401@quentustech.com>
	<965759A53E43FE439E43565A7715E5F058F4156D32@oa-exchange1.oa.com.au>
	<CAKzWOxWFh3RNYg9NY4Dqa-=f9KZOaXTvM8NGzHpvEvzV16VUWA@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156DD8@oa-exchange1.oa.com.au>
	<CAKzWOxUYyfZ_8dLgy_t-Ov2a+YXEDj04dDkdLSn14Q-tZk-8pQ@mail.gmail.com>
Message-ID: <965759A53E43FE439E43565A7715E5F058F4156DF7@oa-exchange1.oa.com.au>

Hi Michael,

>> How are you physically connecting from FreeSWITCH to the PBX?

I connect this via E1 link - I have a Sangoma card installed on the FreeSWITCH machine.

With regards,
Sirish

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Wednesday, 19 December 2012 1:31 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

To trigger SIP registrations you'd need the PBX to have a SIP client. I'm assuming this is possible, but maybe that's a false assumption. How are you physically connecting from FreeSWITCH to the PBX?

-MC
On Tue, Dec 18, 2012 at 2:36 PM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au<mailto:Sirish.MasurMohan at oa.com.au>> wrote:
Hey Guys,

Would really appreciate if you could help me out here - isn't there a way to handle concurrent calls in the dial plan, especially when Lua scripts, accessing global variables, are executed on receiving calls?

Is mod_fifo the closest I could get to handle concurrency (as Michael has explained)? If yes, how do I trigger SIP registrations, especially working with a PBX which I don't have full control of?

With regards,
Sirish

From: Sirish Masur Mohan
Sent: Tuesday, 18 December 2012 1:02 PM
To: FreeSWITCH Users Help
Subject: RE: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

Hi Michael,

Thanks for the reply.

>> You would need a SIP registration from the PBX to FreeSWITCH for each of the modem extensions
I am seen SIP clients such as X-Lite sending out the SIP registrations, but could you please clarify as to how this can be achieved in the PBX? The final production environment would be out in the customer's PBX, which I may not have complete control of..

>> What application are you building?
I may not be able to provide the details because of the NDA with customer, but what I am trying to achieve is, to replace an existing IVR with FreeSWITCH in an old existing setup of the customer - that's the reason why we continue working with dialup modems!

With regards,
Sirish

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: Tuesday, 18 December 2012 6:23 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

You don't have to have actual human agents for mod_fifo. You could define a user for each modem and then manually "log in" those "agents" on the command line using the fifo_member API command. Something like this:

fifo_member add fifo_name {fifo_member_wait=nowait}user/1234

Where 1234 is the user id of one of the modems. You would need a SIP registration from the PBX to FreeSWITCH for each of the modem extensions.

Having modems go through a VoIP system sounds a bit scary. What application are you building?
-MC
On Sun, Dec 16, 2012 at 5:41 PM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au<mailto:Sirish.MasurMohan at oa.com.au>> wrote:
Hi William,

Thanks for the reply.

My setup is as follows:
Client(Caller)->dialup modem->PBX->FreeSWITCH->PBX-> 4 dialup modems->Server(Receiver)

I basically need FreeSWITCH to bridge the incoming call to the best external destination (out of the 4 available), so that the modem training, connection etc can take place smoothly, before exchange of data. I am not sure if mod_fifo would help me in this scenario, as, I would require an agent to dial in and read the fifo. Could you please clarify?

Thanks!

With regards,
Sirish

-----Original Message-----
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 William King
Sent: Monday, 17 December 2012 10:47 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

Sounds like you want to take a look into mod_fifo.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337<tel:%28877%29%20211-9337>
Office: (206) 388-4772<tel:%28206%29%20388-4772>
Cell:   (253) 686-5518<tel:%28253%29%20686-5518>
william.king at quentustech.com<mailto:william.king at quentustech.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

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From nandy1925 at gmail.com  Wed Dec 19 06:46:12 2012
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Wed, 19 Dec 2012 11:46:12 +0800
Subject: [Freeswitch-users] Mutiple GSMopen USB modem setup
In-Reply-To: <CAMjaK3D0Xw4cnXeO1gQr-9+MD1JCuq1m6Xk3qKS0YFQDm6=z5g@mail.gmail.com>
References: <CAHKqMb6OBRVtOkEepF-a5fVXz_x-NbvhDijvU+dgCBsGcQ5sKg@mail.gmail.com>
	<CALXCt0pT2UoMWZ0dUPC2zKTbm_7zd2Z_2UAr1e8i78JO24uXTw@mail.gmail.com>
	<CAHKqMb69d22pUBqDEw16MwEh_EDR-wMG=j78zYKGFq1ubGZ0DA@mail.gmail.com>
	<CAMjaK3D0Xw4cnXeO1gQr-9+MD1JCuq1m6Xk3qKS0YFQDm6=z5g@mail.gmail.com>
Message-ID: <CAHKqMb5h3VzjqHiUUPeicuNh1U-a+WnjaPf-RL1GiS51YtXhMw@mail.gmail.com>

Hi Giovanni,

I have not tested it yet. As mentioned by Dmitry, modem serial numbers are
not accessible. Perhaps, mapping the serial port based on the USB port
where they're connected would, at least, a workable solution somewhat as
discussed in this post:

http://stackoverflow.com/questions/4800099/how-to-identify-multiple-usb-serial-adapters-under-ubuntu-10-1

Rgds,
Nandy

On Tue, Dec 18, 2012 at 9:03 PM, Dmitry Lysenko
<dvl36.ripe.nick at gmail.com>wrote:

> Hi!
> Huawei USB modems does not have accessible via USB protocol serial
> numbers. At least E1550, E153 and E171.
> Best wishes,
>  Dmitry.
>
>
> 2012/12/16 Nandy Dagondon <nandy1925 at gmail.com>
>
>> Thanks for the feedback Giovanni.  Re fixing the ttyUSB port assignment,
>> I have searched a guy made a script to scan USB serial numbers using udev,
>> then created symbolic links. Will try it out later.
>> /Nandy
>>
>>
>>
>> On Thu, Dec 13, 2012 at 7:02 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>>
>>> On Thu, Dec 13, 2012 at 4:44 AM, Nandy Dagondon <nandy1925 at gmail.com>
>>> wrote:
>>> > I have run GSMOpen on CentOS 5.7 defying the Wiki advisory. Questions:
>>>
>>> I would invite you to go for the OS distros detailed in the wiki page.
>>> You'll probably encounter problems with different distros, and you're
>>> on your own to solve it.
>>>
>>> > 1. What is the maximum number of USB modems tested? Can we get the
>>> numbers
>>> > and the CPU used?
>>>
>>> I've heard about 48 concurrent, and 64. Me personally have tested with
>>> 5. No CPU consumption. The critical part is the USB BUS. So use
>>> cascading and POWERED good usb 2.0 hubs
>>>
>>> > 2. I'll be installing multiple modems each connected to a different
>>> mobile
>>> > network. Is the /dev/ttyUSB assignments  constant for every modem?
>>> Meaning
>>> > it doesn't change if I plug it on different USB jacks.
>>>
>>> it will change not only if you change USB port, but also randomly if
>>> you stay on the same USB port and reboot (and sometimes also without
>>> rebooting). That's a "feature" of Linux distros (a demented one,
>>> cannot understand why they choose this behavior).
>>>
>>> Soon or later I'll look into this, and come out with a solution (I've
>>> made some preliminary research and reasoning about in the past).
>>>
>>> If you have a commercial interest in that, and a real budget for it,
>>> contact me in private as consultant, or put a public bounty on it.
>>>
>>> -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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From babak.freeswitch at gmail.com  Wed Dec 19 08:33:58 2012
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Wed, 19 Dec 2012 09:03:58 +0330
Subject: [Freeswitch-users] attended transfer to rxfax
In-Reply-To: <CAKzWOxVxXcmfY+y-3iE89HDFS7qXX66pycSv12z4SkhvPni0zg@mail.gmail.com>
References: <CAEVfcaZukd+gs3ReiS=ro-K3rTGKgEoHmNn_QWqwgN+Uks2iOw@mail.gmail.com>
	<CAKzWOxVxXcmfY+y-3iE89HDFS7qXX66pycSv12z4SkhvPni0zg@mail.gmail.com>
Message-ID: <CAEVfcaacGan3A8m=iKONXaBoGqpuUen9PcjkSACptbV2OfkVDw@mail.gmail.com>

user is using cisco ip phones which only support attended transfer

On Wed, Dec 19, 2012 at 6:09 AM, Michael Collins <msc at freeswitch.org> wrote:

> Why do you need to do an attended transfer? If the fax extension is just
> going to do rxfax then perform a blind transfer and be done with it.
> -MC
>
> On Tue, Dec 18, 2012 at 2:56 AM, Babak Yakhchali <
> babak.freeswitch at gmail.com> wrote:
>
>> Hi
>> How can I implement this:
>> user calls to a number which starts sending fax after user presses 1
>> (channel A)
>> after user hears fax signaling starts to attended transfer to a local
>> extension which executes rxfax (Channel B)
>> after user hears local extension rxfax signaling presses transfer to
>> complete the transfer
>> but no fax is received!!
>> I trace logs and see that channel B ends after user presses transfer. it
>> seems I need to create a C channel which is running txfax but I donno how
>> to do it
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 andrew at cassidywebservices.co.uk  Wed Dec 19 11:53:09 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 19 Dec 2012 08:53:09 +0000
Subject: [Freeswitch-users] attended transfer to rxfax
In-Reply-To: <CAEVfcaacGan3A8m=iKONXaBoGqpuUen9PcjkSACptbV2OfkVDw@mail.gmail.com>
References: <CAEVfcaZukd+gs3ReiS=ro-K3rTGKgEoHmNn_QWqwgN+Uks2iOw@mail.gmail.com>
	<CAKzWOxVxXcmfY+y-3iE89HDFS7qXX66pycSv12z4SkhvPni0zg@mail.gmail.com>
	<CAEVfcaacGan3A8m=iKONXaBoGqpuUen9PcjkSACptbV2OfkVDw@mail.gmail.com>
Message-ID: <CAMbW9oL+mfp4KmHynLbYb=SJ=qOPzH84EgKHXZvD+X1ptn0VCQ@mail.gmail.com>

Which model? All the phones I've ever used support both attended and blind
transfer. You may need to press the right arrow button on SPA50x models to
see the bxfer button.

You can also use detect_fax_tones to have freeswitch automatically transfer
the call for you.

On 19 December 2012 05:33, Babak Yakhchali <babak.freeswitch at gmail.com>wrote:

> user is using cisco ip phones which only support attended transfer
>
>
> On Wed, Dec 19, 2012 at 6:09 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Why do you need to do an attended transfer? If the fax extension is just
>> going to do rxfax then perform a blind transfer and be done with it.
>> -MC
>>
>> On Tue, Dec 18, 2012 at 2:56 AM, Babak Yakhchali <
>> babak.freeswitch at gmail.com> wrote:
>>
>>> Hi
>>> How can I implement this:
>>> user calls to a number which starts sending fax after user presses 1
>>> (channel A)
>>> after user hears fax signaling starts to attended transfer to a local
>>> extension which executes rxfax (Channel B)
>>> after user hears local extension rxfax signaling presses transfer to
>>> complete the transfer
>>> but no fax is received!!
>>> I trace logs and see that channel B ends after user presses transfer. it
>>> seems I need to create a C channel which is running txfax but I donno how
>>> to do it
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 steveayre at gmail.com  Wed Dec 19 12:13:18 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 19 Dec 2012 09:13:18 +0000
Subject: [Freeswitch-users] New install 1.2.Stable blows up compiling
	mod_perl.so
In-Reply-To: <CCF67B69.62430%krice@freeswitch.org>
References: <CAKzWOxV6-8tu3a5A6n47hG=upWBGimJT+cL3mXSO+caFercZbQ@mail.gmail.com>
	<CCF67B69.62430%krice@freeswitch.org>
Message-ID: <CAFiqYumD67ok6XVrPzkNwMmVH-w56UK-0-XZ4oQ-0Qt-=71WVQ@mail.gmail.com>

>
> /usr/bin/ld: cannot find -lgdbm

**

You're missing the gdbm-devel dependancy.

If you see 'cannot find -lXXX' you will usually need to install XXX-devel
or libXXX-devel.

-Steve


On 19 December 2012 01:58, Ken Rice <krice at freeswitch.org> wrote:

>  Back in the day I ran a large multi-line pay BBS/ISP.... We used this
> software called MajorBBS to run the whole thing... It used these loadable
> modules, not much different then freeswitch modules we use today, the only
> problem was they needed to be loaded in a specific order, the only way to
> get that order correct was moving the modules out of and back into the
> directory they lived in until you got them to load right (they loaded based
> on unsorted directory order)... So as with most hard drives back in the
> day, I had to run a defrag regularly which always screwed up the load
> order... After a few pints one night I decide its a good time to do the
> routine defrag, etc...
>
> Needless to say after about an hour of screwing with the system, I?ve
> decided I had enough, I would just restore the backup I made of the FAT
> tables and let it sit til later... Only problem I had defragged since the
> last FAT back up... ie: the files were no longer in the same spot on hdd...
> Needless to say there were a lotta people angry they couldn?t get onto the
> system for about 3 days while I unscrewed things and tried to work my day
> job also
>
> K
>
>
> On 12/18/12 7:47 PM, "Michael Collins" <msc at freeswitch.org> wrote:
>
>
>
> On Tue, Dec 18, 2012 at 5:32 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>
> LOL  live and learn.  They are rebuilding my Centos 5.8 now.
> Thanks.
>
>
> Just remember we're laughing *with* you, not at you, because pretty much
> everyone one of us has a story to tell. "One time I had this critical
> server in place and I typed rm -fr * when I meant to type cd /tmp && rm -fr
> * ..."
>
> In fact, on tomorrow's conference call we should have a contest to see who
> has done the biggest CLI woopsie daisy of them all. My money's on Ken Rice
> - that guy has some mad skillz. :)
>
> -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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at celliax.org  Wed Dec 19 12:20:58 2012
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 19 Dec 2012 10:20:58 +0100
Subject: [Freeswitch-users] Mutiple GSMopen USB modem setup
In-Reply-To: <CAHKqMb5h3VzjqHiUUPeicuNh1U-a+WnjaPf-RL1GiS51YtXhMw@mail.gmail.com>
References: <CAHKqMb6OBRVtOkEepF-a5fVXz_x-NbvhDijvU+dgCBsGcQ5sKg@mail.gmail.com>
	<CALXCt0pT2UoMWZ0dUPC2zKTbm_7zd2Z_2UAr1e8i78JO24uXTw@mail.gmail.com>
	<CAHKqMb69d22pUBqDEw16MwEh_EDR-wMG=j78zYKGFq1ubGZ0DA@mail.gmail.com>
	<CAMjaK3D0Xw4cnXeO1gQr-9+MD1JCuq1m6Xk3qKS0YFQDm6=z5g@mail.gmail.com>
	<CAHKqMb5h3VzjqHiUUPeicuNh1U-a+WnjaPf-RL1GiS51YtXhMw@mail.gmail.com>
Message-ID: <CALXCt0rbArYwSOhfKYNv2oXWNe7gmUUObUONTJ8Fa=w-ct205Q@mail.gmail.com>

Nandy and Dimitry,

please let us know your findings.

Thanks for sharing!

-giovanni

On Wed, Dec 19, 2012 at 4:46 AM, Nandy Dagondon <nandy1925 at gmail.com> wrote:
> Hi Giovanni,
>
> I have not tested it yet. As mentioned by Dmitry, modem serial numbers are
> not accessible. Perhaps, mapping the serial port based on the USB port where
> they're connected would, at least, a workable solution somewhat as discussed
> in this post:
>
> http://stackoverflow.com/questions/4800099/how-to-identify-multiple-usb-serial-adapters-under-ubuntu-10-1
>
> Rgds,
> Nandy
>
>
> On Tue, Dec 18, 2012 at 9:03 PM, Dmitry Lysenko <dvl36.ripe.nick at gmail.com>
> wrote:
>>
>> Hi!
>> Huawei USB modems does not have accessible via USB protocol serial
>> numbers. At least E1550, E153 and E171.
>> Best wishes,
>>  Dmitry.
>>
>>
>> 2012/12/16 Nandy Dagondon <nandy1925 at gmail.com>
>>>
>>> Thanks for the feedback Giovanni.  Re fixing the ttyUSB port assignment,
>>> I have searched a guy made a script to scan USB serial numbers using udev,
>>> then created symbolic links. Will try it out later.
>>> /Nandy
>>>
>>>
>>>
>>> On Thu, Dec 13, 2012 at 7:02 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>>
>>>> On Thu, Dec 13, 2012 at 4:44 AM, Nandy Dagondon <nandy1925 at gmail.com>
>>>> wrote:
>>>> > I have run GSMOpen on CentOS 5.7 defying the Wiki advisory. Questions:
>>>>
>>>> I would invite you to go for the OS distros detailed in the wiki page.
>>>> You'll probably encounter problems with different distros, and you're
>>>> on your own to solve it.
>>>>
>>>> > 1. What is the maximum number of USB modems tested? Can we get the
>>>> > numbers
>>>> > and the CPU used?
>>>>
>>>> I've heard about 48 concurrent, and 64. Me personally have tested with
>>>> 5. No CPU consumption. The critical part is the USB BUS. So use
>>>> cascading and POWERED good usb 2.0 hubs
>>>>
>>>> > 2. I'll be installing multiple modems each connected to a different
>>>> > mobile
>>>> > network. Is the /dev/ttyUSB assignments  constant for every modem?
>>>> > Meaning
>>>> > it doesn't change if I plug it on different USB jacks.
>>>>
>>>> it will change not only if you change USB port, but also randomly if
>>>> you stay on the same USB port and reboot (and sometimes also without
>>>> rebooting). That's a "feature" of Linux distros (a demented one,
>>>> cannot understand why they choose this behavior).
>>>>
>>>> Soon or later I'll look into this, and come out with a solution (I've
>>>> made some preliminary research and reasoning about in the past).
>>>>
>>>> If you have a commercial interest in that, and a real budget for it,
>>>> contact me in private as consultant, or put a public bounty on it.
>>>>
>>>> -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
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From babak.freeswitch at gmail.com  Wed Dec 19 13:06:40 2012
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Wed, 19 Dec 2012 13:36:40 +0330
Subject: [Freeswitch-users] attended transfer to rxfax
In-Reply-To: <CAMbW9oL+mfp4KmHynLbYb=SJ=qOPzH84EgKHXZvD+X1ptn0VCQ@mail.gmail.com>
References: <CAEVfcaZukd+gs3ReiS=ro-K3rTGKgEoHmNn_QWqwgN+Uks2iOw@mail.gmail.com>
	<CAKzWOxVxXcmfY+y-3iE89HDFS7qXX66pycSv12z4SkhvPni0zg@mail.gmail.com>
	<CAEVfcaacGan3A8m=iKONXaBoGqpuUen9PcjkSACptbV2OfkVDw@mail.gmail.com>
	<CAMbW9oL+mfp4KmHynLbYb=SJ=qOPzH84EgKHXZvD+X1ptn0VCQ@mail.gmail.com>
Message-ID: <CAEVfcaYjoH8K2uyRS20wFWFXg4cKj41FDP2ZOr+GG19gPemgBg@mail.gmail.com>

79xx series. like 7911 7942 ...

On Wed, Dec 19, 2012 at 12:23 PM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> Which model? All the phones I've ever used support both attended and blind
> transfer. You may need to press the right arrow button on SPA50x models to
> see the bxfer button.
>
> You can also use detect_fax_tones to have freeswitch automatically
> transfer the call for you.
>
>
> On 19 December 2012 05:33, Babak Yakhchali <babak.freeswitch at gmail.com>wrote:
>
>> user is using cisco ip phones which only support attended transfer
>>
>>
>> On Wed, Dec 19, 2012 at 6:09 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Why do you need to do an attended transfer? If the fax extension is just
>>> going to do rxfax then perform a blind transfer and be done with it.
>>> -MC
>>>
>>> On Tue, Dec 18, 2012 at 2:56 AM, Babak Yakhchali <
>>> babak.freeswitch at gmail.com> wrote:
>>>
>>>> Hi
>>>> How can I implement this:
>>>> user calls to a number which starts sending fax after user presses 1
>>>> (channel A)
>>>> after user hears fax signaling starts to attended transfer to a local
>>>> extension which executes rxfax (Channel B)
>>>> after user hears local extension rxfax signaling presses transfer to
>>>> complete the transfer
>>>> but no fax is received!!
>>>> I trace logs and see that channel B ends after user presses transfer.
>>>> it seems I need to create a C channel which is running txfax but I donno
>>>> how to do it
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>>>
>>>>
>>>
>>>
>>> --
>>> Michael S Collins
>>> Twitter: @mercutioviz
>>> http://www.FreeSWITCH.org
>>> http://www.ClueCon.com
>>> http://www.OSTAG.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Dec 19 13:24:31 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 19 Dec 2012 10:24:31 +0000
Subject: [Freeswitch-users] attended transfer to rxfax
In-Reply-To: <CAEVfcaYjoH8K2uyRS20wFWFXg4cKj41FDP2ZOr+GG19gPemgBg@mail.gmail.com>
References: <CAEVfcaZukd+gs3ReiS=ro-K3rTGKgEoHmNn_QWqwgN+Uks2iOw@mail.gmail.com>
	<CAKzWOxVxXcmfY+y-3iE89HDFS7qXX66pycSv12z4SkhvPni0zg@mail.gmail.com>
	<CAEVfcaacGan3A8m=iKONXaBoGqpuUen9PcjkSACptbV2OfkVDw@mail.gmail.com>
	<CAMbW9oL+mfp4KmHynLbYb=SJ=qOPzH84EgKHXZvD+X1ptn0VCQ@mail.gmail.com>
	<CAEVfcaYjoH8K2uyRS20wFWFXg4cKj41FDP2ZOr+GG19gPemgBg@mail.gmail.com>
Message-ID: <CAMbW9oLGjiwqkcyGLeSKsH4+1mVfW7x893yubkWmJwW402_wPA@mail.gmail.com>

I have a 7912 at home, I'll have a look later for you.

On 19 December 2012 10:06, Babak Yakhchali <babak.freeswitch at gmail.com>wrote:

> 79xx series. like 7911 7942 ...
>
>
> On Wed, Dec 19, 2012 at 12:23 PM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
>> Which model? All the phones I've ever used support both attended and
>> blind transfer. You may need to press the right arrow button on SPA50x
>> models to see the bxfer button.
>>
>> You can also use detect_fax_tones to have freeswitch automatically
>> transfer the call for you.
>>
>>
>> On 19 December 2012 05:33, Babak Yakhchali <babak.freeswitch at gmail.com>wrote:
>>
>>> user is using cisco ip phones which only support attended transfer
>>>
>>>
>>> On Wed, Dec 19, 2012 at 6:09 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Why do you need to do an attended transfer? If the fax extension is
>>>> just going to do rxfax then perform a blind transfer and be done with it.
>>>> -MC
>>>>
>>>> On Tue, Dec 18, 2012 at 2:56 AM, Babak Yakhchali <
>>>> babak.freeswitch at gmail.com> wrote:
>>>>
>>>>> Hi
>>>>> How can I implement this:
>>>>> user calls to a number which starts sending fax after user presses 1
>>>>> (channel A)
>>>>> after user hears fax signaling starts to attended transfer to a local
>>>>> extension which executes rxfax (Channel B)
>>>>> after user hears local extension rxfax signaling presses transfer to
>>>>> complete the transfer
>>>>> but no fax is received!!
>>>>> I trace logs and see that channel B ends after user presses transfer.
>>>>> it seems I need to create a C channel which is running txfax but I donno
>>>>> how to do it
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting 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
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Michael S Collins
>>>> Twitter: @mercutioviz
>>>> http://www.FreeSWITCH.org
>>>> http://www.ClueCon.com
>>>> http://www.OSTAG.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
*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 mickstevens at yahoo.com  Wed Dec 19 13:38:28 2012
From: mickstevens at yahoo.com (Mick Stevens)
Date: Wed, 19 Dec 2012 02:38:28 -0800 (PST)
Subject: [Freeswitch-users] Fax *.tiff file deleted after call completion?
Message-ID: <1355913508.48869.YahooMailNeo@web160802.mail.bf1.yahoo.com>

Hi FS Folks,

When testing fax receipt as per the wiki (?http://wiki.freeswitch.org/wiki/Mod_spandsp ) I "see/hear" the fax being sent & the *.tif file appear (briefly! :-) ?) in /tmp... But when the call has completed the *.tif file in /tmp disappears!?

I have tried incoming DDI's configured to my fax extension from two different carriers with the same result, both in terms of user experience & associated traces, so am "assuming" it's an FS problem (or, more accurately, a problem with MY fs config).

Any ideas guys, anybody else experienced this before?

Assoc. info...

fs extension config:

<extension name="test_rxfax_stream">
 ? ? <condition field="destination_number" expression="0123456789">
? ? ? ?<action application="log" data="Test CarrierX DDI for Fax ${destination_number}"/>
 ? ? ? <action application="answer" />
 ? ? ? <action application="playback" data="silence_stream://2000"/>
? ? ? ?<action application="set" data="fax_enable_t38_request=true"/>
? ? ? ?<action application="set" data="fax_enable_t38=true"/>
 ? ? ? <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/>
 ? ? ? <action application="hangup"/>
 ? ? </condition>
 ? </extension>

In /tmp one minute... (observation: the fax file appears very small?)

root at fs:/tmp# ls -la
total 12412
drwxrwxrwt ?4 root ? ? root ? ? ? ? 4096 2012-12-19 10:07 .
drwxr-xr-x 27 root ? ? root ? ? ? ? 4096 2012-12-17 12:27 ..
-rw-r--r-- ?1 www-data www-data ? ? 1422 2012-12-19 10:07 213.146.146.70-mysql_cacti_stats.txt
-rw-r--r-- ?1 root ? ? root ? ? ? ?12892 2012-12-17 11:34 a_b2747e50-e438-455f-8364-65ad580891ee.cdr.xml
-rw-r--r-- ?1 root ? ? root ? ? 12652544 2012-12-19 10:07 fax_191212.pcap
-rw-r--r-- ?1 root ? ? root ? ? ? ? ? ?8 2012-12-19 10:07 FAX-fc034321-a3be-495e-ac42-655f16fceea2.tif
drwxrwxrwt ?2 root ? ? root ? ? ? ? 4096 2012-12-10 09:12 .ICE-unix
-rw-r--r-- ?1 root ? ? root ? ? ? ? ? ?2 2012-12-19 10:00 Last_IO_Errno.txt
-rw-r--r-- ?1 root ? ? root ? ? ? ? ? ?2 2012-12-19 10:00 Last_SQL_Errno.txt
-rw-r--r-- ?1 root ? ? root ? ? ? ? ? ?4 2012-12-19 10:00 Slave_IO_Running.txt
-rw-r--r-- ?1 root ? ? root ? ? ? ? ? ?4 2012-12-19 10:00 Slave_SQL_Running.txt
drwxrwxrwt ?2 root ? ? root ? ? ? ? 4096 2012-12-10 09:12 .X11-unix

In /tmp the next! has?FAX-fc034321-a3be-495e-ac42-655f16fceea2.tif?disappeared!? :-(

root at n01s01:/tmp# ls -la
total 12412
drwxrwxrwt ?4 root ? ? root ? ? ? ? 4096 2012-12-19 10:07 .
drwxr-xr-x 27 root ? ? root ? ? ? ? 4096 2012-12-17 12:27 ..
-rw-r--r-- ?1 www-data www-data ? ? 1422 2012-12-19 10:07 213.146.146.70-mysql_cacti_stats.txt
-rw-r--r-- ?1 root ? ? root ? ? ? ?12892 2012-12-17 11:34 a_b2747e50-e438-455f-8364-65ad580891ee.cdr.xml
-rw-r--r-- ?1 root ? ? root ? ? 12656640 2012-12-19 10:07 fax_191212.pcap
drwxrwxrwt ?2 root ? ? root ? ? ? ? 4096 2012-12-10 09:12 .ICE-unix
-rw-r--r-- ?1 root ? ? root ? ? ? ? ? ?2 2012-12-19 10:00 Last_IO_Errno.txt
-rw-r--r-- ?1 root ? ? root ? ? ? ? ? ?2 2012-12-19 10:00 Last_SQL_Errno.txt
-rw-r--r-- ?1 root ? ? root ? ? ? ? ? ?4 2012-12-19 10:00 Slave_IO_Running.txt
-rw-r--r-- ?1 root ? ? root ? ? ? ? ? ?4 2012-12-19 10:00 Slave_SQL_Running.txt
drwxrwxrwt ?2 root ? ? root ? ? ? ? 4096 2012-12-10 09:12 .X11-unix

pcap traces available on request - I see lots of t30 indicators in the traces & also data:v21? Not sure whether this is correct or not? I have tried enabling/disabling T38 to see if this makes any difference to no avail.

Any guidance gratefully received!

Festiveness....

?
Rgds, Mick
Tel/SMS. +44(0)7967 594432
Fax. +44(0)7053 452429

Email/IM. mickstevens at yahoo.com
Skype: mick_stevens
www.facebook.com/mickstevens
www.twitter.com/mickstevens
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From Alexander.Haugg at c4b.de  Wed Dec 19 14:13:52 2012
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Wed, 19 Dec 2012 11:13:52 +0000
Subject: [Freeswitch-users] build error with new Freeswitch 'nametab.h':
 No such file or directory
In-Reply-To: <1355844703594-7585673.post@n2.nabble.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC837D409@vc4bex2.corp.c4b.de>
	<1355844703594-7585673.post@n2.nabble.com>
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC837D614@vc4bex2.corp.c4b.de>

No, 

i checked out in a new folder with the command "git clone -v git://git.freeswitch.org/freeswitch.git C:/my_fs/"

thanks

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jeff Lenk
Gesendet: Dienstag, 18. Dezember 2012 16:32
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] build error with new Freeswitch 'nametab.h': No such file or directory

Was an older build present before updating. You may need to do a git clean -fdx.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/build-error-with-new-Freeswitch-nametab-h-No-such-file-or-directory-tp7585669p7585673.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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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  Wed Dec 19 14:52:27 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 19 Dec 2012 06:52:27 -0500
Subject: [Freeswitch-users] Install 1.2.Stable still fails.  Ver 2
In-Reply-To: <5fdf01cddd96$9670f330$c352d990$@bizfocused.com>
References: <5fdf01cddd96$9670f330$c352d990$@bizfocused.com>
Message-ID: <61ad01cddddf$51cfa2a0$f56ee7e0$@bizfocused.com>

Anybody?

 

This is a brand new install on a new install of Centos5.8

 

The error is:

libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object

/usr/local/src/freeswitch/libs/apr/ passwd/apr_getpass.lo exists but libtool
does not like it.

 

Full error listed below.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Devoy
Sent: Tuesday, December 18, 2012 10:12 PM
To: FreeSWITCH-users at lists.freeswitch.org
Subject: [Freeswitch-users] Install 1.2.Stable still fails. Ver 2

 

OK, I had my Centos restored to base install 5.8.

 

The install fails.  Here is EXACTLY what I did:

 

Login to root
cd /usr/local/src

rpm -Uvh
http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm


git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git

cd /usr/local/src/freeswitch

./bootstrap.sh

./configure

make & make install

 

 

It fails with:

/bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link  gcc
-g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
-I./include -I/usr/local/src/freeswitch/libs/apr/include/arch/unix
-I./include/arch/unix -I/usr/local/src/freeswitch/libs/apr/include
-version-info 2:8:2    -o libapr-1.la -rpath /usr/local/freeswitch/lib
passwd/apr_getpass.lo strings/apr_cpystrn.lo strings/apr_strnatcmp.lo
strings/apr_strings.lo strings/apr_strtok.lo strings/apr_fnmatch.lo
strings/apr_snprintf.lo tables/apr_tables.lo tables/apr_hash.lo
atomic/unix/apr_atomic.lo dso/unix/dso.lo file_io/unix/flock.lo
file_io/unix/readwrite.lo file_io/unix/filepath_util.lo file_io/unix/seek.lo
file_io/unix/dir.lo file_io/unix/mktemp.lo file_io/unix/filedup.lo
file_io/unix/tempdir.lo file_io/unix/filepath.lo file_io/unix/pipe.lo
file_io/unix/open.lo file_io/unix/filestat.lo file_io/unix/copy.lo
file_io/unix/fileacc.lo file_io/unix/fullrw.lo locks/unix/thread_rwlock.lo
locks/unix/thread_mutex.lo locks/unix/thread_cond.lo
locks/unix/proc_mutex.lo locks/unix/global_mutex.lo memory/unix/apr_pools.lo
misc/unix/charset.lo misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo
misc/unix/start.lo misc/unix/errorcodes.lo misc/unix/getopt.lo
misc/unix/otherchild.lo mmap/unix/mmap.lo mmap/unix/common.lo
network_io/unix/sockaddr.lo network_io/unix/sockopt.lo
network_io/unix/sendrecv.lo network_io/unix/multicast.lo
network_io/unix/sockets.lo network_io/unix/inet_ntop.lo
network_io/unix/inet_pton.lo poll/unix/epoll.lo poll/unix/select.lo
poll/unix/poll.lo poll/unix/port.lo poll/unix/kqueue.lo random/unix/sha2.lo
random/unix/apr_random.lo random/unix/sha2_glue.lo shmem/unix/shm.lo
support/unix/waitio.lo threadproc/unix/procsup.lo threadproc/unix/thread.lo
threadproc/unix/signals.lo threadproc/unix/proc.lo
threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
-lpthread

libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object

make[2]: *** [libapr-1.la] Error 1

make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make[1]: *** [all-recursive] Error 1

make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make: *** [libs/apr/libapr-1.la] Error 2

/bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link  gcc
-g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
-I./include -I/usr/local/src/freeswitch/libs/apr/include/arch/unix
-I./include/arch/unix -I/usr/local/src/freeswitch/libs/apr/include
-version-info 2:8:2    -o libapr-1.la -rpath /usr/local/freeswitch/lib
passwd/apr_getpass.lo strings/apr_cpystrn.lo strings/apr_strnatcmp.lo
strings/apr_strings.lo strings/apr_strtok.lo strings/apr_fnmatch.lo
strings/apr_snprintf.lo tables/apr_tables.lo tables/apr_hash.lo
atomic/unix/apr_atomic.lo dso/unix/dso.lo file_io/unix/flock.lo
file_io/unix/readwrite.lo file_io/unix/filepath_util.lo file_io/unix/seek.lo
file_io/unix/dir.lo file_io/unix/mktemp.lo file_io/unix/filedup.lo
file_io/unix/tempdir.lo file_io/unix/filepath.lo file_io/unix/pipe.lo
file_io/unix/open.lo file_io/unix/filestat.lo file_io/unix/copy.lo
file_io/unix/fileacc.lo file_io/unix/fullrw.lo locks/unix/thread_rwlock.lo
locks/unix/thread_mutex.lo locks/unix/thread_cond.lo
locks/unix/proc_mutex.lo locks/unix/global_mutex.lo memory/unix/apr_pools.lo
misc/unix/charset.lo misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo
misc/unix/start.lo misc/unix/errorcodes.lo misc/unix/getopt.lo
misc/unix/otherchild.lo mmap/unix/mmap.lo mmap/unix/common.lo
network_io/unix/sockaddr.lo network_io/unix/sockopt.lo
network_io/unix/sendrecv.lo network_io/unix/multicast.lo
network_io/unix/sockets.lo network_io/unix/inet_ntop.lo
network_io/unix/inet_pton.lo poll/unix/epoll.lo poll/unix/select.lo
poll/unix/poll.lo poll/unix/port.lo poll/unix/kqueue.lo random/unix/sha2.lo
random/unix/apr_random.lo random/unix/sha2_glue.lo shmem/unix/shm.lo
support/unix/waitio.lo threadproc/unix/procsup.lo threadproc/unix/thread.lo
threadproc/unix/signals.lo threadproc/unix/proc.lo
threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
-lpthread

libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object

make[2]: *** [libapr-1.la] Error 1

make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make[1]: *** [all-recursive] Error 1

make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make: *** [libs/apr/libapr-1.la] Error 2

[1]+  Exit 2                  make

 

What next guys?

 

Sean

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From dvl36.ripe.nick at gmail.com  Wed Dec 19 15:35:23 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Wed, 19 Dec 2012 14:35:23 +0200
Subject: [Freeswitch-users] Mutiple GSMopen USB modem setup
In-Reply-To: <CALXCt0rbArYwSOhfKYNv2oXWNe7gmUUObUONTJ8Fa=w-ct205Q@mail.gmail.com>
References: <CAHKqMb6OBRVtOkEepF-a5fVXz_x-NbvhDijvU+dgCBsGcQ5sKg@mail.gmail.com>
	<CALXCt0pT2UoMWZ0dUPC2zKTbm_7zd2Z_2UAr1e8i78JO24uXTw@mail.gmail.com>
	<CAHKqMb69d22pUBqDEw16MwEh_EDR-wMG=j78zYKGFq1ubGZ0DA@mail.gmail.com>
	<CAMjaK3D0Xw4cnXeO1gQr-9+MD1JCuq1m6Xk3qKS0YFQDm6=z5g@mail.gmail.com>
	<CAHKqMb5h3VzjqHiUUPeicuNh1U-a+WnjaPf-RL1GiS51YtXhMw@mail.gmail.com>
	<CALXCt0rbArYwSOhfKYNv2oXWNe7gmUUObUONTJ8Fa=w-ct205Q@mail.gmail.com>
Message-ID: <CAMjaK3B=cVhjNFD6ymyN5SyVESWUW4kqhKLhhZBw8Cn4QM80+Q@mail.gmail.com>

Here (http://wiki.e1550.mobi/doku.php?id=troubleshooting) you can find some
useful info and udev rules file.
Do not tried it yet, but it seems it can work. Maybe after slightly
patching.


2012/12/19 Giovanni Maruzzelli <gmaruzz at celliax.org>

> Nandy and Dimitry,
>
> please let us know your findings.
>
> Thanks for sharing!
>
> -giovanni
>
> On Wed, Dec 19, 2012 at 4:46 AM, Nandy Dagondon <nandy1925 at gmail.com>
> wrote:
> > Hi Giovanni,
> >
> > I have not tested it yet. As mentioned by Dmitry, modem serial numbers
> are
> > not accessible. Perhaps, mapping the serial port based on the USB port
> where
> > they're connected would, at least, a workable solution somewhat as
> discussed
> > in this post:
> >
> >
> http://stackoverflow.com/questions/4800099/how-to-identify-multiple-usb-serial-adapters-under-ubuntu-10-1
> >
> > Rgds,
> > Nandy
> >
> >
> > On Tue, Dec 18, 2012 at 9:03 PM, Dmitry Lysenko <
> dvl36.ripe.nick at gmail.com>
> > wrote:
> >>
> >> Hi!
> >> Huawei USB modems does not have accessible via USB protocol serial
> >> numbers. At least E1550, E153 and E171.
> >> Best wishes,
> >>  Dmitry.
> >>
> >>
> >> 2012/12/16 Nandy Dagondon <nandy1925 at gmail.com>
> >>>
> >>> Thanks for the feedback Giovanni.  Re fixing the ttyUSB port
> assignment,
> >>> I have searched a guy made a script to scan USB serial numbers using
> udev,
> >>> then created symbolic links. Will try it out later.
> >>> /Nandy
> >>>
> >>>
> >>>
> >>> On Thu, Dec 13, 2012 at 7:02 PM, Giovanni Maruzzelli <
> gmaruzz at gmail.com>
> >>> wrote:
> >>>>
> >>>> On Thu, Dec 13, 2012 at 4:44 AM, Nandy Dagondon <nandy1925 at gmail.com>
> >>>> wrote:
> >>>> > I have run GSMOpen on CentOS 5.7 defying the Wiki advisory.
> Questions:
> >>>>
> >>>> I would invite you to go for the OS distros detailed in the wiki page.
> >>>> You'll probably encounter problems with different distros, and you're
> >>>> on your own to solve it.
> >>>>
> >>>> > 1. What is the maximum number of USB modems tested? Can we get the
> >>>> > numbers
> >>>> > and the CPU used?
> >>>>
> >>>> I've heard about 48 concurrent, and 64. Me personally have tested with
> >>>> 5. No CPU consumption. The critical part is the USB BUS. So use
> >>>> cascading and POWERED good usb 2.0 hubs
> >>>>
> >>>> > 2. I'll be installing multiple modems each connected to a different
> >>>> > mobile
> >>>> > network. Is the /dev/ttyUSB assignments  constant for every modem?
> >>>> > Meaning
> >>>> > it doesn't change if I plug it on different USB jacks.
> >>>>
> >>>> it will change not only if you change USB port, but also randomly if
> >>>> you stay on the same USB port and reboot (and sometimes also without
> >>>> rebooting). That's a "feature" of Linux distros (a demented one,
> >>>> cannot understand why they choose this behavior).
> >>>>
> >>>> Soon or later I'll look into this, and come out with a solution (I've
> >>>> made some preliminary research and reasoning about in the past).
> >>>>
> >>>> If you have a commercial interest in that, and a real budget for it,
> >>>> contact me in private as consultant, or put a public bounty on it.
> >>>>
> >>>> -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
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting 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
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
> >
>
>
>
> --
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at celliax.org  Wed Dec 19 16:03:16 2012
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 19 Dec 2012 14:03:16 +0100
Subject: [Freeswitch-users] Mutiple GSMopen USB modem setup
In-Reply-To: <CAMjaK3B=cVhjNFD6ymyN5SyVESWUW4kqhKLhhZBw8Cn4QM80+Q@mail.gmail.com>
References: <CAHKqMb6OBRVtOkEepF-a5fVXz_x-NbvhDijvU+dgCBsGcQ5sKg@mail.gmail.com>
	<CALXCt0pT2UoMWZ0dUPC2zKTbm_7zd2Z_2UAr1e8i78JO24uXTw@mail.gmail.com>
	<CAHKqMb69d22pUBqDEw16MwEh_EDR-wMG=j78zYKGFq1ubGZ0DA@mail.gmail.com>
	<CAMjaK3D0Xw4cnXeO1gQr-9+MD1JCuq1m6Xk3qKS0YFQDm6=z5g@mail.gmail.com>
	<CAHKqMb5h3VzjqHiUUPeicuNh1U-a+WnjaPf-RL1GiS51YtXhMw@mail.gmail.com>
	<CALXCt0rbArYwSOhfKYNv2oXWNe7gmUUObUONTJ8Fa=w-ct205Q@mail.gmail.com>
	<CAMjaK3B=cVhjNFD6ymyN5SyVESWUW4kqhKLhhZBw8Cn4QM80+Q@mail.gmail.com>
Message-ID: <CALXCt0rR5nNhWjsFDKJJJu2GfPPh7AncS5jXgU-iU7F+JgPJBA@mail.gmail.com>

yep, seen that.

I was hoping for something tried and true :).

On Wed, Dec 19, 2012 at 1:35 PM, Dmitry Lysenko
<dvl36.ripe.nick at gmail.com> wrote:
> Here (http://wiki.e1550.mobi/doku.php?id=troubleshooting) you can find some
> useful info and udev rules file.
> Do not tried it yet, but it seems it can work. Maybe after slightly
> patching.
>
>
> 2012/12/19 Giovanni Maruzzelli <gmaruzz at celliax.org>
>>
>> Nandy and Dimitry,
>>
>> please let us know your findings.
>>
>> Thanks for sharing!
>>
>> -giovanni
>>
>> On Wed, Dec 19, 2012 at 4:46 AM, Nandy Dagondon <nandy1925 at gmail.com>
>> wrote:
>> > Hi Giovanni,
>> >
>> > I have not tested it yet. As mentioned by Dmitry, modem serial numbers
>> > are
>> > not accessible. Perhaps, mapping the serial port based on the USB port
>> > where
>> > they're connected would, at least, a workable solution somewhat as
>> > discussed
>> > in this post:
>> >
>> >
>> > http://stackoverflow.com/questions/4800099/how-to-identify-multiple-usb-serial-adapters-under-ubuntu-10-1
>> >
>> > Rgds,
>> > Nandy
>> >
>> >
>> > On Tue, Dec 18, 2012 at 9:03 PM, Dmitry Lysenko
>> > <dvl36.ripe.nick at gmail.com>
>> > wrote:
>> >>
>> >> Hi!
>> >> Huawei USB modems does not have accessible via USB protocol serial
>> >> numbers. At least E1550, E153 and E171.
>> >> Best wishes,
>> >>  Dmitry.
>> >>
>> >>
>> >> 2012/12/16 Nandy Dagondon <nandy1925 at gmail.com>
>> >>>
>> >>> Thanks for the feedback Giovanni.  Re fixing the ttyUSB port
>> >>> assignment,
>> >>> I have searched a guy made a script to scan USB serial numbers using
>> >>> udev,
>> >>> then created symbolic links. Will try it out later.
>> >>> /Nandy
>> >>>
>> >>>
>> >>>
>> >>> On Thu, Dec 13, 2012 at 7:02 PM, Giovanni Maruzzelli
>> >>> <gmaruzz at gmail.com>
>> >>> wrote:
>> >>>>
>> >>>> On Thu, Dec 13, 2012 at 4:44 AM, Nandy Dagondon <nandy1925 at gmail.com>
>> >>>> wrote:
>> >>>> > I have run GSMOpen on CentOS 5.7 defying the Wiki advisory.
>> >>>> > Questions:
>> >>>>
>> >>>> I would invite you to go for the OS distros detailed in the wiki
>> >>>> page.
>> >>>> You'll probably encounter problems with different distros, and you're
>> >>>> on your own to solve it.
>> >>>>
>> >>>> > 1. What is the maximum number of USB modems tested? Can we get the
>> >>>> > numbers
>> >>>> > and the CPU used?
>> >>>>
>> >>>> I've heard about 48 concurrent, and 64. Me personally have tested
>> >>>> with
>> >>>> 5. No CPU consumption. The critical part is the USB BUS. So use
>> >>>> cascading and POWERED good usb 2.0 hubs
>> >>>>
>> >>>> > 2. I'll be installing multiple modems each connected to a different
>> >>>> > mobile
>> >>>> > network. Is the /dev/ttyUSB assignments  constant for every modem?
>> >>>> > Meaning
>> >>>> > it doesn't change if I plug it on different USB jacks.
>> >>>>
>> >>>> it will change not only if you change USB port, but also randomly if
>> >>>> you stay on the same USB port and reboot (and sometimes also without
>> >>>> rebooting). That's a "feature" of Linux distros (a demented one,
>> >>>> cannot understand why they choose this behavior).
>> >>>>
>> >>>> Soon or later I'll look into this, and come out with a solution (I've
>> >>>> made some preliminary research and reasoning about in the past).
>> >>>>
>> >>>> If you have a commercial interest in that, and a real budget for it,
>> >>>> contact me in private as consultant, or put a public bounty on it.
>> >>>>
>> >>>> -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
>> >>>>
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>
>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>>
>> >>>
>> >>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting 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
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting 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
>> >
>>
>>
>>
>> --
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From pm_zefman_r at mail.ru  Wed Dec 19 16:18:36 2012
From: pm_zefman_r at mail.ru (=?UTF-8?B?RG1pdHJpeSBTaHVtYWV2?=)
Date: Wed, 19 Dec 2012 17:18:36 +0400
Subject: [Freeswitch-users] =?utf-8?q?Support_MD5-authentication_with_qop?=
 =?utf-8?b?PSJhdXRoX2ludCIgKFJGQyAyNjE3KQ==?=
Message-ID: <1355923116.673609307@f188.mail.ru>


Does FreeSWITCH support MD5-authentication with qop="auth_int" (RFC 2617)(differs from qop="auth" by the presence of MD5(entityBody) in authenticate response calculation)?
If so, How can I configure?FSW to use this authenticate mechanism?


With best regards, Shumaev DA, KBR technology.
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From sdevoy at bizfocused.com  Wed Dec 19 17:27:31 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 19 Dec 2012 09:27:31 -0500
Subject: [Freeswitch-users] Install 1.2.Stable still fails.  Ver 2
In-Reply-To: <61ad01cddddf$51cfa2a0$f56ee7e0$@bizfocused.com>
References: <5fdf01cddd96$9670f330$c352d990$@bizfocused.com>
	<61ad01cddddf$51cfa2a0$f56ee7e0$@bizfocused.com>
Message-ID: <634e01cdddf4$fb25f560$f171e020$@bizfocused.com>

Can someone at least tell me the GIT command to go back one or more versions
in the "Stable" build that is clearly not stable?

 

Thanks

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Devoy
Sent: Wednesday, December 19, 2012 6:52 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Install 1.2.Stable still fails. Ver 2

 

Anybody?

 

This is a brand new install on a new install of Centos5.8

 

The error is:

libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object

/usr/local/src/freeswitch/libs/apr/ passwd/apr_getpass.lo exists but libtool
does not like it.

 

Full error listed below.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Devoy
Sent: Tuesday, December 18, 2012 10:12 PM
To: FreeSWITCH-users at lists.freeswitch.org
Subject: [Freeswitch-users] Install 1.2.Stable still fails. Ver 2

 

OK, I had my Centos restored to base install 5.8.

 

The install fails.  Here is EXACTLY what I did:

 

Login to root
cd /usr/local/src

rpm -Uvh
http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm


git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git

cd /usr/local/src/freeswitch

./bootstrap.sh

./configure

make & make install

 

 

It fails with:

/bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link  gcc
-g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
-I./include -I/usr/local/src/freeswitch/libs/apr/include/arch/unix
-I./include/arch/unix -I/usr/local/src/freeswitch/libs/apr/include
-version-info 2:8:2    -o libapr-1.la -rpath /usr/local/freeswitch/lib
passwd/apr_getpass.lo strings/apr_cpystrn.lo strings/apr_strnatcmp.lo
strings/apr_strings.lo strings/apr_strtok.lo strings/apr_fnmatch.lo
strings/apr_snprintf.lo tables/apr_tables.lo tables/apr_hash.lo
atomic/unix/apr_atomic.lo dso/unix/dso.lo file_io/unix/flock.lo
file_io/unix/readwrite.lo file_io/unix/filepath_util.lo file_io/unix/seek.lo
file_io/unix/dir.lo file_io/unix/mktemp.lo file_io/unix/filedup.lo
file_io/unix/tempdir.lo file_io/unix/filepath.lo file_io/unix/pipe.lo
file_io/unix/open.lo file_io/unix/filestat.lo file_io/unix/copy.lo
file_io/unix/fileacc.lo file_io/unix/fullrw.lo locks/unix/thread_rwlock.lo
locks/unix/thread_mutex.lo locks/unix/thread_cond.lo
locks/unix/proc_mutex.lo locks/unix/global_mutex.lo memory/unix/apr_pools.lo
misc/unix/charset.lo misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo
misc/unix/start.lo misc/unix/errorcodes.lo misc/unix/getopt.lo
misc/unix/otherchild.lo mmap/unix/mmap.lo mmap/unix/common.lo
network_io/unix/sockaddr.lo network_io/unix/sockopt.lo
network_io/unix/sendrecv.lo network_io/unix/multicast.lo
network_io/unix/sockets.lo network_io/unix/inet_ntop.lo
network_io/unix/inet_pton.lo poll/unix/epoll.lo poll/unix/select.lo
poll/unix/poll.lo poll/unix/port.lo poll/unix/kqueue.lo random/unix/sha2.lo
random/unix/apr_random.lo random/unix/sha2_glue.lo shmem/unix/shm.lo
support/unix/waitio.lo threadproc/unix/procsup.lo threadproc/unix/thread.lo
threadproc/unix/signals.lo threadproc/unix/proc.lo
threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
-lpthread

libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object

make[2]: *** [libapr-1.la] Error 1

make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make[1]: *** [all-recursive] Error 1

make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make: *** [libs/apr/libapr-1.la] Error 2

/bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link  gcc
-g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
-I./include -I/usr/local/src/freeswitch/libs/apr/include/arch/unix
-I./include/arch/unix -I/usr/local/src/freeswitch/libs/apr/include
-version-info 2:8:2    -o libapr-1.la -rpath /usr/local/freeswitch/lib
passwd/apr_getpass.lo strings/apr_cpystrn.lo strings/apr_strnatcmp.lo
strings/apr_strings.lo strings/apr_strtok.lo strings/apr_fnmatch.lo
strings/apr_snprintf.lo tables/apr_tables.lo tables/apr_hash.lo
atomic/unix/apr_atomic.lo dso/unix/dso.lo file_io/unix/flock.lo
file_io/unix/readwrite.lo file_io/unix/filepath_util.lo file_io/unix/seek.lo
file_io/unix/dir.lo file_io/unix/mktemp.lo file_io/unix/filedup.lo
file_io/unix/tempdir.lo file_io/unix/filepath.lo file_io/unix/pipe.lo
file_io/unix/open.lo file_io/unix/filestat.lo file_io/unix/copy.lo
file_io/unix/fileacc.lo file_io/unix/fullrw.lo locks/unix/thread_rwlock.lo
locks/unix/thread_mutex.lo locks/unix/thread_cond.lo
locks/unix/proc_mutex.lo locks/unix/global_mutex.lo memory/unix/apr_pools.lo
misc/unix/charset.lo misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo
misc/unix/start.lo misc/unix/errorcodes.lo misc/unix/getopt.lo
misc/unix/otherchild.lo mmap/unix/mmap.lo mmap/unix/common.lo
network_io/unix/sockaddr.lo network_io/unix/sockopt.lo
network_io/unix/sendrecv.lo network_io/unix/multicast.lo
network_io/unix/sockets.lo network_io/unix/inet_ntop.lo
network_io/unix/inet_pton.lo poll/unix/epoll.lo poll/unix/select.lo
poll/unix/poll.lo poll/unix/port.lo poll/unix/kqueue.lo random/unix/sha2.lo
random/unix/apr_random.lo random/unix/sha2_glue.lo shmem/unix/shm.lo
support/unix/waitio.lo threadproc/unix/procsup.lo threadproc/unix/thread.lo
threadproc/unix/signals.lo threadproc/unix/proc.lo
threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
-lpthread

libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object

make[2]: *** [libapr-1.la] Error 1

make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make[1]: *** [all-recursive] Error 1

make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'

make: *** [libs/apr/libapr-1.la] Error 2

[1]+  Exit 2                  make

 

What next guys?

 

Sean

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From stkn at openisdn.net  Wed Dec 19 17:42:53 2012
From: stkn at openisdn.net (Stefan Knoblich)
Date: Wed, 19 Dec 2012 15:42:53 +0100
Subject: [Freeswitch-users] Install 1.2.Stable still fails.  Ver 2
In-Reply-To: <5fdf01cddd96$9670f330$c352d990$@bizfocused.com>
References: <5fdf01cddd96$9670f330$c352d990$@bizfocused.com>
Message-ID: <50D1D26D.806@openisdn.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 19.12.2012 04:11, Sean Devoy wrote:
> OK, I had my Centos restored to base install 5.8.
> 
> 
> 
> The install fails.  Here is EXACTLY what I did:
> 
> 
> 
> Login to root cd /usr/local/src
> 
> rpm -Uvh http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm
> 
> 
> git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
> 
> cd /usr/local/src/freeswitch
> 
> ./bootstrap.sh
> 
> ./configure
> 
> make & make install


make && make install

SOLVED

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Comment: Using GnuPG with undefined - http://www.enigmail.net/

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k8EAn3ka7YKZfaxLRQy2BmIUtPjN8599
=YaQU
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From moises.silva at gmail.com  Wed Dec 19 17:47:03 2012
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 19 Dec 2012 09:47:03 -0500
Subject: [Freeswitch-users] Install 1.2.Stable still fails. Ver 2
In-Reply-To: <634e01cdddf4$fb25f560$f171e020$@bizfocused.com>
References: <5fdf01cddd96$9670f330$c352d990$@bizfocused.com>
	<61ad01cddddf$51cfa2a0$f56ee7e0$@bizfocused.com>
	<634e01cdddf4$fb25f560$f171e020$@bizfocused.com>
Message-ID: <CAA4nhyC0M+YgrVbpS3L9v-YQ1VJggt4A6h-Co44jBRV1HHZQJQ@mail.gmail.com>

On Wed, Dec 19, 2012 at 9:27 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Can someone at least tell me the GIT command to go back one or more
> versions in the ?Stable? build that is clearly not stable?****
>
> **
>
When asking for help I'd not at the same time nag about how things are not
"stable", it gives a rather demanding tone for someone asking for help.

If you are using git, even when checking out a stable branch, small
mistakes are to be expected. Most likely full QA will not be run on every
commit, but at some specific points in time, like before making a new
release out of that branch. Also the issue could very well be due to your
specific build environment.

Anyhow, all you have to do to go back 1 commit is:

# git reset --hard HEAD~1

Change 1 for any number of commits you want to go back.

You could also checkout a specific commit:

# git checkout <commit-hash-id>

I personally prefer the former as doing git checkout leaves you in a
detached state (unless you create a new branch out of that commit)


*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 jnvines at gmail.com  Wed Dec 19 17:50:20 2012
From: jnvines at gmail.com (Nick Vines)
Date: Wed, 19 Dec 2012 09:50:20 -0500
Subject: [Freeswitch-users] Install 1.2.Stable still fails. Ver 2
In-Reply-To: <634e01cdddf4$fb25f560$f171e020$@bizfocused.com>
References: <5fdf01cddd96$9670f330$c352d990$@bizfocused.com>
	<61ad01cddddf$51cfa2a0$f56ee7e0$@bizfocused.com>
	<634e01cdddf4$fb25f560$f171e020$@bizfocused.com>
Message-ID: <CA+NjZ2vJVKMOA-wffkb2MSeL7gCWDq6_r-Dr8d4-QqaGPGiRUw@mail.gmail.com>

Just got Stefan's message, but already typed this up.

To go to a different commit in the v1.2.stable branch, you can do either
after you run:
> git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git

Option 1:
> git checkout v1.2.5.2 (this is the previous named rev)

Option 2:
> git checkout [commit string]
i.e. git checkout aa0815748ed909bd1be5ec204afcdb1866693a83 OR git checkout
aa08157
You can find the commit string if you click on commit message at the link
below. You can do either of the commands, but if you are lazy just copy and
paste the whole string.

See versions here:
http://git.freeswitch.org/git/freeswitch/log/?h=v1.2.stable

Git wiki entry: http://wiki.freeswitch.org/wiki/Git_Tips


On Wed, Dec 19, 2012 at 9:27 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Can someone at least tell me the GIT command to go back one or more
> versions in the ?Stable? build that is clearly not stable?****
>
> ** **
>
> Thanks****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sean Devoy
> *Sent:* Wednesday, December 19, 2012 6:52 AM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] Install 1.2.Stable still fails. Ver 2***
> *
>
> ** **
>
> Anybody?****
>
> ** **
>
> This is a brand new install on a new install of Centos5.8****
>
> ** **
>
> The error is:****
>
> libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object****
>
> /usr/local/src/freeswitch/libs/apr/ passwd/apr_getpass.lo exists but
> libtool does not like it.****
>
> ** **
>
> Full error listed below.****
>
> ** **
>
> Sean****
>
> ** **
>
> *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 *Sean Devoy
> *Sent:* Tuesday, December 18, 2012 10:12 PM
> *To:* FreeSWITCH-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Install 1.2.Stable still fails. Ver 2****
>
> ** **
>
> OK, I had my Centos restored to base install 5.8.****
>
> ** **
>
> The install fails.  Here is EXACTLY what I did:****
>
> ** **
>
> Login to root
> cd /usr/local/src****
>
> rpm -Uvh
> http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm***
> *
>
>
> git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git****
>
> cd /usr/local/src/freeswitch****
>
> ./bootstrap.sh****
>
> ./configure****
>
> make & make install****
>
> ** **
>
> ** **
>
> It fails with:****
>
> /bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link
> gcc -g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT
> -D_GNU_SOURCE   -I./include
> -I/usr/local/src/freeswitch/libs/apr/include/arch/unix
> -I./include/arch/unix -I/usr/local/src/freeswitch/libs/apr/include
> -version-info 2:8:2    -o libapr-1.la -rpath /usr/local/freeswitch/lib
> passwd/apr_getpass.lo strings/apr_cpystrn.lo strings/apr_strnatcmp.lo
> strings/apr_strings.lo strings/apr_strtok.lo strings/apr_fnmatch.lo
> strings/apr_snprintf.lo tables/apr_tables.lo tables/apr_hash.lo
> atomic/unix/apr_atomic.lo dso/unix/dso.lo file_io/unix/flock.lo
> file_io/unix/readwrite.lo file_io/unix/filepath_util.lo
> file_io/unix/seek.lo file_io/unix/dir.lo file_io/unix/mktemp.lo
> file_io/unix/filedup.lo file_io/unix/tempdir.lo file_io/unix/filepath.lo
> file_io/unix/pipe.lo file_io/unix/open.lo file_io/unix/filestat.lo
> file_io/unix/copy.lo file_io/unix/fileacc.lo file_io/unix/fullrw.lo
> locks/unix/thread_rwlock.lo locks/unix/thread_mutex.lo
> locks/unix/thread_cond.lo locks/unix/proc_mutex.lo
> locks/unix/global_mutex.lo memory/unix/apr_pools.lo misc/unix/charset.lo
> misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo misc/unix/start.lo
> misc/unix/errorcodes.lo misc/unix/getopt.lo misc/unix/otherchild.lo
> mmap/unix/mmap.lo mmap/unix/common.lo network_io/unix/sockaddr.lo
> network_io/unix/sockopt.lo network_io/unix/sendrecv.lo
> network_io/unix/multicast.lo network_io/unix/sockets.lo
> network_io/unix/inet_ntop.lo network_io/unix/inet_pton.lo
> poll/unix/epoll.lo poll/unix/select.lo poll/unix/poll.lo poll/unix/port.lo
> poll/unix/kqueue.lo random/unix/sha2.lo random/unix/apr_random.lo
> random/unix/sha2_glue.lo shmem/unix/shm.lo support/unix/waitio.lo
> threadproc/unix/procsup.lo threadproc/unix/thread.lo
> threadproc/unix/signals.lo threadproc/unix/proc.lo
> threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
> user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
> -lpthread****
>
> libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object****
>
> make[2]: *** [libapr-1.la] Error 1****
>
> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'****
>
> make[1]: *** [all-recursive] Error 1****
>
> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'****
>
> make: *** [libs/apr/libapr-1.la] Error 2****
>
> /bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link
> gcc -g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT
> -D_GNU_SOURCE   -I./include
> -I/usr/local/src/freeswitch/libs/apr/include/arch/unix
> -I./include/arch/unix -I/usr/local/src/freeswitch/libs/apr/include
> -version-info 2:8:2    -o libapr-1.la -rpath /usr/local/freeswitch/lib
> passwd/apr_getpass.lo strings/apr_cpystrn.lo strings/apr_strnatcmp.lo
> strings/apr_strings.lo strings/apr_strtok.lo strings/apr_fnmatch.lo
> strings/apr_snprintf.lo tables/apr_tables.lo tables/apr_hash.lo
> atomic/unix/apr_atomic.lo dso/unix/dso.lo file_io/unix/flock.lo
> file_io/unix/readwrite.lo file_io/unix/filepath_util.lo
> file_io/unix/seek.lo file_io/unix/dir.lo file_io/unix/mktemp.lo
> file_io/unix/filedup.lo file_io/unix/tempdir.lo file_io/unix/filepath.lo
> file_io/unix/pipe.lo file_io/unix/open.lo file_io/unix/filestat.lo
> file_io/unix/copy.lo file_io/unix/fileacc.lo file_io/unix/fullrw.lo
> locks/unix/thread_rwlock.lo locks/unix/thread_mutex.lo
> locks/unix/thread_cond.lo locks/unix/proc_mutex.lo
> locks/unix/global_mutex.lo memory/unix/apr_pools.lo misc/unix/charset.lo
> misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo misc/unix/start.lo
> misc/unix/errorcodes.lo misc/unix/getopt.lo misc/unix/otherchild.lo
> mmap/unix/mmap.lo mmap/unix/common.lo network_io/unix/sockaddr.lo
> network_io/unix/sockopt.lo network_io/unix/sendrecv.lo
> network_io/unix/multicast.lo network_io/unix/sockets.lo
> network_io/unix/inet_ntop.lo network_io/unix/inet_pton.lo
> poll/unix/epoll.lo poll/unix/select.lo poll/unix/poll.lo poll/unix/port.lo
> poll/unix/kqueue.lo random/unix/sha2.lo random/unix/apr_random.lo
> random/unix/sha2_glue.lo shmem/unix/shm.lo support/unix/waitio.lo
> threadproc/unix/procsup.lo threadproc/unix/thread.lo
> threadproc/unix/signals.lo threadproc/unix/proc.lo
> threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
> user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
> -lpthread****
>
> libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object****
>
> make[2]: *** [libapr-1.la] Error 1****
>
> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'****
>
> make[1]: *** [all-recursive] Error 1****
>
> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'****
>
> make: *** [libs/apr/libapr-1.la] Error 2****
>
> [1]+  Exit 2                  make****
>
> ** **
>
> What next guys?****
>
> ** **
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gvvsubhashkumar at gmail.com  Wed Dec 19 18:43:09 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Wed, 19 Dec 2012 21:13:09 +0530
Subject: [Freeswitch-users] How to Set URI in the from header
Message-ID: <CAFvodxVg=uMpqnqSfW8w1iffX_Ans9k8Snfj+Au71EoGGGZdjw@mail.gmail.com>

Hi all,

        I am trying to bridge two calls while bridging the call want to set
the TO and FROM header in the INVITE message sending to B-leg. I found
example how to set TO uri but did not get an example setting FROM header.

      To uri:

           originate sofia/internal/33334444 at 192.168.4.6^11112222 &park


Please help me in how to set FROM header in INVITE message



Thanks,
Subhash.
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From dvl36.ripe.nick at gmail.com  Wed Dec 19 18:55:35 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Wed, 19 Dec 2012 17:55:35 +0200
Subject: [Freeswitch-users] Mutiple GSMopen USB modem setup
In-Reply-To: <CALXCt0rR5nNhWjsFDKJJJu2GfPPh7AncS5jXgU-iU7F+JgPJBA@mail.gmail.com>
References: <CAHKqMb6OBRVtOkEepF-a5fVXz_x-NbvhDijvU+dgCBsGcQ5sKg@mail.gmail.com>
	<CALXCt0pT2UoMWZ0dUPC2zKTbm_7zd2Z_2UAr1e8i78JO24uXTw@mail.gmail.com>
	<CAHKqMb69d22pUBqDEw16MwEh_EDR-wMG=j78zYKGFq1ubGZ0DA@mail.gmail.com>
	<CAMjaK3D0Xw4cnXeO1gQr-9+MD1JCuq1m6Xk3qKS0YFQDm6=z5g@mail.gmail.com>
	<CAHKqMb5h3VzjqHiUUPeicuNh1U-a+WnjaPf-RL1GiS51YtXhMw@mail.gmail.com>
	<CALXCt0rbArYwSOhfKYNv2oXWNe7gmUUObUONTJ8Fa=w-ct205Q@mail.gmail.com>
	<CAMjaK3B=cVhjNFD6ymyN5SyVESWUW4kqhKLhhZBw8Cn4QM80+Q@mail.gmail.com>
	<CALXCt0rR5nNhWjsFDKJJJu2GfPPh7AncS5jXgU-iU7F+JgPJBA@mail.gmail.com>
Message-ID: <CAMjaK3CxZxZpWguoUeZZuLLKhT8fxRpSnwge4KFoLO5PGk5uHw@mail.gmail.com>

I tried, not so hard, but it does not work. :( At least on Debian Wheezy.
Linux Udev system is changing too often, so I decided to use
/dev/serial/by-path/* for persistence.


2012/12/19 Giovanni Maruzzelli <gmaruzz at celliax.org>

> yep, seen that.
>
> I was hoping for something tried and true :).
>
> On Wed, Dec 19, 2012 at 1:35 PM, Dmitry Lysenko
> <dvl36.ripe.nick at gmail.com> wrote:
> > Here (http://wiki.e1550.mobi/doku.php?id=troubleshooting) you can find
> some
> > useful info and udev rules file.
> > Do not tried it yet, but it seems it can work. Maybe after slightly
> > patching.
> >
> >
> > 2012/12/19 Giovanni Maruzzelli <gmaruzz at celliax.org>
> >>
> >> Nandy and Dimitry,
> >>
> >> please let us know your findings.
> >>
> >> Thanks for sharing!
> >>
> >> -giovanni
> >>
> >> On Wed, Dec 19, 2012 at 4:46 AM, Nandy Dagondon <nandy1925 at gmail.com>
> >> wrote:
> >> > Hi Giovanni,
> >> >
> >> > I have not tested it yet. As mentioned by Dmitry, modem serial numbers
> >> > are
> >> > not accessible. Perhaps, mapping the serial port based on the USB port
> >> > where
> >> > they're connected would, at least, a workable solution somewhat as
> >> > discussed
> >> > in this post:
> >> >
> >> >
> >> >
> http://stackoverflow.com/questions/4800099/how-to-identify-multiple-usb-serial-adapters-under-ubuntu-10-1
> >> >
> >> > Rgds,
> >> > Nandy
> >> >
> >> >
> >> > On Tue, Dec 18, 2012 at 9:03 PM, Dmitry Lysenko
> >> > <dvl36.ripe.nick at gmail.com>
> >> > wrote:
> >> >>
> >> >> Hi!
> >> >> Huawei USB modems does not have accessible via USB protocol serial
> >> >> numbers. At least E1550, E153 and E171.
> >> >> Best wishes,
> >> >>  Dmitry.
> >> >>
> >> >>
> >> >> 2012/12/16 Nandy Dagondon <nandy1925 at gmail.com>
> >> >>>
> >> >>> Thanks for the feedback Giovanni.  Re fixing the ttyUSB port
> >> >>> assignment,
> >> >>> I have searched a guy made a script to scan USB serial numbers using
> >> >>> udev,
> >> >>> then created symbolic links. Will try it out later.
> >> >>> /Nandy
> >> >>>
> >> >>>
> >> >>>
> >> >>> On Thu, Dec 13, 2012 at 7:02 PM, Giovanni Maruzzelli
> >> >>> <gmaruzz at gmail.com>
> >> >>> wrote:
> >> >>>>
> >> >>>> On Thu, Dec 13, 2012 at 4:44 AM, Nandy Dagondon <
> nandy1925 at gmail.com>
> >> >>>> wrote:
> >> >>>> > I have run GSMOpen on CentOS 5.7 defying the Wiki advisory.
> >> >>>> > Questions:
> >> >>>>
> >> >>>> I would invite you to go for the OS distros detailed in the wiki
> >> >>>> page.
> >> >>>> You'll probably encounter problems with different distros, and
> you're
> >> >>>> on your own to solve it.
> >> >>>>
> >> >>>> > 1. What is the maximum number of USB modems tested? Can we get
> the
> >> >>>> > numbers
> >> >>>> > and the CPU used?
> >> >>>>
> >> >>>> I've heard about 48 concurrent, and 64. Me personally have tested
> >> >>>> with
> >> >>>> 5. No CPU consumption. The critical part is the USB BUS. So use
> >> >>>> cascading and POWERED good usb 2.0 hubs
> >> >>>>
> >> >>>> > 2. I'll be installing multiple modems each connected to a
> different
> >> >>>> > mobile
> >> >>>> > network. Is the /dev/ttyUSB assignments  constant for every
> modem?
> >> >>>> > Meaning
> >> >>>> > it doesn't change if I plug it on different USB jacks.
> >> >>>>
> >> >>>> it will change not only if you change USB port, but also randomly
> if
> >> >>>> you stay on the same USB port and reboot (and sometimes also
> without
> >> >>>> rebooting). That's a "feature" of Linux distros (a demented one,
> >> >>>> cannot understand why they choose this behavior).
> >> >>>>
> >> >>>> Soon or later I'll look into this, and come out with a solution
> (I've
> >> >>>> made some preliminary research and reasoning about in the past).
> >> >>>>
> >> >>>> If you have a commercial interest in that, and a real budget for
> it,
> >> >>>> contact me in private as consultant, or put a public bounty on it.
> >> >>>>
> >> >>>> -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
> >> >>>>
> >> >>>> FreeSWITCH-users mailing list
> >> >>>> FreeSWITCH-users at lists.freeswitch.org
> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>>
> >> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>>> http://www.freeswitch.org
> >> >>>
> >> >>>
> >> >>>
> >> >>>
> >> >>>
> _________________________________________________________________________
> >> >>> Professional FreeSWITCH Consulting 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
> >> >>
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> >
> >> >
> _________________________________________________________________________
> >> > Professional FreeSWITCH Consulting 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
> >> >
> >>
> >>
> >>
> >> --
> >> 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
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
> >
>
>
>
> --
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 19 19:03:50 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 19 Dec 2012 17:03:50 +0100
Subject: [Freeswitch-users] Mutiple GSMopen USB modem setup
In-Reply-To: <CAMjaK3CxZxZpWguoUeZZuLLKhT8fxRpSnwge4KFoLO5PGk5uHw@mail.gmail.com>
References: <CAHKqMb6OBRVtOkEepF-a5fVXz_x-NbvhDijvU+dgCBsGcQ5sKg@mail.gmail.com>
	<CALXCt0pT2UoMWZ0dUPC2zKTbm_7zd2Z_2UAr1e8i78JO24uXTw@mail.gmail.com>
	<CAHKqMb69d22pUBqDEw16MwEh_EDR-wMG=j78zYKGFq1ubGZ0DA@mail.gmail.com>
	<CAMjaK3D0Xw4cnXeO1gQr-9+MD1JCuq1m6Xk3qKS0YFQDm6=z5g@mail.gmail.com>
	<CAHKqMb5h3VzjqHiUUPeicuNh1U-a+WnjaPf-RL1GiS51YtXhMw@mail.gmail.com>
	<CALXCt0rbArYwSOhfKYNv2oXWNe7gmUUObUONTJ8Fa=w-ct205Q@mail.gmail.com>
	<CAMjaK3B=cVhjNFD6ymyN5SyVESWUW4kqhKLhhZBw8Cn4QM80+Q@mail.gmail.com>
	<CALXCt0rR5nNhWjsFDKJJJu2GfPPh7AncS5jXgU-iU7F+JgPJBA@mail.gmail.com>
	<CAMjaK3CxZxZpWguoUeZZuLLKhT8fxRpSnwge4KFoLO5PGk5uHw@mail.gmail.com>
Message-ID: <CALXCt0oW33s0ztcqb0a-4gYqjAFQwxFYMSTs7kFz6fX6Y0HxEA@mail.gmail.com>

On Wed, Dec 19, 2012 at 4:55 PM, Dmitry Lysenko
<dvl36.ripe.nick at gmail.com> wrote:
> I tried, not so hard, but it does not work. :( At least on Debian Wheezy.
> Linux Udev system is changing too often, so I decided to use
> /dev/serial/by-path/* for persistence.

can you please detail how you use /dev/serial/by-path/* for persistence ?
(btw, the patch for long device names you sent is in mainline since some weeks)

-giovanni


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From krice at freeswitch.org  Wed Dec 19 19:08:06 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 19 Dec 2012 10:08:06 -0600
Subject: [Freeswitch-users] Install 1.2.Stable still fails.  Ver 2
In-Reply-To: <5fdf01cddd96$9670f330$c352d990$@bizfocused.com>
Message-ID: <CCF74286.6246F%krice@freeswitch.org>

Your make command is wrong...

Its make && make install...

Notice the two (2) &?s there you have one in the command you stated...

 make & make install  (1 &) tries to run 2 makes concurrently... The first
make will get backgrounded, and the 2nd make will run in the foreground....

This will screw things up royally...

As for Centos5 the entire tree is tested routinely on Centos5... 99% of the
development occurs on centos5, so I hate to say it but if you are having a
problem building on that specific platform it is almost always (99.999% of
the time) a problem with your installation environment...

So sit back and relax... Also keep in mind that it seems we have the highest
concentration of FreeSWITCH users in the West (ie North and South America)
and not all of them are around all the time... So when you ask a question it
make take several hours or even a day before someone with the right
knowledge can answer your query....

If you are deploying in a commercial environment where you might need
priority support, I would suggest contacting consulting at freeswitch.org where
you can get someone from the FreeSWITCH team to help you via a support
agreement... 

K


On 12/18/12 9:11 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> OK, I had my Centos restored to base install 5.8.
>  
> The install fails.  Here is EXACTLY what I did:
>  
> Login to root
> cd /usr/local/src
> rpm -Uvh 
> http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm
> 
> git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
> cd /usr/local/src/freeswitch
> ./bootstrap.sh
> ./configure
> make & make install
>  
>  
> It fails with:
> /bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link  gcc
> -g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
> -I./include -I/usr/local/src/freeswitch/libs/apr/include/arch/unix
> -I./include/arch/unix -I/usr/local/src/freeswitch/libs/apr/include
> -version-info 2:8:2    -o libapr-1.la -rpath /usr/local/freeswitch/lib
> passwd/apr_getpass.lo strings/apr_cpystrn.lo strings/apr_strnatcmp.lo
> strings/apr_strings.lo strings/apr_strtok.lo strings/apr_fnmatch.lo
> strings/apr_snprintf.lo tables/apr_tables.lo tables/apr_hash.lo
> atomic/unix/apr_atomic.lo dso/unix/dso.lo file_io/unix/flock.lo
> file_io/unix/readwrite.lo file_io/unix/filepath_util.lo file_io/unix/seek.lo
> file_io/unix/dir.lo file_io/unix/mktemp.lo file_io/unix/filedup.lo
> file_io/unix/tempdir.lo file_io/unix/filepath.lo file_io/unix/pipe.lo
> file_io/unix/open.lo file_io/unix/filestat.lo file_io/unix/copy.lo
> file_io/unix/fileacc.lo file_io/unix/fullrw.lo locks/unix/thread_rwlock.lo
> locks/unix/thread_mutex.lo locks/unix/thread_cond.lo locks/unix/proc_mutex.lo
> locks/unix/global_mutex.lo memory/unix/apr_pools.lo misc/unix/charset.lo
> misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo misc/unix/start.lo
> misc/unix/errorcodes.lo misc/unix/getopt.lo misc/unix/otherchild.lo
> mmap/unix/mmap.lo mmap/unix/common.lo network_io/unix/sockaddr.lo
> network_io/unix/sockopt.lo network_io/unix/sendrecv.lo
> network_io/unix/multicast.lo network_io/unix/sockets.lo
> network_io/unix/inet_ntop.lo network_io/unix/inet_pton.lo poll/unix/epoll.lo
> poll/unix/select.lo poll/unix/poll.lo poll/unix/port.lo poll/unix/kqueue.lo
> random/unix/sha2.lo random/unix/apr_random.lo random/unix/sha2_glue.lo
> shmem/unix/shm.lo support/unix/waitio.lo threadproc/unix/procsup.lo
> threadproc/unix/thread.lo threadproc/unix/signals.lo threadproc/unix/proc.lo
> threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
> user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
> -lpthread
> libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object
> make[2]: *** [libapr-1.la] Error 1
> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'
> make: *** [libs/apr/libapr-1.la] Error 2
> /bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link  gcc
> -g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
> -I./include -I/usr/local/src/freeswitch/libs/apr/include/arch/unix
> -I./include/arch/unix -I/usr/local/src/freeswitch/libs/apr/include
> -version-info 2:8:2    -o libapr-1.la -rpath /usr/local/freeswitch/lib
> passwd/apr_getpass.lo strings/apr_cpystrn.lo strings/apr_strnatcmp.lo
> strings/apr_strings.lo strings/apr_strtok.lo strings/apr_fnmatch.lo
> strings/apr_snprintf.lo tables/apr_tables.lo tables/apr_hash.lo
> atomic/unix/apr_atomic.lo dso/unix/dso.lo file_io/unix/flock.lo
> file_io/unix/readwrite.lo file_io/unix/filepath_util.lo file_io/unix/seek.lo
> file_io/unix/dir.lo file_io/unix/mktemp.lo file_io/unix/filedup.lo
> file_io/unix/tempdir.lo file_io/unix/filepath.lo file_io/unix/pipe.lo
> file_io/unix/open.lo file_io/unix/filestat.lo file_io/unix/copy.lo
> file_io/unix/fileacc.lo file_io/unix/fullrw.lo locks/unix/thread_rwlock.lo
> locks/unix/thread_mutex.lo locks/unix/thread_cond.lo locks/unix/proc_mutex.lo
> locks/unix/global_mutex.lo memory/unix/apr_pools.lo misc/unix/charset.lo
> misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo misc/unix/start.lo
> misc/unix/errorcodes.lo misc/unix/getopt.lo misc/unix/otherchild.lo
> mmap/unix/mmap.lo mmap/unix/common.lo network_io/unix/sockaddr.lo
> network_io/unix/sockopt.lo network_io/unix/sendrecv.lo
> network_io/unix/multicast.lo network_io/unix/sockets.lo
> network_io/unix/inet_ntop.lo network_io/unix/inet_pton.lo poll/unix/epoll.lo
> poll/unix/select.lo poll/unix/poll.lo poll/unix/port.lo poll/unix/kqueue.lo
> random/unix/sha2.lo random/unix/apr_random.lo random/unix/sha2_glue.lo
> shmem/unix/shm.lo support/unix/waitio.lo threadproc/unix/procsup.lo
> threadproc/unix/thread.lo threadproc/unix/signals.lo threadproc/unix/proc.lo
> threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
> user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
> -lpthread
> libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object
> make[2]: *** [libapr-1.la] Error 1
> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'
> make: *** [libs/apr/libapr-1.la] Error 2
> [1]+  Exit 2                  make
>  
> What next guys?
>  
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From dvl36.ripe.nick at gmail.com  Wed Dec 19 19:46:00 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Wed, 19 Dec 2012 18:46:00 +0200
Subject: [Freeswitch-users] Mutiple GSMopen USB modem setup
In-Reply-To: <CALXCt0oW33s0ztcqb0a-4gYqjAFQwxFYMSTs7kFz6fX6Y0HxEA@mail.gmail.com>
References: <CAHKqMb6OBRVtOkEepF-a5fVXz_x-NbvhDijvU+dgCBsGcQ5sKg@mail.gmail.com>
	<CALXCt0pT2UoMWZ0dUPC2zKTbm_7zd2Z_2UAr1e8i78JO24uXTw@mail.gmail.com>
	<CAHKqMb69d22pUBqDEw16MwEh_EDR-wMG=j78zYKGFq1ubGZ0DA@mail.gmail.com>
	<CAMjaK3D0Xw4cnXeO1gQr-9+MD1JCuq1m6Xk3qKS0YFQDm6=z5g@mail.gmail.com>
	<CAHKqMb5h3VzjqHiUUPeicuNh1U-a+WnjaPf-RL1GiS51YtXhMw@mail.gmail.com>
	<CALXCt0rbArYwSOhfKYNv2oXWNe7gmUUObUONTJ8Fa=w-ct205Q@mail.gmail.com>
	<CAMjaK3B=cVhjNFD6ymyN5SyVESWUW4kqhKLhhZBw8Cn4QM80+Q@mail.gmail.com>
	<CALXCt0rR5nNhWjsFDKJJJu2GfPPh7AncS5jXgU-iU7F+JgPJBA@mail.gmail.com>
	<CAMjaK3CxZxZpWguoUeZZuLLKhT8fxRpSnwge4KFoLO5PGk5uHw@mail.gmail.com>
	<CALXCt0oW33s0ztcqb0a-4gYqjAFQwxFYMSTs7kFz6fX6Y0HxEA@mail.gmail.com>
Message-ID: <CAMjaK3CbriWMWzQWVFY4xtRYh+pOW=oe4oQkwgqeqraS2xfQZg@mail.gmail.com>

Below snippet of my gsmopen config for 1-st and 4-st USB hub ports with
Huawei modems in it.
USB hub port numbers marked as bold.
-----
<!-- output of "ls -1" at /dev/serial/by-path
platform-orion-ehci.0-usb-0:1.*1*:1.0-port0
platform-orion-ehci.0-usb-0:1.*1*:1.1-port0
platform-orion-ehci.0-usb-0:1.*1*:1.2-port0
platform-orion-ehci.0-usb-0:1.*4*:1.0-port0
platform-orion-ehci.0-usb-0:1.*4*:1.1-port0
platform-orion-ehci.0-usb-0:1.*4*:1.2-port0
platform-orion-ehci.0-usb-0:1.*6*:1.0-port0
platform-orion-ehci.0-usb-0:1.*6*:1.1-port0
platform-orion-ehci.0-usb-0:1.*6*:1.2-port0
platform-orion-ehci.0-usb-0:1.*7*:1.0-port0
platform-orion-ehci.0-usb-0:1.*7*:1.1-port0
platform-orion-ehci.0-usb-0:1.*7*:1.2-port0
-->
  <per_interface_settings>
    <interface id="1" name="gsm01">
      <param name="controldevice_name"
value="/dev/serial/by-path/platform-orion-ehci.0-usb-0:1.*1*:1.2-port0"/>
      <param name="controldevice_audio_name"
value="/dev/serial/by-path/platform-orion-ehci.0-usb-0:1.*1*:1.1-port0"/>
      <param name="capture_boost" value="+4"/>
      <param name="playback_boost" value="+6"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <param name="hold-music" value="$${moh_uri}"/>
      <param name="destination" value="1000"/>
    </interface>
    <interface id="2" name="gsm02">
      <param name="controldevice_name"
value="/dev/serial/by-path/platform-orion-ehci.0-usb-0:1.*4*:1.2-port0"/>
      <param name="controldevice_audio_name"
value="/dev/serial/by-path/platform-orion-ehci.0-usb-0:1.*4*:1.1-port0"/>
      <param name="capture_boost" value="+4"/>
      <param name="playback_boost" value="+6"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <param name="hold-music" value="$${moh_uri}"/>
      <param name="destination" value="1000"/>
---
BTW, the patch for long device names is not mine.


2012/12/19 Giovanni Maruzzelli <gmaruzz at gmail.com>

> On Wed, Dec 19, 2012 at 4:55 PM, Dmitry Lysenko
> <dvl36.ripe.nick at gmail.com> wrote:
> > I tried, not so hard, but it does not work. :( At least on Debian Wheezy.
> > Linux Udev system is changing too often, so I decided to use
> > /dev/serial/by-path/* for persistence.
>
> can you please detail how you use /dev/serial/by-path/* for persistence ?
> (btw, the patch for long device names you sent is in mainline since some
> weeks)
>
> -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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Dec 19 19:48:24 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 19 Dec 2012 08:48:24 -0800 (PST)
Subject: [Freeswitch-users] build error with new Freeswitch 'nametab.h':
 No such file or directory
In-Reply-To: <B60DA91F5EA0E24A96E73CDE43CC9EC837D614@vc4bex2.corp.c4b.de>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC837D409@vc4bex2.corp.c4b.de>
	<1355844703594-7585673.post@n2.nabble.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC837D614@vc4bex2.corp.c4b.de>
Message-ID: <1355935704180-7585726.post@n2.nabble.com>

Ok thanks, I think this might be cause because a code generator is failing
because of a space in the path.

Can you try building from a directory without spaces a test to verify. Then
I will look to fix.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/build-error-with-new-Freeswitch-nametab-h-No-such-file-or-directory-tp7585669p7585726.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From 8f27e956 at gmail.com  Wed Dec 19 20:08:49 2012
From: 8f27e956 at gmail.com (S. Scott)
Date: Wed, 19 Dec 2012 12:08:49 -0500
Subject: [Freeswitch-users] Install 1.2.Stable still fails. Ver 2
In-Reply-To: <CCF74286.6246F%krice@freeswitch.org>
References: <CCF74286.6246F%krice@freeswitch.org>
Message-ID: <6909951704489875258@unknownmsgid>

I'm not that strong in these areas, but i too tried EPEL flavor AND i too
ran into freeswitch build issues. I manually yum'd the fs dependancies and
hit issues there too.  After googling around, I flushed and went back to
the centos iso AND

Do you epel pull,

THEN THE FIRST THING YOU MUST DO BEFORE UPDATING OR OTHER YUM ACTION IS
(was for me)

yum install yum-plugin-protectbase.noarch

The readings i found assert you MUST protect the base repo's from the epel
repo's.  The above plugin does this repo boundry.  Everything -- dependancy
chain and freeSWITCH -- yum'ed and make'd just fine thereafter.

Dont know if this is your issue,,, like i said im not that strong in these
areas, but it solved my epel-version issues.

goodluck,

?????
iThing:  Big thumbs & little keys. Please excuse typo, spelling and grammar
errors ? Last night I played a blank CD at full blast. The Mime next door
went nuts.


On 2012-12-19, at 11:11, Ken Rice <krice at freeswitch.org> wrote:

Re: [Freeswitch-users] Install 1.2.Stable still fails.  Ver 2 Your make
command is wrong...

Its make && make install...

Notice the two (2) &?s there you have one in the command you stated...

 make & make install  (1 &) tries to run 2 makes concurrently... The first
make will get backgrounded, and the 2nd make will run in the foreground....

This will screw things up royally...

As for Centos5 the entire tree is tested routinely on Centos5... 99% of the
development occurs on centos5, so I hate to say it but if you are having a
problem building on that specific platform it is almost always (99.999% of
the time) a problem with your installation environment...

So sit back and relax... Also keep in mind that it seems we have the
highest concentration of FreeSWITCH users in the West (ie North and South
America) and not all of them are around all the time... So when you ask a
question it make take several hours or even a day before someone with the
right knowledge can answer your query....

If you are deploying in a commercial environment where you might need
priority support, I would suggest contacting
consulting at freeswitch.orgwhere you can get someone from the FreeSWITCH
team to help you via a
support agreement...

K


On 12/18/12 9:11 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

OK, I had my Centos restored to base install 5.8.

The install fails.  Here is EXACTLY what I did:

Login to root
cd /usr/local/src
rpm -Uvh
http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm

git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
cd /usr/local/src/freeswitch
./bootstrap.sh
./configure
make & make install


It fails with:
/bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link
 gcc -g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT
-D_GNU_SOURCE   -I./include
-I/usr/local/src/freeswitch/libs/apr/include/arch/unix
-I./include/arch/unix -I/usr/local/src/freeswitch/libs/apr/include
 -version-info 2:8:2    -o libapr-1.la -rpath /usr/local/freeswitch/lib
passwd/apr_getpass.lo strings/apr_cpystrn.lo strings/apr_strnatcmp.lo
strings/apr_strings.lo strings/apr_strtok.lo strings/apr_fnmatch.lo
strings/apr_snprintf.lo tables/apr_tables.lo tables/apr_hash.lo
atomic/unix/apr_atomic.lo dso/unix/dso.lo file_io/unix/flock.lo
file_io/unix/readwrite.lo file_io/unix/filepath_util.lo
file_io/unix/seek.lo file_io/unix/dir.lo file_io/unix/mktemp.lo
file_io/unix/filedup.lo file_io/unix/tempdir.lo file_io/unix/filepath.lo
file_io/unix/pipe.lo file_io/unix/open.lo file_io/unix/filestat.lo
file_io/unix/copy.lo file_io/unix/fileacc.lo file_io/unix/fullrw.lo
locks/unix/thread_rwlock.lo locks/unix/thread_mutex.lo
locks/unix/thread_cond.lo locks/unix/proc_mutex.lo
locks/unix/global_mutex.lo memory/unix/apr_pools.lo misc/unix/charset.lo
misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo misc/unix/start.lo
misc/unix/errorcodes.lo misc/unix/getopt.lo misc/unix/otherchild.lo
mmap/unix/mmap.lo mmap/unix/common.lo network_io/unix/sockaddr.lo
network_io/unix/sockopt.lo network_io/unix/sendrecv.lo
network_io/unix/multicast.lo network_io/unix/sockets.lo
network_io/unix/inet_ntop.lo network_io/unix/inet_pton.lo
poll/unix/epoll.lo poll/unix/select.lo poll/unix/poll.lo poll/unix/port.lo
poll/unix/kqueue.lo random/unix/sha2.lo random/unix/apr_random.lo
random/unix/sha2_glue.lo shmem/unix/shm.lo support/unix/waitio.lo
threadproc/unix/procsup.lo threadproc/unix/thread.lo
threadproc/unix/signals.lo threadproc/unix/proc.lo
threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
 -lpthread
libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object
make[2]: *** [libapr-1.la] Error 1
make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'
make: *** [libs/apr/libapr-1.la] Error 2
/bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link
 gcc -g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT
-D_GNU_SOURCE   -I./include
-I/usr/local/src/freeswitch/libs/apr/include/arch/unix
-I./include/arch/unix -I/usr/local/src/freeswitch/libs/apr/include
 -version-info 2:8:2    -o libapr-1.la -rpath /usr/local/freeswitch/lib
passwd/apr_getpass.lo strings/apr_cpystrn.lo strings/apr_strnatcmp.lo
strings/apr_strings.lo strings/apr_strtok.lo strings/apr_fnmatch.lo
strings/apr_snprintf.lo tables/apr_tables.lo tables/apr_hash.lo
atomic/unix/apr_atomic.lo dso/unix/dso.lo file_io/unix/flock.lo
file_io/unix/readwrite.lo file_io/unix/filepath_util.lo
file_io/unix/seek.lo file_io/unix/dir.lo file_io/unix/mktemp.lo
file_io/unix/filedup.lo file_io/unix/tempdir.lo file_io/unix/filepath.lo
file_io/unix/pipe.lo file_io/unix/open.lo file_io/unix/filestat.lo
file_io/unix/copy.lo file_io/unix/fileacc.lo file_io/unix/fullrw.lo
locks/unix/thread_rwlock.lo locks/unix/thread_mutex.lo
locks/unix/thread_cond.lo locks/unix/proc_mutex.lo
locks/unix/global_mutex.lo memory/unix/apr_pools.lo misc/unix/charset.lo
misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo misc/unix/start.lo
misc/unix/errorcodes.lo misc/unix/getopt.lo misc/unix/otherchild.lo
mmap/unix/mmap.lo mmap/unix/common.lo network_io/unix/sockaddr.lo
network_io/unix/sockopt.lo network_io/unix/sendrecv.lo
network_io/unix/multicast.lo network_io/unix/sockets.lo
network_io/unix/inet_ntop.lo network_io/unix/inet_pton.lo
poll/unix/epoll.lo poll/unix/select.lo poll/unix/poll.lo poll/unix/port.lo
poll/unix/kqueue.lo random/unix/sha2.lo random/unix/apr_random.lo
random/unix/sha2_glue.lo shmem/unix/shm.lo support/unix/waitio.lo
threadproc/unix/procsup.lo threadproc/unix/thread.lo
threadproc/unix/signals.lo threadproc/unix/proc.lo
threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
 -lpthread
libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object
make[2]: *** [libapr-1.la] Error 1
make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'
make: *** [libs/apr/libapr-1.la] Error 2
[1]+  Exit 2                  make

What next guys?

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

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


-- 
Ken
*http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
*irc.freenode.net #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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 Dec 19 20:19:44 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Dec 2012 09:19:44 -0800
Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Today
Message-ID: <CAKzWOxVo=-xZ=oEdnByLuGpyKevL_hJqxqFS5v0a=t5WqYJz5A@mail.gmail.com>

Hello all!

We have the weekly conference call today:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_19

We'll be taking a look at the new nested conditions feature that was
recently added to the XML dialplan.

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From steveayre at gmail.com  Wed Dec 19 20:38:45 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 19 Dec 2012 17:38:45 +0000
Subject: [Freeswitch-users] How to Set URI in the from header
In-Reply-To: <CAFvodxVg=uMpqnqSfW8w1iffX_Ans9k8Snfj+Au71EoGGGZdjw@mail.gmail.com>
References: <CAFvodxVg=uMpqnqSfW8w1iffX_Ans9k8Snfj+Au71EoGGGZdjw@mail.gmail.com>
Message-ID: <CAFiqYunWQ_NCMe=31KJV2Wsvei5RVhUpX03fRznbbB7uA9akYw@mail.gmail.com>

Try these variables:
http://wiki.freeswitch.org/wiki/Channel_Variables#Caller_ID_Related

Either use 'export' or set them within the dialstring itself.

-Steve



On 19 December 2012 15:43, Subhash <gvvsubhashkumar at gmail.com> wrote:

> Hi all,
>
>         I am trying to bridge two calls while bridging the call want to
> set the TO and FROM header in the INVITE message sending to B-leg. I found
> example how to set TO uri but did not get an example setting FROM header.
>
>       To uri:
>
>            originate sofia/internal/33334444 at 192.168.4.6^11112222 &park
>
>
> Please help me in how to set FROM header in INVITE message
>
>
>
> Thanks,
> Subhash.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 steveayre at gmail.com  Wed Dec 19 20:43:12 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 19 Dec 2012 17:43:12 +0000
Subject: [Freeswitch-users] Support MD5-authentication with
 qop="auth_int" (RFC 2617)
In-Reply-To: <1355923116.673609307@f188.mail.ru>
References: <1355923116.673609307@f188.mail.ru>
Message-ID: <CAFiqYumkknD24MZRCyrbDHp85qgkQb+Sn5RhC_STUeLg02SVxA@mail.gmail.com>

auth-int appears in sofia-sip, but I don't think it's exposed to FS.

I think it'll work for outgoing calls due to the challenge in the 401.

For incoming calls I think it would need a patch to FS to set AUTHTAG_QOP

-Steve



On 19 December 2012 13:18, Dmitriy Shumaev <pm_zefman_r at mail.ru> wrote:

> Does FreeSWITCH support MD5-authentication with qop="auth_int" (RFC
> 2617)(differs from qop="auth" by the presence of MD5(entityBody) in
> authenticate response calculation)?
> If so, How can I configure FSW to use this authenticate mechanism?
>
>
> With best regards, Shumaev DA, KBR technology.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 anthony.minessale at gmail.com  Wed Dec 19 21:26:50 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 19 Dec 2012 12:26:50 -0600
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <5da601cddd67$dc76b6f0$956424d0$@bizfocused.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>
	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>
	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>
	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>
	<541b01cddcb1$72330200$56990600$@bizfocused.com>
	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>
	<5da601cddd67$dc76b6f0$956424d0$@bizfocused.com>
Message-ID: <CAKbxfG_eXUY=eLnp-_xLks4-bXaoR3YiwqvsJ0teV15AF8w6vA@mail.gmail.com>

it gives 404 err


On Tue, Dec 18, 2012 at 3:37 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> OK, I got one the logs!  I have trimmed the file down, but it is still
> over 5 MB and pastebin just chokes on it.****
>
> ** **
>
> It is on my server:  http://www.bizfocused.com/service_not_found.log****
>
> ** **
>
> Thanks for looking.****
>
> ** **
>
> If you really need pcap, I am afraid I need detailed instructions on
> installing/running it.****
>
> ** **
>
> Sean****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 17, 2012 7:00 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
> ** **
>
> usually the 406 is done in an established call to refuse a codec change
> during re-invite.****
>
> Its possible the other end thinks we want to change the codec when we do
> the session-timer re-invite but I'm sure we don't but the sip trace can
> help shed some light.  You can run a pcap too at the same time so when we
> find the bad call in the logs we can filter it out of the pcap too.  To
> avoid it getting too big you can just restart it every so often or use
> sippcapdump and delete calls that are not affected.****
>
> ** **
>
> ** **
>
> ** **
>
> On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> Waiting for another failure with siptrace REALLY on this time.****
>
>  ****
>
> If the user has clicked DND on these cisco phones, could that cause these
> messages?****
>
>  ****
>
> Sean****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 17, 2012 12:32 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> The siptrace is not in here?  Did you enable it on the console and then
> capture to the log file perhaps?****
>
> you should do "sofia tracelevel debug" too to route traces to the log file.
> ****
>
>  ****
>
> Its hard to say for sure with no sip trace but it seems like the far end
> is rejecting the session timer re-invite causing the call to end.****
>
>  ****
>
>  ****
>
> On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com>
> wrote:****
>
> http://pastebin.freeswitch.org/20342****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
> ** **
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
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From sdevoy at bizfocused.com  Thu Dec 20 00:43:06 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 19 Dec 2012 16:43:06 -0500
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <CAKbxfG_eXUY=eLnp-_xLks4-bXaoR3YiwqvsJ0teV15AF8w6vA@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>	<541b01cddcb1$72330200$56990600$@bizfocused.com>	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>	<5da601cddd67$dc76b6f0$956424d0$@bizfocused.com>
	<CAKbxfG_eXUY=eLnp-_xLks4-bXaoR3YiwqvsJ0teV15AF8w6vA@mail.gmail.com>
Message-ID: <681801cdde31$d4fe1ec0$7efa5c40$@bizfocused.com>

Whoa.   404 a.k.a device not found?

 

This could be as I suspected, router capacity exceeded or general NAT issue
with this router?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, December 19, 2012 1:27 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

it gives 404 err

 

On Tue, Dec 18, 2012 at 3:37 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

OK, I got one the logs!  I have trimmed the file down, but it is still over
5 MB and pastebin just chokes on it.

 

It is on my server:  http://www.bizfocused.com/service_not_found.log

 

Thanks for looking.

 

If you really need pcap, I am afraid I need detailed instructions on
installing/running it.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 7:00 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

usually the 406 is done in an established call to refuse a codec change
during re-invite.

Its possible the other end thinks we want to change the codec when we do the
session-timer re-invite but I'm sure we don't but the sip trace can help
shed some light.  You can run a pcap too at the same time so when we find
the bad call in the logs we can filter it out of the pcap too.  To avoid it
getting too big you can just restart it every so often or use sippcapdump
and delete calls that are not affected.

 

 

 

On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Waiting for another failure with siptrace REALLY on this time.

 

If the user has clicked DND on these cisco phones, could that cause these
messages?

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 12:32 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

The siptrace is not in here?  Did you enable it on the console and then
capture to the log file perhaps?

you should do "sofia tracelevel debug" too to route traces to the log file.

 

Its hard to say for sure with no sip trace but it seems like the far end is
rejecting the session timer re-invite causing the call to end.

 

 

On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

http://pastebin.freeswitch.org/20342





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900> 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900> 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 

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From anthony.minessale at gmail.com  Thu Dec 20 00:49:51 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 19 Dec 2012 15:49:51 -0600
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <681801cdde31$d4fe1ec0$7efa5c40$@bizfocused.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>
	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>
	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>
	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>
	<541b01cddcb1$72330200$56990600$@bizfocused.com>
	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>
	<5da601cddd67$dc76b6f0$956424d0$@bizfocused.com>
	<CAKbxfG_eXUY=eLnp-_xLks4-bXaoR3YiwqvsJ0teV15AF8w6vA@mail.gmail.com>
	<681801cdde31$d4fe1ec0$7efa5c40$@bizfocused.com>
Message-ID: <CAKbxfG_4Xp=DwOnq6zFo=S-30BpN_BME+wSq8pzho4VusuQCpg@mail.gmail.com>

No the url is 404 so I can't see the file.



On Wed, Dec 19, 2012 at 3:43 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Whoa.   404 a.k.a device not found?****
>
> ** **
>
> This could be as I suspected, router capacity exceeded or general NAT
> issue with this router?****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, December 19, 2012 1:27 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
> ** **
>
> it gives 404 err****
>
> ** **
>
> On Tue, Dec 18, 2012 at 3:37 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> OK, I got one the logs!  I have trimmed the file down, but it is still
> over 5 MB and pastebin just chokes on it.****
>
>  ****
>
> It is on my server:  http://www.bizfocused.com/service_not_found.log****
>
>  ****
>
> Thanks for looking.****
>
>  ****
>
> If you really need pcap, I am afraid I need detailed instructions on
> installing/running it.****
>
>  ****
>
> Sean****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 17, 2012 7:00 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> usually the 406 is done in an established call to refuse a codec change
> during re-invite.****
>
> Its possible the other end thinks we want to change the codec when we do
> the session-timer re-invite but I'm sure we don't but the sip trace can
> help shed some light.  You can run a pcap too at the same time so when we
> find the bad call in the logs we can filter it out of the pcap too.  To
> avoid it getting too big you can just restart it every so often or use
> sippcapdump and delete calls that are not affected.****
>
>  ****
>
>  ****
>
>  ****
>
> On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> Waiting for another failure with siptrace REALLY on this time.****
>
>  ****
>
> If the user has clicked DND on these cisco phones, could that cause these
> messages?****
>
>  ****
>
> Sean****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 17, 2012 12:32 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> The siptrace is not in here?  Did you enable it on the console and then
> capture to the log file perhaps?****
>
> you should do "sofia tracelevel debug" too to route traces to the log file.
> ****
>
>  ****
>
> Its hard to say for sure with no sip trace but it seems like the far end
> is rejecting the session timer re-invite causing the call to end.****
>
>  ****
>
>  ****
>
> On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com>
> wrote:****
>
> http://pastebin.freeswitch.org/20342****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
> ** **
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
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From jpablolorenzetti at hotmail.com  Thu Dec 20 00:50:34 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Wed, 19 Dec 2012 21:50:34 +0000
Subject: [Freeswitch-users] call_timeout fails for callcenter agents
Message-ID: <DUB002-W1033E5834F19C765571A6B1D9300@phx.gbl>

Hi, i m trying to configure a callcenter, i need the application to ring the agentsfor 10 seconds and then move on to the next agent in the list, everything works ok except that the agents ring for more than 10 seconds, so the channel var call_timeout is not being used bythe callcenter application, probably due to a misconfiguration somewhere but i cant see where ... this is my config for the agents:
<agent name="2 at callcenter" type="callback" contact=" ignore_early_media=true,originate_timeout=10,call_timeout=10]sofia/gateway/huawei_csoft/6611290" status="Available" max-no-answer="3" wrap-up-time="10" reject- delay-time="5" busy-delay-time="5" no-answer-delay-time="5" />
and this is my config in the dialplan (also tried with exports):
<action application="set" data="hangup_after_bridge=true"/><action application="set" data="call_timeout=10"/><action application="set" data="originate_timeout=10"/><action application="set" data="ignore_early_media=true"/><action application="callcenter" data="callcenter at default"/>
i can not see what i m doing wrong so i added that variables in both places (dialplan and the agents defs)and i see in the logs that they are being set in both places:
EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(hangup_after_bridge=true)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [hangup_after_bridge]=[true]EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(call_timeout=10)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [call_timeout]=[10]EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(originate_timeout=10)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [originate_timeout]=[10]EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(ignore_early_media=true)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [ignore_early_media]=[true]EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 callcenter(callcenter at default)2012-12-19 15:38:16.716876 [DEBUG] mod_callcenter.c:2508 Member 2802018 <2802018> joining queue callcenter at default2012-12-19 15:38:16.736878 [DEBUG] mod_local_stream.c:417 Opening Stream [moh/8000] 8000hz2012-12-19 15:38:16.736878 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms2012-12-19 15:38:16.776878 [DEBUG] mod_callcenter.c:1049 Updated Agent 2 at callcenter set state = Receiving2012-12-19 15:38:16.786922 [DEBUG] switch_ivr_originate.c:2005 Parsing global variables2012-12-19 15:38:16.786922 [DEBUG] switch_ivr_originate.c:2425 Parsing session specific variables2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [ignore_early_media]=[true]2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [originate_timeout]=[10]2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [call_timeout]=[10]2012-12-19 15:38:16.786922 [NOTICE] switch_channel.c:951 New Channel sofia/external-huawei_gw/6611290 [09d48fc6-d133-4673-9f12-074af601f1f2]

kindly give me a hint what i m doing wrong, thanks a lot for your help.
 		 	   		  
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From sdevoy at bizfocused.com  Thu Dec 20 01:33:55 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 19 Dec 2012 17:33:55 -0500
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <CAKbxfG_4Xp=DwOnq6zFo=S-30BpN_BME+wSq8pzho4VusuQCpg@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>	<541b01cddcb1$72330200$56990600$@bizfocused.com>	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>	<5da601cddd67$dc76b6f0$956424d0$@bizfocused.com>	<CAKbxfG_eXUY=eLnp-_xLks4-bXaoR3YiwqvsJ0teV15AF8w6vA@mail.gmail.com>	<681801cdde31$d4fe1ec0$7efa5c40$@bizfocused.com>
	<CAKbxfG_4Xp=DwOnq6zFo=S-30BpN_BME+wSq8pzho4VusuQCpg@mail.gmail.com>
Message-ID: <68b001cdde38$f31f6bf0$d95e43d0$@bizfocused.com>

Sorry, server won't serve .log files!

http://www.bizfocused.com/service_not_found.txt

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, December 19, 2012 4:50 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

No the url is 404 so I can't see the file.

 

 

On Wed, Dec 19, 2012 at 3:43 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Whoa.   404 a.k.a device not found?

 

This could be as I suspected, router capacity exceeded or general NAT issue
with this router?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, December 19, 2012 1:27 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

it gives 404 err

 

On Tue, Dec 18, 2012 at 3:37 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

OK, I got one the logs!  I have trimmed the file down, but it is still over
5 MB and pastebin just chokes on it.

 

It is on my server:  http://www.bizfocused.com/service_not_found.log

 

Thanks for looking.

 

If you really need pcap, I am afraid I need detailed instructions on
installing/running it.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 7:00 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

usually the 406 is done in an established call to refuse a codec change
during re-invite.

Its possible the other end thinks we want to change the codec when we do the
session-timer re-invite but I'm sure we don't but the sip trace can help
shed some light.  You can run a pcap too at the same time so when we find
the bad call in the logs we can filter it out of the pcap too.  To avoid it
getting too big you can just restart it every so often or use sippcapdump
and delete calls that are not affected.

 

 

 

On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Waiting for another failure with siptrace REALLY on this time.

 

If the user has clicked DND on these cisco phones, could that cause these
messages?

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 12:32 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

The siptrace is not in here?  Did you enable it on the console and then
capture to the log file perhaps?

you should do "sofia tracelevel debug" too to route traces to the log file.

 

Its hard to say for sure with no sip trace but it seems like the far end is
rejecting the session timer re-invite causing the call to end.

 

 

On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

http://pastebin.freeswitch.org/20342





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900> 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900> 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900>  


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 

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From sdevoy at bizfocused.com  Thu Dec 20 01:36:10 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 19 Dec 2012 17:36:10 -0500
Subject: [Freeswitch-users] Install 1.2.Stable still fails. Ver 2
In-Reply-To: <6909951704489875258@unknownmsgid>
References: <CCF74286.6246F%krice@freeswitch.org>
	<6909951704489875258@unknownmsgid>
Message-ID: <68d001cdde39$3eddd540$bc997fc0$@bizfocused.com>

In my case it was just the stupid & versus &&.

I split it into:

make

make install.

Worked perfectly.

 

Thanks all.

Sean

(I hate Unix)

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of S. Scott
Sent: Wednesday, December 19, 2012 12:09 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Install 1.2.Stable still fails. Ver 2

 

I'm not that strong in these areas, but i too tried EPEL flavor AND i too
ran into freeswitch build issues. I manually yum'd the fs dependancies and
hit issues there too.  After googling around, I flushed and went back to the
centos iso AND

 

Do you epel pull,

 

THEN THE FIRST THING YOU MUST DO BEFORE UPDATING OR OTHER YUM ACTION IS (was
for me)

 

yum install yum-plugin-protectbase.noarch

 

The readings i found assert you MUST protect the base repo's from the epel
repo's.  The above plugin does this repo boundry.  Everything -- dependancy
chain and freeSWITCH -- yum'ed and make'd just fine thereafter.

 

Dont know if this is your issue,,, like i said im not that strong in these
areas, but it solved my epel-version issues. 

goodluck,


-----

iThing:  Big thumbs & little keys. Please excuse typo, spelling and grammar
errors . Last night I played a blank CD at full blast. The Mime next door
went nuts.

 


On 2012-12-19, at 11:11, Ken Rice <krice at freeswitch.org> wrote:

Your make command is wrong...

Its make && make install...

Notice the two (2) &'s there you have one in the command you stated...

 make & make install  (1 &) tries to run 2 makes concurrently... The first
make will get backgrounded, and the 2nd make will run in the foreground....

This will screw things up royally...

As for Centos5 the entire tree is tested routinely on Centos5... 99% of the
development occurs on centos5, so I hate to say it but if you are having a
problem building on that specific platform it is almost always (99.999% of
the time) a problem with your installation environment...

So sit back and relax... Also keep in mind that it seems we have the highest
concentration of FreeSWITCH users in the West (ie North and South America)
and not all of them are around all the time... So when you ask a question it
make take several hours or even a day before someone with the right
knowledge can answer your query....

If you are deploying in a commercial environment where you might need
priority support, I would suggest contacting consulting at freeswitch.org where
you can get someone from the FreeSWITCH team to help you via a support
agreement... 

K


On 12/18/12 9:11 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

OK, I had my Centos restored to base install 5.8.
 
The install fails.  Here is EXACTLY what I did:
 
Login to root
cd /usr/local/src
rpm -Uvh
http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm

git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
cd /usr/local/src/freeswitch
./bootstrap.sh
./configure
make & make install
 
 
It fails with:
/bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link  gcc
-g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
-I./include -I/usr/local/src/freeswitch/libs/apr/include/arch/unix
-I./include/arch/unix -I/usr/local/src/freeswitch/libs/apr/include
-version-info 2:8:2    -o libapr-1.la -rpath /usr/local/freeswitch/lib
passwd/apr_getpass.lo strings/apr_cpystrn.lo strings/apr_strnatcmp.lo
strings/apr_strings.lo strings/apr_strtok.lo strings/apr_fnmatch.lo
strings/apr_snprintf.lo tables/apr_tables.lo tables/apr_hash.lo
atomic/unix/apr_atomic.lo dso/unix/dso.lo file_io/unix/flock.lo
file_io/unix/readwrite.lo file_io/unix/filepath_util.lo file_io/unix/seek.lo
file_io/unix/dir.lo file_io/unix/mktemp.lo file_io/unix/filedup.lo
file_io/unix/tempdir.lo file_io/unix/filepath.lo file_io/unix/pipe.lo
file_io/unix/open.lo file_io/unix/filestat.lo file_io/unix/copy.lo
file_io/unix/fileacc.lo file_io/unix/fullrw.lo locks/unix/thread_rwlock.lo
locks/unix/thread_mutex.lo locks/unix/thread_cond.lo
locks/unix/proc_mutex.lo locks/unix/global_mutex.lo memory/unix/apr_pools.lo
misc/unix/charset.lo misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo
misc/unix/start.lo misc/unix/errorcodes.lo misc/unix/getopt.lo
misc/unix/otherchild.lo mmap/unix/mmap.lo mmap/unix/common.lo
network_io/unix/sockaddr.lo network_io/unix/sockopt.lo
network_io/unix/sendrecv.lo network_io/unix/multicast.lo
network_io/unix/sockets.lo network_io/unix/inet_ntop.lo
network_io/unix/inet_pton.lo poll/unix/epoll.lo poll/unix/select.lo
poll/unix/poll.lo poll/unix/port.lo poll/unix/kqueue.lo random/unix/sha2.lo
random/unix/apr_random.lo random/unix/sha2_glue.lo shmem/unix/shm.lo
support/unix/waitio.lo threadproc/unix/procsup.lo threadproc/unix/thread.lo
threadproc/unix/signals.lo threadproc/unix/proc.lo
threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
-lpthread
libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object
make[2]: *** [libapr-1.la] Error 1
make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'
make: *** [libs/apr/libapr-1.la] Error 2
/bin/sh /usr/local/src/freeswitch/libs/apr/libtool --silent --mode=link  gcc
-g -O2 -pthread  -g -O2 -DHAVE_CONFIG_H -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
-I./include -I/usr/local/src/freeswitch/libs/apr/include/arch/unix
-I./include/arch/unix -I/usr/local/src/freeswitch/libs/apr/include
-version-info 2:8:2    -o libapr-1.la -rpath /usr/local/freeswitch/lib
passwd/apr_getpass.lo strings/apr_cpystrn.lo strings/apr_strnatcmp.lo
strings/apr_strings.lo strings/apr_strtok.lo strings/apr_fnmatch.lo
strings/apr_snprintf.lo tables/apr_tables.lo tables/apr_hash.lo
atomic/unix/apr_atomic.lo dso/unix/dso.lo file_io/unix/flock.lo
file_io/unix/readwrite.lo file_io/unix/filepath_util.lo file_io/unix/seek.lo
file_io/unix/dir.lo file_io/unix/mktemp.lo file_io/unix/filedup.lo
file_io/unix/tempdir.lo file_io/unix/filepath.lo file_io/unix/pipe.lo
file_io/unix/open.lo file_io/unix/filestat.lo file_io/unix/copy.lo
file_io/unix/fileacc.lo file_io/unix/fullrw.lo locks/unix/thread_rwlock.lo
locks/unix/thread_mutex.lo locks/unix/thread_cond.lo
locks/unix/proc_mutex.lo locks/unix/global_mutex.lo memory/unix/apr_pools.lo
misc/unix/charset.lo misc/unix/env.lo misc/unix/version.lo misc/unix/rand.lo
misc/unix/start.lo misc/unix/errorcodes.lo misc/unix/getopt.lo
misc/unix/otherchild.lo mmap/unix/mmap.lo mmap/unix/common.lo
network_io/unix/sockaddr.lo network_io/unix/sockopt.lo
network_io/unix/sendrecv.lo network_io/unix/multicast.lo
network_io/unix/sockets.lo network_io/unix/inet_ntop.lo
network_io/unix/inet_pton.lo poll/unix/epoll.lo poll/unix/select.lo
poll/unix/poll.lo poll/unix/port.lo poll/unix/kqueue.lo random/unix/sha2.lo
random/unix/apr_random.lo random/unix/sha2_glue.lo shmem/unix/shm.lo
support/unix/waitio.lo threadproc/unix/procsup.lo threadproc/unix/thread.lo
threadproc/unix/signals.lo threadproc/unix/proc.lo
threadproc/unix/threadpriv.lo time/unix/time.lo time/unix/timestr.lo
user/unix/userinfo.lo user/unix/groupinfo.lo   -luuid -lrt -ldl -lcrypt
-lpthread
libtool: link: `passwd/apr_getpass.lo' is not a valid libtool object
make[2]: *** [libapr-1.la] Error 1
make[2]: Leaving directory `/usr/local/src/freeswitch/libs/apr'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch/libs/apr'
make: *** [libs/apr/libapr-1.la] Error 2
[1]+  Exit 2                  make
 
What next guys?
 
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

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


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #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

FreeSWITCH-users mailing 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 yungwei at resolvity.com  Thu Dec 20 02:31:24 2012
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 19 Dec 2012 18:31:24 -0500
Subject: [Freeswitch-users] Installing freeswitch-1.2.5.3 on CentOS 5
Message-ID: <33095823FD21DF429B481B5163264B799F6D8753B3@VMBX102.ihostexchange.net>

Hi,

I am having trouble installing freeswitch-1.2.5.3 on CentOS 5.5 from freeswitch-1.2.5.3.tar.bz2.
The following error shows up during make. I installed autoconf26x-2.63-4.ius.el5.
How can I tell make to use autoconf26x instead of autoconf? Thanks.

make[3]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/lib'
make[2]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/lib'
Making all in src
make[2]: Entering directory `/var/install/freeswitch-1.2.5.3/libs/curl/src'
(CDPATH="${ZSH_VERSION+.}:" && cd .. && autoheader)
aclocal.m4:20: warning: this file was generated for autoconf 2.67.
You have another version of autoconf.  It may work, but is not guaranteed to.
If you have problems, you may need to regenerate the build system entirely.
To do so, use the procedure documented by the package, typically `autoreconf'.
configure.ac:65: error: Autoconf version 2.62 or higher is required
aclocal.m4:8484: AM_INIT_AUTOMAKE is expanded from...
configure.ac:65: the top level
autom4te: /usr/bin/m4 failed with exit status: 63
autoheader: /usr/bin/autom4te failed with exit status: 63
make[2]: *** [config.h.in] Error 63
make[2]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl'
make: *** [libs/curl/lib/libcurl.la] Error 2




From sdevoy at bizfocused.com  Thu Dec 20 03:05:53 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 19 Dec 2012 19:05:53 -0500
Subject: [Freeswitch-users] Installing freeswitch-1.2.5.3 on CentOS 5
In-Reply-To: <33095823FD21DF429B481B5163264B799F6D8753B3@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B799F6D8753B3@VMBX102.ihostexchange.net>
Message-ID: <698601cdde45$c732d740$559885c0$@bizfocused.com>

HI Chen,

I am certainly no expert, but I just completed installing FS 1.2.5.3 on
Centos 5.8 today.  The first issue I had was EPEL must be added when
installing on Cenots 5.n

My second error was I used "make & make install" <== WRONG.  It should have
been "make && make install"

Here are the exact commands I used ,  comments in ():
login root
	cd /usr/local/src
(Add EPEL for 32 bit Centos 5.n):  	rpm -Uvh
http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm

Install pre-reqs:
	yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool
make ncurses-devel pkgconfig unixODBC-devel openssl-devel libogg-devel
libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel
expat-devel zlib zlib-devel bzip2 which gdbm-devel db4-devel

	git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git
	cd /usr/local/src/freeswitch
	./bootstrap.sh                (takes a while)
	./configure
	make  	            (takes 20 minutes or more)
	make install  

I 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 Yungwei
Chen
Sent: Wednesday, December 19, 2012 6:31 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Installing freeswitch-1.2.5.3 on CentOS 5

Hi,

I am having trouble installing freeswitch-1.2.5.3 on CentOS 5.5 from
freeswitch-1.2.5.3.tar.bz2.
The following error shows up during make. I installed
autoconf26x-2.63-4.ius.el5.
How can I tell make to use autoconf26x instead of autoconf? Thanks.

make[3]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/lib'
make[2]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/lib'
Making all in src
make[2]: Entering directory `/var/install/freeswitch-1.2.5.3/libs/curl/src'
(CDPATH="${ZSH_VERSION+.}:" && cd .. && autoheader)
aclocal.m4:20: warning: this file was generated for autoconf 2.67.
You have another version of autoconf.  It may work, but is not guaranteed
to.
If you have problems, you may need to regenerate the build system entirely.
To do so, use the procedure documented by the package, typically
`autoreconf'.
configure.ac:65: error: Autoconf version 2.62 or higher is required
aclocal.m4:8484: AM_INIT_AUTOMAKE is expanded from...
configure.ac:65: the top level
autom4te: /usr/bin/m4 failed with exit status: 63
autoheader: /usr/bin/autom4te failed with exit status: 63
make[2]: *** [config.h.in] Error 63
make[2]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl'
make: *** [libs/curl/lib/libcurl.la] 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

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




From yungwei at resolvity.com  Thu Dec 20 03:14:07 2012
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 19 Dec 2012 19:14:07 -0500
Subject: [Freeswitch-users] voices in the recordings are out of sync
In-Reply-To: <CAKzWOxVMYHSv5yOEiRdrAUGWOniwLfM-VEjzdtGZDyW2oTCHNw@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B799F6D8752B6@VMBX102.ihostexchange.net>
	<CAKzWOxVMYHSv5yOEiRdrAUGWOniwLfM-VEjzdtGZDyW2oTCHNw@mail.gmail.com>
Message-ID: <33095823FD21DF429B481B5163264B799F6D8753B5@VMBX102.ihostexchange.net>

Please tell me the list of files affected by this fix or the bug id. Thanks.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, December 18, 2012 8:26 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] voices in the recordings are out of sync

Latest version of FreeSWITCH has some updates that may fix this issue. I would update to 1.2.5.3 ASAP.
-MC
On Tue, Dec 18, 2012 at 1:56 PM, Yungwei Chen <yungwei at resolvity.com<mailto:yungwei at resolvity.com>> wrote:
Hi,

I found one issue that voices are always out of sync in the recordings.
I am running freeswitch-1.2.5.1-1 on CentOS 5 (i386), which was installed from yum.
I am having trouble installing the latest version from source due to an error: Autoconf version 2.62 or higher is required.
It would be nice if someone can reproduce this issue against HEAD. Thanks.

Here're the steps to reproduce it. The idea is to call a phone number and then bridge to another phone number while the entire session is being recorded.
1. In dialplan/public.xml, make sure you have a dialplan to handle any 10 digit phone numbers.
    <extension name="public_extensions">
      <condition field="destination_number" expression="^\d{10}$">
        <action application="transfer" data="main XML default"/>
      </condition>
    </extension>

2. In dialplan/default/main.xml, make sure you have an extension to handle the call in the default context.
  <extension name="test">
      <condition field="${destination_number}" expression="^main$">
        <action application="set" data="RECORD_STEREO=false"/>
        <action application="answer" />
        <action application="record_session" data="/tmp/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
        <action application="bridge" data="{ignore_early_media=false}[leg_timeout=60]sofia/gateway/gw1/1234567890"/>
      </condition>
  </extension>

3. In sip_profiles/external/gateways.xml, make sure you have a gateway that allows you to make an outbound call.
    <include>
        <gateway name="gw1">
          <param name="username" value=""/>
          <param name="password" value=""/>
          <param name="realm" value=""/>
          <param name="from-domain" value=""/>
          <param name="extension" value=""/>
          <param name="expire-seconds" value="60"/>
          <param name="register" value="false"/>
          <param name="retry-seconds" value="60"/>
        </gateway>
    </include>

4. make a call to one of the allowed 10-digit phone numbers in your environment.
5. Once the call is answered, the caller shall start to count from 1 to 60 with some pause after each number.
6. The callee shall repeat each number he/she heard from the caller.
7. You should be able to hear that 2 voices in the recoridng (/tmp/rec.wav) are out of sync.

_________________________________________________________________________
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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

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From msc at freeswitch.org  Thu Dec 20 03:34:52 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Dec 2012 16:34:52 -0800
Subject: [Freeswitch-users] voices in the recordings are out of sync
In-Reply-To: <33095823FD21DF429B481B5163264B799F6D8753B5@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B799F6D8752B6@VMBX102.ihostexchange.net>
	<CAKzWOxVMYHSv5yOEiRdrAUGWOniwLfM-VEjzdtGZDyW2oTCHNw@mail.gmail.com>
	<33095823FD21DF429B481B5163264B799F6D8753B5@VMBX102.ihostexchange.net>
Message-ID: <CAKzWOxVZmzXe3e2ZXj9oDw8AX0gC4b=OMoJ74EeDCxDNwTRw+Q@mail.gmail.com>

Use fisheye.freeswitch.org or do a git checkout and "git log -p"
-MC

On Wed, Dec 19, 2012 at 4:14 PM, Yungwei Chen <yungwei at resolvity.com> wrote:

> Please tell me the list of files affected by this fix or the bug id.
> Thanks.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Tuesday, December 18, 2012 8:26 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] voices in the recordings are out of sync
> ****
>
> ** **
>
> Latest version of FreeSWITCH has some updates that may fix this issue. I
> would update to 1.2.5.3 ASAP.
> -MC****
>
> On Tue, Dec 18, 2012 at 1:56 PM, Yungwei Chen <yungwei at resolvity.com>
> wrote:****
>
> Hi,
>
> I found one issue that voices are always out of sync in the recordings.
> I am running freeswitch-1.2.5.1-1 on CentOS 5 (i386), which was installed
> from yum.
> I am having trouble installing the latest version from source due to an
> error: Autoconf version 2.62 or higher is required.
> It would be nice if someone can reproduce this issue against HEAD. Thanks.
>
> Here're the steps to reproduce it. The idea is to call a phone number and
> then bridge to another phone number while the entire session is being
> recorded.
> 1. In dialplan/public.xml, make sure you have a dialplan to handle any 10
> digit phone numbers.
>     <extension name="public_extensions">
>       <condition field="destination_number" expression="^\d{10}$">
>         <action application="transfer" data="main XML default"/>
>       </condition>
>     </extension>
>
> 2. In dialplan/default/main.xml, make sure you have an extension to handle
> the call in the default context.
>   <extension name="test">
>       <condition field="${destination_number}" expression="^main$">
>         <action application="set" data="RECORD_STEREO=false"/>
>         <action application="answer" />
>         <action application="record_session"
> data="/tmp/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
>         <action application="bridge"
> data="{ignore_early_media=false}[leg_timeout=60]sofia/gateway/gw1/1234567890"/>
>       </condition>
>   </extension>
>
> 3. In sip_profiles/external/gateways.xml, make sure you have a gateway
> that allows you to make an outbound call.
>     <include>
>         <gateway name="gw1">
>           <param name="username" value=""/>
>           <param name="password" value=""/>
>           <param name="realm" value=""/>
>           <param name="from-domain" value=""/>
>           <param name="extension" value=""/>
>           <param name="expire-seconds" value="60"/>
>           <param name="register" value="false"/>
>           <param name="retry-seconds" value="60"/>
>         </gateway>
>     </include>
>
> 4. make a call to one of the allowed 10-digit phone numbers in your
> environment.
> 5. Once the call is answered, the caller shall start to count from 1 to 60
> with some pause after each number.
> 6. The callee shall repeat each number he/she heard from the caller.
> 7. You should be able to hear that 2 voices in the recoridng
> (/tmp/rec.wav) are out of sync.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
> ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From steveayre at gmail.com  Thu Dec 20 03:41:31 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Dec 2012 00:41:31 +0000
Subject: [Freeswitch-users] Installing freeswitch-1.2.5.3 on CentOS 5
In-Reply-To: <33095823FD21DF429B481B5163264B799F6D8753B3@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B799F6D8753B3@VMBX102.ihostexchange.net>
Message-ID: <CAFiqYumTnU3HV2hrTeFJRZSwO6vHHGTnou6ybX6UN4FkHmxNBg@mail.gmail.com>

Install the autoconf-2.67 package

On 19 December 2012 23:31, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> I am having trouble installing freeswitch-1.2.5.3 on CentOS 5.5 from
> freeswitch-1.2.5.3.tar.bz2.
> The following error shows up during make. I installed
> autoconf26x-2.63-4.ius.el5.
> How can I tell make to use autoconf26x instead of autoconf? Thanks.
>
> make[3]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/lib'
> make[2]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/lib'
> Making all in src
> make[2]: Entering directory `/var/install/freeswitch-1.2.5.3/libs/curl/src'
> (CDPATH="${ZSH_VERSION+.}:" && cd .. && autoheader)
> aclocal.m4:20: warning: this file was generated for autoconf 2.67.
> You have another version of autoconf.  It may work, but is not guaranteed
> to.
> If you have problems, you may need to regenerate the build system entirely.
> To do so, use the procedure documented by the package, typically
> `autoreconf'.
> configure.ac:65: error: Autoconf version 2.62 or higher is required
> aclocal.m4:8484: AM_INIT_AUTOMAKE is expanded from...
> configure.ac:65: the top level
> autom4te: /usr/bin/m4 failed with exit status: 63
> autoheader: /usr/bin/autom4te failed with exit status: 63
> make[2]: *** [config.h.in] Error 63
> make[2]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/src'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl'
> make: *** [libs/curl/lib/libcurl.la] 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Thu Dec 20 03:45:14 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 19 Dec 2012 18:45:14 -0600
Subject: [Freeswitch-users] Installing freeswitch-1.2.5.3 on CentOS 5
In-Reply-To: <33095823FD21DF429B481B5163264B799F6D8753B3@VMBX102.ihostexchange.net>
Message-ID: <CCF7BBBA.62506%krice@freeswitch.org>

Did you install all the pre-req's as stated on the wiki page?

K

On 12/19/12 5:31 PM, "Yungwei Chen" <yungwei at resolvity.com> wrote:

> Hi,
> 
> I am having trouble installing freeswitch-1.2.5.3 on CentOS 5.5 from
> freeswitch-1.2.5.3.tar.bz2.
> The following error shows up during make. I installed
> autoconf26x-2.63-4.ius.el5.
> How can I tell make to use autoconf26x instead of autoconf? Thanks.
> 
> make[3]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/lib'
> make[2]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/lib'
> Making all in src
> make[2]: Entering directory `/var/install/freeswitch-1.2.5.3/libs/curl/src'
> (CDPATH="${ZSH_VERSION+.}:" && cd .. && autoheader)
> aclocal.m4:20: warning: this file was generated for autoconf 2.67.
> You have another version of autoconf.  It may work, but is not guaranteed to.
> If you have problems, you may need to regenerate the build system entirely.
> To do so, use the procedure documented by the package, typically `autoreconf'.
> configure.ac:65: error: Autoconf version 2.62 or higher is required
> aclocal.m4:8484: AM_INIT_AUTOMAKE is expanded from...
> configure.ac:65: the top level
> autom4te: /usr/bin/m4 failed with exit status: 63
> autoheader: /usr/bin/autom4te failed with exit status: 63
> make[2]: *** [config.h.in] Error 63
> make[2]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/src'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl'
> make: *** [libs/curl/lib/libcurl.la] 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From krice at freeswitch.org  Thu Dec 20 03:54:09 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 19 Dec 2012 18:54:09 -0600
Subject: [Freeswitch-users] Installing freeswitch-1.2.5.3 on CentOS 5
In-Reply-To: <CAFiqYumTnU3HV2hrTeFJRZSwO6vHHGTnou6ybX6UN4FkHmxNBg@mail.gmail.com>
Message-ID: <CCF7BDD1.62509%krice@freeswitch.org>

Autoconf should be fine, if you keep getting complaints like this run the
rebootstrap script


On 12/19/12 6:41 PM, "Steven Ayre" <steveayre at gmail.com> wrote:

> Install the autoconf-2.67 package
> 
> On 19 December 2012 23:31, Yungwei Chen <yungwei at resolvity.com> wrote:
>> Hi,
>> 
>> I am having trouble installing freeswitch-1.2.5.3 on CentOS 5.5 from
>> freeswitch-1.2.5.3.tar.bz2.
>> The following error shows up during make. I installed
>> autoconf26x-2.63-4.ius.el5.
>> How can I tell make to use autoconf26x instead of autoconf? Thanks.
>> 
>> make[3]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/lib'
>> make[2]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/lib'
>> Making all in src
>> make[2]: Entering directory `/var/install/freeswitch-1.2.5.3/libs/curl/src'
>> (CDPATH="${ZSH_VERSION+.}:" && cd .. && autoheader)
>> aclocal.m4:20: warning: this file was generated for autoconf 2.67.
>> You have another version of autoconf. ?It may work, but is not guaranteed to.
>> If you have problems, you may need to regenerate the build system entirely.
>> To do so, use the procedure documented by the package, typically
>> `autoreconf'.
>> configure.ac:65 <http://configure.ac:65> : error: Autoconf version 2.62 or
>> higher is required
>> aclocal.m4:8484: AM_INIT_AUTOMAKE is expanded from...
>> configure.ac:65 <http://configure.ac:65> : the top level
>> autom4te: /usr/bin/m4 failed with exit status: 63
>> autoheader: /usr/bin/autom4te failed with exit status: 63
>> make[2]: *** [config.h.in <http://config.h.in> ] Error 63
>> make[2]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl/src'
>> make[1]: *** [all-recursive] Error 1
>> make[1]: Leaving directory `/var/install/freeswitch-1.2.5.3/libs/curl'
>> make: *** [libs/curl/lib/libcurl.la <http://libcurl.la> ] 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From lloyd.aloysius at gmail.com  Thu Dec 20 04:32:44 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Wed, 19 Dec 2012 20:32:44 -0500
Subject: [Freeswitch-users] git - llibs/sipcc/
Message-ID: <CAGvuWYZ20JvufciRep9y6BcoED+YyW47VE-o3f_wi2WD37HD_w@mail.gmail.com>

Hello All:

Today I did a git pull and see lots of libs/sipcc/ - What is the new
feature or enhancement added to the tree?

Thanks
Lloyd
*
*
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From krice at freeswitch.org  Thu Dec 20 04:36:58 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 19 Dec 2012 19:36:58 -0600
Subject: [Freeswitch-users] git - llibs/sipcc/
In-Reply-To: <CAGvuWYZ20JvufciRep9y6BcoED+YyW47VE-o3f_wi2WD37HD_w@mail.gmail.com>
Message-ID: <CCF7C7DA.6250D%krice@freeswitch.org>

Nothing yet... Sipcc is a sdp parser a couple of the core devs are looking
into


On 12/19/12 7:32 PM, "Lloyd Aloysius" <lloyd.aloysius at gmail.com> wrote:

> Hello All:
> 
> Today I did a git pull and see lots of?libs/sipcc/ - What is the new feature
> or enhancement added to the tree?
> 
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From lloyd.aloysius at gmail.com  Thu Dec 20 04:59:24 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Wed, 19 Dec 2012 20:59:24 -0500
Subject: [Freeswitch-users] git - llibs/sipcc/
In-Reply-To: <CCF7C7DA.6250D%krice@freeswitch.org>
References: <CAGvuWYZ20JvufciRep9y6BcoED+YyW47VE-o3f_wi2WD37HD_w@mail.gmail.com>
	<CCF7C7DA.6250D%krice@freeswitch.org>
Message-ID: <CAGvuWYZL-p0yj-G3zYJPnAy1dhT9nAikg9nVw4WCuihwQ-hfdA@mail.gmail.com>

Ken, Thank you for the info.




On Wed, Dec 19, 2012 at 8:36 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Nothing yet... Sipcc is a sdp parser a couple of the core devs are
> looking into
>
>
>
> On 12/19/12 7:32 PM, "Lloyd Aloysius" <lloyd.aloysius at gmail.com> wrote:
>
> Hello All:
>
> Today I did a git pull and see lots of libs/sipcc/ - What is the new
> feature or enhancement added to the tree?
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Thu Dec 20 06:41:39 2012
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 20 Dec 2012 11:41:39 +0800
Subject: [Freeswitch-users] fifo questions
In-Reply-To: <CAKzWOxXQpU5Lr7BYK2WUM_2kmSF-=z2cMVEWDk4MQ3_4iNJ+Kw@mail.gmail.com>
References: <E75D158C1EE546CEBF617224942B26FF@gmail.com>
	<CAKzWOxXQpU5Lr7BYK2WUM_2kmSF-=z2cMVEWDk4MQ3_4iNJ+Kw@mail.gmail.com>
Message-ID: <EB33B9615493423BAE671E21ACEAB7BA@gmail.com>

I didn't config anything, I created on-hook agents with fifo_member add and it's automatically created and ring all when someone drops into the fifo. Which is the default I think.

I answered my own question by reading the code and get more info missed on wiki and I will update when I found some time.

Thanks.


On Wednesday, December 19, 2012 at 10:42 AM, Michael Collins wrote:

> Seven,
> 
> Could you share with us your fifo config? Others have asked about how to ring all agents and it seems our documentation is lacking.
> 
> Thanks!
> -MC
> 
> On Fri, Dec 14, 2012 at 8:43 PM, Seven Du <dujinfang at gmail.com (mailto:dujinfang at gmail.com)> wrote:
> > Hi, 
> > 
> > 
> > I'm using mod_fifo with onhook agents, and when a caller in it will ring all agents. The problem is that if an agent is placing an outbound call and the fifo still ring it regardless it's "busy". I manually set on the phone to accept only one channel solved the problem. But mod_fifo still try to ring it, is it possible to not ring the "busy" agent? I found fifo_track_calls, might work with this? 
> > 
> > also, what's the purpose of fifo_add_outbound? should it have a difference with fifo_member add ?
> > 
> > 
> > Thanks.
> > 
> > 
> > -- 
> > Seven Du
> > Sent with Sparrow (http://www.sparrowmailapp.com/?sig)
> > 
> > 
> > _________________________________________________________________________
> > 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
> > 
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.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
> 
> 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 regis.freeswitch.org at tornad.net  Thu Dec 20 10:24:08 2012
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Thu, 20 Dec 2012 08:24:08 +0100
Subject: [Freeswitch-users] call_timeout fails for callcenter agents
In-Reply-To: <DUB002-W1033E5834F19C765571A6B1D9300@phx.gbl>
References: <DUB002-W1033E5834F19C765571A6B1D9300@phx.gbl>
Message-ID: <CAHWtr1QoU6pc8Z1E1Oo4NLa+=FtMQqE8UYwdvHUxC1h9g9k3NA@mail.gmail.com>

Hello.
I think you must use leg_timeout instead of call_timeout.

Regards


2012/12/19 Juan Pablo L. <jpablolorenzetti at hotmail.com>

> Hi, i m trying to configure a callcenter, i need the application to ring
> the agentsfor 10 seconds and then move on to the next agent in the list,
> everything works ok except that the agents ring for more than 10 seconds,
> so the channel var call_timeout is not being used by
> the callcenter application, probably due to a misconfiguration somewhere
> but i cant see where ... this is my config for the agents:
>
> <agent name="2 at callcenter" type="callback" contact="
> ignore_early_media=true,originate_timeout=10,call_timeout=10]sofia/gateway/huawei_csoft/
> 6611290" status="Availabl
> e" max-no-answer="3" wrap-up-time="10" reject- delay-time="5"
> busy-delay-time="5" no-answer-delay-time="5" />
>
> and this is my config in the dialplan (also tried with exports):
>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set" data="call_timeout=10"/>
> <action application="set" data="originate_timeout=10"/>
> <action application="set" data="ignore_early_media=true"/>
> <action application="callcenter" data="callcenter at default"/>
>
> i can not see what i m doing wrong so i added that variables in both
> places (dialplan and the agents defs)
> and i see in the logs that they are being set in both places:
>
> EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062set(hangup_after_bridge=true)
> 2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335
> sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET
> [hangup_after_bridge]=[true]
> EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062set(call_timeout=10)
> 2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335
> sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [call_timeout]=[10]
> EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062set(originate_timeout=10)
> 2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335
> sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET
> [originate_timeout]=[10]
> EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062set(ignore_early_media=true)
> 2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335
> sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET
> [ignore_early_media]=[true]
> EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062callcenter(callcenter at default
> )
> 2012-12-19 15:38:16.716876 [DEBUG] mod_callcenter.c:2508 Member 2802018
> <2802018> joining queue callcenter at default
> 2012-12-19 15:38:16.736878 [DEBUG] mod_local_stream.c:417 Opening Stream
> [moh/8000] 8000hz
> 2012-12-19 15:38:16.736878 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2012-12-19 15:38:16.776878 [DEBUG] mod_callcenter.c:1049 Updated Agent
> 2 at callcenter set state = Receiving
> 2012-12-19 15:38:16.786922 [DEBUG] switch_ivr_originate.c:2005 Parsing
> global variables
> 2012-12-19 15:38:16.786922 [DEBUG] switch_ivr_originate.c:2425 Parsing
> session specific variables
> 2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable
> [ignore_early_media]=[true]
> 2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable
> [originate_timeout]=[10]
> 2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable
> [call_timeout]=[10]
> 2012-12-19 15:38:16.786922 [NOTICE] switch_channel.c:951 New Channel
> sofia/external-huawei_gw/6611290 [09d48fc6-d133-4673-9f12-074af601f1f2]
>
>
> kindly give me a hint what i m doing wrong, thanks a lot for your help.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 miha at softnet.si  Thu Dec 20 10:44:37 2012
From: miha at softnet.si (Miha)
Date: Thu, 20 Dec 2012 08:44:37 +0100
Subject: [Freeswitch-users] Tracing
Message-ID: <50D2C1E5.10708@softnet.si>

Hi,

I am experiencing some problems related with media with some users. It 
is almost impossible to figure it out where is a problem because this 
problem is not presented all day.
I need some sip trace dump for a specific user if it is possible due to 
a lot of traffic. It would be nice that this would not cause to much CPU 
load and disk consumption as this would need to be running and tracing 24/7.

Users are reporting the thy exeriacing noise in conversation and 
sometime thy do not hear other side.

What would be the best way?

Thanks!

Miha


From steveayre at gmail.com  Thu Dec 20 11:16:50 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Dec 2012 08:16:50 +0000
Subject: [Freeswitch-users] git - llibs/sipcc/
In-Reply-To: <CAGvuWYZ20JvufciRep9y6BcoED+YyW47VE-o3f_wi2WD37HD_w@mail.gmail.com>
References: <CAGvuWYZ20JvufciRep9y6BcoED+YyW47VE-o3f_wi2WD37HD_w@mail.gmail.com>
Message-ID: <CAFiqYunfRjzvox4rKEiXUszfhAjYYzYgU_2cZiRz2MOmmHzJAA@mail.gmail.com>

SDP currently is handled by sofia in mod_sofia SIP, I believe they're
looking at using sipcc to provide SDP support within the core itself which
would allow it to be reused by multiple endpoints (eg for WebRTC).

But not in use just yet

On 20 December 2012 01:32, Lloyd Aloysius <lloyd.aloysius at gmail.com> wrote:

> Hello All:
>
> Today I did a git pull and see lots of libs/sipcc/ - What is the new
> feature or enhancement added to the tree?
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ntomer at newgen.co.in  Thu Dec 20 11:33:09 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Thu, 20 Dec 2012 14:03:09 +0530
Subject: [Freeswitch-users] Quick help needed
Message-ID: <030101cdde8c$a44f2390$eced6ab0$@co.in>

Hi,

 

In my application, I am parking a call to an extension using valet-park.
Before this I want to read  a message to caller, but am not able to. I tried
to use say and phrase but am not able to get the syntax right. Please help
me.

 

Dialplan -

 

<extension>

                        <condition field="destination_number"
expression="^(450)$">
                                    <action application="set"
data="caller_id_name=Account Opening Request" />
                                    <action application="set"
data="call_timeout=60" />
                                    <action application="set"
data="originate_timeout=60" />
                                    <action inline="true" application="lua"
data="accountopening.lua"/>
                                    <action application="db"
data="insert/accountopening/${parkednumber}/nonumber" />    
                                    <action application="say" data="en
messages pronounced 'Please wait for an agent to pick your call.'" /> 
                                    <action application="valet_park"
data="my_lot ${parkednumber}" />          

              </condition>
</extension>

 

Regards

 

Nitin


Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From avi at avimarcus.net  Thu Dec 20 11:42:00 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 20 Dec 2012 10:42:00 +0200
Subject: [Freeswitch-users] Quick help needed
In-Reply-To: <030101cdde8c$a44f2390$eced6ab0$@co.in>
References: <030101cdde8c$a44f2390$eced6ab0$@co.in>
Message-ID: <CALNZuwkisdhY_FUA9AiTca2rYTGYPgFcUN_C96N-w3DTyQi20w@mail.gmail.com>

SAY isn't for dynamic text, but rather to join together pre-recorded files
for saying numbers, dates, etc. see: SAY on the
wiki<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say>

You either want:
1) playback <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_playback> of
a pre-recorded file.
2) use TTS with the free flite e.g. with
SPEAK<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_speak> (not
say) e.g.:

<action application="speak" data="flite|kal|FreeSWITCH is awesome"/>


-Avi


On Thu, Dec 20, 2012 at 10:33 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:

>  Hi,****
>
> ** **
>
> In my application, I am parking a call to an extension using valet-park.
> Before this I want to read  a message to caller, but am not able to. I
> tried to use say and phrase but am not able to get the syntax right. Please
> help me.****
>
> ** **
>
> Dialplan ?****
>
> ** **
>
> <extension>****
>
>                         <condition field="destination_number"
> expression="^(450)$">
>                                     <action application="set"
> data="caller_id_name=Account Opening Request" />
>                                     <action application="set"
> data="call_timeout=60" />
>                                     <action application="set"
> data="originate_timeout=60" />
>                                     <action inline="true"
> application="lua" data="accountopening.lua"/>
>                                     <action application="db"
> data="insert/accountopening/${parkednumber}/nonumber" />
>                                     <action application="say" data="en
> messages pronounced 'Please wait for an agent to pick your call.'" />
>                                     <action application="valet_park"
> data="my_lot ${parkednumber}" />          ****
>
>               </condition>
> </extension>****
>
> ** **
>
> Regards****
>
> ** **
>
> Nitin****
>
> Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ntomer at newgen.co.in  Thu Dec 20 11:56:39 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Thu, 20 Dec 2012 14:26:39 +0530
Subject: [Freeswitch-users] Quick help needed
In-Reply-To: <CALNZuwkisdhY_FUA9AiTca2rYTGYPgFcUN_C96N-w3DTyQi20w@mail.gmail.com>
References: <030101cdde8c$a44f2390$eced6ab0$@co.in>
	<CALNZuwkisdhY_FUA9AiTca2rYTGYPgFcUN_C96N-w3DTyQi20w@mail.gmail.com>
Message-ID: <031501cdde8f$ed006c90$c70145b0$@co.in>

Thanks Avi, it worked.

 

But I am using Say to dynamic text in following instances and it is working
fine -

 

<action application="play_and_get_digits" data="10 16 3 300 # say:'Press
your account number, followed by hash key' say:'Wrong Input' res \d+" />

<menu name="nitin_ivr"
                greet-long="say:Welcome to Newgen Bank Limited. Press 1 for
Account Opening, 2 for Fixed Deposits, 3 for Address Change or 4 for Closing
Account."
                greet-short="say:Welcome to Newgen. Press 1 for Account
Opening, 2 for Fixed Deposits, 3 for Address Change or 4 for Closing
Account."
                invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
                exit-sound="voicemail/vm-goodbye.wav"
                confirm-macro=""
                confirm-key=""
                tts-engine="flite"
                tts-voice="slt"
                confirm-attempts="3"
                timeout="3000"
                inter-digit-timeout="2000"
                max-failures="3"
                max-timeouts="3"
                digit-len="4">

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Thursday, December 20, 2012 2:12 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Quick help needed

 

SAY isn't for dynamic text, but rather to join together pre-recorded files
for saying numbers, dates, etc. see: SAY on the wiki
<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say> 

 

You either want:

1) playback <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_playback>
of a pre-recorded file.

2) use TTS with the free flite e.g. with SPEAK
<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_speak>  (not say)
e.g.:

<action application="speak" data="flite|kal|FreeSWITCH is awesome"/>




-Avi

 

On Thu, Dec 20, 2012 at 10:33 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:

Hi,

 

In my application, I am parking a call to an extension using valet-park.
Before this I want to read  a message to caller, but am not able to. I tried
to use say and phrase but am not able to get the syntax right. Please help
me.

 

Dialplan -

 

<extension>

                        <condition field="destination_number"
expression="^(450)$">
                                    <action application="set"
data="caller_id_name=Account Opening Request" />
                                    <action application="set"
data="call_timeout=60" />
                                    <action application="set"
data="originate_timeout=60" />
                                    <action inline="true" application="lua"
data="accountopening.lua"/>
                                    <action application="db"
data="insert/accountopening/${parkednumber}/nonumber" />    
                                    <action application="say" data="en
messages pronounced 'Please wait for an agent to pick your call.'" /> 
                                    <action application="valet_park"
data="my_lot ${parkednumber}" />          

              </condition>
</extension>

 

Regards

 

Nitin

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 


_________________________________________________________________________
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/> 

FreeSWITCH-users mailing list
FreeSWITCH-users at 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/> 

 


Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From steveayre at gmail.com  Thu Dec 20 16:16:02 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Dec 2012 13:16:02 +0000
Subject: [Freeswitch-users] Quick help needed
In-Reply-To: <CALNZuwkisdhY_FUA9AiTca2rYTGYPgFcUN_C96N-w3DTyQi20w@mail.gmail.com>
References: <030101cdde8c$a44f2390$eced6ab0$@co.in>
	<CALNZuwkisdhY_FUA9AiTca2rYTGYPgFcUN_C96N-w3DTyQi20w@mail.gmail.com>
Message-ID: <CAFiqYumnWWJh2=Se=xC+Y5sM-beXd=P1QafSUNVyLp+MqUCFtw@mail.gmail.com>

Bear in mind a prerecorded file will consume less CPU than performing TTS
if this is a prompt that'll be played a lot.

If you don't want to pay a voice actor or do it yourself, you could always
generate a prerecorded file using TTS.


On 20 December 2012 08:42, Avi Marcus <avi at avimarcus.net> wrote:

> SAY isn't for dynamic text, but rather to join together pre-recorded files
> for saying numbers, dates, etc. see: SAY on the wiki<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say>
>
> You either want:
> 1) playback<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_playback> of
> a pre-recorded file.
> 2) use TTS with the free flite e.g. with SPEAK<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_speak> (not
> say) e.g.:
>
> <action application="speak" data="flite|kal|FreeSWITCH is awesome"/>
>
>
> -Avi
>
>
> On Thu, Dec 20, 2012 at 10:33 AM, Nitin Tomer <ntomer at newgen.co.in> wrote:
>
>>  Hi,****
>>
>> ** **
>>
>> In my application, I am parking a call to an extension using valet-park.
>> Before this I want to read  a message to caller, but am not able to. I
>> tried to use say and phrase but am not able to get the syntax right. Please
>> help me.****
>>
>> ** **
>>
>> Dialplan ?****
>>
>> ** **
>>
>> <extension>****
>>
>>                         <condition field="destination_number"
>> expression="^(450)$">
>>                                     <action application="set"
>> data="caller_id_name=Account Opening Request" />
>>                                     <action application="set"
>> data="call_timeout=60" />
>>                                     <action application="set"
>> data="originate_timeout=60" />
>>                                     <action inline="true"
>> application="lua" data="accountopening.lua"/>
>>                                     <action application="db"
>> data="insert/accountopening/${parkednumber}/nonumber" />
>>                                     <action application="say" data="en
>> messages pronounced 'Please wait for an agent to pick your call.'" />
>>                                     <action application="valet_park"
>> data="my_lot ${parkednumber}" />          ****
>>
>>               </condition>
>> </extension>****
>>
>> ** **
>>
>> Regards****
>>
>> ** **
>>
>> Nitin****
>>
>> Disclaimer :- This e-mail and any attachment may contain confidential,
>> proprietary or legally privileged information. If you are not the original
>> intended recipient and have erroneously received this message, you are
>> prohibited from using, copying, altering or disclosing the content of this
>> message. Please delete it immediately and notify the sender. Newgen
>> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
>> damage arising from the use of the information transmitted by this email
>> including damages from virus and further acknowledges that no binding
>> nature of the message shall be implied or assumed unless the sender does so
>> expressly with due authority of NSTL.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cal.leeming at simplicitymedialtd.co.uk  Thu Dec 20 16:32:33 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Thu, 20 Dec 2012 13:32:33 +0000
Subject: [Freeswitch-users] Quick help needed
In-Reply-To: <CAFiqYumnWWJh2=Se=xC+Y5sM-beXd=P1QafSUNVyLp+MqUCFtw@mail.gmail.com>
References: <030101cdde8c$a44f2390$eced6ab0$@co.in>
	<CALNZuwkisdhY_FUA9AiTca2rYTGYPgFcUN_C96N-w3DTyQi20w@mail.gmail.com>
	<CAFiqYumnWWJh2=Se=xC+Y5sM-beXd=P1QafSUNVyLp+MqUCFtw@mail.gmail.com>
Message-ID: <CAHKQagGkrFq5FpwWW4fSY-DrF4bXx01E_hQq=OjpaRakEFFnJQ@mail.gmail.com>

Also fyi, during our own tests, Flite caused FreeSWITCH 1.2.3 to crash
under heavy load (60+ calls/sec).

Cal

On Thu, Dec 20, 2012 at 1:16 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Bear in mind a prerecorded file will consume less CPU than performing TTS
> if this is a prompt that'll be played a lot.
>
> If you don't want to pay a voice actor or do it yourself, you could always
> generate a prerecorded file using TTS.
>
>
> On 20 December 2012 08:42, Avi Marcus <avi at avimarcus.net> wrote:
>
>> SAY isn't for dynamic text, but rather to join together pre-recorded
>> files for saying numbers, dates, etc. see: SAY on the wiki<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say>
>>
>> You either want:
>> 1) playback<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_playback> of
>> a pre-recorded file.
>> 2) use TTS with the free flite e.g. with SPEAK<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_speak> (not
>> say) e.g.:
>>
>> <action application="speak" data="flite|kal|FreeSWITCH is awesome"/>
>>
>>
>> -Avi
>>
>>
>> On Thu, Dec 20, 2012 at 10:33 AM, Nitin Tomer <ntomer at newgen.co.in>wrote:
>>
>>>  Hi,****
>>>
>>> ** **
>>>
>>> In my application, I am parking a call to an extension using valet-park.
>>> Before this I want to read  a message to caller, but am not able to. I
>>> tried to use say and phrase but am not able to get the syntax right. Please
>>> help me.****
>>>
>>> ** **
>>>
>>> Dialplan ?****
>>>
>>> ** **
>>>
>>> <extension>****
>>>
>>>                         <condition field="destination_number"
>>> expression="^(450)$">
>>>                                     <action application="set"
>>> data="caller_id_name=Account Opening Request" />
>>>                                     <action application="set"
>>> data="call_timeout=60" />
>>>                                     <action application="set"
>>> data="originate_timeout=60" />
>>>                                     <action inline="true"
>>> application="lua" data="accountopening.lua"/>
>>>                                     <action application="db"
>>> data="insert/accountopening/${parkednumber}/nonumber" />
>>>                                     <action application="say" data="en
>>> messages pronounced 'Please wait for an agent to pick your call.'" />
>>>                                     <action application="valet_park"
>>> data="my_lot ${parkednumber}" />          ****
>>>
>>>               </condition>
>>> </extension>****
>>>
>>> ** **
>>>
>>> Regards****
>>>
>>> ** **
>>>
>>> Nitin****
>>>
>>> Disclaimer :- This e-mail and any attachment may contain confidential,
>>> proprietary or legally privileged information. If you are not the original
>>> intended recipient and have erroneously received this message, you are
>>> prohibited from using, copying, altering or disclosing the content of this
>>> message. Please delete it immediately and notify the sender. Newgen
>>> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
>>> damage arising from the use of the information transmitted by this email
>>> including damages from virus and further acknowledges that no binding
>>> nature of the message shall be implied or assumed unless the sender does so
>>> expressly with due authority of NSTL.
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 Alexander.Haugg at c4b.de  Thu Dec 20 16:55:53 2012
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Thu, 20 Dec 2012 13:55:53 +0000
Subject: [Freeswitch-users] build error with new Freeswitch 'nametab.h':
 No such file or directory
In-Reply-To: <1355935704180-7585726.post@n2.nabble.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC837D409@vc4bex2.corp.c4b.de>
	<1355844703594-7585673.post@n2.nabble.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC837D614@vc4bex2.corp.c4b.de>
	<1355935704180-7585726.post@n2.nabble.com>
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC837D838@vc4bex2.corp.c4b.de>

i tried it again with a new git clone in the Folder e:\GIT_FS\.

At the first run i had the same error, but if start the build (NOT rebuild) again it works successfully.

I hope that help. Is it possible to fix it that the build run successfully at the first run?

For information i build on windows (I'm afraid) with the Freeswitch.2008.sln.

thanks

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jeff Lenk
Gesendet: Mittwoch, 19. Dezember 2012 17:48
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] build error with new Freeswitch 'nametab.h': No such file or directory

Ok thanks, I think this might be cause because a code generator is failing because of a space in the path.

Can you try building from a directory without spaces a test to verify. Then I will look to fix.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/build-error-with-new-Freeswitch-nametab-h-No-such-file-or-directory-tp7585669p7585726.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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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  Thu Dec 20 17:50:01 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Dec 2012 14:50:01 +0000
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <68b001cdde38$f31f6bf0$d95e43d0$@bizfocused.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>
	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>
	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>
	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>
	<541b01cddcb1$72330200$56990600$@bizfocused.com>
	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>
	<5da601cddd67$dc76b6f0$956424d0$@bizfocused.com>
	<CAKbxfG_eXUY=eLnp-_xLks4-bXaoR3YiwqvsJ0teV15AF8w6vA@mail.gmail.com>
	<681801cdde31$d4fe1ec0$7efa5c40$@bizfocused.com>
	<CAKbxfG_4Xp=DwOnq6zFo=S-30BpN_BME+wSq8pzho4VusuQCpg@mail.gmail.com>
	<68b001cdde38$f31f6bf0$d95e43d0$@bizfocused.com>
Message-ID: <CAFiqYumb-6T767WKsmyzyoA+-84Ct+M2Fbx6RJD9DYsHCW=+Ag@mail.gmail.com>

The Cisco SPA504G doesn't like something in this request and gives the 406
Not Acceptable response (SERVICE_NOT_IMPLEMENTED is the Q.931 mapping of
SIP 406).
   INVITE sip:302 at 69.251.170.6:1085 SIP/2.0
   Via: SIP/2.0/UDP 204.62.15.231;rport;branch=z9hG4bKaFrHeKgQ6F2Ue
   Max-Forwards: 70
   From: "anonymous" <sip:4104850152 at 204.62.15.231>;tag=aK7263m4r24Hj
   To: <sip:302 at 69.251.170.6:1085>
   Call-ID: 8308fa5b-c3ed-1230-899f-03777f384a29
   CSeq: 37593727 INVITE
   Contact: <sip:mod_sofia at 204.62.15.231:5060>
   User-Agent:
FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: precondition, path, replaces
   Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 209
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "anonymous" <sip:4104850152 at 204.62.15.231
>;party=calling;screen=yes;privacy=full

   v=0
   o=FreeSWITCH 1355837829 1355837830 IN IP4 204.62.15.231
   s=FreeSWITCH
   c=IN IP4 204.62.15.231
   t=0 0
   m=audio 21754 RTP/AVP 9 0 8 3 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20

This site suggests that it could be blocking the call because the callerid
is "anonymous""
http://www.mysysadmintips.com/linux/servers/309-linksys-spa-voip-phone-rejects-calls-from-skype-sip-trunk-on-asterisk
That page shows a setting to disable that, can see you if that helps?
You could also try setting effective_caller_id_name to send something other
than anonymous.

-Steve


On 19 December 2012 22:33, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Sorry, server won?t serve .log files!****
>
> http://www.bizfocused.com/service_not_found.txt****
>
> ** **
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, December 19, 2012 4:50 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
> ** **
>
> No the url is 404 so I can't see the file.****
>
> ** **
>
> ** **
>
> On Wed, Dec 19, 2012 at 3:43 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> Whoa.   404 a.k.a device not found?****
>
>  ****
>
> This could be as I suspected, router capacity exceeded or general NAT
> issue with this router?****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, December 19, 2012 1:27 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> it gives 404 err****
>
>  ****
>
> On Tue, Dec 18, 2012 at 3:37 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> OK, I got one the logs!  I have trimmed the file down, but it is still
> over 5 MB and pastebin just chokes on it.****
>
>  ****
>
> It is on my server:  http://www.bizfocused.com/service_not_found.log****
>
>  ****
>
> Thanks for looking.****
>
>  ****
>
> If you really need pcap, I am afraid I need detailed instructions on
> installing/running it.****
>
>  ****
>
> Sean****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 17, 2012 7:00 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> usually the 406 is done in an established call to refuse a codec change
> during re-invite.****
>
> Its possible the other end thinks we want to change the codec when we do
> the session-timer re-invite but I'm sure we don't but the sip trace can
> help shed some light.  You can run a pcap too at the same time so when we
> find the bad call in the logs we can filter it out of the pcap too.  To
> avoid it getting too big you can just restart it every so often or use
> sippcapdump and delete calls that are not affected.****
>
>  ****
>
>  ****
>
>  ****
>
> On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> Waiting for another failure with siptrace REALLY on this time.****
>
>  ****
>
> If the user has clicked DND on these cisco phones, could that cause these
> messages?****
>
>  ****
>
> Sean****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 17, 2012 12:32 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> The siptrace is not in here?  Did you enable it on the console and then
> capture to the log file perhaps?****
>
> you should do "sofia tracelevel debug" too to route traces to the log file.
> ****
>
>  ****
>
> Its hard to say for sure with no sip trace but it seems like the far end
> is rejecting the session timer re-invite causing the call to end.****
>
>  ****
>
>  ****
>
> On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com>
> wrote:****
>
> http://pastebin.freeswitch.org/20342****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
> ** **
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Thu Dec 20 18:00:15 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 20 Dec 2012 07:00:15 -0800 (PST)
Subject: [Freeswitch-users] build error with new Freeswitch 'nametab.h':
 No such file or directory
In-Reply-To: <B60DA91F5EA0E24A96E73CDE43CC9EC837D838@vc4bex2.corp.c4b.de>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC837D409@vc4bex2.corp.c4b.de>
	<1355844703594-7585673.post@n2.nabble.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC837D614@vc4bex2.corp.c4b.de>
	<1355935704180-7585726.post@n2.nabble.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC837D838@vc4bex2.corp.c4b.de>
Message-ID: <1356015615778-7585759.post@n2.nabble.com>

Ok I made a correction to the solution file that should correct the
dependency problem. Also the project policy is to open Jiras
(http://jira.freeswitch.org)for possible problems. Thanks for bringing this
issue attention.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/build-error-with-new-Freeswitch-nametab-h-No-such-file-or-directory-tp7585669p7585759.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sdevoy at bizfocused.com  Thu Dec 20 18:39:41 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 20 Dec 2012 10:39:41 -0500
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <CAFiqYumb-6T767WKsmyzyoA+-84Ct+M2Fbx6RJD9DYsHCW=+Ag@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>	<541b01cddcb1$72330200$56990600$@bizfocused.com>	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>	<5da601cddd67$dc76b6f0$956424d0$@bizfocused.com>	<CAKbxfG_eXUY=eLnp-_xLks4-bXaoR3YiwqvsJ0teV15AF8w6vA@mail.gmail.com>	<681801cdde31$d4fe1ec0$7efa5c40$@bizfocused.com>	<CAKbxfG_4Xp=DwOnq6zFo=S-30BpN_BME+wSq8pzho4VusuQCpg@mail.gmail.com>	<68b001cdde38$f31f6bf0$d95e43d0$@bizfocused.com>
	<CAFiqYumb-6T767WKsmyzyoA+-84Ct+M2Fbx6RJD9DYsHCW=+Ag@mail.gmail.com>
Message-ID: <6ded01cddec8$3a7e17a0$af7a46e0$@bizfocused.com>

Thank you very much for your response Steven.  I have checked the Block
Anonymous Call and it is set to NO.

 

But, at least I know it is the 504G's issue and I can dig deeper there.  I
will certainly start with the firmware versions, etc.

 

Can you take a guess at something for me?

                Why does it only happen occasionally (even from this same
caller)?

                Is this the same SIP response to a phone on Do Not Disturb?

 

Thanks again.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: Thursday, December 20, 2012 9:50 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

 

The Cisco SPA504G doesn't like something in this request and gives the 406
Not Acceptable response (SERVICE_NOT_IMPLEMENTED is the Q.931 mapping of SIP
406).

   INVITE sip:302 at 69.251.170.6:1085 SIP/2.0

   Via: SIP/2.0/UDP 204.62.15.231;rport;branch=z9hG4bKaFrHeKgQ6F2Ue

   Max-Forwards: 70

   From: "anonymous" <sip:4104850152 at 204.62.15.231
<mailto:sip%3A4104850152 at 204.62.15.231> >;tag=aK7263m4r24Hj

   To: <sip:302 at 69.251.170.6:1085>

   Call-ID: 8308fa5b-c3ed-1230-899f-03777f384a29

   CSeq: 37593727 INVITE

   Contact: <sip:mod_sofia at 204.62.15.231:5060>

   User-Agent:
FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120
712T101002Z

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: precondition, path, replaces

   Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer

   Content-Type: application/sdp

   Content-Disposition: session

   Content-Length: 209

   X-FS-Support: update_display,send_info

   Remote-Party-ID: "anonymous" <sip:4104850152 at 204.62.15.231
<mailto:sip%3A4104850152 at 204.62.15.231>
>;party=calling;screen=yes;privacy=full

   

   v=0

   o=FreeSWITCH 1355837829 1355837830 IN IP4 204.62.15.231

   s=FreeSWITCH

   c=IN IP4 204.62.15.231

   t=0 0

   m=audio 21754 RTP/AVP 9 0 8 3 101 13

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-16

   a=ptime:20

 

This site suggests that it could be blocking the call because the callerid
is "anonymous""

http://www.mysysadmintips.com/linux/servers/309-linksys-spa-voip-phone-rejec
ts-calls-from-skype-sip-trunk-on-asterisk

That page shows a setting to disable that, can see you if that helps?

You could also try setting effective_caller_id_name to send something other
than anonymous.

 

-Steve

 

 

On 19 December 2012 22:33, Sean Devoy <sdevoy at bizfocused.com> wrote:

Sorry, server won't serve .log files!

http://www.bizfocused.com/service_not_found.txt

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, December 19, 2012 4:50 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

No the url is 404 so I can't see the file.

 

 

On Wed, Dec 19, 2012 at 3:43 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Whoa.   404 a.k.a device not found?

 

This could be as I suspected, router capacity exceeded or general NAT issue
with this router?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, December 19, 2012 1:27 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

it gives 404 err

 

On Tue, Dec 18, 2012 at 3:37 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

OK, I got one the logs!  I have trimmed the file down, but it is still over
5 MB and pastebin just chokes on it.

 

It is on my server:  http://www.bizfocused.com/service_not_found.log

 

Thanks for looking.

 

If you really need pcap, I am afraid I need detailed instructions on
installing/running it.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 7:00 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

usually the 406 is done in an established call to refuse a codec change
during re-invite.

Its possible the other end thinks we want to change the codec when we do the
session-timer re-invite but I'm sure we don't but the sip trace can help
shed some light.  You can run a pcap too at the same time so when we find
the bad call in the logs we can filter it out of the pcap too.  To avoid it
getting too big you can just restart it every so often or use sippcapdump
and delete calls that are not affected.

 

 

 

On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Waiting for another failure with siptrace REALLY on this time.

 

If the user has clicked DND on these cisco phones, could that cause these
messages?

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 12:32 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

The siptrace is not in here?  Did you enable it on the console and then
capture to the log file perhaps?

you should do "sofia tracelevel debug" too to route traces to the log file.

 

Its hard to say for sure with no sip trace but it seems like the far end is
rejecting the session timer re-invite causing the call to end.

 

 

On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

http://pastebin.freeswitch.org/20342





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900> 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900> 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900>  


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900>  


_________________________________________________________________________
Professional FreeSWITCH Consulting 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 andrew at cassidywebservices.co.uk  Thu Dec 20 19:27:08 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 20 Dec 2012 16:27:08 +0000
Subject: [Freeswitch-users] Quick help needed
In-Reply-To: <CAHKQagGkrFq5FpwWW4fSY-DrF4bXx01E_hQq=OjpaRakEFFnJQ@mail.gmail.com>
References: <030101cdde8c$a44f2390$eced6ab0$@co.in>
	<CALNZuwkisdhY_FUA9AiTca2rYTGYPgFcUN_C96N-w3DTyQi20w@mail.gmail.com>
	<CAFiqYumnWWJh2=Se=xC+Y5sM-beXd=P1QafSUNVyLp+MqUCFtw@mail.gmail.com>
	<CAHKQagGkrFq5FpwWW4fSY-DrF4bXx01E_hQq=OjpaRakEFFnJQ@mail.gmail.com>
Message-ID: <CAMbW9oK=Ha2VfhtyMCHoV1xZDY9ONR-To_fyhLMnwC3xnO-u6w@mail.gmail.com>

+1 for prerecorded files, I've not found 1 TTS I've liked.

On 20 December 2012 13:32, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Also fyi, during our own tests, Flite caused FreeSWITCH 1.2.3 to crash
> under heavy load (60+ calls/sec).
>
> Cal
>
>
> On Thu, Dec 20, 2012 at 1:16 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Bear in mind a prerecorded file will consume less CPU than performing TTS
>> if this is a prompt that'll be played a lot.
>>
>> If you don't want to pay a voice actor or do it yourself, you could
>> always generate a prerecorded file using TTS.
>>
>>
>> On 20 December 2012 08:42, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> SAY isn't for dynamic text, but rather to join together pre-recorded
>>> files for saying numbers, dates, etc. see: SAY on the wiki<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say>
>>>
>>> You either want:
>>> 1) playback<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_playback> of
>>> a pre-recorded file.
>>> 2) use TTS with the free flite e.g. with SPEAK<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_speak> (not
>>> say) e.g.:
>>>
>>> <action application="speak" data="flite|kal|FreeSWITCH is awesome"/>
>>>
>>>
>>> -Avi
>>>
>>>
>>> On Thu, Dec 20, 2012 at 10:33 AM, Nitin Tomer <ntomer at newgen.co.in>wrote:
>>>
>>>>  Hi,****
>>>>
>>>> ** **
>>>>
>>>> In my application, I am parking a call to an extension using
>>>> valet-park. Before this I want to read  a message to caller, but am not
>>>> able to. I tried to use say and phrase but am not able to get the syntax
>>>> right. Please help me.****
>>>>
>>>> ** **
>>>>
>>>> Dialplan ?****
>>>>
>>>> ** **
>>>>
>>>> <extension>****
>>>>
>>>>                         <condition field="destination_number"
>>>> expression="^(450)$">
>>>>                                     <action application="set"
>>>> data="caller_id_name=Account Opening Request" />
>>>>                                     <action application="set"
>>>> data="call_timeout=60" />
>>>>                                     <action application="set"
>>>> data="originate_timeout=60" />
>>>>                                     <action inline="true"
>>>> application="lua" data="accountopening.lua"/>
>>>>                                     <action application="db"
>>>> data="insert/accountopening/${parkednumber}/nonumber" />
>>>>                                     <action application="say" data="en
>>>> messages pronounced 'Please wait for an agent to pick your call.'" />
>>>>                                     <action application="valet_park"
>>>> data="my_lot ${parkednumber}" />          ****
>>>>
>>>>               </condition>
>>>> </extension>****
>>>>
>>>> ** **
>>>>
>>>> Regards****
>>>>
>>>> ** **
>>>>
>>>> Nitin****
>>>>
>>>> Disclaimer :- This e-mail and any attachment may contain confidential,
>>>> proprietary or legally privileged information. If you are not the original
>>>> intended recipient and have erroneously received this message, you are
>>>> prohibited from using, copying, altering or disclosing the content of this
>>>> message. Please delete it immediately and notify the sender. Newgen
>>>> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
>>>> damage arising from the use of the information transmitted by this email
>>>> including damages from virus and further acknowledges that no binding
>>>> nature of the message shall be implied or assumed unless the sender does so
>>>> expressly with due authority of NSTL.
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From steveayre at gmail.com  Thu Dec 20 19:36:59 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Dec 2012 16:36:59 +0000
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <6ded01cddec8$3a7e17a0$af7a46e0$@bizfocused.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>
	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>
	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>
	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>
	<541b01cddcb1$72330200$56990600$@bizfocused.com>
	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>
	<5da601cddd67$dc76b6f0$956424d0$@bizfocused.com>
	<CAKbxfG_eXUY=eLnp-_xLks4-bXaoR3YiwqvsJ0teV15AF8w6vA@mail.gmail.com>
	<681801cdde31$d4fe1ec0$7efa5c40$@bizfocused.com>
	<CAKbxfG_4Xp=DwOnq6zFo=S-30BpN_BME+wSq8pzho4VusuQCpg@mail.gmail.com>
	<68b001cdde38$f31f6bf0$d95e43d0$@bizfocused.com>
	<CAFiqYumb-6T767WKsmyzyoA+-84Ct+M2Fbx6RJD9DYsHCW=+Ag@mail.gmail.com>
	<6ded01cddec8$3a7e17a0$af7a46e0$@bizfocused.com>
Message-ID: <CAFiqYu=RTcO60HmQ1YTyDcSRHcRdioBXG2ydTXoDpXtt8Jb3Gw@mail.gmail.com>

>
> Why does it only happen occasionally (even from this same caller)?

That makes me wonder if there's a bug that rarely treats Block Anonymous
Call as Yes. Cisco has been known for buggy SIP implementations in the
past. See if there's a firmware upgrade available.

Otherwise it's also possible it's disliking something set in the
Allow/Supported/Allow-Events headers. I would take a siptrace of a working
call to the same user and compare the INVITE packets to see if there's a
difference.

You don't need the sofia stack debugging on by the way, which'll make your
logs much smaller.

-Steve



On 20 December 2012 15:39, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Thank you very much for your response Steven.  I have checked the Block
> Anonymous Call and it is set to NO.****
>
> ** **
>
> But, at least I know it is the 504G?s issue and I can dig deeper there.  I
> will certainly start with the firmware versions, etc.****
>
> ** **
>
> Can you take a guess at something for me?****
>
>                 Why does it only happen occasionally (even from this same
> caller)?****
>
>                 Is this the same SIP response to a phone on Do Not Disturb?
> ****
>
> ** **
>
> Thanks again.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* Thursday, December 20, 2012 9:50 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
> ** **
>
> ** **
>
> The Cisco SPA504G doesn't like something in this request and gives the 406
> Not Acceptable response (SERVICE_NOT_IMPLEMENTED is the Q.931 mapping of
> SIP 406).****
>
>    INVITE sip:302 at 69.251.170.6:1085 SIP/2.0****
>
>    Via: SIP/2.0/UDP 204.62.15.231;rport;branch=z9hG4bKaFrHeKgQ6F2Ue****
>
>    Max-Forwards: 70****
>
>    From: "anonymous" <sip:4104850152 at 204.62.15.231>;tag=aK7263m4r24Hj****
>
>    To: <sip:302 at 69.251.170.6:1085>****
>
>    Call-ID: 8308fa5b-c3ed-1230-899f-03777f384a29****
>
>    CSeq: 37593727 INVITE****
>
>    Contact: <sip:mod_sofia at 204.62.15.231:5060>****
>
>    User-Agent:
> FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
> ****
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE****
>
>    Supported: precondition, path, replaces****
>
>    Allow-Events: talk, hold, conference, presence, dialog, line-seize,
> call-info, sla, include-session-description, presence.winfo,
> message-summary, refer****
>
>    Content-Type: application/sdp****
>
>    Content-Disposition: session****
>
>    Content-Length: 209****
>
>    X-FS-Support: update_display,send_info****
>
>    Remote-Party-ID: "anonymous" <sip:4104850152 at 204.62.15.231
> >;party=calling;screen=yes;privacy=full****
>
>    ****
>
>    v=0****
>
>    o=FreeSWITCH 1355837829 1355837830 IN IP4 204.62.15.231****
>
>    s=FreeSWITCH****
>
>    c=IN IP4 204.62.15.231****
>
>    t=0 0****
>
>    m=audio 21754 RTP/AVP 9 0 8 3 101 13****
>
>    a=rtpmap:101 telephone-event/8000****
>
>    a=fmtp:101 0-16****
>
>    a=ptime:20****
>
> ** **
>
> This site suggests that it could be blocking the call because the callerid
> is "anonymous""****
>
>
> http://www.mysysadmintips.com/linux/servers/309-linksys-spa-voip-phone-rejects-calls-from-skype-sip-trunk-on-asterisk
> ****
>
> That page shows a setting to disable that, can see you if that helps?****
>
> You could also try setting effective_caller_id_name to send something
> other than anonymous.****
>
> ** **
>
> -Steve****
>
> ** **
>
> ** **
>
> On 19 December 2012 22:33, Sean Devoy <sdevoy at bizfocused.com> wrote:****
>
> Sorry, server won?t serve .log files!****
>
> http://www.bizfocused.com/service_not_found.txt****
>
>  ****
>
>  ****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, December 19, 2012 4:50 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> No the url is 404 so I can't see the file.****
>
>  ****
>
>  ****
>
> On Wed, Dec 19, 2012 at 3:43 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> Whoa.   404 a.k.a device not found?****
>
>  ****
>
> This could be as I suspected, router capacity exceeded or general NAT
> issue with this router?****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, December 19, 2012 1:27 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> it gives 404 err****
>
>  ****
>
> On Tue, Dec 18, 2012 at 3:37 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> OK, I got one the logs!  I have trimmed the file down, but it is still
> over 5 MB and pastebin just chokes on it.****
>
>  ****
>
> It is on my server:  http://www.bizfocused.com/service_not_found.log****
>
>  ****
>
> Thanks for looking.****
>
>  ****
>
> If you really need pcap, I am afraid I need detailed instructions on
> installing/running it.****
>
>  ****
>
> Sean****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 17, 2012 7:00 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> usually the 406 is done in an established call to refuse a codec change
> during re-invite.****
>
> Its possible the other end thinks we want to change the codec when we do
> the session-timer re-invite but I'm sure we don't but the sip trace can
> help shed some light.  You can run a pcap too at the same time so when we
> find the bad call in the logs we can filter it out of the pcap too.  To
> avoid it getting too big you can just restart it every so often or use
> sippcapdump and delete calls that are not affected.****
>
>  ****
>
>  ****
>
>  ****
>
> On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> Waiting for another failure with siptrace REALLY on this time.****
>
>  ****
>
> If the user has clicked DND on these cisco phones, could that cause these
> messages?****
>
>  ****
>
> Sean****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 17, 2012 12:32 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> The siptrace is not in here?  Did you enable it on the console and then
> capture to the log file perhaps?****
>
> you should do "sofia tracelevel debug" too to route traces to the log file.
> ****
>
>  ****
>
> Its hard to say for sure with no sip trace but it seems like the far end
> is rejecting the session timer re-invite causing the call to end.****
>
>  ****
>
>  ****
>
> On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com>
> wrote:****
>
> http://pastebin.freeswitch.org/20342****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 msc at freeswitch.org  Thu Dec 20 19:42:16 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Dec 2012 08:42:16 -0800
Subject: [Freeswitch-users] Newbie question: Executing Lua scripts for
 incoming calls, how concurrency is to be handled?
In-Reply-To: <965759A53E43FE439E43565A7715E5F058F4156DF7@oa-exchange1.oa.com.au>
References: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au>
	<50CE5D6E.2020401@quentustech.com>
	<965759A53E43FE439E43565A7715E5F058F4156D32@oa-exchange1.oa.com.au>
	<CAKzWOxWFh3RNYg9NY4Dqa-=f9KZOaXTvM8NGzHpvEvzV16VUWA@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156DD8@oa-exchange1.oa.com.au>
	<CAKzWOxUYyfZ_8dLgy_t-Ov2a+YXEDj04dDkdLSn14Q-tZk-8pQ@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156DF7@oa-exchange1.oa.com.au>
Message-ID: <CAKzWOxUrdVNqqDJxQ8x4_OaE=oEOrGzdEb2D+XmvC-0tW_yfjA@mail.gmail.com>

Sirish,

Since you're using an E1 to connect to the PBX then really all you need to
do is keep track of the last time each port was hung up and whether or not
a given port is currently in use. I would use api_on_answer to launch a
simple script to set a flag to say that a particular port is in use and
then use the api_hangup_hook to launch another script when the call ends.

The channel variables page on the wiki has some examples of how to use
these. I recommend that you write simple Lua scripts that use the "hash"
API to store information in the local database. Also, check out the
"hash_dump" API as it is a useful way to quickly see what all is stored
there.

For an example of how to add, remove, and read information from the local
database using the "hash" API please see conf/dialplan/default.xml. Search
for "hash" and you'll see all sorts of examples of how the example dialplan
uses the local database to store useful information that allows us to
implement features like call return, call intercept, etc.

-MC

On Tue, Dec 18, 2012 at 7:39 PM, Sirish Masur Mohan <
Sirish.MasurMohan at oa.com.au> wrote:

> Hi Michael,****
>
> ** **
>
> >> How are you physically connecting from FreeSWITCH to the PBX?****
>
> ** **
>
> I connect this via E1 link ? I have a Sangoma card installed on the
> FreeSWITCH machine.****
>
> ** **
>
> With regards,****
>
> Sirish****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Wednesday, 19 December 2012 1:31 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Newbie question: Executing Lua scripts
> for incoming calls, how concurrency is to be handled?****
>
> ** **
>
> To trigger SIP registrations you'd need the PBX to have a SIP client. I'm
> assuming this is possible, but maybe that's a false assumption. How are you
> physically connecting from FreeSWITCH to the PBX?
>
> -MC****
>
> On Tue, Dec 18, 2012 at 2:36 PM, Sirish Masur Mohan <
> Sirish.MasurMohan at oa.com.au> wrote:****
>
> Hey Guys,****
>
>  ****
>
> Would really appreciate if you could help me out here ? isn?t there a way
> to handle concurrent calls in the dial plan, especially when Lua scripts,
> accessing global variables, are executed on receiving calls? ****
>
>  ****
>
> Is mod_fifo the closest I could get to handle concurrency (as Michael has
> explained)? If yes, how do I trigger SIP registrations, especially working
> with a PBX which I don?t have full control of?****
>
>  ****
>
> With regards,****
>
> Sirish****
>
>  ****
>
> *From:* Sirish Masur Mohan
> *Sent:* Tuesday, 18 December 2012 1:02 PM
> *To:* FreeSWITCH Users Help
> *Subject:* RE: [Freeswitch-users] Newbie question: Executing Lua scripts
> for incoming calls, how concurrency is to be handled?****
>
>  ****
>
> Hi Michael,****
>
>  ****
>
> Thanks for the reply. ****
>
>  ****
>
> >> You would need a SIP registration from the PBX to FreeSWITCH for each
> of the modem extensions****
>
> I am seen SIP clients such as X-Lite sending out the SIP registrations,
> but could you please clarify as to how this can be achieved in the PBX? The
> final production environment would be out in the customer?s PBX, which I
> may not have complete control of.. ****
>
>  ****
>
> >> What application are you building?****
>
> I may not be able to provide the details because of the NDA with customer,
> but what I am trying to achieve is, to replace an existing IVR with
> FreeSWITCH in an old existing setup of the customer ? that?s the reason why
> we continue working with dialup modems!****
>
>  ****
>
> With regards,****
>
> Sirish****
>
>  ****
>
> *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:* Tuesday, 18 December 2012 6:23 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Newbie question: Executing Lua scripts
> for incoming calls, how concurrency is to be handled?****
>
>  ****
>
> You don't have to have actual human agents for mod_fifo. You could define
> a user for each modem and then manually "log in" those "agents" on the
> command line using the fifo_member API command. Something like this:
>
> fifo_member add fifo_name {fifo_member_wait=nowait}user/1234
>
> Where 1234 is the user id of one of the modems. You would need a SIP
> registration from the PBX to FreeSWITCH for each of the modem extensions.
>
> Having modems go through a VoIP system sounds a bit scary. What
> application are you building?
> -MC****
>
> On Sun, Dec 16, 2012 at 5:41 PM, Sirish Masur Mohan <
> Sirish.MasurMohan at oa.com.au> wrote:****
>
> Hi William,
>
> Thanks for the reply.
>
> My setup is as follows:
> Client(Caller)->dialup modem->PBX->FreeSWITCH->PBX-> 4 dialup
> modems->Server(Receiver)
>
> I basically need FreeSWITCH to bridge the incoming call to the best
> external destination (out of the 4 available), so that the modem training,
> connection etc can take place smoothly, before exchange of data. I am not
> sure if mod_fifo would help me in this scenario, as, I would require an
> agent to dial in and read the fifo. Could you please clarify?
>
> Thanks!
>
> With regards,
> Sirish****
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William King
> Sent: Monday, 17 December 2012 10:47 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for
> incoming calls, how concurrency is to be handled?
>
> Sounds like you want to take a look into mod_fifo.
>
> 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****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
> ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From steveayre at gmail.com  Thu Dec 20 19:45:50 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Dec 2012 16:45:50 +0000
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <6ded01cddec8$3a7e17a0$af7a46e0$@bizfocused.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>
	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>
	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>
	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>
	<541b01cddcb1$72330200$56990600$@bizfocused.com>
	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>
	<5da601cddd67$dc76b6f0$956424d0$@bizfocused.com>
	<CAKbxfG_eXUY=eLnp-_xLks4-bXaoR3YiwqvsJ0teV15AF8w6vA@mail.gmail.com>
	<681801cdde31$d4fe1ec0$7efa5c40$@bizfocused.com>
	<CAKbxfG_4Xp=DwOnq6zFo=S-30BpN_BME+wSq8pzho4VusuQCpg@mail.gmail.com>
	<68b001cdde38$f31f6bf0$d95e43d0$@bizfocused.com>
	<CAFiqYumb-6T767WKsmyzyoA+-84Ct+M2Fbx6RJD9DYsHCW=+Ag@mail.gmail.com>
	<6ded01cddec8$3a7e17a0$af7a46e0$@bizfocused.com>
Message-ID: <CAFiqYunqSpqY9iz8Tyq0f4cFEvvh1jWNkpmDqYXoQhwqtk9H8w@mail.gmail.com>

>
> Can you take a guess at something for me?

   Is this the same SIP response to a phone on Do Not Disturb?


It is a guess because I don't have or have used a SPA, but I think the
phone still rings, just on silent (line flashing or something similar).

I'd probably expect 480 Temporarily Unavailable eventually for no answer,
but not immediately.

-Steve



On 20 December 2012 15:39, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Thank you very much for your response Steven.  I have checked the Block
> Anonymous Call and it is set to NO.****
>
> ** **
>
> But, at least I know it is the 504G?s issue and I can dig deeper there.  I
> will certainly start with the firmware versions, etc.****
>
> ** **
>
> Can you take a guess at something for me?****
>
>                 Why does it only happen occasionally (even from this same
> caller)?****
>
>                 Is this the same SIP response to a phone on Do Not Disturb?
> ****
>
> ** **
>
> Thanks again.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* Thursday, December 20, 2012 9:50 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
> ** **
>
> ** **
>
> The Cisco SPA504G doesn't like something in this request and gives the 406
> Not Acceptable response (SERVICE_NOT_IMPLEMENTED is the Q.931 mapping of
> SIP 406).****
>
>    INVITE sip:302 at 69.251.170.6:1085 SIP/2.0****
>
>    Via: SIP/2.0/UDP 204.62.15.231;rport;branch=z9hG4bKaFrHeKgQ6F2Ue****
>
>    Max-Forwards: 70****
>
>    From: "anonymous" <sip:4104850152 at 204.62.15.231>;tag=aK7263m4r24Hj****
>
>    To: <sip:302 at 69.251.170.6:1085>****
>
>    Call-ID: 8308fa5b-c3ed-1230-899f-03777f384a29****
>
>    CSeq: 37593727 INVITE****
>
>    Contact: <sip:mod_sofia at 204.62.15.231:5060>****
>
>    User-Agent:
> FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
> ****
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE****
>
>    Supported: precondition, path, replaces****
>
>    Allow-Events: talk, hold, conference, presence, dialog, line-seize,
> call-info, sla, include-session-description, presence.winfo,
> message-summary, refer****
>
>    Content-Type: application/sdp****
>
>    Content-Disposition: session****
>
>    Content-Length: 209****
>
>    X-FS-Support: update_display,send_info****
>
>    Remote-Party-ID: "anonymous" <sip:4104850152 at 204.62.15.231
> >;party=calling;screen=yes;privacy=full****
>
>    ****
>
>    v=0****
>
>    o=FreeSWITCH 1355837829 1355837830 IN IP4 204.62.15.231****
>
>    s=FreeSWITCH****
>
>    c=IN IP4 204.62.15.231****
>
>    t=0 0****
>
>    m=audio 21754 RTP/AVP 9 0 8 3 101 13****
>
>    a=rtpmap:101 telephone-event/8000****
>
>    a=fmtp:101 0-16****
>
>    a=ptime:20****
>
> ** **
>
> This site suggests that it could be blocking the call because the callerid
> is "anonymous""****
>
>
> http://www.mysysadmintips.com/linux/servers/309-linksys-spa-voip-phone-rejects-calls-from-skype-sip-trunk-on-asterisk
> ****
>
> That page shows a setting to disable that, can see you if that helps?****
>
> You could also try setting effective_caller_id_name to send something
> other than anonymous.****
>
> ** **
>
> -Steve****
>
> ** **
>
> ** **
>
> On 19 December 2012 22:33, Sean Devoy <sdevoy at bizfocused.com> wrote:****
>
> Sorry, server won?t serve .log files!****
>
> http://www.bizfocused.com/service_not_found.txt****
>
>  ****
>
>  ****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, December 19, 2012 4:50 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> No the url is 404 so I can't see the file.****
>
>  ****
>
>  ****
>
> On Wed, Dec 19, 2012 at 3:43 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> Whoa.   404 a.k.a device not found?****
>
>  ****
>
> This could be as I suspected, router capacity exceeded or general NAT
> issue with this router?****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, December 19, 2012 1:27 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> it gives 404 err****
>
>  ****
>
> On Tue, Dec 18, 2012 at 3:37 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> OK, I got one the logs!  I have trimmed the file down, but it is still
> over 5 MB and pastebin just chokes on it.****
>
>  ****
>
> It is on my server:  http://www.bizfocused.com/service_not_found.log****
>
>  ****
>
> Thanks for looking.****
>
>  ****
>
> If you really need pcap, I am afraid I need detailed instructions on
> installing/running it.****
>
>  ****
>
> Sean****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 17, 2012 7:00 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> usually the 406 is done in an established call to refuse a codec change
> during re-invite.****
>
> Its possible the other end thinks we want to change the codec when we do
> the session-timer re-invite but I'm sure we don't but the sip trace can
> help shed some light.  You can run a pcap too at the same time so when we
> find the bad call in the logs we can filter it out of the pcap too.  To
> avoid it getting too big you can just restart it every so often or use
> sippcapdump and delete calls that are not affected.****
>
>  ****
>
>  ****
>
>  ****
>
> On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> Waiting for another failure with siptrace REALLY on this time.****
>
>  ****
>
> If the user has clicked DND on these cisco phones, could that cause these
> messages?****
>
>  ****
>
> Sean****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 17, 2012 12:32 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> The siptrace is not in here?  Did you enable it on the console and then
> capture to the log file perhaps?****
>
> you should do "sofia tracelevel debug" too to route traces to the log file.
> ****
>
>  ****
>
> Its hard to say for sure with no sip trace but it seems like the far end
> is rejecting the session timer re-invite causing the call to end.****
>
>  ****
>
>  ****
>
> On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com>
> wrote:****
>
> http://pastebin.freeswitch.org/20342****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 msc at freeswitch.org  Thu Dec 20 19:50:06 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Dec 2012 08:50:06 -0800
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <CAFiqYu=RTcO60HmQ1YTyDcSRHcRdioBXG2ydTXoDpXtt8Jb3Gw@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>
	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>
	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>
	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>
	<541b01cddcb1$72330200$56990600$@bizfocused.com>
	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>
	<5da601cddd67$dc76b6f0$956424d0$@bizfocused.com>
	<CAKbxfG_eXUY=eLnp-_xLks4-bXaoR3YiwqvsJ0teV15AF8w6vA@mail.gmail.com>
	<681801cdde31$d4fe1ec0$7efa5c40$@bizfocused.com>
	<CAKbxfG_4Xp=DwOnq6zFo=S-30BpN_BME+wSq8pzho4VusuQCpg@mail.gmail.com>
	<68b001cdde38$f31f6bf0$d95e43d0$@bizfocused.com>
	<CAFiqYumb-6T767WKsmyzyoA+-84Ct+M2Fbx6RJD9DYsHCW=+Ag@mail.gmail.com>
	<6ded01cddec8$3a7e17a0$af7a46e0$@bizfocused.com>
	<CAFiqYu=RTcO60HmQ1YTyDcSRHcRdioBXG2ydTXoDpXtt8Jb3Gw@mail.gmail.com>
Message-ID: <CAKzWOxUBq8k+fzURNFBc6gx+348upE1H-kY=rR3M_kGqj6XpHg@mail.gmail.com>

On Thu, Dec 20, 2012 at 8:36 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Why does it only happen occasionally (even from this same caller)?
>
> That makes me wonder if there's a bug that rarely treats Block Anonymous
> Call as Yes. Cisco has been known for buggy SIP implementations in the
> past. See if there's a firmware upgrade available.
>
> Otherwise it's also possible it's disliking something set in the
> Allow/Supported/Allow-Events headers. I would take a siptrace of a working
> call to the same user and compare the INVITE packets to see if there's a
> difference.
>
> You don't need the sofia stack debugging on by the way, which'll make your
> logs much smaller.
>
Pro tip:
For SIP trace just use something like:
sofia global siptrace on
-OR-
sofia profile internal siptrace on

Don't do "sofia loglevel xxx" if you're just wanting a simple SIP trace.
-MC

>
> -Steve
>
>
>
> On 20 December 2012 15:39, Sean Devoy <sdevoy at bizfocused.com> wrote:
>
>> Thank you very much for your response Steven.  I have checked the Block
>> Anonymous Call and it is set to NO.****
>>
>> ** **
>>
>> But, at least I know it is the 504G?s issue and I can dig deeper there.
>> I will certainly start with the firmware versions, etc.****
>>
>> ** **
>>
>> Can you take a guess at something for me?****
>>
>>                 Why does it only happen occasionally (even from this same
>> caller)?****
>>
>>                 Is this the same SIP response to a phone on Do Not
>> Disturb?****
>>
>> ** **
>>
>> Thanks again.****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
>> *Sent:* Thursday, December 20, 2012 9:50 AM
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
>> [SERVICE_NOT_IMPLEMENTED]****
>>
>> ** **
>>
>> ** **
>>
>> The Cisco SPA504G doesn't like something in this request and gives the
>> 406 Not Acceptable response (SERVICE_NOT_IMPLEMENTED is the Q.931 mapping
>> of SIP 406).****
>>
>>    INVITE sip:302 at 69.251.170.6:1085 SIP/2.0****
>>
>>    Via: SIP/2.0/UDP 204.62.15.231;rport;branch=z9hG4bKaFrHeKgQ6F2Ue****
>>
>>    Max-Forwards: 70****
>>
>>    From: "anonymous" <sip:4104850152 at 204.62.15.231>;tag=aK7263m4r24Hj****
>>
>>    To: <sip:302 at 69.251.170.6:1085>****
>>
>>    Call-ID: 8308fa5b-c3ed-1230-899f-03777f384a29****
>>
>>    CSeq: 37593727 INVITE****
>>
>>    Contact: <sip:mod_sofia at 204.62.15.231:5060>****
>>
>>    User-Agent:
>> FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
>> ****
>>
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE****
>>
>>    Supported: precondition, path, replaces****
>>
>>    Allow-Events: talk, hold, conference, presence, dialog, line-seize,
>> call-info, sla, include-session-description, presence.winfo,
>> message-summary, refer****
>>
>>    Content-Type: application/sdp****
>>
>>    Content-Disposition: session****
>>
>>    Content-Length: 209****
>>
>>    X-FS-Support: update_display,send_info****
>>
>>    Remote-Party-ID: "anonymous" <sip:4104850152 at 204.62.15.231
>> >;party=calling;screen=yes;privacy=full****
>>
>>    ****
>>
>>    v=0****
>>
>>    o=FreeSWITCH 1355837829 1355837830 IN IP4 204.62.15.231****
>>
>>    s=FreeSWITCH****
>>
>>    c=IN IP4 204.62.15.231****
>>
>>    t=0 0****
>>
>>    m=audio 21754 RTP/AVP 9 0 8 3 101 13****
>>
>>    a=rtpmap:101 telephone-event/8000****
>>
>>    a=fmtp:101 0-16****
>>
>>    a=ptime:20****
>>
>> ** **
>>
>> This site suggests that it could be blocking the call because the
>> callerid is "anonymous""****
>>
>>
>> http://www.mysysadmintips.com/linux/servers/309-linksys-spa-voip-phone-rejects-calls-from-skype-sip-trunk-on-asterisk
>> ****
>>
>> That page shows a setting to disable that, can see you if that helps?****
>>
>> You could also try setting effective_caller_id_name to send something
>> other than anonymous.****
>>
>> ** **
>>
>> -Steve****
>>
>> ** **
>>
>> ** **
>>
>> On 19 December 2012 22:33, Sean Devoy <sdevoy at bizfocused.com> wrote:****
>>
>> Sorry, server won?t serve .log files!****
>>
>> http://www.bizfocused.com/service_not_found.txt****
>>
>>  ****
>>
>>  ****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
>> Minessale
>> *Sent:* Wednesday, December 19, 2012 4:50 PM****
>>
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
>> [SERVICE_NOT_IMPLEMENTED]****
>>
>>  ****
>>
>> No the url is 404 so I can't see the file.****
>>
>>  ****
>>
>>  ****
>>
>> On Wed, Dec 19, 2012 at 3:43 PM, Sean Devoy <sdevoy at bizfocused.com>
>> wrote:****
>>
>> Whoa.   404 a.k.a device not found?****
>>
>>  ****
>>
>> This could be as I suspected, router capacity exceeded or general NAT
>> issue with this router?****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
>> Minessale
>> *Sent:* Wednesday, December 19, 2012 1:27 PM****
>>
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
>> [SERVICE_NOT_IMPLEMENTED]****
>>
>>  ****
>>
>> it gives 404 err****
>>
>>  ****
>>
>> On Tue, Dec 18, 2012 at 3:37 PM, Sean Devoy <sdevoy at bizfocused.com>
>> wrote:****
>>
>> OK, I got one the logs!  I have trimmed the file down, but it is still
>> over 5 MB and pastebin just chokes on it.****
>>
>>  ****
>>
>> It is on my server:  http://www.bizfocused.com/service_not_found.log****
>>
>>  ****
>>
>> Thanks for looking.****
>>
>>  ****
>>
>> If you really need pcap, I am afraid I need detailed instructions on
>> installing/running it.****
>>
>>  ****
>>
>> Sean****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
>> Minessale
>> *Sent:* Monday, December 17, 2012 7:00 PM****
>>
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
>> [SERVICE_NOT_IMPLEMENTED]****
>>
>>  ****
>>
>> usually the 406 is done in an established call to refuse a codec change
>> during re-invite.****
>>
>> Its possible the other end thinks we want to change the codec when we do
>> the session-timer re-invite but I'm sure we don't but the sip trace can
>> help shed some light.  You can run a pcap too at the same time so when we
>> find the bad call in the logs we can filter it out of the pcap too.  To
>> avoid it getting too big you can just restart it every so often or use
>> sippcapdump and delete calls that are not affected.****
>>
>>  ****
>>
>>  ****
>>
>>  ****
>>
>> On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com>
>> wrote:****
>>
>> Waiting for another failure with siptrace REALLY on this time.****
>>
>>  ****
>>
>> If the user has clicked DND on these cisco phones, could that cause these
>> messages?****
>>
>>  ****
>>
>> Sean****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
>> Minessale
>> *Sent:* Monday, December 17, 2012 12:32 PM****
>>
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
>> [SERVICE_NOT_IMPLEMENTED]****
>>
>>  ****
>>
>> The siptrace is not in here?  Did you enable it on the console and then
>> capture to the log file perhaps?****
>>
>> you should do "sofia tracelevel debug" too to route traces to the log
>> file.****
>>
>>  ****
>>
>> Its hard to say for sure with no sip trace but it seems like the far end
>> is rejecting the session timer re-invite causing the call to end.****
>>
>>  ****
>>
>>  ****
>>
>> On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com>
>> wrote:****
>>
>> http://pastebin.freeswitch.org/20342****
>>
>>
>>
>> ****
>>
>>  ****
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>>
>>
>> ****
>>
>>  ****
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>>
>>
>> ****
>>
>>  ****
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>>
>>
>> ****
>>
>>  ****
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>> ** **
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From cal.leeming at simplicitymedialtd.co.uk  Thu Dec 20 20:17:47 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Thu, 20 Dec 2012 17:17:47 +0000
Subject: [Freeswitch-users] Quick help needed
In-Reply-To: <CAMbW9oK=Ha2VfhtyMCHoV1xZDY9ONR-To_fyhLMnwC3xnO-u6w@mail.gmail.com>
References: <030101cdde8c$a44f2390$eced6ab0$@co.in>
	<CALNZuwkisdhY_FUA9AiTca2rYTGYPgFcUN_C96N-w3DTyQi20w@mail.gmail.com>
	<CAFiqYumnWWJh2=Se=xC+Y5sM-beXd=P1QafSUNVyLp+MqUCFtw@mail.gmail.com>
	<CAHKQagGkrFq5FpwWW4fSY-DrF4bXx01E_hQq=OjpaRakEFFnJQ@mail.gmail.com>
	<CAMbW9oK=Ha2VfhtyMCHoV1xZDY9ONR-To_fyhLMnwC3xnO-u6w@mail.gmail.com>
Message-ID: <CAHKQagHGyeaqNQ8DMC3rWbczGwLUZUdw4rYeqGMOzR4QaPLhyg@mail.gmail.com>

If you're looking for a decent TTS, you may want to consider this;

http://www.nuance.co.uk/for-business/by-solution/customer-service-solutions/solutions-services/inbound-solutions/self-service-automation/vocalizer/index.htm

They are pretty much the best on the market right now - but I still don't
think it's great.

Another option is to go on fiverr.com and get some voice work done for 5
bucks, plenty of voice artists there!!

Cal

On Thu, Dec 20, 2012 at 4:27 PM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> +1 for prerecorded files, I've not found 1 TTS I've liked.
>
> On 20 December 2012 13:32, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> Also fyi, during our own tests, Flite caused FreeSWITCH 1.2.3 to crash
>> under heavy load (60+ calls/sec).
>>
>> Cal
>>
>>
>> On Thu, Dec 20, 2012 at 1:16 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Bear in mind a prerecorded file will consume less CPU than performing
>>> TTS if this is a prompt that'll be played a lot.
>>>
>>> If you don't want to pay a voice actor or do it yourself, you could
>>> always generate a prerecorded file using TTS.
>>>
>>>
>>> On 20 December 2012 08:42, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> SAY isn't for dynamic text, but rather to join together pre-recorded
>>>> files for saying numbers, dates, etc. see: SAY on the wiki<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say>
>>>>
>>>> You either want:
>>>> 1) playback<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_playback> of
>>>> a pre-recorded file.
>>>> 2) use TTS with the free flite e.g. with SPEAK<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_speak> (not
>>>> say) e.g.:
>>>>
>>>> <action application="speak" data="flite|kal|FreeSWITCH is awesome"/>
>>>>
>>>>
>>>> -Avi
>>>>
>>>>
>>>> On Thu, Dec 20, 2012 at 10:33 AM, Nitin Tomer <ntomer at newgen.co.in>wrote:
>>>>
>>>>>  Hi,****
>>>>>
>>>>> ** **
>>>>>
>>>>> In my application, I am parking a call to an extension using
>>>>> valet-park. Before this I want to read  a message to caller, but am not
>>>>> able to. I tried to use say and phrase but am not able to get the syntax
>>>>> right. Please help me.****
>>>>>
>>>>> ** **
>>>>>
>>>>> Dialplan ?****
>>>>>
>>>>> ** **
>>>>>
>>>>> <extension>****
>>>>>
>>>>>                         <condition field="destination_number"
>>>>> expression="^(450)$">
>>>>>                                     <action application="set"
>>>>> data="caller_id_name=Account Opening Request" />
>>>>>                                     <action application="set"
>>>>> data="call_timeout=60" />
>>>>>                                     <action application="set"
>>>>> data="originate_timeout=60" />
>>>>>                                     <action inline="true"
>>>>> application="lua" data="accountopening.lua"/>
>>>>>                                     <action application="db"
>>>>> data="insert/accountopening/${parkednumber}/nonumber" />
>>>>>                                     <action application="say" data="en
>>>>> messages pronounced 'Please wait for an agent to pick your call.'" />
>>>>>                                     <action application="valet_park"
>>>>> data="my_lot ${parkednumber}" />          ****
>>>>>
>>>>>               </condition>
>>>>> </extension>****
>>>>>
>>>>> ** **
>>>>>
>>>>> Regards****
>>>>>
>>>>> ** **
>>>>>
>>>>> Nitin****
>>>>>
>>>>> Disclaimer :- This e-mail and any attachment may contain confidential,
>>>>> proprietary or legally privileged information. If you are not the original
>>>>> intended recipient and have erroneously received this message, you are
>>>>> prohibited from using, copying, altering or disclosing the content of this
>>>>> message. Please delete it immediately and notify the sender. Newgen
>>>>> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
>>>>> damage arising from the use of the information transmitted by this email
>>>>> including damages from virus and further acknowledges that no binding
>>>>> nature of the message shall be implied or assumed unless the sender does so
>>>>> expressly with due authority of NSTL.
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting 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
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 20 20:31:35 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 20 Dec 2012 12:31:35 -0500
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <CAKzWOxUBq8k+fzURNFBc6gx+348upE1H-kY=rR3M_kGqj6XpHg@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>	<541b01cddcb1$72330200$56990600$@bizfocused.com>	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>	<5da601cddd67$dc76b6f0$956424d0$@bizfocused.com>	<CAKbxfG_eXUY=eLnp-_xLks4-bXaoR3YiwqvsJ0teV15AF8w6vA@mail.gmail.com>	<681801cdde31$d4fe1ec0$7efa5c40$@bizfocused.com>	<CAKbxfG_4Xp=DwOnq6zFo=S-30BpN_BME+wSq8pzho4VusuQCpg@mail.gmail.com>	<68b001cdde38$f31f6bf0$d95e43d0$@bizfocused.com>	<CAFiqYumb-6T767WKsmyzyoA+-84Ct+M2Fbx6RJD9DYsHCW=+Ag@mail.gmail.com>	<6ded01cddec8$3a7e17a0$af7a46e0$@bizfocused.com>	<CAFiqY
	u=RTcO60HmQ1YTyDcSRHcRdioBXG2ydTXoDpXtt8Jb3Gw@mail.gmail.com>
	<CAKzWOxUBq8k+fzURNFBc6gx+348upE1H-kY=rR3M_kGqj6XpHg@mail.gmail.com>
Message-ID: <6f4001cdded7$dc6df120$9549d360$@bizfocused.com>

First, THANKS to all for ideas.

Second, I am at firmware 7.4.n  Cisco has jumped to 7.5.3  I had no idea it
was so volatile. I will upgrade on Saturday.

Third, Anthony M asked for the sip trace in the file.  I do ANYTHING he
wants!

 

Can someone spend 3 minutes and explain the relationships and functions of:

1.       Sofia loglevel

2.       Sofia tracelevel

3.       Console loglevel

4.       Sofia global siptrace  (and sofia prfile XYZ siptrace on|off)

Thanks,

Sean

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, December 20, 2012 11:50 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

 

On Thu, Dec 20, 2012 at 8:36 AM, Steven Ayre <steveayre at gmail.com> wrote:

Why does it only happen occasionally (even from this same caller)?

That makes me wonder if there's a bug that rarely treats Block Anonymous
Call as Yes. Cisco has been known for buggy SIP implementations in the past.
See if there's a firmware upgrade available.

 

Otherwise it's also possible it's disliking something set in the
Allow/Supported/Allow-Events headers. I would take a siptrace of a working
call to the same user and compare the INVITE packets to see if there's a
difference.

 

You don't need the sofia stack debugging on by the way, which'll make your
logs much smaller.

Pro tip:
For SIP trace just use something like:
sofia global siptrace on
-OR-
sofia profile internal siptrace on

Don't do "sofia loglevel xxx" if you're just wanting a simple SIP trace.
-MC 

 

-Steve

 

 

 

On 20 December 2012 15:39, Sean Devoy <sdevoy at bizfocused.com> wrote:

Thank you very much for your response Steven.  I have checked the Block
Anonymous Call and it is set to NO.

 

But, at least I know it is the 504G's issue and I can dig deeper there.  I
will certainly start with the firmware versions, etc.

 

Can you take a guess at something for me?

                Why does it only happen occasionally (even from this same
caller)?

                Is this the same SIP response to a phone on Do Not Disturb?

 

Thanks again.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: Thursday, December 20, 2012 9:50 AM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

 

The Cisco SPA504G doesn't like something in this request and gives the 406
Not Acceptable response (SERVICE_NOT_IMPLEMENTED is the Q.931 mapping of SIP
406).

   INVITE sip:302 at 69.251.170.6:1085 SIP/2.0

   Via: SIP/2.0/UDP 204.62.15.231;rport;branch=z9hG4bKaFrHeKgQ6F2Ue

   Max-Forwards: 70

   From: "anonymous" <sip:4104850152 at 204.62.15.231
<mailto:sip%3A4104850152 at 204.62.15.231> >;tag=aK7263m4r24Hj

   To: <sip:302 at 69.251.170.6:1085>

   Call-ID: 8308fa5b-c3ed-1230-899f-03777f384a29

   CSeq: 37593727 INVITE

   Contact: <sip:mod_sofia at 204.62.15.231:5060>

   User-Agent:
FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120
712T101002Z

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: precondition, path, replaces

   Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer

   Content-Type: application/sdp

   Content-Disposition: session

   Content-Length: 209

   X-FS-Support: update_display,send_info

   Remote-Party-ID: "anonymous" <sip:4104850152 at 204.62.15.231
<mailto:sip%3A4104850152 at 204.62.15.231>
>;party=calling;screen=yes;privacy=full

   

   v=0

   o=FreeSWITCH 1355837829 1355837830 IN IP4 204.62.15.231

   s=FreeSWITCH

   c=IN IP4 204.62.15.231

   t=0 0

   m=audio 21754 RTP/AVP 9 0 8 3 101 13

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-16

   a=ptime:20

 

This site suggests that it could be blocking the call because the callerid
is "anonymous""

http://www.mysysadmintips.com/linux/servers/309-linksys-spa-voip-phone-rejec
ts-calls-from-skype-sip-trunk-on-asterisk

That page shows a setting to disable that, can see you if that helps?

You could also try setting effective_caller_id_name to send something other
than anonymous.

 

-Steve

 

 

On 19 December 2012 22:33, Sean Devoy <sdevoy at bizfocused.com> wrote:

Sorry, server won't serve .log files!

http://www.bizfocused.com/service_not_found.txt

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, December 19, 2012 4:50 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

No the url is 404 so I can't see the file.

 

 

On Wed, Dec 19, 2012 at 3:43 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Whoa.   404 a.k.a device not found?

 

This could be as I suspected, router capacity exceeded or general NAT issue
with this router?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, December 19, 2012 1:27 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

it gives 404 err

 

On Tue, Dec 18, 2012 at 3:37 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

OK, I got one the logs!  I have trimmed the file down, but it is still over
5 MB and pastebin just chokes on it.

 

It is on my server:  http://www.bizfocused.com/service_not_found.log

 

Thanks for looking.

 

If you really need pcap, I am afraid I need detailed instructions on
installing/running it.

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 7:00 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

usually the 406 is done in an established call to refuse a codec change
during re-invite.

Its possible the other end thinks we want to change the codec when we do the
session-timer re-invite but I'm sure we don't but the sip trace can help
shed some light.  You can run a pcap too at the same time so when we find
the bad call in the logs we can filter it out of the pcap too.  To avoid it
getting too big you can just restart it every so often or use sippcapdump
and delete calls that are not affected.

 

 

 

On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Waiting for another failure with siptrace REALLY on this time.

 

If the user has clicked DND on these cisco phones, could that cause these
messages?

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 12:32 PM


To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

The siptrace is not in here?  Did you enable it on the console and then
capture to the log file perhaps?

you should do "sofia tracelevel debug" too to route traces to the log file.

 

Its hard to say for sure with no sip trace but it seems like the far end is
rejecting the session timer re-invite causing the call to end.

 

 

On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

http://pastebin.freeswitch.org/20342





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900> 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900> 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900>  


_________________________________________________________________________
Professional FreeSWITCH Consulting 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





 

-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900 <tel:%2B19193869900>  


_________________________________________________________________________
Professional FreeSWITCH Consulting 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

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

 


_________________________________________________________________________
Professional FreeSWITCH Consulting 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




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org



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From Ryan at ocens.com  Thu Dec 20 20:41:09 2012
From: Ryan at ocens.com (Ryan Watkins)
Date: Thu, 20 Dec 2012 17:41:09 +0000
Subject: [Freeswitch-users] Codec usage between FS and Gateway
Message-ID: <44E5C0A9D48A3246966A4AE04692014D550E1DC6@BN1PRD0612MB650.namprd06.prod.outlook.com>

Hello all!

Got a question that I hope someone can help with? I'm trying to test Codec2 usage (mod_codec2) however, since there doesn't seem to be any hard/softphones out that use the codec, I've setup two FS servers in an attempt to do the following:

SIP Phone < -- any codec --> FS Server 1 < -- codec2 --> FS Server 2 < -- any codec --> SIP Trunk < -- --> endpoint

So my question is how do you enforce the use of mod_codec2 between the two FS servers, while still allowing the endpoints to use any codec they support?

Thanks!

Ryan Watkins
Networking & Customer Support

OCENS
22608 Marine View Drive South Suite 300
Des Moines, WA 98198

Satellite Systems and Services for Iridium, Inmarsat, Globalstar, KVH
________________________________________________________
www.ocens.com<http://www.ocens.com> | support.ocens.com<http://support.ocens.com>
Office: (206) 878-8270 x1004 | Cell: (360) 521-7334 | Fax: (206) 878-8314
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From sdevoy at bizfocused.com  Thu Dec 20 20:47:05 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 20 Dec 2012 12:47:05 -0500
Subject: [Freeswitch-users] Yet another NAT question
Message-ID: <6f6c01cddeda$06e96bd0$14bc4370$@bizfocused.com>

Hi,

 

Since several people have suggested to avoid SIP ALG at all costs, I have
disabled it and banged my head some more!  Without SIP ALG, only one of my
Polycom 335s can Register.  The other 5 fail. (With SIP ALG they all
register fine).

 

Now I have found that changing:

SETTINGS>SIP>LOCAL SIP PORT> to a UNIQUE port (5066 in this case)

And

SETTINGS>NETWORK>NAT>SIGNALING PORT> to the same UNIQUE port

I got a second phone to REGISTER and exchange calls!

 

When I say unique port, I mean unique to all the phones on this NAT Router.

 

My question is "Is that fine?"

I just want to know if there are limitations on the SIP Port number?  Are
there issues I should watch out for.

 

Thanks,

Sean

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From steveayre at gmail.com  Thu Dec 20 20:56:25 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Dec 2012 17:56:25 +0000
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <6f4001cdded7$dc6df120$9549d360$@bizfocused.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>
	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>
	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>
	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>
	<541b01cddcb1$72330200$56990600$@bizfocused.com>
	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>
	<5da601cddd67$dc76b6f0$956424d0$@bizfocused.com>
	<CAKbxfG_eXUY=eLnp-_xLks4-bXaoR3YiwqvsJ0teV15AF8w6vA@mail.gmail.com>
	<681801cdde31$d4fe1ec0$7efa5c40$@bizfocused.com>
	<CAKbxfG_4Xp=DwOnq6zFo=S-30BpN_BME+wSq8pzho4VusuQCpg@mail.gmail.com>
	<68b001cdde38$f31f6bf0$d95e43d0$@bizfocused.com>
	<CAFiqYumb-6T767WKsmyzyoA+-84Ct+M2Fbx6RJD9DYsHCW=+Ag@mail.gmail.com>
	<6ded01cddec8$3a7e17a0$af7a46e0$@bizfocused.com>
	<CAKzWOxUBq8k+fzURNFBc6gx+348upE1H-kY=rR3M_kGqj6XpHg@mail.gmail.com>
	<6f4001cdded7$dc6df120$9549d360$@bizfocused.com>
Message-ID: <CAFiqYu=+8kenVacCSkH=-H62h=8ijQBybAnCPtx_mWCxM6GbJQ@mail.gmail.com>

1.       **Sofia loglevel****

Turns on *extremely* verbose debugging of what's happening within the sofia
stack components. You'll rarely need this, if ever.

**2.       **Sofia tracelevel****

Sets sets the level at which the siptrace writes to the log file.

**3.       **Console loglevel****

Sets the log level you see at the console. That's the console when running
freeswitch in the foreground, which is distinct from fs_cli where the
equivalent is the /log command.

**4.       **Sofia global siptrace  (and sofia prfile XYZ siptrace on|off)
Enable logging of SIP packets globally or for a specific profile.




On 20 December 2012 17:31, Sean Devoy <sdevoy at bizfocused.com> wrote:

> First, THANKS to all for ideas.****
>
> Second, I am at firmware 7.4.n  Cisco has jumped to 7.5.3  I had no idea
> it was so volatile. I will upgrade on Saturday.****
>
> Third, Anthony M asked for the sip trace in the file.  I do ANYTHING he
> wants!****
>
> ** **
>
> Can someone spend 3 minutes and explain the relationships and functions of:
> ****
>
> **1.       **Sofia loglevel****
>
> **2.       **Sofia tracelevel****
>
> **3.       **Console loglevel****
>
> **4.       **Sofia global siptrace  (and sofia prfile XYZ siptrace on|off)
> ****
>
> Thanks,****
>
> Sean****
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, December 20, 2012 11:50 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
> ** **
>
> ** **
>
> On Thu, Dec 20, 2012 at 8:36 AM, Steven Ayre <steveayre at gmail.com> wrote:*
> ***
>
> Why does it only happen occasionally (even from this same caller)?****
>
> That makes me wonder if there's a bug that rarely treats Block Anonymous
> Call as Yes. Cisco has been known for buggy SIP implementations in the
> past. See if there's a firmware upgrade available.****
>
> ** **
>
> Otherwise it's also possible it's disliking something set in the
> Allow/Supported/Allow-Events headers. I would take a siptrace of a working
> call to the same user and compare the INVITE packets to see if there's a
> difference.****
>
> ** **
>
> You don't need the sofia stack debugging on by the way, which'll make your
> logs much smaller.****
>
> Pro tip:
> For SIP trace just use something like:
> sofia global siptrace on
> -OR-
> sofia profile internal siptrace on
>
> Don't do "sofia loglevel xxx" if you're just wanting a simple SIP trace.
> -MC ****
>
> ** **
>
> -Steve****
>
> ** **
>
>  ****
>
> ** **
>
> On 20 December 2012 15:39, Sean Devoy <sdevoy at bizfocused.com> wrote:****
>
> Thank you very much for your response Steven.  I have checked the Block
> Anonymous Call and it is set to NO.****
>
>  ****
>
> But, at least I know it is the 504G?s issue and I can dig deeper there.  I
> will certainly start with the firmware versions, etc.****
>
>  ****
>
> Can you take a guess at something for me?****
>
>                 Why does it only happen occasionally (even from this same
> caller)?****
>
>                 Is this the same SIP response to a phone on Do Not Disturb?
> ****
>
>  ****
>
> Thanks again.****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* Thursday, December 20, 2012 9:50 AM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
>  ****
>
> The Cisco SPA504G doesn't like something in this request and gives the 406
> Not Acceptable response (SERVICE_NOT_IMPLEMENTED is the Q.931 mapping of
> SIP 406).****
>
>    INVITE sip:302 at 69.251.170.6:1085 SIP/2.0****
>
>    Via: SIP/2.0/UDP 204.62.15.231;rport;branch=z9hG4bKaFrHeKgQ6F2Ue****
>
>    Max-Forwards: 70****
>
>    From: "anonymous" <sip:4104850152 at 204.62.15.231>;tag=aK7263m4r24Hj****
>
>    To: <sip:302 at 69.251.170.6:1085>****
>
>    Call-ID: 8308fa5b-c3ed-1230-899f-03777f384a29****
>
>    CSeq: 37593727 INVITE****
>
>    Contact: <sip:mod_sofia at 204.62.15.231:5060>****
>
>    User-Agent:
> FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
> ****
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE****
>
>    Supported: precondition, path, replaces****
>
>    Allow-Events: talk, hold, conference, presence, dialog, line-seize,
> call-info, sla, include-session-description, presence.winfo,
> message-summary, refer****
>
>    Content-Type: application/sdp****
>
>    Content-Disposition: session****
>
>    Content-Length: 209****
>
>    X-FS-Support: update_display,send_info****
>
>    Remote-Party-ID: "anonymous" <sip:4104850152 at 204.62.15.231
> >;party=calling;screen=yes;privacy=full****
>
>    ****
>
>    v=0****
>
>    o=FreeSWITCH 1355837829 1355837830 IN IP4 204.62.15.231****
>
>    s=FreeSWITCH****
>
>    c=IN IP4 204.62.15.231****
>
>    t=0 0****
>
>    m=audio 21754 RTP/AVP 9 0 8 3 101 13****
>
>    a=rtpmap:101 telephone-event/8000****
>
>    a=fmtp:101 0-16****
>
>    a=ptime:20****
>
>  ****
>
> This site suggests that it could be blocking the call because the callerid
> is "anonymous""****
>
>
> http://www.mysysadmintips.com/linux/servers/309-linksys-spa-voip-phone-rejects-calls-from-skype-sip-trunk-on-asterisk
> ****
>
> That page shows a setting to disable that, can see you if that helps?****
>
> You could also try setting effective_caller_id_name to send something
> other than anonymous.****
>
>  ****
>
> -Steve****
>
>  ****
>
>  ****
>
> On 19 December 2012 22:33, Sean Devoy <sdevoy at bizfocused.com> wrote:****
>
> Sorry, server won?t serve .log files!****
>
> http://www.bizfocused.com/service_not_found.txt****
>
>  ****
>
>  ****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, December 19, 2012 4:50 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> No the url is 404 so I can't see the file.****
>
>  ****
>
>  ****
>
> On Wed, Dec 19, 2012 at 3:43 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> Whoa.   404 a.k.a device not found?****
>
>  ****
>
> This could be as I suspected, router capacity exceeded or general NAT
> issue with this router?****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, December 19, 2012 1:27 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> it gives 404 err****
>
>  ****
>
> On Tue, Dec 18, 2012 at 3:37 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> OK, I got one the logs!  I have trimmed the file down, but it is still
> over 5 MB and pastebin just chokes on it.****
>
>  ****
>
> It is on my server:  http://www.bizfocused.com/service_not_found.log****
>
>  ****
>
> Thanks for looking.****
>
>  ****
>
> If you really need pcap, I am afraid I need detailed instructions on
> installing/running it.****
>
>  ****
>
> Sean****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 17, 2012 7:00 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> usually the 406 is done in an established call to refuse a codec change
> during re-invite.****
>
> Its possible the other end thinks we want to change the codec when we do
> the session-timer re-invite but I'm sure we don't but the sip trace can
> help shed some light.  You can run a pcap too at the same time so when we
> find the bad call in the logs we can filter it out of the pcap too.  To
> avoid it getting too big you can just restart it every so often or use
> sippcapdump and delete calls that are not affected.****
>
>  ****
>
>  ****
>
>  ****
>
> On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> Waiting for another failure with siptrace REALLY on this time.****
>
>  ****
>
> If the user has clicked DND on these cisco phones, could that cause these
> messages?****
>
>  ****
>
> Sean****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 17, 2012 12:32 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]****
>
>  ****
>
> The siptrace is not in here?  Did you enable it on the console and then
> capture to the log file perhaps?****
>
> you should do "sofia tracelevel debug" too to route traces to the log file.
> ****
>
>  ****
>
> Its hard to say for sure with no sip trace but it seems like the far end
> is rejecting the session timer re-invite causing the call to end.****
>
>  ****
>
>  ****
>
> On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com>
> wrote:****
>
> http://pastebin.freeswitch.org/20342****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
>  ****
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
> ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 anthony.minessale at gmail.com  Thu Dec 20 21:04:52 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 20 Dec 2012 12:04:52 -0600
Subject: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
In-Reply-To: <CAFiqYu=+8kenVacCSkH=-H62h=8ijQBybAnCPtx_mWCxM6GbJQ@mail.gmail.com>
References: <31c601cdd6f7$7a4f5ca0$6eee15e0$@bizfocused.com>
	<CAKbxfG-r1+PSAPHAikbJdqvK5tbLAcueFzrs0gucX2229+XZmQ@mail.gmail.com>
	<0d3301cdd953$108e5a10$31ab0e30$@bizfocused.com>
	<CAKbxfG_Kfv4RAszTb4jFciovXKgAwoYqVb71DBF_KoFtY6OKQw@mail.gmail.com>
	<0d8901cdd956$f5c6be30$e1543a90$@bizfocused.com>
	<CAKbxfG847gSvz0HcVNtJZ_kiJ=Eiw8x5eV8qxNWvciCpuWawfA@mail.gmail.com>
	<500e01cddc78$beb4f950$3c1eebf0$@bizfocused.com>
	<CAKbxfG-CYsGoLiTsZY4TQG7z5-7JD3=XUFgv8Y5KitjCog6pTQ@mail.gmail.com>
	<541b01cddcb1$72330200$56990600$@bizfocused.com>
	<CAKbxfG9hx4J-pXo4gERW49iirFdr9Q-WeKFe0n_KB+pqZWJVSQ@mail.gmail.com>
	<5da601cddd67$dc76b6f0$956424d0$@bizfocused.com>
	<CAKbxfG_eXUY=eLnp-_xLks4-bXaoR3YiwqvsJ0teV15AF8w6vA@mail.gmail.com>
	<681801cdde31$d4fe1ec0$7efa5c40$@bizfocused.com>
	<CAKbxfG_4Xp=DwOnq6zFo=S-30BpN_BME+wSq8pzho4VusuQCpg@mail.gmail.com>
	<68b001cdde38$f31f6bf0$d95e43d0$@bizfocused.com>
	<CAFiqYumb-6T767WKsmyzyoA+-84Ct+M2Fbx6RJD9DYsHCW=+Ag@mail.gmail.com>
	<6ded01cddec8$3a7e17a0$af7a46e0$@bizfocused.com>
	<CAKzWOxUBq8k+fzURNFBc6gx+348upE1H-kY=rR3M_kGqj6XpHg@mail.gmail.com>
	<6f4001cdded7$dc6df120$9549d360$@bizfocused.com>
	<CAFiqYu=+8kenVacCSkH=-H62h=8ijQBybAnCPtx_mWCxM6GbJQ@mail.gmail.com>
Message-ID: <CAKbxfG_GJESr-qjLOoSOfqSNyR_drUzca1PkBCkDQTna-qXkWA@mail.gmail.com>

sdp says audio 21754 RTP/AVP 9 0 8 3 101 13

maybe its something silly about the codec, you could try putting

{absolute_codec_string=PCMU} in your dial sting to rule out any odd bug.

also you could try adding

{verbose_sdp=true} to enable bigger sdp for some challenged devices who
cant parse it right.





On Thu, Dec 20, 2012 at 11:56 AM, Steven Ayre <steveayre at gmail.com> wrote:

> 1.       **Sofia loglevel****
>
> Turns on *extremely* verbose debugging of what's happening within the
> sofia stack components. You'll rarely need this, if ever.
>
> **2.       **Sofia tracelevel****
>
> Sets sets the level at which the siptrace writes to the log file.
>
> **3.       **Console loglevel****
>
> Sets the log level you see at the console. That's the console when running
> freeswitch in the foreground, which is distinct from fs_cli where the
> equivalent is the /log command.
>
> **4.       **Sofia global siptrace  (and sofia prfile XYZ siptrace on|off)
> Enable logging of SIP packets globally or for a specific profile.
>
>
>
>
> On 20 December 2012 17:31, Sean Devoy <sdevoy at bizfocused.com> wrote:
>
>> First, THANKS to all for ideas.****
>>
>> Second, I am at firmware 7.4.n  Cisco has jumped to 7.5.3  I had no idea
>> it was so volatile. I will upgrade on Saturday.****
>>
>> Third, Anthony M asked for the sip trace in the file.  I do ANYTHING he
>> wants!****
>>
>> ** **
>>
>> Can someone spend 3 minutes and explain the relationships and functions
>> of:****
>>
>> **1.       **Sofia loglevel****
>>
>> **2.       **Sofia tracelevel****
>>
>> **3.       **Console loglevel****
>>
>> **4.       **Sofia global siptrace  (and sofia prfile XYZ siptrace
>> on|off)****
>>
>> Thanks,****
>>
>> Sean****
>>
>> ** **
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
>> Collins
>> *Sent:* Thursday, December 20, 2012 11:50 AM
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
>> [SERVICE_NOT_IMPLEMENTED]****
>>
>> ** **
>>
>> ** **
>>
>> On Thu, Dec 20, 2012 at 8:36 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> ****
>>
>> Why does it only happen occasionally (even from this same caller)?****
>>
>> That makes me wonder if there's a bug that rarely treats Block Anonymous
>> Call as Yes. Cisco has been known for buggy SIP implementations in the
>> past. See if there's a firmware upgrade available.****
>>
>> ** **
>>
>> Otherwise it's also possible it's disliking something set in the
>> Allow/Supported/Allow-Events headers. I would take a siptrace of a working
>> call to the same user and compare the INVITE packets to see if there's a
>> difference.****
>>
>> ** **
>>
>> You don't need the sofia stack debugging on by the way, which'll make
>> your logs much smaller.****
>>
>> Pro tip:
>> For SIP trace just use something like:
>> sofia global siptrace on
>> -OR-
>> sofia profile internal siptrace on
>>
>> Don't do "sofia loglevel xxx" if you're just wanting a simple SIP trace.
>> -MC ****
>>
>> ** **
>>
>> -Steve****
>>
>> ** **
>>
>>  ****
>>
>> ** **
>>
>> On 20 December 2012 15:39, Sean Devoy <sdevoy at bizfocused.com> wrote:****
>>
>> Thank you very much for your response Steven.  I have checked the Block
>> Anonymous Call and it is set to NO.****
>>
>>  ****
>>
>> But, at least I know it is the 504G?s issue and I can dig deeper there.
>> I will certainly start with the firmware versions, etc.****
>>
>>  ****
>>
>> Can you take a guess at something for me?****
>>
>>                 Why does it only happen occasionally (even from this same
>> caller)?****
>>
>>                 Is this the same SIP response to a phone on Do Not
>> Disturb?****
>>
>>  ****
>>
>> Thanks again.****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
>> *Sent:* Thursday, December 20, 2012 9:50 AM****
>>
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
>> [SERVICE_NOT_IMPLEMENTED]****
>>
>>  ****
>>
>>  ****
>>
>> The Cisco SPA504G doesn't like something in this request and gives the
>> 406 Not Acceptable response (SERVICE_NOT_IMPLEMENTED is the Q.931 mapping
>> of SIP 406).****
>>
>>    INVITE sip:302 at 69.251.170.6:1085 SIP/2.0****
>>
>>    Via: SIP/2.0/UDP 204.62.15.231;rport;branch=z9hG4bKaFrHeKgQ6F2Ue****
>>
>>    Max-Forwards: 70****
>>
>>    From: "anonymous" <sip:4104850152 at 204.62.15.231>;tag=aK7263m4r24Hj****
>>
>>    To: <sip:302 at 69.251.170.6:1085>****
>>
>>    Call-ID: 8308fa5b-c3ed-1230-899f-03777f384a29****
>>
>>    CSeq: 37593727 INVITE****
>>
>>    Contact: <sip:mod_sofia at 204.62.15.231:5060>****
>>
>>    User-Agent:
>> FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z
>> ****
>>
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE****
>>
>>    Supported: precondition, path, replaces****
>>
>>    Allow-Events: talk, hold, conference, presence, dialog, line-seize,
>> call-info, sla, include-session-description, presence.winfo,
>> message-summary, refer****
>>
>>    Content-Type: application/sdp****
>>
>>    Content-Disposition: session****
>>
>>    Content-Length: 209****
>>
>>    X-FS-Support: update_display,send_info****
>>
>>    Remote-Party-ID: "anonymous" <sip:4104850152 at 204.62.15.231
>> >;party=calling;screen=yes;privacy=full****
>>
>>    ****
>>
>>    v=0****
>>
>>    o=FreeSWITCH 1355837829 1355837830 IN IP4 204.62.15.231****
>>
>>    s=FreeSWITCH****
>>
>>    c=IN IP4 204.62.15.231****
>>
>>    t=0 0****
>>
>>    m=audio 21754 RTP/AVP 9 0 8 3 101 13****
>>
>>    a=rtpmap:101 telephone-event/8000****
>>
>>    a=fmtp:101 0-16****
>>
>>    a=ptime:20****
>>
>>  ****
>>
>> This site suggests that it could be blocking the call because the
>> callerid is "anonymous""****
>>
>>
>> http://www.mysysadmintips.com/linux/servers/309-linksys-spa-voip-phone-rejects-calls-from-skype-sip-trunk-on-asterisk
>> ****
>>
>> That page shows a setting to disable that, can see you if that helps?****
>>
>> You could also try setting effective_caller_id_name to send something
>> other than anonymous.****
>>
>>  ****
>>
>> -Steve****
>>
>>  ****
>>
>>  ****
>>
>> On 19 December 2012 22:33, Sean Devoy <sdevoy at bizfocused.com> wrote:****
>>
>> Sorry, server won?t serve .log files!****
>>
>> http://www.bizfocused.com/service_not_found.txt****
>>
>>  ****
>>
>>  ****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
>> Minessale
>> *Sent:* Wednesday, December 19, 2012 4:50 PM****
>>
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
>> [SERVICE_NOT_IMPLEMENTED]****
>>
>>  ****
>>
>> No the url is 404 so I can't see the file.****
>>
>>  ****
>>
>>  ****
>>
>> On Wed, Dec 19, 2012 at 3:43 PM, Sean Devoy <sdevoy at bizfocused.com>
>> wrote:****
>>
>> Whoa.   404 a.k.a device not found?****
>>
>>  ****
>>
>> This could be as I suspected, router capacity exceeded or general NAT
>> issue with this router?****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
>> Minessale
>> *Sent:* Wednesday, December 19, 2012 1:27 PM****
>>
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
>> [SERVICE_NOT_IMPLEMENTED]****
>>
>>  ****
>>
>> it gives 404 err****
>>
>>  ****
>>
>> On Tue, Dec 18, 2012 at 3:37 PM, Sean Devoy <sdevoy at bizfocused.com>
>> wrote:****
>>
>> OK, I got one the logs!  I have trimmed the file down, but it is still
>> over 5 MB and pastebin just chokes on it.****
>>
>>  ****
>>
>> It is on my server:  http://www.bizfocused.com/service_not_found.log****
>>
>>  ****
>>
>> Thanks for looking.****
>>
>>  ****
>>
>> If you really need pcap, I am afraid I need detailed instructions on
>> installing/running it.****
>>
>>  ****
>>
>> Sean****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
>> Minessale
>> *Sent:* Monday, December 17, 2012 7:00 PM****
>>
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
>> [SERVICE_NOT_IMPLEMENTED]****
>>
>>  ****
>>
>> usually the 406 is done in an established call to refuse a codec change
>> during re-invite.****
>>
>> Its possible the other end thinks we want to change the codec when we do
>> the session-timer re-invite but I'm sure we don't but the sip trace can
>> help shed some light.  You can run a pcap too at the same time so when we
>> find the bad call in the logs we can filter it out of the pcap too.  To
>> avoid it getting too big you can just restart it every so often or use
>> sippcapdump and delete calls that are not affected.****
>>
>>  ****
>>
>>  ****
>>
>>  ****
>>
>> On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com>
>> wrote:****
>>
>> Waiting for another failure with siptrace REALLY on this time.****
>>
>>  ****
>>
>> If the user has clicked DND on these cisco phones, could that cause these
>> messages?****
>>
>>  ****
>>
>> Sean****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
>> Minessale
>> *Sent:* Monday, December 17, 2012 12:32 PM****
>>
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA]
>> [SERVICE_NOT_IMPLEMENTED]****
>>
>>  ****
>>
>> The siptrace is not in here?  Did you enable it on the console and then
>> capture to the log file perhaps?****
>>
>> you should do "sofia tracelevel debug" too to route traces to the log
>> file.****
>>
>>  ****
>>
>> Its hard to say for sure with no sip trace but it seems like the far end
>> is rejecting the session timer re-invite causing the call to end.****
>>
>>  ****
>>
>>  ****
>>
>> On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com>
>> wrote:****
>>
>> http://pastebin.freeswitch.org/20342****
>>
>>
>>
>> ****
>>
>>  ****
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>>
>>
>> ****
>>
>>  ****
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>>
>>
>> ****
>>
>>  ****
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>>
>>
>> ****
>>
>>  ****
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>>  ****
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>> ****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
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From jpablolorenzetti at hotmail.com  Thu Dec 20 21:09:26 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Thu, 20 Dec 2012 18:09:26 +0000
Subject: [Freeswitch-users] call_timeout fails for callcenter agents
In-Reply-To: <CAHWtr1QoU6pc8Z1E1Oo4NLa+=FtMQqE8UYwdvHUxC1h9g9k3NA@mail.gmail.com>
References: <DUB002-W1033E5834F19C765571A6B1D9300@phx.gbl>,
	<CAHWtr1QoU6pc8Z1E1Oo4NLa+=FtMQqE8UYwdvHUxC1h9g9k3NA@mail.gmail.com>
Message-ID: <DUB002-W866D933ECBA654F1EA6126D9370@phx.gbl>

Regis thank you very much for your answer, actually leg_timeout worked. i think the wiki and samples, tutorials, etc need to be updated because all of them mention that call_timeout is the variable to set to accomplish what i was looking for, even the sample configuration file that comes with the callcenter says so ... in any case it works now .. thanks!!

Date: Thu, 20 Dec 2012 08:24:08 +0100
From: regis.freeswitch.org at tornad.net
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] call_timeout fails for callcenter agents

Hello.
I think you must use leg_timeout instead of call_timeout.

Regards


2012/12/19 Juan Pablo L. <jpablolorenzetti at hotmail.com>




Hi, i m trying to configure a callcenter, i need the application to ring the agentsfor 10 seconds and then move on to the next agent in the list, everything works ok except that the agents ring for more than 10 seconds, so the channel var call_timeout is not being used by
the callcenter application, probably due to a misconfiguration somewhere but i cant see where ... this is my config for the agents:
<agent name="2 at callcenter" type="callback" contact=" ignore_early_media=true,originate_timeout=10,call_timeout=10]sofia/gateway/huawei_csoft/6611290" status="Availabl
e" max-no-answer="3" wrap-up-time="10" reject- delay-time="5" busy-delay-time="5" no-answer-delay-time="5" />
and this is my config in the dialplan (also tried with exports):

<action application="set" data="hangup_after_bridge=true"/><action application="set" data="call_timeout=10"/>
<action application="set" data="originate_timeout=10"/><action application="set" data="ignore_early_media=true"/><action application="callcenter" data="callcenter at default"/>

i can not see what i m doing wrong so i added that variables in both places (dialplan and the agents defs)and i see in the logs that they are being set in both places:

EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(hangup_after_bridge=true)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [hangup_after_bridge]=[true]
EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(call_timeout=10)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [call_timeout]=[10]
EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(originate_timeout=10)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [originate_timeout]=[10]
EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(ignore_early_media=true)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [ignore_early_media]=[true]
EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 callcenter(callcenter at default)
2012-12-19 15:38:16.716876 [DEBUG] mod_callcenter.c:2508 Member 2802018 <2802018> joining queue callcenter at default2012-12-19 15:38:16.736878 [DEBUG] mod_local_stream.c:417 Opening Stream [moh/8000] 8000hz
2012-12-19 15:38:16.736878 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms2012-12-19 15:38:16.776878 [DEBUG] mod_callcenter.c:1049 Updated Agent 2 at callcenter set state = Receiving
2012-12-19 15:38:16.786922 [DEBUG] switch_ivr_originate.c:2005 Parsing global variables2012-12-19 15:38:16.786922 [DEBUG] switch_ivr_originate.c:2425 Parsing session specific variables2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [ignore_early_media]=[true]
2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [originate_timeout]=[10]2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [call_timeout]=[10]2012-12-19 15:38:16.786922 [NOTICE] switch_channel.c:951 New Channel sofia/external-huawei_gw/6611290 [09d48fc6-d133-4673-9f12-074af601f1f2]


kindly give me a hint what i m doing wrong, thanks a lot for your help.
 		 	   		  

_________________________________________________________________________

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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http://www.freeswitch.org





_________________________________________________________________________
Professional FreeSWITCH Consulting 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
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From steveayre at gmail.com  Thu Dec 20 22:01:59 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Dec 2012 19:01:59 +0000
Subject: [Freeswitch-users] Codec usage between FS and Gateway
In-Reply-To: <44E5C0A9D48A3246966A4AE04692014D550E1DC6@BN1PRD0612MB650.namprd06.prod.outlook.com>
References: <44E5C0A9D48A3246966A4AE04692014D550E1DC6@BN1PRD0612MB650.namprd06.prod.outlook.com>
Message-ID: <CAFiqYunM8v06k0UHmjx_VeqUdEN9NJRxhvd3kft95qY-_x0Muw@mail.gmail.com>

1. Configure profile incoming/outgoing codecs to all those you want to
support.
2. Use {absolute_codec_string=codec2} in the dialstring between FS servers.
This will override the outgoing codec setting.

-Steve



On 20 December 2012 17:41, Ryan Watkins <Ryan at ocens.com> wrote:

>  Hello all!
>
>  Got a question that I hope someone can help with? I'm trying to test
> Codec2 usage (mod_codec2) however, since there doesn't seem to be any
> hard/softphones out that use the codec, I've setup two FS servers in an
> attempt to do the following:
>
>  *SIP Phone* < -- any codec --> *FS Server 1* < -- codec2 --> *FS Server 2
> * < -- any codec --> *SIP Trunk* < -- --> *endpoint*
>
>  So my question is how do you enforce the use of mod_codec2 between the
> two FS servers, while still allowing the endpoints to use any codec they
> support?
>
>  Thanks!
>
>  *Ryan Watkins *
> *Networking & Customer Support*
>
> *OCENS*
> *22608 Marine View Drive South Suite 300
> Des Moines, WA 98198*
>
> Satellite Systems and Services for Iridium, Inmarsat, Globalstar, KVH
> ________________________________________________________
> www.ocens.com | support.ocens.com
> *Office*: (206) 878-8270 x1004 | *Cell*: (360) 521-7334 | *Fax*: (206)
> 878-8314
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 bdfoster at endigotech.com  Thu Dec 20 22:09:43 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 20 Dec 2012 14:09:43 -0500
Subject: [Freeswitch-users] call_timeout fails for callcenter agents
In-Reply-To: <DUB002-W866D933ECBA654F1EA6126D9370@phx.gbl>
References: <DUB002-W1033E5834F19C765571A6B1D9300@phx.gbl>
	<CAHWtr1QoU6pc8Z1E1Oo4NLa+=FtMQqE8UYwdvHUxC1h9g9k3NA@mail.gmail.com>
	<DUB002-W866D933ECBA654F1EA6126D9370@phx.gbl>
Message-ID: <5FCD045B-4DE7-4696-814F-94492286FC67@endigotech.com>

Ready to pay the wiki tax? ;)

Sent from my iPhone

On Dec 20, 2012, at 1:09 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:

> Regis thank you very much for your answer, actually leg_timeout worked. i think the wiki and samples, tutorials, etc need to be updated because all of them mention that call_timeout is the variable to set to accomplish what i was looking for, even the sample configuration file that comes with the callcenter says so ... in any case it works now .. thanks!!
> 
> Date: Thu, 20 Dec 2012 08:24:08 +0100
> From: regis.freeswitch.org at tornad.net
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] call_timeout fails for callcenter agents
> 
> Hello.
> I think you must use leg_timeout instead of call_timeout.
> 
> Regards
> 
> 
> 2012/12/19 Juan Pablo L. <jpablolorenzetti at hotmail.com>
> Hi, i m trying to configure a callcenter, i need the application to ring the agentsfor 10 seconds and then move on to the next agent in the list, 
> everything works ok except that the agents ring for more than 10 seconds, so the channel var call_timeout is not being used by
> the callcenter application, probably due to a misconfiguration somewhere but i cant see where ... this is my config for the agents:
> 
> <agent name="2 at callcenter" type="callback" contact=" ignore_early_media=true,originate_timeout=10,call_timeout=10]sofia/gateway/huawei_csoft/6611290" status="Availabl
> e" max-no-answer="3" wrap-up-time="10" reject- delay-time="5" busy-delay-time="5" no-answer-delay-time="5" />
> 
> and this is my config in the dialplan (also tried with exports):
> 
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set" data="call_timeout=10"/>
> <action application="set" data="originate_timeout=10"/>
> <action application="set" data="ignore_early_media=true"/>
> <action application="callcenter" data="callcenter at default"/>
> 
> i can not see what i m doing wrong so i added that variables in both places (dialplan and the agents defs)
> and i see in the logs that they are being set in both places:
> 
> EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(hangup_after_bridge=true)
> 2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [hangup_after_bridge]=[true]
> EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(call_timeout=10)
> 2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [call_timeout]=[10]
> EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(originate_timeout=10)
> 2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [originate_timeout]=[10]
> EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(ignore_early_media=true)
> 2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [ignore_early_media]=[true]
> EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 callcenter(callcenter at default)
> 2012-12-19 15:38:16.716876 [DEBUG] mod_callcenter.c:2508 Member 2802018 <2802018> joining queue callcenter at default
> 2012-12-19 15:38:16.736878 [DEBUG] mod_local_stream.c:417 Opening Stream [moh/8000] 8000hz
> 2012-12-19 15:38:16.736878 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
> 2012-12-19 15:38:16.776878 [DEBUG] mod_callcenter.c:1049 Updated Agent 2 at callcenter set state = Receiving
> 2012-12-19 15:38:16.786922 [DEBUG] switch_ivr_originate.c:2005 Parsing global variables
> 2012-12-19 15:38:16.786922 [DEBUG] switch_ivr_originate.c:2425 Parsing session specific variables
> 2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [ignore_early_media]=[true]
> 2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [originate_timeout]=[10]
> 2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [call_timeout]=[10]
> 2012-12-19 15:38:16.786922 [NOTICE] switch_channel.c:951 New Channel sofia/external-huawei_gw/6611290 [09d48fc6-d133-4673-9f12-074af601f1f2]
> 
> 
> kindly give me a hint what i m doing wrong, thanks a lot for your help.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________ Professional FreeSWITCH Consulting 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
> 
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From spencer at 5ninesolutions.com  Thu Dec 20 22:14:11 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 20 Dec 2012 11:14:11 -0800
Subject: [Freeswitch-users] t.38 Call Flow Information
Message-ID: <AAD54836-F121-4190-A473-B64D9A9E4612@5ninesolutions.com>

Hello,
I've been analyzing some Wireshark traces to get a better grasp on the t.38 gateway process.  I'm using t38_gateway to convert SIP t38 to TDM audio on the receiving end.  I'm struggling to understand the following call flow.  I've excluded the audio portion for brevity but t38 is negotiated correctly by both endpoints (FS on our end and Acme on the other).  Note the absence of a DIS message and the PPS prior to CFR.  Can anyone shed any light on this?

Thanks in advance,
Spencer


Acme                                                                                FreeSWITCH
|1716.691635|         INVITE SDP (t38)               |                   |SIP Request
|         |(5060)   <------------------  (5070)   |                   |
|1716.719556|         100 trying -- your call is important to us          |                   |SIP Status
|         |(5060)   ------------------>  (5070)   |                   |
|1716.830431|                   |         no-signal |                   |t38:t30 Ind:no-signal
|         |                   |(21440)  <------------------  (15876)  |
|1716.906132|         200 OK SDP (t38)              |                   |SIP Status
|         |(5060)   ------------------>  (5070)   |                   |
|1716.912839|         ACK       |                   |                   |SIP Request
|         |(5060)   <------------------  (5070)   |                   |
|1716.973011|                   |         no-signal |                   |t38:t30 Ind:no-signal
|         |                   |(21440)  ------------------>  (15876)  |
|1717.450595|                   |         v21-preamble                  |t38:t30 Ind:v21-preamble
|         |                   |(21440)  <------------------  (15876)  |
|1717.730869|                   |         PPS       |                   |t38:v21:HDLC:Partial Page Signal
|         |                   |(21440)  <------------------  (15876)  |
|1717.780811|                   |         no-signal |                   |t38:t30 Ind:no-signal
|         |                   |(21440)  <------------------  (15876)  |
|1718.130798|                   |         v17-14400-long-training          |t38:t30 Ind:v17-14400-long-training
|         |                   |(21440)  <------------------  (15876)  |
|1719.541095|                   |         t4-non-ecm-data:v17-14400          |t38:t4-non-ecm-data:v17-14400 Duration: 1.51s No packet lost
|         |                   |(21440)  <------------------  (15876)  |
|1721.060850|                   |         no-signal |                   |t38:t30 Ind:no-signal
|         |                   |(21440)  <------------------  (15876)  |
|1722.573572|                   |         v21-preamble                  |t38:t30 Ind:v21-preamble
|         |                   |(21440)  ------------------>  (15876)  |
|1723.593652|                   |         CFR       |                   |t38:v21:HDLC:Confirmation To Receive

From msc at freeswitch.org  Fri Dec 21 00:01:29 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Dec 2012 13:01:29 -0800
Subject: [Freeswitch-users] Tracing
In-Reply-To: <50D2C1E5.10708@softnet.si>
References: <50D2C1E5.10708@softnet.si>
Message-ID: <CAKzWOxWmLF05rzRCVjCCaQxqX2v6hahdve+PpuLtdts0C--xeg@mail.gmail.com>

Your best bet would be to use pcapsipdump and manually or via script clear
out the old pcap files each day/week/month/whatever.

Also, make sure that you are rotating your FreeSWITCH log files on a
regular basis. Also, be sure to enable uuid logging in your
logfile.conf.xml file. Having the uuid of the call in the fs log file is
really helpful on a busy system.

-MC

On Wed, Dec 19, 2012 at 11:44 PM, Miha <miha at softnet.si> wrote:

> Hi,
>
> I am experiencing some problems related with media with some users. It
> is almost impossible to figure it out where is a problem because this
> problem is not presented all day.
> I need some sip trace dump for a specific user if it is possible due to
> a lot of traffic. It would be nice that this would not cause to much CPU
> load and disk consumption as this would need to be running and tracing
> 24/7.
>
> Users are reporting the thy exeriacing noise in conversation and
> sometime thy do not hear other side.
>
> What would be the best way?
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Fri Dec 21 00:11:17 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Dec 2012 13:11:17 -0800
Subject: [Freeswitch-users] Yet another NAT question
In-Reply-To: <6f6c01cddeda$06e96bd0$14bc4370$@bizfocused.com>
References: <6f6c01cddeda$06e96bd0$14bc4370$@bizfocused.com>
Message-ID: <CAKzWOxVGxa8phbbt9E+tyVVJ3u7oGaz+GDoMJRrrDdFGeH+K7w@mail.gmail.com>

Using a "unique" SIP port for each phone is fine. Lots of people do that
for various reasons, one of which is, naturally, busting through NAT. FS
doesn't really care what SIP port the device is using as long as it knows
where to send the SIP traffic.

-MC

On Thu, Dec 20, 2012 at 9:47 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Hi,****
>
> ** **
>
> Since several people have suggested to avoid SIP ALG at all costs, I have
> disabled it and banged my head some more!  Without SIP ALG, only one of my
> Polycom 335s can Register.  The other 5 fail. (With SIP ALG they all
> register fine).****
>
> ** **
>
> Now I have found that changing:****
>
> *SETTINGS>SIP>LOCAL SIP PORT>* to a UNIQUE port (5066 in this case)****
>
> And****
>
> *SETTINGS>NETWORK>NAT>SIGNALING PORT>* to the same UNIQUE port****
>
> I got a second phone to REGISTER and exchange calls!****
>
> ** **
>
> When I say unique port, I mean unique to all the phones on this NAT Router.
> ****
>
> ** **
>
> My question is ?Is that fine??****
>
> I just want to know if there are limitations on the SIP Port number?  Are
> there issues I should watch out for.****
>
> ** **
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jpablolorenzetti at hotmail.com  Fri Dec 21 01:36:23 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Thu, 20 Dec 2012 22:36:23 +0000
Subject: [Freeswitch-users] call_timeout fails for callcenter agents
In-Reply-To: <5FCD045B-4DE7-4696-814F-94492286FC67@endigotech.com>
References: <DUB002-W1033E5834F19C765571A6B1D9300@phx.gbl>,
	<CAHWtr1QoU6pc8Z1E1Oo4NLa+=FtMQqE8UYwdvHUxC1h9g9k3NA@mail.gmail.com>,
	<DUB002-W866D933ECBA654F1EA6126D9370@phx.gbl>,
	<5FCD045B-4DE7-4696-814F-94492286FC67@endigotech.com>
Message-ID: <DUB002-W99CF867CF961444EE8E856D9370@phx.gbl>

lol ...

From: bdfoster at endigotech.com
Date: Thu, 20 Dec 2012 14:09:43 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] call_timeout fails for callcenter agents

Ready to pay the wiki tax? ;)

Sent from my iPhone
On Dec 20, 2012, at 1:09 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:




Regis thank you very much for your answer, actually leg_timeout worked. i think the wiki and samples, tutorials, etc need to be updated because all of them mention that call_timeout is the variable to set to accomplish what i was looking for, even the sample configuration file that comes with the callcenter says so ... in any case it works now .. thanks!!

Date: Thu, 20 Dec 2012 08:24:08 +0100
From: regis.freeswitch.org at tornad.net
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] call_timeout fails for callcenter agents

Hello.
I think you must use leg_timeout instead of call_timeout.

Regards


2012/12/19 Juan Pablo L. <jpablolorenzetti at hotmail.com>




Hi, i m trying to configure a callcenter, i need the application to ring the agentsfor 10 seconds and then move on to the next agent in the list, everything works ok except that the agents ring for more than 10 seconds, so the channel var call_timeout is not being used by
the callcenter application, probably due to a misconfiguration somewhere but i cant see where ... this is my config for the agents:
<agent name="2 at callcenter" type="callback" contact=" ignore_early_media=true,originate_timeout=10,call_timeout=10]sofia/gateway/huawei_csoft/6611290" status="Availabl
e" max-no-answer="3" wrap-up-time="10" reject- delay-time="5" busy-delay-time="5" no-answer-delay-time="5" />
and this is my config in the dialplan (also tried with exports):

<action application="set" data="hangup_after_bridge=true"/><action application="set" data="call_timeout=10"/>
<action application="set" data="originate_timeout=10"/><action application="set" data="ignore_early_media=true"/><action application="callcenter" data="callcenter at default"/>

i can not see what i m doing wrong so i added that variables in both places (dialplan and the agents defs)and i see in the logs that they are being set in both places:

EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(hangup_after_bridge=true)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [hangup_after_bridge]=[true]
EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(call_timeout=10)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [call_timeout]=[10]
EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(originate_timeout=10)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [originate_timeout]=[10]
EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 set(ignore_early_media=true)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 SET [ignore_early_media]=[true]
EXECUTE sofia/external-huawei_gw/2802018 at 10.49.0.2:5062 callcenter(callcenter at default)
2012-12-19 15:38:16.716876 [DEBUG] mod_callcenter.c:2508 Member 2802018 <2802018> joining queue callcenter at default2012-12-19 15:38:16.736878 [DEBUG] mod_local_stream.c:417 Opening Stream [moh/8000] 8000hz
2012-12-19 15:38:16.736878 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms2012-12-19 15:38:16.776878 [DEBUG] mod_callcenter.c:1049 Updated Agent 2 at callcenter set state = Receiving
2012-12-19 15:38:16.786922 [DEBUG] switch_ivr_originate.c:2005 Parsing global variables2012-12-19 15:38:16.786922 [DEBUG] switch_ivr_originate.c:2425 Parsing session specific variables2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [ignore_early_media]=[true]
2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [originate_timeout]=[10]2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [call_timeout]=[10]2012-12-19 15:38:16.786922 [NOTICE] switch_channel.c:951 New Channel sofia/external-huawei_gw/6611290 [09d48fc6-d133-4673-9f12-074af601f1f2]


kindly give me a hint what i m doing wrong, thanks a lot for your help.
 		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

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_________________________________________________________________________
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http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
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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

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

_________________________________________________________________________
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http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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FreeSWITCH-users mailing list
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From jpablolorenzetti at hotmail.com  Fri Dec 21 01:41:05 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Thu, 20 Dec 2012 22:41:05 +0000
Subject: [Freeswitch-users] core dump when compiling master git
Message-ID: <DUB002-W7263E86E1BE8C91318247CD9370@phx.gbl>

Hi, i m trying to update my development server (VM) by making a clean installation from git master, but i m getting a core dump while compiling:
/bin/sh: line 1:  9386 Segmentation fault      (core dumped) ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.hmake[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139make[6]: *** [all] Error 2make[5]: *** [all-recursive] Error 1make[4]: *** [/home/fs/freeswitch/libs/spandsp/src/libspandsp.la] Error 2make[3]: *** [mod_spandsp-all] Error 1make[2]: *** [all-recursive] Error 1make[1]: *** [all-recursive] Error 1make: *** [all] Error 2

i followed the classic steps (bootstrap,configure and make) is anyone experimenting the same issues ???my info is:Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011 i686 Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux
thanks! 		 	   		  
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From bdfoster at endigotech.com  Fri Dec 21 01:59:46 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 20 Dec 2012 17:59:46 -0500
Subject: [Freeswitch-users] core dump when compiling master git
In-Reply-To: <DUB002-W7263E86E1BE8C91318247CD9370@phx.gbl>
References: <DUB002-W7263E86E1BE8C91318247CD9370@phx.gbl>
Message-ID: <CAKV-jFzEhNMRqpeQ9+7y61w74k-6qum-RWfJBDFgwPQO_ZG8zQ@mail.gmail.com>

*Dear OP,*
*
*
*We've detected that this is a potential bug in FreeSWITCH. *Because this
is a potential bug, the mailing list is really not the place to report this.

*Before you file a bug report, and if you haven't done this already, *please
make sure that the symptoms you are experiencing do not concur with another
previously submitted bug report. Search JIRA <http://jira.freeswitch.org> to
ensure this. You might want to check the Wiki just in case this is a
feature, not a bug.

*You can make a difference. *Bugs can be reported to the FreeSWITCH
JIRA<http://jira.freeswitch.org>.
By doing this one simple task, you can save the FreeSWITCH developers the
blood, sweat, and tears that are needed in order to gather information
related to your bug report and track the commits needed to fix the problem.

*By filing a JIRA, *you also help the developers
gather relevant information so they can take the best course of action.
Developers ask for several pieces of information when filing a report. Take
a look at the Reporting
Bugs<http://wiki.freeswitch.org/wiki/Reporting_Bugs>page for more
details and instructions. The more information you can
gather, the easier it will be to fix the issue and the faster a commit can
be made.

*Remember that FreeSWITCH is an open source project.* FreeSWITCH Developers
do not get paid to fix bugs, they simply do it for the sake of the
community. They sacrifice sleep, family time, brain cells, and Advil to
help YOU. Show them that you care. *Report bugs the correct way.*



On Thu, Dec 20, 2012 at 5:41 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com
> wrote:

> Hi, i m trying to update my development server (VM) by making a clean
> installation from git master,
> but i m getting a core dump while compiling:
>
> /bin/sh: line 1:  9386 Segmentation fault      (core dumped)
> ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.h
> make[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139
> make[6]: *** [all] Error 2
> make[5]: *** [all-recursive] Error 1
> make[4]: *** [/home/fs/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
>
>
> i followed the classic steps (bootstrap,configure and make) is anyone
> experimenting the same issues ???
> my info is:
> Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011 i686
> Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
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.
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From msc at freeswitch.org  Fri Dec 21 02:06:30 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Dec 2012 15:06:30 -0800
Subject: [Freeswitch-users] core dump when compiling master git
In-Reply-To: <CAKV-jFzEhNMRqpeQ9+7y61w74k-6qum-RWfJBDFgwPQO_ZG8zQ@mail.gmail.com>
References: <DUB002-W7263E86E1BE8C91318247CD9370@phx.gbl>
	<CAKV-jFzEhNMRqpeQ9+7y61w74k-6qum-RWfJBDFgwPQO_ZG8zQ@mail.gmail.com>
Message-ID: <CAKzWOxUufkOJkeR3j=6JX8xAw2F7Cze-=P=u0zuDPN_iJn_hAA@mail.gmail.com>

Brian, have you typed this one before? :D

Juan, yes, definitely file a Jira on this. Also, collect the backtrace on
that segfault if you can and attach as a txt file to the Jira case.
-MC

On Thu, Dec 20, 2012 at 2:59 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> *Dear OP,*
> *
> *
> *We've detected that this is a potential bug in FreeSWITCH. *Because this
> is a potential bug, the mailing list is really not the place to report this.
>
> *Before you file a bug report, and if you haven't done this already, *please
> make sure that the symptoms you are experiencing do not concur with another
> previously submitted bug report. Search JIRA <http://jira.freeswitch.org> to
> ensure this. You might want to check the Wiki just in case this is a
> feature, not a bug.
>
> *You can make a difference. *Bugs can be reported to the FreeSWITCH JIRA<http://jira.freeswitch.org>.
> By doing this one simple task, you can save the FreeSWITCH developers the
> blood, sweat, and tears that are needed in order to gather information
> related to your bug report and track the commits needed to fix the problem.
>
> *By filing a JIRA, *you also help the developers
> gather relevant information so they can take the best course of action.
> Developers ask for several pieces of information when filing a report. Take
> a look at the Reporting Bugs<http://wiki.freeswitch.org/wiki/Reporting_Bugs>page for more details and instructions. The more information you can
> gather, the easier it will be to fix the issue and the faster a commit can
> be made.
>
> *Remember that FreeSWITCH is an open source project.* FreeSWITCH
> Developers do not get paid to fix bugs, they simply do it for the sake of
> the community. They sacrifice sleep, family time, brain cells, and Advil to
> help YOU. Show them that you care. *Report bugs the correct way.*
>
>
>
> On Thu, Dec 20, 2012 at 5:41 PM, Juan Pablo L. <
> jpablolorenzetti at hotmail.com> wrote:
>
>> Hi, i m trying to update my development server (VM) by making a clean
>> installation from git master,
>> but i m getting a core dump while compiling:
>>
>> /bin/sh: line 1:  9386 Segmentation fault      (core dumped)
>> ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.h
>> make[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139
>> make[6]: *** [all] Error 2
>> make[5]: *** [all-recursive] Error 1
>> make[4]: *** [/home/fs/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
>>
>>
>> i followed the classic steps (bootstrap,configure and make) is anyone
>> experimenting the same issues ???
>> my info is:
>> Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011 i686
>> Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From krice at freeswitch.org  Fri Dec 21 02:16:26 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 20 Dec 2012 17:16:26 -0600
Subject: [Freeswitch-users] core dump when compiling master git
In-Reply-To: <CAKzWOxUufkOJkeR3j=6JX8xAw2F7Cze-=P=u0zuDPN_iJn_hAA@mail.gmail.com>
Message-ID: <CCF8F86A.62599%krice@freeswitch.org>

This is not a Segfault in FreeSWITCH you both fail....

This is probably a build system issue....

I would recommend pastebinning and a full trace from a clean tree of the
bootstrap, configure and build process...



On 12/20/12 5:06 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> Brian, have you typed this one before? :D
> 
> Juan, yes, definitely file a Jira on this. Also, collect the backtrace on that
> segfault if you can and attach as a txt file to the Jira case.
> -MC
> 
> On Thu, Dec 20, 2012 at 2:59 PM, Brian Foster <bdfoster at endigotech.com> wrote:
>> Dear OP,
>> 
>> We've detected that this is a potential bug in FreeSWITCH. Because this is a
>> potential bug, the mailing list is really not the place to report this.
>> 
>> Before you file a bug report, and if you haven't done this already, please
>> make sure that the symptoms you are experiencing do not concur with another
>> previously submitted bug report. Search JIRA <http://jira.freeswitch.org> ?to
>> ensure this. You might want to check the Wiki just in case this is a feature,
>> not a bug.
>> 
>> You can make a difference. Bugs can be reported to the FreeSWITCH JIRA
>> <http://jira.freeswitch.org> . By doing this one simple task, you can save
>> the FreeSWITCH developers the blood, sweat, and tears that are needed in
>> order to gather information related to your bug report and track the commits
>> needed to fix the problem.
>> 
>> By filing a JIRA, you also help the developers gather?relevant?information so
>> they can take the best course of action. Developers ask for several pieces of
>> information when filing a report. Take a look at the Reporting Bugs
>> <http://wiki.freeswitch.org/wiki/Reporting_Bugs>  page for more details and
>> instructions. The more information you can gather, the easier it will be to
>> fix the issue and the faster a commit can be made.
>> 
>> Remember that FreeSWITCH is an open source project.?FreeSWITCH Developers do
>> not get paid to fix bugs, they simply do it for the sake of the community.
>> They sacrifice sleep, family time, brain cells, and Advil to help YOU. Show
>> them that you care. Report bugs the correct way.
>> 
>> 
>> 
>> On Thu, Dec 20, 2012 at 5:41 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com>
>> wrote:
>>> Hi, i m trying to update my development server (VM) by making a clean
>>> installation from git master,?
>>> but i m getting a core dump while compiling:
>>> 
>>> /bin/sh: line 1: ?9386 Segmentation fault ? ? ?(core dumped)
>>> ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.h
>>> make[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139
>>> make[6]: *** [all] Error 2
>>> make[5]: *** [all-recursive] Error 1
>>> make[4]: *** [/home/fs/freeswitch/libs/spandsp/src/libspandsp.la
>>> <http://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
>>> 
>>> 
>>> i followed the?classic?steps (bootstrap,configure and make) is anyone
>>> experimenting the same issues ???
>>> my info is:
>>> Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011 i686
>>> Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux
>>> 
>>> 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
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From bdfoster at endigotech.com  Fri Dec 21 02:20:34 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 20 Dec 2012 18:20:34 -0500
Subject: [Freeswitch-users] core dump when compiling master git
In-Reply-To: <CAKzWOxUufkOJkeR3j=6JX8xAw2F7Cze-=P=u0zuDPN_iJn_hAA@mail.gmail.com>
References: <DUB002-W7263E86E1BE8C91318247CD9370@phx.gbl>
	<CAKV-jFzEhNMRqpeQ9+7y61w74k-6qum-RWfJBDFgwPQO_ZG8zQ@mail.gmail.com>
	<CAKzWOxUufkOJkeR3j=6JX8xAw2F7Cze-=P=u0zuDPN_iJn_hAA@mail.gmail.com>
Message-ID: <CAKV-jFwyCzHxmV32iS-bH0VMLO3EYMgkd+OnKSwYd3XT3BOOEQ@mail.gmail.com>

I have a bunch of canned responses and they are taken from the typical
responses that are made when a particular situation occurs... :)




On Thu, Dec 20, 2012 at 6:06 PM, Michael Collins <msc at freeswitch.org> wrote:

> Brian, have you typed this one before? :D
>
> Juan, yes, definitely file a Jira on this. Also, collect the backtrace on
> that segfault if you can and attach as a txt file to the Jira case.
> -MC
>
>
> On Thu, Dec 20, 2012 at 2:59 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> *Dear OP,*
>> *
>> *
>> *We've detected that this is a potential bug in FreeSWITCH. *Because
>> this is a potential bug, the mailing list is really not the place to report
>> this.
>>
>> *Before you file a bug report, and if you haven't done this already, *please
>> make sure that the symptoms you are experiencing do not concur with another
>> previously submitted bug report. Search JIRA <http://jira.freeswitch.org> to
>> ensure this. You might want to check the Wiki just in case this is a
>> feature, not a bug.
>>
>> *You can make a difference. *Bugs can be reported to the FreeSWITCH JIRA<http://jira.freeswitch.org>.
>> By doing this one simple task, you can save the FreeSWITCH developers the
>> blood, sweat, and tears that are needed in order to gather information
>> related to your bug report and track the commits needed to fix the problem.
>>
>> *By filing a JIRA, *you also help the developers
>> gather relevant information so they can take the best course of action.
>> Developers ask for several pieces of information when filing a report. Take
>> a look at the Reporting Bugs<http://wiki.freeswitch.org/wiki/Reporting_Bugs>page for more details and instructions. The more information you can
>> gather, the easier it will be to fix the issue and the faster a commit can
>> be made.
>>
>> *Remember that FreeSWITCH is an open source project.* FreeSWITCH
>> Developers do not get paid to fix bugs, they simply do it for the sake of
>> the community. They sacrifice sleep, family time, brain cells, and Advil to
>> help YOU. Show them that you care. *Report bugs the correct way.*
>>
>>
>>
>> On Thu, Dec 20, 2012 at 5:41 PM, Juan Pablo L. <
>> jpablolorenzetti at hotmail.com> wrote:
>>
>>> Hi, i m trying to update my development server (VM) by making a clean
>>> installation from git master,
>>> but i m getting a core dump while compiling:
>>>
>>> /bin/sh: line 1:  9386 Segmentation fault      (core dumped)
>>> ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.h
>>> make[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139
>>> make[6]: *** [all] Error 2
>>> make[5]: *** [all-recursive] Error 1
>>> make[4]: *** [/home/fs/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
>>>
>>>
>>> i followed the classic steps (bootstrap,configure and make) is anyone
>>> experimenting the same issues ???
>>> my info is:
>>> Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011
>>> i686 Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux
>>>
>>> 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> 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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
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.
-------------- next part --------------
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From Sirish.MasurMohan at oa.com.au  Fri Dec 21 02:23:01 2012
From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan)
Date: Fri, 21 Dec 2012 10:23:01 +1100
Subject: [Freeswitch-users] Newbie question: Executing Lua scripts for
 incoming calls, how concurrency is to be handled?
In-Reply-To: <CAKzWOxUrdVNqqDJxQ8x4_OaE=oEOrGzdEb2D+XmvC-0tW_yfjA@mail.gmail.com>
References: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au>
	<50CE5D6E.2020401@quentustech.com>
	<965759A53E43FE439E43565A7715E5F058F4156D32@oa-exchange1.oa.com.au>
	<CAKzWOxWFh3RNYg9NY4Dqa-=f9KZOaXTvM8NGzHpvEvzV16VUWA@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156DD8@oa-exchange1.oa.com.au>
	<CAKzWOxUYyfZ_8dLgy_t-Ov2a+YXEDj04dDkdLSn14Q-tZk-8pQ@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156DF7@oa-exchange1.oa.com.au>
	<CAKzWOxUrdVNqqDJxQ8x4_OaE=oEOrGzdEb2D+XmvC-0tW_yfjA@mail.gmail.com>
Message-ID: <965759A53E43FE439E43565A7715E5F058F4156E6C@oa-exchange1.oa.com.au>

Hi Michael,

Thanks for the reply, and the suggestions.

My current implementation is a similar to what you have suggested, i.e.:

1.       I have defined 4 sets of global variables in vars.xml, where each set keeps a track of port's status and last used timestamp

2.       In the dialplan, I execute a simple Lua script which 'chooses' the line based on global variables status and timestamp (using "global_getvar") and update the status of the line to be used (using "global_setvar")

3.       Bridge to the line returned by the above script

4.       Execute another Lua script on hang-up, which updates the global variables (status and timestamp)

If I were to implement the hash way, I would have to probably do the following:

1.       In the dialplan, I execute a simple Lua script which 'chooses' the line by reading the hash values of the line status and timestamp and update the hash status of the line to be used

2.       Bridge to the line returned by the above script

3.       Execute another Lua script on hang-up, which updates the hash values (status and timestamp)

But I am still not clear on the concurrency question - if FreeSWITCH has received 2 calls at the same time on the E1 line, should I be worried about protecting the script that gets executed to access the hash, decide on the line to be used, and then update the hash?

Thanks again!

With regards,
Sirish

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, 21 December 2012 3:42 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

Sirish,

Since you're using an E1 to connect to the PBX then really all you need to do is keep track of the last time each port was hung up and whether or not a given port is currently in use. I would use api_on_answer to launch a simple script to set a flag to say that a particular port is in use and then use the api_hangup_hook to launch another script when the call ends.

The channel variables page on the wiki has some examples of how to use these. I recommend that you write simple Lua scripts that use the "hash" API to store information in the local database. Also, check out the "hash_dump" API as it is a useful way to quickly see what all is stored there.

For an example of how to add, remove, and read information from the local database using the "hash" API please see conf/dialplan/default.xml. Search for "hash" and you'll see all sorts of examples of how the example dialplan uses the local database to store useful information that allows us to implement features like call return, call intercept, etc.

-MC
On Tue, Dec 18, 2012 at 7:39 PM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au<mailto:Sirish.MasurMohan at oa.com.au>> wrote:
Hi Michael,

>> How are you physically connecting from FreeSWITCH to the PBX?

I connect this via E1 link - I have a Sangoma card installed on the FreeSWITCH machine.

With regards,
Sirish

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, 19 December 2012 1:31 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

To trigger SIP registrations you'd need the PBX to have a SIP client. I'm assuming this is possible, but maybe that's a false assumption. How are you physically connecting from FreeSWITCH to the PBX?

-MC
On Tue, Dec 18, 2012 at 2:36 PM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au<mailto:Sirish.MasurMohan at oa.com.au>> wrote:
Hey Guys,

Would really appreciate if you could help me out here - isn't there a way to handle concurrent calls in the dial plan, especially when Lua scripts, accessing global variables, are executed on receiving calls?

Is mod_fifo the closest I could get to handle concurrency (as Michael has explained)? If yes, how do I trigger SIP registrations, especially working with a PBX which I don't have full control of?

With regards,
Sirish

From: Sirish Masur Mohan
Sent: Tuesday, 18 December 2012 1:02 PM
To: FreeSWITCH Users Help
Subject: RE: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

Hi Michael,

Thanks for the reply.

>> You would need a SIP registration from the PBX to FreeSWITCH for each of the modem extensions
I am seen SIP clients such as X-Lite sending out the SIP registrations, but could you please clarify as to how this can be achieved in the PBX? The final production environment would be out in the customer's PBX, which I may not have complete control of..

>> What application are you building?
I may not be able to provide the details because of the NDA with customer, but what I am trying to achieve is, to replace an existing IVR with FreeSWITCH in an old existing setup of the customer - that's the reason why we continue working with dialup modems!

With regards,
Sirish

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: Tuesday, 18 December 2012 6:23 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

You don't have to have actual human agents for mod_fifo. You could define a user for each modem and then manually "log in" those "agents" on the command line using the fifo_member API command. Something like this:

fifo_member add fifo_name {fifo_member_wait=nowait}user/1234

Where 1234 is the user id of one of the modems. You would need a SIP registration from the PBX to FreeSWITCH for each of the modem extensions.

Having modems go through a VoIP system sounds a bit scary. What application are you building?
-MC
On Sun, Dec 16, 2012 at 5:41 PM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au<mailto:Sirish.MasurMohan at oa.com.au>> wrote:
Hi William,

Thanks for the reply.

My setup is as follows:
Client(Caller)->dialup modem->PBX->FreeSWITCH->PBX-> 4 dialup modems->Server(Receiver)

I basically need FreeSWITCH to bridge the incoming call to the best external destination (out of the 4 available), so that the modem training, connection etc can take place smoothly, before exchange of data. I am not sure if mod_fifo would help me in this scenario, as, I would require an agent to dial in and read the fifo. Could you please clarify?

Thanks!

With regards,
Sirish

-----Original Message-----
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 William King
Sent: Monday, 17 December 2012 10:47 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

Sounds like you want to take a look into mod_fifo.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337<tel:%28877%29%20211-9337>
Office: (206) 388-4772<tel:%28206%29%20388-4772>
Cell:   (253) 686-5518<tel:%28253%29%20686-5518>
william.king at quentustech.com<mailto:william.king at quentustech.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

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From bdfoster at endigotech.com  Fri Dec 21 02:22:54 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 20 Dec 2012 18:22:54 -0500
Subject: [Freeswitch-users] core dump when compiling master git
In-Reply-To: <CCF8F86A.62599%krice@freeswitch.org>
References: <CAKzWOxUufkOJkeR3j=6JX8xAw2F7Cze-=P=u0zuDPN_iJn_hAA@mail.gmail.com>
	<CCF8F86A.62599%krice@freeswitch.org>
Message-ID: <CAKV-jFwUH=S-fMhf_Dq+kqL2FwXTABgSDjRLhaA6ekFFQ4WaYQ@mail.gmail.com>

Technically, yes, SpanDSP but I thought Steve floats around JIRA?


On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice at freeswitch.org> wrote:

>  This is not a Segfault in FreeSWITCH you both fail....
>
> This is probably a build system issue....
>
> I would recommend pastebinning and a full trace from a clean tree of the
> bootstrap, configure and build process...
>
>
>
>
> On 12/20/12 5:06 PM, "Michael Collins" <msc at freeswitch.org> wrote:
>
> Brian, have you typed this one before? :D
>
> Juan, yes, definitely file a Jira on this. Also, collect the backtrace on
> that segfault if you can and attach as a txt file to the Jira case.
> -MC
>
> On Thu, Dec 20, 2012 at 2:59 PM, Brian Foster <bdfoster at endigotech.com>
> wrote:
>
> *Dear OP,
>
> We've detected that this is a potential bug in FreeSWITCH. *Because this
> is a potential bug, the mailing list is really not the place to report this.
>
> *Before you file a bug report, and if you haven't done this already, *please
> make sure that the symptoms you are experiencing do not concur with another
> previously submitted bug report. Search JIRA <http://jira.freeswitch.org>
>  to ensure this. You might want to check the Wiki just in case this is a
> feature, not a bug.
>
> *You can make a difference. *Bugs can be reported to the FreeSWITCH JIRA <
> http://jira.freeswitch.org> . By doing this one simple task, you can save
> the FreeSWITCH developers the blood, sweat, and tears that are needed in
> order to gather information related to your bug report and track the
> commits needed to fix the problem.
>
> *By filing a JIRA, *you also help the developers
> gather relevant information so they can take the best course of action.
> Developers ask for several pieces of information when filing a report. Take
> a look at the Reporting Bugs <
> http://wiki.freeswitch.org/wiki/Reporting_Bugs>  page for more details
> and instructions. The more information you can gather, the easier it will
> be to fix the issue and the faster a commit can be made.
>
>
> *Remember that FreeSWITCH is an open source project.* FreeSWITCH
> Developers do not get paid to fix bugs, they simply do it for the sake of
> the community. They sacrifice sleep, family time, brain cells, and Advil to
> help YOU. Show them that you care. *Report bugs the correct way.
> *
>
>
> On Thu, Dec 20, 2012 at 5:41 PM, Juan Pablo L. <
> jpablolorenzetti at hotmail.com> wrote:
>
> Hi, i m trying to update my development server (VM) by making a clean
> installation from git master,
> but i m getting a core dump while compiling:
>
> /bin/sh: line 1:  9386 Segmentation fault      (core dumped)
> ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.h
> make[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139
> make[6]: *** [all] Error 2
> make[5]: *** [all-recursive] Error 1
> make[4]: *** [/home/fs/freeswitch/libs/spandsp/src/libspandsp.la <
> http://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
>
>
> i followed the classic steps (bootstrap,configure and make) is anyone
> experimenting the same issues ???
> my info is:
> Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011 i686
> Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Ken
>
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *
> irc.freenode.net #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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
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
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From steveayre at gmail.com  Fri Dec 21 02:44:09 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 20 Dec 2012 23:44:09 +0000
Subject: [Freeswitch-users] core dump when compiling master git
In-Reply-To: <CAKV-jFwUH=S-fMhf_Dq+kqL2FwXTABgSDjRLhaA6ekFFQ4WaYQ@mail.gmail.com>
References: <CAKzWOxUufkOJkeR3j=6JX8xAw2F7Cze-=P=u0zuDPN_iJn_hAA@mail.gmail.com>
	<CCF8F86A.62599%krice@freeswitch.org>
	<CAKV-jFwUH=S-fMhf_Dq+kqL2FwXTABgSDjRLhaA6ekFFQ4WaYQ@mail.gmail.com>
Message-ID: <CAFiqYumG5urgZHmQ5HM3axWyUkgi72kvFYwwoX+2Ra1gvp92fg@mail.gmail.com>

Searching Jira turns up a similar segfault in the same program
(make_modem_filter).
http://jira.freeswitch.org/browse/FS-3787

It's tagged as Fixed, although I can't find the jira number in the commit
logs on fisheye.

Not sure then if it was resolved, or just never tracked down. It could be
something related to the system you're building on rather than the source,
or it could be a bug in that program that only presents in certain
situations.

Juan, can you fill us in on what FS branch/version you're using, and what
the host system is? And out of curiosity, is it 32bit like in that old jira?

I'd file a new Jira on it, and reference FS-3787. Gives somewhere to track
the issue, and it costs nothing to close it if it's found to be a build
system tools not FS issue.

-Steve




On 20 December 2012 23:22, Brian Foster <bdfoster at endigotech.com> wrote:

> Technically, yes, SpanDSP but I thought Steve floats around JIRA?
>
>
> On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>>  This is not a Segfault in FreeSWITCH you both fail....
>>
>> This is probably a build system issue....
>>
>> I would recommend pastebinning and a full trace from a clean tree of the
>> bootstrap, configure and build process...
>>
>>
>>
>>
>> On 12/20/12 5:06 PM, "Michael Collins" <msc at freeswitch.org> wrote:
>>
>> Brian, have you typed this one before? :D
>>
>> Juan, yes, definitely file a Jira on this. Also, collect the backtrace on
>> that segfault if you can and attach as a txt file to the Jira case.
>> -MC
>>
>> On Thu, Dec 20, 2012 at 2:59 PM, Brian Foster <bdfoster at endigotech.com>
>> wrote:
>>
>> *Dear OP,
>>
>> We've detected that this is a potential bug in FreeSWITCH. *Because this
>> is a potential bug, the mailing list is really not the place to report this.
>>
>> *Before you file a bug report, and if you haven't done this already, *please
>> make sure that the symptoms you are experiencing do not concur with another
>> previously submitted bug report. Search JIRA <http://jira.freeswitch.org>
>>  to ensure this. You might want to check the Wiki just in case this is a
>> feature, not a bug.
>>
>> *You can make a difference. *Bugs can be reported to the FreeSWITCH JIRA
>> <http://jira.freeswitch.org> . By doing this one simple task, you can
>> save the FreeSWITCH developers the blood, sweat, and tears that are needed
>> in order to gather information related to your bug report and track the
>> commits needed to fix the problem.
>>
>> *By filing a JIRA, *you also help the developers
>> gather relevant information so they can take the best course of action.
>> Developers ask for several pieces of information when filing a report. Take
>> a look at the Reporting Bugs <
>> http://wiki.freeswitch.org/wiki/Reporting_Bugs>  page for more details
>> and instructions. The more information you can gather, the easier it will
>> be to fix the issue and the faster a commit can be made.
>>
>>
>> *Remember that FreeSWITCH is an open source project.* FreeSWITCH
>> Developers do not get paid to fix bugs, they simply do it for the sake of
>> the community. They sacrifice sleep, family time, brain cells, and Advil to
>> help YOU. Show them that you care. *Report bugs the correct way.
>> *
>>
>>
>> On Thu, Dec 20, 2012 at 5:41 PM, Juan Pablo L. <
>> jpablolorenzetti at hotmail.com> wrote:
>>
>> Hi, i m trying to update my development server (VM) by making a clean
>> installation from git master,
>> but i m getting a core dump while compiling:
>>
>> /bin/sh: line 1:  9386 Segmentation fault      (core dumped)
>> ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.h
>> make[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139
>> make[6]: *** [all] Error 2
>> make[5]: *** [all-recursive] Error 1
>> make[4]: *** [/home/fs/freeswitch/libs/spandsp/src/libspandsp.la <
>> http://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
>>
>>
>> i followed the classic steps (bootstrap,configure and make) is anyone
>> experimenting the same issues ???
>> my info is:
>> Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011 i686
>> Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> Ken
>>
>> *http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> *
>> irc.freenode.net #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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> 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 krice at freeswitch.org  Fri Dec 21 03:16:11 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 20 Dec 2012 18:16:11 -0600
Subject: [Freeswitch-users] core dump when compiling master git
In-Reply-To: <CAKV-jFwUH=S-fMhf_Dq+kqL2FwXTABgSDjRLhaA6ekFFQ4WaYQ@mail.gmail.com>
Message-ID: <CCF9066B.625B5%krice@freeswitch.org>

No its not even in spandsp... This happens when he is building spandsp...
This means something like gcc is segfaulting...


On 12/20/12 5:22 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> Technically, yes, SpanDSP but I thought Steve floats around JIRA?
> 
> 
> On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice at freeswitch.org> wrote:
>> This is not a Segfault in FreeSWITCH you both fail....
>> 
>> This is probably a build system issue....
>> 
>> I would recommend pastebinning and a full trace from a clean tree of the
>> bootstrap, configure and build process...
>> 
>> 
>> 
>> 
>> On 12/20/12 5:06 PM, "Michael Collins" <msc at freeswitch.org
>> <http://msc at freeswitch.org> > wrote:
>> 
>>> Brian, have you typed this one before? :D
>>> 
>>> Juan, yes, definitely file a Jira on this. Also, collect the backtrace on
>>> that segfault if you can and attach as a txt file to the Jira case.
>>> -MC
>>> 
>>> On Thu, Dec 20, 2012 at 2:59 PM, Brian Foster <bdfoster at endigotech.com
>>> <http://bdfoster at endigotech.com> > wrote:
>>>> Dear OP,
>>>> 
>>>> We've detected that this is a potential bug in FreeSWITCH. Because this is
>>>> a potential bug, the mailing list is really not the place to report this.
>>>> 
>>>> Before you file a bug report, and if you haven't done this already, please
>>>> make sure that the symptoms you are experiencing do not concur with another
>>>> previously submitted bug report. Search JIRA <http://jira.freeswitch.org>
>>>> ?to ensure this. You might want to check the Wiki just in case this is a
>>>> feature, not a bug.
>>>> 
>>>> You can make a difference. Bugs can be reported to the FreeSWITCH JIRA
>>>> <http://jira.freeswitch.org> . By doing this one simple task, you can save
>>>> the FreeSWITCH developers the blood, sweat, and tears that are needed in
>>>> order to gather information related to your bug report and track the
>>>> commits needed to fix the problem.
>>>> 
>>>> By filing a JIRA, you also help the developers gather?relevant?information
>>>> so they can take the best course of action. Developers ask for several
>>>> pieces of information when filing a report. Take a look at the Reporting
>>>> Bugs <http://wiki.freeswitch.org/wiki/Reporting_Bugs> ?page for more
>>>> details and instructions. The more information you can gather, the easier
>>>> it will be to fix the issue and the faster a commit can be made.
>>>> 
>>>> 
>>>> Remember that FreeSWITCH is an open source project.?FreeSWITCH Developers
>>>> do not get paid to fix bugs, they simply do it for the sake of the
>>>> community. They sacrifice sleep, family time, brain cells, and Advil to
>>>> help YOU. Show them that you care. Report bugs the correct way.
>>>> 
>>>> 
>>>> 
>>>> On Thu, Dec 20, 2012 at 5:41 PM, Juan Pablo L.
>>>> <jpablolorenzetti at hotmail.com <http://jpablolorenzetti at hotmail.com> >
>>>> wrote:
>>>>> Hi, i m trying to update my development server (VM) by making a clean
>>>>> installation from git master,?
>>>>> but i m getting a core dump while compiling:
>>>>> 
>>>>> /bin/sh: line 1: ?9386 Segmentation fault ? ? ?(core dumped)
>>>>> ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.h
>>>>> make[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139
>>>>> make[6]: *** [all] Error 2
>>>>> make[5]: *** [all-recursive] Error 1
>>>>> make[4]: *** [/home/fs/freeswitch/libs/spandsp/src/libspandsp.la
>>>>> <http://libspandsp.la>  <http://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
>>>>> 
>>>>> 
>>>>> i followed the?classic?steps (bootstrap,configure and make) is anyone
>>>>> experimenting the same issues ???
>>>>> my info is:
>>>>> Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011 i686
>>>>> Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux
>>>>> 
>>>>> thanks!
>>>>> ???????
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org <http://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://FreeSWITCH-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>> 
>>>> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From anthony.minessale at gmail.com  Fri Dec 21 03:34:21 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 20 Dec 2012 18:34:21 -0600
Subject: [Freeswitch-users] core dump when compiling master git
In-Reply-To: <CCF9066B.625B5%krice@freeswitch.org>
References: <CAKV-jFwUH=S-fMhf_Dq+kqL2FwXTABgSDjRLhaA6ekFFQ4WaYQ@mail.gmail.com>
	<CCF9066B.625B5%krice@freeswitch.org>
Message-ID: <CAKbxfG-G+PjaSbq039YKdfeko4dhnoOXBLpM91t_wC2uvyZxZA@mail.gmail.com>

It looks like a seg in the helper app that generates part of the code in
the spandsp build so I can see it being a real bug but really it may be a
bug in a compliler or something too since we don't all suffer from this
same problem.  It must be partially environmental.

+10 points for the form letter!!!!

w00t JIRA FTW!

Also we brought back "make sure" by popular demand that nukes the whole
tree and recompiles.




On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice at freeswitch.org> wrote:

>  No its not even in spandsp... This happens when he is building
> spandsp... This means something like gcc is segfaulting...
>
>
>
> On 12/20/12 5:22 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>
> Technically, yes, SpanDSP but I thought Steve floats around JIRA?
>
>
> On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> This is not a Segfault in FreeSWITCH you both fail....
>
> This is probably a build system issue....
>
> I would recommend pastebinning and a full trace from a clean tree of the
> bootstrap, configure and build process...
>
>
>
>
> On 12/20/12 5:06 PM, "Michael Collins" <msc at freeswitch.org <
> http://msc at freeswitch.org> > wrote:
>
> Brian, have you typed this one before? :D
>
> Juan, yes, definitely file a Jira on this. Also, collect the backtrace on
> that segfault if you can and attach as a txt file to the Jira case.
> -MC
>
> On Thu, Dec 20, 2012 at 2:59 PM, Brian Foster <bdfoster at endigotech.com <
> http://bdfoster at endigotech.com> > wrote:
>
> *Dear OP,
>
> We've detected that this is a potential bug in FreeSWITCH. *Because this
> is a potential bug, the mailing list is really not the place to report this.
>
> *Before you file a bug report, and if you haven't done this already, *please
> make sure that the symptoms you are experiencing do not concur with another
> previously submitted bug report. Search JIRA <http://jira.freeswitch.org>
>  to ensure this. You might want to check the Wiki just in case this is a
> feature, not a bug.
>
> *You can make a difference. *Bugs can be reported to the FreeSWITCH JIRA <
> http://jira.freeswitch.org> . By doing this one simple task, you can save
> the FreeSWITCH developers the blood, sweat, and tears that are needed in
> order to gather information related to your bug report and track the
> commits needed to fix the problem.
>
> *By filing a JIRA, *you also help the developers
> gather relevant information so they can take the best course of action.
> Developers ask for several pieces of information when filing a report. Take
> a look at the Reporting Bugs <
> http://wiki.freeswitch.org/wiki/Reporting_Bugs>  page for more details
> and instructions. The more information you can gather, the easier it will
> be to fix the issue and the faster a commit can be made.
>
>
> *Remember that FreeSWITCH is an open source project.* FreeSWITCH
> Developers do not get paid to fix bugs, they simply do it for the sake of
> the community. They sacrifice sleep, family time, brain cells, and Advil to
> help YOU. Show them that you care. *Report bugs the correct way.
> *
>
>
> On Thu, Dec 20, 2012 at 5:41 PM, Juan Pablo L. <
> jpablolorenzetti at hotmail.com <http://jpablolorenzetti at hotmail.com> >
> wrote:
>
> Hi, i m trying to update my development server (VM) by making a clean
> installation from git master,
> but i m getting a core dump while compiling:
>
> /bin/sh: line 1:  9386 Segmentation fault      (core dumped)
> ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.h
> make[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139
> make[6]: *** [all] Error 2
> make[5]: *** [all-recursive] Error 1
> make[4]: *** [/home/fs/freeswitch/libs/spandsp/src/libspandsp.la <
> http://libspandsp.la>  <http://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
>
>
> i followed the classic steps (bootstrap,configure and make) is anyone
> experimenting the same issues ???
> my info is:
> Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011 i686
> Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux
>
> thanks!
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://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://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
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From jpablolorenzetti at hotmail.com  Fri Dec 21 05:16:37 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Fri, 21 Dec 2012 02:16:37 +0000
Subject: [Freeswitch-users] core dump when compiling master git
In-Reply-To: <CAKbxfG-G+PjaSbq039YKdfeko4dhnoOXBLpM91t_wC2uvyZxZA@mail.gmail.com>
References: <CAKV-jFwUH=S-fMhf_Dq+kqL2FwXTABgSDjRLhaA6ekFFQ4WaYQ@mail.gmail.com>,
	<CCF9066B.625B5%krice@freeswitch.org>,
	<CAKbxfG-G+PjaSbq039YKdfeko4dhnoOXBLpM91t_wC2uvyZxZA@mail.gmail.com>
Message-ID: <DUB002-W94F9E2FA82F9273BC3F554D9360@phx.gbl>

Hi, i m having a little trouble installing gdb in my testing machine, i have a very slim machine which mirrors my production servers andto install gdb i need to upgrade my testing environment and that would make it different from my production servers ... is there any other way i couldget the backtrace of this issue ? maybe turning a flag while making the package ... ? 

Date: Thu, 20 Dec 2012 18:34:21 -0600
From: anthony.minessale at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] core dump when compiling master git

It looks like a seg in the helper app that generates part of the code in the spandsp build so I can see it being a real bug but really it may be a bug in a compliler or something too since we don't all suffer from this same problem.  It must be partially environmental.

+10 points for the form letter!!!!
w00t JIRA FTW!
Also we brought back "make sure" by popular demand that nukes the whole tree and recompiles.




On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice at freeswitch.org> wrote:






No its not even in spandsp... This happens when he is building spandsp... This means something like gcc is segfaulting...





On 12/20/12 5:22 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:



Technically, yes, SpanDSP but I thought Steve floats around JIRA?





On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice at freeswitch.org> wrote:

This is not a Segfault in FreeSWITCH you both fail....



This is probably a build system issue.... 



I would recommend pastebinning and a full trace from a clean tree of the bootstrap, configure and build process...









On 12/20/12 5:06 PM, "Michael Collins" <msc at freeswitch.org <http://msc at freeswitch.org> > wrote:




Brian, have you typed this one before? :D



Juan, yes, definitely file a Jira on this. Also, collect the backtrace on that segfault if you can and attach as a txt file to the Jira case.

-MC



On Thu, Dec 20, 2012 at 2:59 PM, Brian Foster <bdfoster at endigotech.com <http://bdfoster at endigotech.com> > wrote:


Dear OP,



We've detected that this is a potential bug in FreeSWITCH. Because this is a potential bug, the mailing list is really not the place to report this.



Before you file a bug report, and if you haven't done this already, please make sure that the symptoms you are experiencing do not concur with another previously submitted bug report. Search JIRA <http://jira.freeswitch.org>  to ensure this. You might want to check the Wiki just in case this is a feature, not a bug.




You can make a difference. Bugs can be reported to the FreeSWITCH JIRA <http://jira.freeswitch.org> . By doing this one simple task, you can save the FreeSWITCH developers the blood, sweat, and tears that are needed in order to gather information related to your bug report and track the commits needed to fix the problem.




By filing a JIRA, you also help the developers gather relevant information so they can take the best course of action. Developers ask for several pieces of information when filing a report. Take a look at the Reporting Bugs <http://wiki.freeswitch.org/wiki/Reporting_Bugs>  page for more details and instructions. The more information you can gather, the easier it will be to fix the issue and the faster a commit can be made.






Remember that FreeSWITCH is an open source project. FreeSWITCH Developers do not get paid to fix bugs, they simply do it for the sake of the community. They sacrifice sleep, family time, brain cells, and Advil to help YOU. Show them that you care. Report bugs the correct way.








On Thu, Dec 20, 2012 at 5:41 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com <http://jpablolorenzetti at hotmail.com> > wrote:


Hi, i m trying to update my development server (VM) by making a clean installation from git master, 

but i m getting a core dump while compiling:



/bin/sh: line 1:  9386 Segmentation fault      (core dumped) ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.h

make[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139

make[6]: *** [all] Error 2

make[5]: *** [all-recursive] Error 1

make[4]: *** [/home/fs/freeswitch/libs/spandsp/src/libspandsp.la <http://libspandsp.la>  <http://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





i followed the classic steps (bootstrap,configure and make) is anyone experimenting the same issues ???

my info is:

Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011 i686 Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux



thanks!

       



_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org <http://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://FreeSWITCH-users at lists.freeswitch.org> 


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

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

http://www.freeswitch.org









-- 

Ken

http://www.FreeSWITCH.org

http://www.ClueCon.com

http://www.OSTAG.org

irc.freenode.net #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



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

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

http://www.freeswitch.org




-- 
Anthony Minessale II

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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  Fri Dec 21 05:32:56 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 21 Dec 2012 02:32:56 +0000
Subject: [Freeswitch-users] Newbie question: Executing Lua scripts for
	incoming calls, how concurrency is to be handled?
In-Reply-To: <965759A53E43FE439E43565A7715E5F058F4156E6C@oa-exchange1.oa.com.au>
References: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au>
	<50CE5D6E.2020401@quentustech.com>
	<965759A53E43FE439E43565A7715E5F058F4156D32@oa-exchange1.oa.com.au>
	<CAKzWOxWFh3RNYg9NY4Dqa-=f9KZOaXTvM8NGzHpvEvzV16VUWA@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156DD8@oa-exchange1.oa.com.au>
	<CAKzWOxUYyfZ_8dLgy_t-Ov2a+YXEDj04dDkdLSn14Q-tZk-8pQ@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156DF7@oa-exchange1.oa.com.au>
	<CAKzWOxUrdVNqqDJxQ8x4_OaE=oEOrGzdEb2D+XmvC-0tW_yfjA@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156E6C@oa-exchange1.oa.com.au>
Message-ID: <849EC2AD-D5F4-471E-B362-480DF549DB62@gmail.com>

If I remember correctly when when anthm added the option by request, global_setvar is NOT safe for setting concurrently. Very occasionally should be ok for config changes without a restart, but it is not safe to use from call dial plans where calls are running alongside each other.

You should look at using mod_hash or mod_db, which are much safer

Sent from my iPad

On 20 Dec 2012, at 23:23, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au> wrote:

> Hi Michael,
>  
> Thanks for the reply, and the suggestions.
>  
> My current implementation is a similar to what you have suggested, i.e.:
> 1.       I have defined 4 sets of global variables in vars.xml, where each set keeps a track of port?s status and last used timestamp
> 2.       In the dialplan, I execute a simple Lua script which ?chooses? the line based on global variables status and timestamp (using "global_getvar") and update the status of the line to be used (using "global_setvar")
> 3.       Bridge to the line returned by the above script
> 4.       Execute another Lua script on hang-up, which updates the global variables (status and timestamp)
>  
> If I were to implement the hash way, I would have to probably do the following:
> 1.       In the dialplan, I execute a simple Lua script which ?chooses? the line by reading the hash values of the line status and timestamp and update the hash status of the line to be used
> 2.       Bridge to the line returned by the above script
> 3.       Execute another Lua script on hang-up, which updates the hash values (status and timestamp)
>  
> But I am still not clear on the concurrency question ? if FreeSWITCH has received 2 calls at the same time on the E1 line, should I be worried about protecting the script that gets executed to access the hash, decide on the line to be used, and then update the hash?
>  
> Thanks again!
>  
> With regards,
> Sirish
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
> Sent: Friday, 21 December 2012 3:42 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?
>  
> Sirish,
> 
> Since you're using an E1 to connect to the PBX then really all you need to do is keep track of the last time each port was hung up and whether or not a given port is currently in use. I would use api_on_answer to launch a simple script to set a flag to say that a particular port is in use and then use the api_hangup_hook to launch another script when the call ends.
> 
> The channel variables page on the wiki has some examples of how to use these. I recommend that you write simple Lua scripts that use the "hash" API to store information in the local database. Also, check out the "hash_dump" API as it is a useful way to quickly see what all is stored there.
> 
> For an example of how to add, remove, and read information from the local database using the "hash" API please see conf/dialplan/default.xml. Search for "hash" and you'll see all sorts of examples of how the example dialplan uses the local database to store useful information that allows us to implement features like call return, call intercept, etc.
> 
> -MC
> 
> On Tue, Dec 18, 2012 at 7:39 PM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au> wrote:
> Hi Michael,
>  
> >> How are you physically connecting from FreeSWITCH to the PBX?
>  
> I connect this via E1 link ? I have a Sangoma card installed on the FreeSWITCH machine.
>  
> With regards,
> Sirish
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
> Sent: Wednesday, 19 December 2012 1:31 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?
>  
> To trigger SIP registrations you'd need the PBX to have a SIP client. I'm assuming this is possible, but maybe that's a false assumption. How are you physically connecting from FreeSWITCH to the PBX?
> 
> -MC
> 
> On Tue, Dec 18, 2012 at 2:36 PM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au> wrote:
> Hey Guys,
>  
> Would really appreciate if you could help me out here ? isn?t there a way to handle concurrent calls in the dial plan, especially when Lua scripts, accessing global variables, are executed on receiving calls?
>  
> Is mod_fifo the closest I could get to handle concurrency (as Michael has explained)? If yes, how do I trigger SIP registrations, especially working with a PBX which I don?t have full control of?
>  
> With regards,
> Sirish
>  
> From: Sirish Masur Mohan 
> Sent: Tuesday, 18 December 2012 1:02 PM
> To: FreeSWITCH Users Help
> Subject: RE: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?
>  
> Hi Michael,
>  
> Thanks for the reply.
>  
> >> You would need a SIP registration from the PBX to FreeSWITCH for each of the modem extensions
> I am seen SIP clients such as X-Lite sending out the SIP registrations, but could you please clarify as to how this can be achieved in the PBX? The final production environment would be out in the customer?s PBX, which I may not have complete control of..
>  
> >> What application are you building?
> I may not be able to provide the details because of the NDA with customer, but what I am trying to achieve is, to replace an existing IVR with FreeSWITCH in an old existing setup of the customer ? that?s the reason why we continue working with dialup modems!
>  
> With regards,
> Sirish
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
> Sent: Tuesday, 18 December 2012 6:23 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?
>  
> You don't have to have actual human agents for mod_fifo. You could define a user for each modem and then manually "log in" those "agents" on the command line using the fifo_member API command. Something like this:
> 
> fifo_member add fifo_name {fifo_member_wait=nowait}user/1234
> 
> Where 1234 is the user id of one of the modems. You would need a SIP registration from the PBX to FreeSWITCH for each of the modem extensions. 
> 
> Having modems go through a VoIP system sounds a bit scary. What application are you building?
> -MC
> 
> On Sun, Dec 16, 2012 at 5:41 PM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au> wrote:
> Hi William,
> 
> Thanks for the reply.
> 
> My setup is as follows:
> Client(Caller)->dialup modem->PBX->FreeSWITCH->PBX-> 4 dialup modems->Server(Receiver)
> 
> I basically need FreeSWITCH to bridge the incoming call to the best external destination (out of the 4 available), so that the modem training, connection etc can take place smoothly, before exchange of data. I am not sure if mod_fifo would help me in this scenario, as, I would require an agent to dial in and read the fifo. Could you please clarify?
> 
> Thanks!
> 
> With regards,
> Sirish
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William King
> Sent: Monday, 17 December 2012 10:47 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?
> 
> Sounds like you want to take a look into mod_fifo.
> 
> 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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 Sirish.MasurMohan at oa.com.au  Fri Dec 21 07:44:58 2012
From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan)
Date: Fri, 21 Dec 2012 15:44:58 +1100
Subject: [Freeswitch-users] Newbie question: Executing Lua scripts
	for	incoming calls, how concurrency is to be handled?
In-Reply-To: <849EC2AD-D5F4-471E-B362-480DF549DB62@gmail.com>
References: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au>
	<50CE5D6E.2020401@quentustech.com>
	<965759A53E43FE439E43565A7715E5F058F4156D32@oa-exchange1.oa.com.au>
	<CAKzWOxWFh3RNYg9NY4Dqa-=f9KZOaXTvM8NGzHpvEvzV16VUWA@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156DD8@oa-exchange1.oa.com.au>
	<CAKzWOxUYyfZ_8dLgy_t-Ov2a+YXEDj04dDkdLSn14Q-tZk-8pQ@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156DF7@oa-exchange1.oa.com.au>
	<CAKzWOxUrdVNqqDJxQ8x4_OaE=oEOrGzdEb2D+XmvC-0tW_yfjA@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156E6C@oa-exchange1.oa.com.au>
	<849EC2AD-D5F4-471E-B362-480DF549DB62@gmail.com>
Message-ID: <965759A53E43FE439E43565A7715E5F058F4156E94@oa-exchange1.oa.com.au>

Thanks Steven, noted!

With regards,
Sirish

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Friday, 21 December 2012 1:33 PM
To: FreeSWITCH Users Help
Cc: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

If I remember correctly when when anthm added the option by request, global_setvar is NOT safe for setting concurrently. Very occasionally should be ok for config changes without a restart, but it is not safe to use from call dial plans where calls are running alongside each other.

You should look at using mod_hash or mod_db, which are much safer

Sent from my iPad

On 20 Dec 2012, at 23:23, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au<mailto:Sirish.MasurMohan at oa.com.au>> wrote:
Hi Michael,

Thanks for the reply, and the suggestions.

My current implementation is a similar to what you have suggested, i.e.:

1.      I have defined 4 sets of global variables in vars.xml, where each set keeps a track of port?s status and last used timestamp

2.      In the dialplan, I execute a simple Lua script which ?chooses? the line based on global variables status and timestamp (using "global_getvar") and update the status of the line to be used (using "global_setvar")

3.      Bridge to the line returned by the above script

4.      Execute another Lua script on hang-up, which updates the global variables (status and timestamp)

If I were to implement the hash way, I would have to probably do the following:

1.      In the dialplan, I execute a simple Lua script which ?chooses? the line by reading the hash values of the line status and timestamp and update the hash status of the line to be used

2.      Bridge to the line returned by the above script

3.      Execute another Lua script on hang-up, which updates the hash values (status and timestamp)

But I am still not clear on the concurrency question ? if FreeSWITCH has received 2 calls at the same time on the E1 line, should I be worried about protecting the script that gets executed to access the hash, decide on the line to be used, and then update the hash?

Thanks again!

With regards,
Sirish

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: Friday, 21 December 2012 3:42 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

Sirish,

Since you're using an E1 to connect to the PBX then really all you need to do is keep track of the last time each port was hung up and whether or not a given port is currently in use. I would use api_on_answer to launch a simple script to set a flag to say that a particular port is in use and then use the api_hangup_hook to launch another script when the call ends.

The channel variables page on the wiki has some examples of how to use these. I recommend that you write simple Lua scripts that use the "hash" API to store information in the local database. Also, check out the "hash_dump" API as it is a useful way to quickly see what all is stored there.

For an example of how to add, remove, and read information from the local database using the "hash" API please see conf/dialplan/default.xml. Search for "hash" and you'll see all sorts of examples of how the example dialplan uses the local database to store useful information that allows us to implement features like call return, call intercept, etc.

-MC
On Tue, Dec 18, 2012 at 7:39 PM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au<mailto:Sirish.MasurMohan at oa.com.au>> wrote:
Hi Michael,

>> How are you physically connecting from FreeSWITCH to the PBX?

I connect this via E1 link ? I have a Sangoma card installed on the FreeSWITCH machine.

With regards,
Sirish

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, 19 December 2012 1:31 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

To trigger SIP registrations you'd need the PBX to have a SIP client. I'm assuming this is possible, but maybe that's a false assumption. How are you physically connecting from FreeSWITCH to the PBX?

-MC
On Tue, Dec 18, 2012 at 2:36 PM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au<mailto:Sirish.MasurMohan at oa.com.au>> wrote:
Hey Guys,

Would really appreciate if you could help me out here ? isn?t there a way to handle concurrent calls in the dial plan, especially when Lua scripts, accessing global variables, are executed on receiving calls?

Is mod_fifo the closest I could get to handle concurrency (as Michael has explained)? If yes, how do I trigger SIP registrations, especially working with a PBX which I don?t have full control of?

With regards,
Sirish

From: Sirish Masur Mohan
Sent: Tuesday, 18 December 2012 1:02 PM
To: FreeSWITCH Users Help
Subject: RE: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

Hi Michael,

Thanks for the reply.

>> You would need a SIP registration from the PBX to FreeSWITCH for each of the modem extensions
I am seen SIP clients such as X-Lite sending out the SIP registrations, but could you please clarify as to how this can be achieved in the PBX? The final production environment would be out in the customer?s PBX, which I may not have complete control of..

>> What application are you building?
I may not be able to provide the details because of the NDA with customer, but what I am trying to achieve is, to replace an existing IVR with FreeSWITCH in an old existing setup of the customer ? that?s the reason why we continue working with dialup modems!

With regards,
Sirish

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: Tuesday, 18 December 2012 6:23 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

You don't have to have actual human agents for mod_fifo. You could define a user for each modem and then manually "log in" those "agents" on the command line using the fifo_member API command. Something like this:

fifo_member add fifo_name {fifo_member_wait=nowait}user/1234

Where 1234 is the user id of one of the modems. You would need a SIP registration from the PBX to FreeSWITCH for each of the modem extensions.

Having modems go through a VoIP system sounds a bit scary. What application are you building?
-MC
On Sun, Dec 16, 2012 at 5:41 PM, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au<mailto:Sirish.MasurMohan at oa.com.au>> wrote:
Hi William,

Thanks for the reply.

My setup is as follows:
Client(Caller)->dialup modem->PBX->FreeSWITCH->PBX-> 4 dialup modems->Server(Receiver)

I basically need FreeSWITCH to bridge the incoming call to the best external destination (out of the 4 available), so that the modem training, connection etc can take place smoothly, before exchange of data. I am not sure if mod_fifo would help me in this scenario, as, I would require an agent to dial in and read the fifo. Could you please clarify?

Thanks!

With regards,
Sirish

-----Original Message-----
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 William King
Sent: Monday, 17 December 2012 10:47 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled?

Sounds like you want to take a look into mod_fifo.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337<tel:%28877%29%20211-9337>
Office: (206) 388-4772<tel:%28206%29%20388-4772>
Cell:   (253) 686-5518<tel:%28253%29%20686-5518>
william.king at quentustech.com<mailto:william.king at quentustech.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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

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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.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

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://lists.freeswitch.org/mailman/options/freeswitch-users>
http://www.freeswitch.org
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From Alexander.Haugg at c4b.de  Fri Dec 21 08:57:06 2012
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Fri, 21 Dec 2012 05:57:06 +0000
Subject: [Freeswitch-users] build error with new Freeswitch 'nametab.h':
 No such file or directory
In-Reply-To: <1356015615778-7585759.post@n2.nabble.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC837D409@vc4bex2.corp.c4b.de>
	<1355844703594-7585673.post@n2.nabble.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC837D614@vc4bex2.corp.c4b.de>
	<1355935704180-7585726.post@n2.nabble.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC837D838@vc4bex2.corp.c4b.de>
	<1356015615778-7585759.post@n2.nabble.com>
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC838468D@vc4bex2.corp.c4b.de>

Thanks for your help!

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jeff Lenk
Gesendet: Donnerstag, 20. Dezember 2012 16:00
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] build error with new Freeswitch 'nametab.h': No such file or directory

Ok I made a correction to the solution file that should correct the dependency problem. Also the project policy is to open Jiras (http://jira.freeswitch.org)for possible problems. Thanks for bringing this issue attention.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/build-error-with-new-Freeswitch-nametab-h-No-such-file-or-directory-tp7585669p7585759.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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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 Dec 21 09:20:43 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 20 Dec 2012 22:20:43 -0800
Subject: [Freeswitch-users] build error with new Freeswitch 'nametab.h':
 No such file or directory
In-Reply-To: <B60DA91F5EA0E24A96E73CDE43CC9EC838468D@vc4bex2.corp.c4b.de>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC837D409@vc4bex2.corp.c4b.de>
	<1355844703594-7585673.post@n2.nabble.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC837D614@vc4bex2.corp.c4b.de>
	<1355935704180-7585726.post@n2.nabble.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC837D838@vc4bex2.corp.c4b.de>
	<1356015615778-7585759.post@n2.nabble.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC838468D@vc4bex2.corp.c4b.de>
Message-ID: <CAMWiMy_vS4XGHzXC63pk0RcktGJW+7JWugyQddmowTeG_pGjjQ@mail.gmail.com>

There were talks about putting VS2008 into an unsupported state
because of what Microsoft did to it a few weeks back. Then again, you
could build FreeSWITCH under VC2010 Express and if you want 64-bit
support, you can download the Windows SDK for Win 7 and use that
64-bit compiler to compile FreeSWITCH.

On Thu, Dec 20, 2012 at 9:57 PM, Alexander Haugg <Alexander.Haugg at c4b.de> wrote:
> Thanks for your help!
>
> -----Urspr?ngliche Nachricht-----
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jeff Lenk
> Gesendet: Donnerstag, 20. Dezember 2012 16:00
> An: freeswitch-users at lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] build error with new Freeswitch 'nametab.h': No such file or directory
>
> Ok I made a correction to the solution file that should correct the dependency problem. Also the project policy is to open Jiras (http://jira.freeswitch.org)for possible problems. Thanks for bringing this issue attention.
>
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/build-error-with-new-Freeswitch-nametab-h-No-such-file-or-directory-tp7585669p7585759.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org


From peter.olsson at visionutveckling.se  Fri Dec 21 10:47:46 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 21 Dec 2012 07:47:46 +0000
Subject: [Freeswitch-users] core dump when compiling master git
Message-ID: <1FFF97C269757C458224B7C895F35F151F113D@cantor.std.visionutv.se>

One common issue is that you've set "ulimit -s 240" to start FS, and then it's active in the running shell. Try to use login into new session and see if it helps.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 21 december 2012 01:34
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] core dump when compiling master git

It looks like a seg in the helper app that generates part of the code in the spandsp build so I can see it being a real bug but really it may be a bug in a compliler or something too since we don't all suffer from this same problem.  It must be partially environmental.

+10 points for the form letter!!!!

w00t JIRA FTW!

Also we brought back "make sure" by popular demand that nukes the whole tree and recompiles.



On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice at freeswitch.org<mailto:krice at freeswitch.org>> wrote:
No its not even in spandsp... This happens when he is building spandsp... This means something like gcc is segfaulting...



On 12/20/12 5:22 PM, "Brian Foster" <bdfoster at endigotech.com<http://bdfoster at endigotech.com>> wrote:
Technically, yes, SpanDSP but I thought Steve floats around JIRA?


On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice at freeswitch.org<http://krice at freeswitch.org>> wrote:
This is not a Segfault in FreeSWITCH you both fail....

This is probably a build system issue....

I would recommend pastebinning and a full trace from a clean tree of the bootstrap, configure and build process...



On 12/20/12 5:06 PM, "Michael Collins" <msc at freeswitch.org<http://msc at freeswitch.org> <http://msc at freeswitch.org> > wrote:
Brian, have you typed this one before? :D

Juan, yes, definitely file a Jira on this. Also, collect the backtrace on that segfault if you can and attach as a txt file to the Jira case.
-MC
On Thu, Dec 20, 2012 at 2:59 PM, Brian Foster <bdfoster at endigotech.com<http://bdfoster at endigotech.com> <http://bdfoster at endigotech.com> > wrote:
Dear OP,

We've detected that this is a potential bug in FreeSWITCH. Because this is a potential bug, the mailing list is really not the place to report this.

Before you file a bug report, and if you haven't done this already, please make sure that the symptoms you are experiencing do not concur with another previously submitted bug report. Search JIRA <http://jira.freeswitch.org>  to ensure this. You might want to check the Wiki just in case this is a feature, not a bug.

You can make a difference. Bugs can be reported to the FreeSWITCH JIRA <http://jira.freeswitch.org> . By doing this one simple task, you can save the FreeSWITCH developers the blood, sweat, and tears that are needed in order to gather information related to your bug report and track the commits needed to fix the problem.

By filing a JIRA, you also help the developers gather relevant information so they can take the best course of action. Developers ask for several pieces of information when filing a report. Take a look at the Reporting Bugs <http://wiki.freeswitch.org/wiki/Reporting_Bugs>  page for more details and instructions. The more information you can gather, the easier it will be to fix the issue and the faster a commit can be made.


Remember that FreeSWITCH is an open source project. FreeSWITCH Developers do not get paid to fix bugs, they simply do it for the sake of the community. They sacrifice sleep, family time, brain cells, and Advil to help YOU. Show them that you care. Report bugs the correct way.


On Thu, Dec 20, 2012 at 5:41 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com<http://jpablolorenzetti at hotmail.com> <http://jpablolorenzetti at hotmail.com> > wrote:
Hi, i m trying to update my development server (VM) by making a clean installation from git master,
but i m getting a core dump while compiling:

/bin/sh: line 1:  9386 Segmentation fault      (core dumped) ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.h
make[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139
make[6]: *** [all] Error 2
make[5]: *** [all-recursive] Error 1
make[4]: *** [/home/fs/freeswitch/libs/spandsp/src/libspandsp.la<http://libspandsp.la> <http://libspandsp.la>  <http://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


i followed the classic steps (bootstrap,configure and make) is anyone experimenting the same issues ???
my info is:
Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011 i686 Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux

thanks!


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<http://consulting at freeswitch.org> <http://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://FreeSWITCH-users at lists.freeswitch.org> <http://FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


--
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net<http://irc.freenode.net> #freeswitch

_________________________________________________________________________
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



--
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
!DSPAM:50d3ac3432766009014104!
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From miha at softnet.si  Fri Dec 21 11:46:47 2012
From: miha at softnet.si (Miha)
Date: Fri, 21 Dec 2012 09:46:47 +0100
Subject: [Freeswitch-users] Tracing
In-Reply-To: <CAKzWOxWmLF05rzRCVjCCaQxqX2v6hahdve+PpuLtdts0C--xeg@mail.gmail.com>
References: <50D2C1E5.10708@softnet.si>
	<CAKzWOxWmLF05rzRCVjCCaQxqX2v6hahdve+PpuLtdts0C--xeg@mail.gmail.com>
Message-ID: <50D421F7.3080601@softnet.si>

Michael, thanks!

I installed pcapsipdump. In long term I thing that cpu load goes in piks 
much higher. I will do port mirroring on switch in trace this on 
different server:)

Thanks!

Dne 12/20/2012 10:01 PM, pis(e Michael Collins:
> Your best bet would be to use pcapsipdump and manually or via script 
> clear out the old pcap files each day/week/month/whatever.
>
> Also, make sure that you are rotating your FreeSWITCH log files on a 
> regular basis. Also, be sure to enable uuid logging in your 
> logfile.conf.xml file. Having the uuid of the call in the fs log file 
> is really helpful on a busy system.
>
> -MC
>
> On Wed, Dec 19, 2012 at 11:44 PM, Miha <miha at softnet.si 
> <mailto:miha at softnet.si>> wrote:
>
>     Hi,
>
>     I am experiencing some problems related with media with some users. It
>     is almost impossible to figure it out where is a problem because this
>     problem is not presented all day.
>     I need some sip trace dump for a specific user if it is possible
>     due to
>     a lot of traffic. It would be nice that this would not cause to
>     much CPU
>     load and disk consumption as this would need to be running and
>     tracing 24/7.
>
>     Users are reporting the thy exeriacing noise in conversation and
>     sometime thy do not hear other side.
>
>     What would be the best way?
>
>     Thanks!
>
>     Miha
>
>     _________________________________________________________________________
>     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
>
>
>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 asilva at wirelessmundi.com  Fri Dec 21 12:35:10 2012
From: asilva at wirelessmundi.com (Antonio)
Date: Fri, 21 Dec 2012 10:35:10 +0100
Subject: [Freeswitch-users] Freeswitch TLS and Yealink t26p
Message-ID: <1356082510.23186.154.camel@marces.madrid.commsmundi.com>

Hi,

I'm trying to register a yealink with TLS, using my one certificates.

I follow the wiki and In fs i have both agent.pem and cafile.pem . I
install in the phone the root certificate.

But when i try to register, i have (tport log):


tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x808fb0) msg
0x7fe9d0aa8180 from (udp/192.168.10.1:5060) has 340 bytes, veclen = 1
tport.c:3004 tport_deliver() tport_deliver(0x808fb0): msg 0x7fe9d0aa8180
(340 bytes) from udp/192.168.10.23:5060/sip next=(nil)
tport.c:4202 tport_release() tport_release(0x808fb0): 0x7fe9d01142f0 by
0x7fe9d025d920 with 0x7fe9d0aa8180
tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fe9c802aad0): events
IN
tport.c:869 tport_alloc_secondary()
tport_alloc_secondary(0x7fe9c802aad0): new secondary tport
0x7fe9c03e8450
tport_type_tls.c:603 tport_tls_accept()
tport_tls_accept(0x7fe9c03e8450): new connection from
tls/192.168.10.36:48754/sips
tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events
NEGOTIATING
tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events
NEGOTIATING
tport_tls.c:526 tls_post_connection_check()
tls_post_connection_check(0x7fe9c03e8450): Peer did not provide X.509
Certificate.



I could make it work and have a register in the tls profile when i check
on the phone the option in Security->Trusted Certificates: "Only Accept
Trusted Certificates: DISABLED".
Could it be some bug in the yealink, or I?m missing something in the
conf...

Another question, is there any problem if i choose to use this
configuration... since is the phone that ignores the certificate and the
validation is done by the server and not by the client. 

Can you help me?

Thanks,
Ant?nio
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From steveu at coppice.org  Fri Dec 21 12:52:25 2012
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 21 Dec 2012 17:52:25 +0800
Subject: [Freeswitch-users] t.38 Call Flow Information
In-Reply-To: <AAD54836-F121-4190-A473-B64D9A9E4612@5ninesolutions.com>
References: <AAD54836-F121-4190-A473-B64D9A9E4612@5ninesolutions.com>
Message-ID: <50D43159.2050108@coppice.org>

Hi,

It looks like your reinvite occurred extremely late, and the FAX was 
already deep in progress as an audio exchange.

Steve


On 12/21/2012 03:14 AM, Spencer Thomason wrote:
> Hello,
> I've been analyzing some Wireshark traces to get a better grasp on the t.38 gateway process.  I'm using t38_gateway to convert SIP t38 to TDM audio on the receiving end.  I'm struggling to understand the following call flow.  I've excluded the audio portion for brevity but t38 is negotiated correctly by both endpoints (FS on our end and Acme on the other).  Note the absence of a DIS message and the PPS prior to CFR.  Can anyone shed any light on this?
>
> Thanks in advance,
> Spencer
>
>
> Acme                                                                                FreeSWITCH
> |1716.691635|         INVITE SDP (t38)               |                   |SIP Request
> |         |(5060)   <------------------  (5070)   |                   |
> |1716.719556|         100 trying -- your call is important to us          |                   |SIP Status
> |         |(5060)   ------------------>  (5070)   |                   |
> |1716.830431|                   |         no-signal |                   |t38:t30 Ind:no-signal
> |         |                   |(21440)  <------------------  (15876)  |
> |1716.906132|         200 OK SDP (t38)              |                   |SIP Status
> |         |(5060)   ------------------>  (5070)   |                   |
> |1716.912839|         ACK       |                   |                   |SIP Request
> |         |(5060)   <------------------  (5070)   |                   |
> |1716.973011|                   |         no-signal |                   |t38:t30 Ind:no-signal
> |         |                   |(21440)  ------------------>  (15876)  |
> |1717.450595|                   |         v21-preamble                  |t38:t30 Ind:v21-preamble
> |         |                   |(21440)  <------------------  (15876)  |
> |1717.730869|                   |         PPS       |                   |t38:v21:HDLC:Partial Page Signal
> |         |                   |(21440)  <------------------  (15876)  |
> |1717.780811|                   |         no-signal |                   |t38:t30 Ind:no-signal
> |         |                   |(21440)  <------------------  (15876)  |
> |1718.130798|                   |         v17-14400-long-training          |t38:t30 Ind:v17-14400-long-training
> |         |                   |(21440)  <------------------  (15876)  |
> |1719.541095|                   |         t4-non-ecm-data:v17-14400          |t38:t4-non-ecm-data:v17-14400 Duration: 1.51s No packet lost
> |         |                   |(21440)  <------------------  (15876)  |
> |1721.060850|                   |         no-signal |                   |t38:t30 Ind:no-signal
> |         |                   |(21440)  <------------------  (15876)  |
> |1722.573572|                   |         v21-preamble                  |t38:t30 Ind:v21-preamble
> |         |                   |(21440)  ------------------>  (15876)  |
> |1723.593652|                   |         CFR       |                   |t38:v21:HDLC:Confirmation To Receive
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



From asilva at wirelessmundi.com  Fri Dec 21 14:54:56 2012
From: asilva at wirelessmundi.com (Antonio)
Date: Fri, 21 Dec 2012 12:54:56 +0100
Subject: [Freeswitch-users] Freeswitch TLS and Yealink t26p
In-Reply-To: <1356082510.23186.154.camel@marces.madrid.commsmundi.com>
References: <1356082510.23186.154.camel@marces.madrid.commsmundi.com>
Message-ID: <1356090896.23186.170.camel@marces.madrid.commsmundi.com>

Answer to myself....

In the yealink configuration, in the account parameters, the "transport"
must be force to TLS.  

I don't know why it just works.... Before i was using DNS-SRV, that
should be the first option, yealink should have some issue here... i
will report to them.


Thanks,
Ant?nio

On Fri, 2012-12-21 at 10:35 +0100, Antonio wrote:

> Hi,
> 
> I'm trying to register a yealink with TLS, using my one certificates.
> 
> I follow the wiki and In fs i have both agent.pem and cafile.pem . I
> install in the phone the root certificate.
> 
> But when i try to register, i have (tport log):
> 
> 
> tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x808fb0) msg
> 0x7fe9d0aa8180 from (udp/192.168.10.1:5060) has 340 bytes, veclen = 1
> tport.c:3004 tport_deliver() tport_deliver(0x808fb0): msg
> 0x7fe9d0aa8180 (340 bytes) from udp/192.168.10.23:5060/sip next=(nil)
> tport.c:4202 tport_release() tport_release(0x808fb0): 0x7fe9d01142f0
> by 0x7fe9d025d920 with 0x7fe9d0aa8180
> tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fe9c802aad0):
> events IN
> tport.c:869 tport_alloc_secondary()
> tport_alloc_secondary(0x7fe9c802aad0): new secondary tport
> 0x7fe9c03e8450
> tport_type_tls.c:603 tport_tls_accept()
> tport_tls_accept(0x7fe9c03e8450): new connection from
> tls/192.168.10.36:48754/sips
> tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events
> NEGOTIATING
> tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events
> NEGOTIATING
> tport_tls.c:526 tls_post_connection_check()
> tls_post_connection_check(0x7fe9c03e8450): Peer did not provide X.509
> Certificate.
> 
> 
> 
> I could make it work and have a register in the tls profile when i
> check on the phone the option in Security->Trusted Certificates: "Only
> Accept Trusted Certificates: DISABLED".
> Could it be some bug in the yealink, or I?m missing something in the
> conf...
> 
> Another question, is there any problem if i choose to use this
> configuration... since is the phone that ignores the certificate and
> the validation is done by the server and not by the client. 
> 
> Can you help me?
> 
> Thanks,
> Ant?nio 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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


-- 

Un cordial saludo / Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com
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From ali.jawad at splendor.net  Fri Dec 21 15:10:06 2012
From: ali.jawad at splendor.net (Ali Jawad)
Date: Fri, 21 Dec 2012 14:10:06 +0200
Subject: [Freeswitch-users] Jingle updates since 2010 ?
Message-ID: <CAG2QdgaLXFA_0QP+oboYUCwi7xWEc+CN5=u_G5O3mqYE3r=NYQ@mail.gmail.com>

Hi
Sometime in 2010, one of my employers wanted to use an xmpp server with
jingle to be able to make pstn calls through freeswitch by sending the
jingle call to freeswitch and freeswitch does convert to SIP. Anthony did
help me at the time and the conclusion was

Now I have to state the following. I have tried Freeswitch in client mode
using the following two approaches


   1.  With A Gtalk account and a gtalk client, this worked flawlessly
   2. With server XMPP servers and jingle clients all register but FS was
   not able to do Jingle to SIP conversion in this case. *The main reason
   as per FS developer Anthony is that jingle is a point to point protocol and
   FS was tested to work with Gtalk and telepathy. Note here that Freeswith
   does not use libjingle it uses it's own special jingle implementation*
   3. Another Approach I use was the component mode with my XMPP server,
   everything worked right expect jingle to sip conversion !! :(.

From
http://www.alijawad.org/cms/index.php?option=com_content&task=view&id=21&Itemid=2

Now, I do have a new employer requesting a similar scenario, and I would
love to know if anything has changed since and if the scenario in question
is now possible.

Regards


*

*
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From sdevoy at bizfocused.com  Fri Dec 21 17:27:46 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 21 Dec 2012 09:27:46 -0500
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <480f01cddbae$6d7d67e0$487837a0$@bizfocused.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<480f01cddbae$6d7d67e0$487837a0$@bizfocused.com>
Message-ID: <047501cddf87$58d11990$0a734cb0$@bizfocused.com>

I just wanted to add a note "for the record".  This is specific to my
Polycom 335s and my FIOS router, but probably has much broader implications
to NAT in general.

 

I was searching for a way to eliminate SIP ALG on my router since it is
disdained by many.  I was successful.  The main difference was that I had to
set each phones local SIP signaling port to a unique number.  When any two
used the same port (like 5060), the second one would not register.  It
appears that some routers (I suspect many routers) do not map a single
external port to multiple local destinations well.  My extensions were 101
to 106, so I set the local SIP ports to 5101 to 5106.  They still all
connect to the same SERVER port (5060).  This is just helping the NAT'ed
Router determine the route of external packets back to the intended device
(phone).  I still had the 2 NDLB settings on for these user profiles as
mentioned below.

 

Just FYI: On the polycom 335s, the setting through the web interface are
SETTINGS=>SIP local port #  and SETTINGS=>NETWORK=>NAT sip signaling port.
They should match.

 

I cannot swear this, but I am pretty sure I had a similar issue with
multiple lines on CISCO 504Gs.  Each line had to have a unique local port.

 

It is certainly easy enough to imagine how it would be much simpler to "map"
multiple "one to one"  external ports to internal ports than to code for a
single external port having multiple internal "candidates" on the same port.
In actuality, the final IP address and PORT should be present and this
should not be a problem - but it is, A LOT of the time. 

 

HTH,

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Devoy
Sent: Sunday, December 16, 2012 11:57 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] NAT traversal - the final say..!

 

I have spent many hours working on NAT issues on client end, my server has a
public address.  

 

With CISCO brand phones I did not need any non-standards compliant settings,
just turning on all the choices in the CISCO web setup NAT section.
However, with Polycom 335 phones (as of Dec 2012) I could not get registered
or get audio without the following:

* NDLB-connectile-dysfunction

* NDLB-force-rport

* Enable SIP ALG on my FIOS router.

With those setting however, this has worked perfectly.  Also note that when
I turned on SIP ALG, my Cisco phones quite working until I added the NDLB
parameter/variable to the Cisco <user> in the directory.   They seem to be
quite complimentary but seem be requirements for each other.

 

I really tried to stay away from SIP ALG because so many posts were so
negative about it.  Without the NDLB "flags" I could never see any
difference when enabling SIP ALG.  The combination for me has been
fantastic.

 

HOWEVER, since there are so many different versions of "success" in the IRC
and Wiki,  I am pretty sure that other router brands with different SIP ALG
implementations and/or other phone brands or even firmware versions may need
different configurations.  It is almost like we just need a checklist that
says try these combinations until you find one that fits your site.

 

HTH,

sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cal
Leeming [Simplicity Media Ltd]
Sent: Sunday, December 16, 2012 11:15 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] NAT traversal - the final say..!

 

Any and all feedback on this thread would be much welcomed.

 

Hello,

 

There seems to be a large number of discussions surrounding NAT traversal,
as well as lots of documentation, but with no concrete answers. 

 

The NAT related wiki documentation is tedious, and depending on the outcome
of this thread, I'd like to spend some time cleaning it up.

 

The most common problem (the same as ours) was having a router with broken
ALG and a softphone that does not seem to work with STUN.

 

The following REGISTER is sent from a phone.

 

REGISTER sip:1.2.3.4:5060 SIP/2.0

Via: SIP/2.0/UDP
192.168.1.102:57787;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>

To: "foxx"<sip:2000 at 1.2.3.4:5060>

From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448

Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.

CSeq: 7 REGISTER

Expires: 120

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
REFER, INFO, MESSAGE

Supported: replaces

User-Agent: 3CXPhone 6.0.25732.0

Content-Length: 0

 

As you can see, the client's public IP is not specified anywhere. FreeSWITCH
offers several ways around this, the main ones being;

 

* NDLB-connectile-dysfunction

* NDLB-force-rport

* apply-nat-acl

* sip-force-contact

 

The one that has worked in our case was "NDLB-connectile-dysfunction"
(otherwise known as NAT HACK), however there seems to be a lot of negative
comments about using this.

 

>From what I can tell, the general argument is that NAT HACK is considered a
non RFC compliant hack, and the SIP phones should be doing a better job of
keeping to the RFCs.

 

In principle, this is a fair argument - but in practise, it's not a
reasonable assumption that all phones are RFC compliant, and (imho) not a
reasonable argument to have this functionality disabled by default.

 

So, I'd like to present the following arguments;

 

* Are there any other negative aspects about using
NDLB-connectile-dysfunction, other than it is a non compliant RFC hack?

 

* Why is NDLB-connectile-dysfunction not enabled by default when certain
conditions are met? In the event that FreeSWITCH receives a REGISTER from a
phone specifying a Contact/Via as 192.168.0.0/16, but received on a public
IP, then it should be obvious that NAT is broken and automatically try to
circumvent it.

 

* People seem to get confused between server side and client side NAT
problems, and that they both need to be resolved in a different way. The
documentation doesn't seem to reflect this clearly.

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From steveayre at gmail.com  Fri Dec 21 17:45:04 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 21 Dec 2012 14:45:04 +0000
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <047501cddf87$58d11990$0a734cb0$@bizfocused.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<480f01cddbae$6d7d67e0$487837a0$@bizfocused.com>
	<047501cddf87$58d11990$0a734cb0$@bizfocused.com>
Message-ID: <CAFiqYukHqkDv+3c+gbxu6DniUbgQdesXJSouZo3OH_hDwqjzuQ@mail.gmail.com>

I wonder whether this router is one that doesn't rewrite the external port,
but accepts traffic to that external port from any IP (allowing hole
punching to work) therefore can only map that external port to a single
internal device.




On 21 December 2012 14:27, Sean Devoy <sdevoy at bizfocused.com> wrote:

> I just wanted to add a note ?for the record?.  This is specific to my
> Polycom 335s and my FIOS router, but probably has much broader implications
> to NAT in general.****
>
> ** **
>
> I was searching for a way to eliminate SIP ALG on my router since it is
> disdained by many.  I was successful.  The main difference was that I had
> to set each phones local SIP signaling port to a unique number.  When any
> two used the same port (like 5060), the second one would not register.  It
> appears that some routers (I suspect many routers) do not map a single
> external port to multiple local destinations well.  My extensions were 101
> to 106, so I set the local SIP ports to 5101 to 5106.  They still all
> connect to the same SERVER port (5060).  This is just helping the NAT?ed
> Router determine the route of external packets back to the intended device
> (phone).  I still had the 2 NDLB settings on for these user profiles as
> mentioned below.****
>
> ** **
>
> Just FYI: On the polycom 335s, the setting through the web interface are
> SETTINGS=>SIP local port #  and SETTINGS=>NETWORK=>NAT sip signaling port.
> They should match.****
>
> ** **
>
> I cannot swear this, but I am pretty sure I had a similar issue with
> multiple lines on CISCO 504Gs.  Each line had to have a unique local port.
> ****
>
> ** **
>
> It is certainly easy enough to imagine how it would be much simpler to
> ?map? multiple ?one to one?  external ports to internal ports than to code
> for a single external port having multiple internal ?candidates? on the
> same port.  In actuality, the final IP address and PORT should be present
> and this should not be a problem ? but it is, A LOT of the time. ****
>
> ** **
>
> HTH,****
>
> Sean****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sean Devoy
> *Sent:* Sunday, December 16, 2012 11:57 AM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] NAT traversal - the final say..!****
>
> ** **
>
> I have spent many hours working on *NAT issues on client end*, my server
> has a public address.  ****
>
> ** **
>
> With CISCO brand phones I did not need any non-standards compliant
> settings, just turning on all the choices in the CISCO web setup NAT
> section.  However, with Polycom 335 phones (as of Dec 2012) I could not get
> registered or get audio without the following:****
>
> * NDLB-connectile-dysfunction****
>
> * NDLB-force-rport****
>
> * Enable SIP ALG on my FIOS router.****
>
> With those setting however, this has worked perfectly.  Also note that
> when I turned on SIP ALG, my Cisco phones quite working until I added the
> NDLB parameter/variable to the Cisco <user> in the directory.   They seem
> to be quite complimentary but seem be requirements for each other.****
>
> ** **
>
> I really tried to stay away from SIP ALG because so many posts were so
> negative about it.  Without the NDLB ?flags? I could never see any
> difference when enabling SIP ALG.  The combination for me has been
> fantastic.****
>
> ** **
>
> HOWEVER, since there are so many different versions of ?success? in the
> IRC and Wiki,  I am pretty sure that other router brands with different SIP
> ALG implementations and/or other phone brands or even firmware versions may
> need different configurations.  It is almost like we just need a checklist
> that says try these combinations until you find one that fits your site.**
> **
>
> ** **
>
> HTH,****
>
> sean****
>
> ** **
>
> *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 *Cal Leeming [Simplicity Media Ltd]
> *Sent:* Sunday, December 16, 2012 11:15 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] NAT traversal - the final say..!****
>
> ** **
>
> *Any and all feedback on this thread would be much welcomed.*****
>
> ** **
>
> Hello,****
>
> ** **
>
> There seems to be a large number of discussions surrounding NAT traversal,
> as well as lots of documentation, but with no concrete answers. ****
>
> ** **
>
> The NAT related wiki documentation is tedious, and depending on the
> outcome of this thread, I'd like to spend some time cleaning it up.****
>
> ** **
>
> The most common problem (the same as ours) was having a router with broken
> ALG and a softphone that does not seem to work with STUN.****
>
> ** **
>
> The following REGISTER is sent from a phone.****
>
> ** **
>
> REGISTER sip:1.2.3.4:5060 SIP/2.0****
>
> Via: SIP/2.0/UDP 192.168.1.102:57787
> ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport****
>
> Max-Forwards: 70****
>
> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>****
>
> To: "foxx"<sip:2000 at 1.2.3.4:5060>****
>
> From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448****
>
> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.****
>
> CSeq: 7 REGISTER****
>
> Expires: 120****
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
> REFER, INFO, MESSAGE****
>
> Supported: replaces****
>
> User-Agent: 3CXPhone 6.0.25732.0****
>
> Content-Length: 0****
>
> ** **
>
> As you can see, the client's public IP is not specified
> anywhere. FreeSWITCH offers several ways around this, the main ones being;
> ****
>
> ** **
>
> * NDLB-connectile-dysfunction****
>
> * NDLB-force-rport****
>
> * apply-nat-acl****
>
> * sip-force-contact****
>
> ** **
>
> The one that has worked in our case was "NDLB-connectile-dysfunction"
> (otherwise known as NAT HACK), however there seems to be a lot of negative
> comments about using this.****
>
> ** **
>
> From what I can tell, the general argument is that NAT HACK is considered
> a non RFC compliant hack, and the SIP phones should be doing a better job
> of keeping to the RFCs.****
>
> ** **
>
> In principle, this is a fair argument - but in practise, it's not a
> reasonable assumption that all phones are RFC compliant, and (imho) not a
> reasonable argument to have this functionality disabled by default.****
>
> ** **
>
> So, I'd like to present the following arguments;****
>
> ** **
>
> * Are there any other negative aspects about
> using NDLB-connectile-dysfunction, other than it is a non compliant RFC
> hack?****
>
> ** **
>
> * Why is NDLB-connectile-dysfunction not enabled by default when certain
> conditions are met? In the event that FreeSWITCH receives a REGISTER from a
> phone specifying a Contact/Via as 192.168.0.0/16, but received on a
> public IP, then it should be obvious that NAT is broken and automatically
> try to circumvent it.****
>
> ** **
>
> * People seem to get confused between server side and client side NAT
> problems, and that they both need to be resolved in a different way. The
> documentation doesn't seem to reflect this clearly.****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 jpablolorenzetti at hotmail.com  Fri Dec 21 19:10:11 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Fri, 21 Dec 2012 16:10:11 +0000
Subject: [Freeswitch-users] core dump when compiling master git
In-Reply-To: <1FFF97C269757C458224B7C895F35F151F113D@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F151F113D@cantor.std.visionutv.se>
Message-ID: <DUB002-W1060B28EA70772D3F5DC19FD9360@phx.gbl>

Hi, trying to install some debugging packages i messed up my OS, so i will re-install everything back and try again .. i will let you know asap. hopefully updating the OS will make the error go away. 
regards! 

From: peter.olsson at visionutveckling.se
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 21 Dec 2012 07:47:46 +0000
Subject: Re: [Freeswitch-users] core dump when compiling master git









One common issue is that you?ve set ?ulimit ?s 240? to start FS, and then it?s active in the running shell. Try to use login into new session and
 see if it helps.
 
/Peter
 
 
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]
F?r Anthony Minessale

Skickat: den 21 december 2012 01:34

Till: FreeSWITCH Users Help

?mne: Re: [Freeswitch-users] core dump when compiling master git
 


It looks like a seg in the helper app that generates part of the code in the spandsp build so I can see it being a real bug but really it may be a bug in a compliler or something too since we don't all suffer from this same problem.  It
 must be partially environmental.


 

+10 points for the form letter!!!!

 


w00t JIRA FTW!


 


Also we brought back "make sure" by popular demand that nukes the whole tree and recompiles.


 


 



 

On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice at freeswitch.org> wrote:

No its not even in spandsp... This happens when he is building spandsp... This means something like gcc is segfaulting...







On 12/20/12 5:22 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:



Technically, yes, SpanDSP but I thought Steve floats around JIRA?





On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice at freeswitch.org> wrote:



This is not a Segfault in FreeSWITCH you both fail....



This is probably a build system issue.... 



I would recommend pastebinning and a full trace from a clean tree of the bootstrap, configure and build process...










On 12/20/12 5:06 PM, "Michael Collins" <msc at freeswitch.org <http://msc at freeswitch.org>
 > wrote:



Brian, have you typed this one before? :D



Juan, yes, definitely file a Jira on this. Also, collect the backtrace on that segfault if you can and attach as a txt file to the Jira case.

-MC


On Thu, Dec 20, 2012 at 2:59 PM, Brian Foster <bdfoster at endigotech.com <http://bdfoster at endigotech.com>
 > wrote:



Dear OP,



We've detected that this is a potential bug in FreeSWITCH. Because this is a potential bug, the mailing list is really not the place to report this.



Before you file a bug report, and if you haven't done this already, please make sure that the symptoms you are experiencing do not concur with another previously submitted bug report. Search JIRA <http://jira.freeswitch.org>
  to ensure this. You might want to check the Wiki just in case this is a feature, not a bug.



You can make a difference. Bugs can be reported to the FreeSWITCH JIRA <http://jira.freeswitch.org> . By doing this one simple task, you can save the FreeSWITCH developers the blood, sweat, and
 tears that are needed in order to gather information related to your bug report and track the commits needed to fix the problem.



By filing a JIRA, you also help the developers gather relevant information so they can take the best course of action. Developers ask for several pieces of information when filing a report. Take a look at the Reporting Bugs <http://wiki.freeswitch.org/wiki/Reporting_Bugs>
  page for more details and instructions. The more information you can gather, the easier it will be to fix the issue and the faster a commit can be made.





Remember that FreeSWITCH is an open source project. FreeSWITCH Developers do not get paid to fix bugs, they simply do it for the sake of the community. They sacrifice sleep, family time, brain cells, and Advil to help YOU. Show them that you care.
Report bugs the correct way.








On Thu, Dec 20, 2012 at 5:41 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com <http://jpablolorenzetti at hotmail.com>
 > wrote:



Hi, i m trying to update my development server (VM) by making a clean installation from git master, 

but i m getting a core dump while compiling:



/bin/sh: line 1:  9386 Segmentation fault      (core dumped) ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.h

make[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139

make[6]: *** [all] Error 2

make[5]: *** [all-recursive] Error 1

make[4]: *** [/home/fs/freeswitch/libs/spandsp/src/libspandsp.la <http://libspandsp.la>
  <http://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





i followed the classic steps (bootstrap,configure and make) is anyone experimenting the same issues ???

my info is:

Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011 i686 Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux



thanks!

       



_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org <http://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://FreeSWITCH-users at lists.freeswitch.org>


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

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

http://www.freeswitch.org


 







-- 

Ken

http://www.FreeSWITCH.org

http://www.ClueCon.com

http://www.OSTAG.org

irc.freenode.net #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



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

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

http://www.freeswitch.org






 

-- 

Anthony Minessale II



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 

!DSPAM:50d3ac3432766009014104! 




_________________________________________________________________________
Professional FreeSWITCH Consulting 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
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From spencer at 5ninesolutions.com  Fri Dec 21 19:42:01 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 21 Dec 2012 08:42:01 -0800
Subject: [Freeswitch-users] t.38 Call Flow Information
In-Reply-To: <50D43159.2050108@coppice.org>
References: <AAD54836-F121-4190-A473-B64D9A9E4612@5ninesolutions.com>
	<50D43159.2050108@coppice.org>
Message-ID: <ca70e0ee-07f4-4dc4-9309-8e7d16ffde39@blur>

Hi Steve,
That was my take as well.   I've posted packet captures and logs here:
http://jira.freeswitch.org/browse/FS-4957

Is there any way to speed up the ReINVITE?

Thanks,
Spencer

-----Original message-----
From: Steve Underwood <steveu at coppice.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Fri, Dec 21, 2012 09:57:07 GMT+00:00
Subject: Re: [Freeswitch-users] t.38 Call Flow Information

Hi,

It looks like your reinvite occurred extremely late, and the FAX was 
already deep in progress as an audio exchange.

Steve


On 12/21/2012 03:14 AM, Spencer Thomason wrote:
> Hello,
> I've been analyzing some Wireshark traces to get a better grasp on the  
t.38 gateway process.  I'm using t38_gateway to convert SIP t38 to TDM audio  
on the receiving end.  I'm struggling to understand the following call flow.  
 I've excluded the audio portion for brevity but t38 is negotiated correctly  
by both endpoints (FS on our end and Acme on the other).  Note the absence  
of a DIS message and the PPS prior to CFR.  Can anyone shed any light on  
this?
>
> Thanks in advance,
> Spencer
>
>
> Acme                                                                        
         FreeSWITCH
> |1716.691635|         INVITE SDP (t38)               |                    
|SIP Request
> |         |(5060)   <------------------  (5070)   |                   |
> |1716.719556|         100 trying -- your call is important to us           
|                   |SIP Status
> |         |(5060)   ------------------>  (5070)   |                   |
> |1716.830431|                   |         no-signal |                    
|t38:t30 Ind:no-signal
> |         |                   |(21440)  <------------------  (15876)  |
> |1716.906132|         200 OK SDP (t38)              |                    
|SIP Status
> |         |(5060)   ------------------>  (5070)   |                   |
> |1716.912839|         ACK       |                   |                    
|SIP Request
> |         |(5060)   <------------------  (5070)   |                   |
> |1716.973011|                   |         no-signal |                    
|t38:t30 Ind:no-signal
> |         |                   |(21440)  ------------------>  (15876)  |
> |1717.450595|                   |         v21-preamble                   
|t38:t30 Ind:v21-preamble
> |         |                   |(21440)  <------------------  (15876)  |
> |1717.730869|                   |         PPS       |                    
|t38:v21:HDLC:Partial Page Signal
> |         |                   |(21440)  <------------------  (15876)  |
> |1717.780811|                   |         no-signal |                    
|t38:t30 Ind:no-signal
> |         |                   |(21440)  <------------------  (15876)  |
> |1718.130798|                   |         v17-14400-long-training           
|t38:t30 Ind:v17-14400-long-training
> |         |                   |(21440)  <------------------  (15876)  |
> |1719.541095|                   |         t4-non-ecm-data:v17-14400         
  |t38:t4-non-ecm-data:v17-14400 Duration: 1.51s No packet lost
> |         |                   |(21440)  <------------------  (15876)  |
> |1721.060850|                   |         no-signal |                    
|t38:t30 Ind:no-signal
> |         |                   |(21440)  <------------------  (15876)  |
> |1722.573572|                   |         v21-preamble                   
|t38:t30 Ind:v21-preamble
> |         |                   |(21440)  ------------------>  (15876)  |
> |1723.593652|                   |         CFR       |                    
|t38:v21:HDLC:Confirmation To Receive
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From anthony.minessale at gmail.com  Fri Dec 21 20:12:18 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Dec 2012 11:12:18 -0600
Subject: [Freeswitch-users] t.38 Call Flow Information
In-Reply-To: <ca70e0ee-07f4-4dc4-9309-8e7d16ffde39@blur>
References: <AAD54836-F121-4190-A473-B64D9A9E4612@5ninesolutions.com>
	<50D43159.2050108@coppice.org>
	<ca70e0ee-07f4-4dc4-9309-8e7d16ffde39@blur>
Message-ID: <CAKbxfG_6dJmuLu0EJb0fVy+Lx=G3UnjuS42MWuhdFNGhn-aDjg@mail.gmail.com>

you can tweak mod_spandsp_fax.c:1391

req_counter is how many packets to wait before sending re-invite.

You can play with reducing this number and if its effective it could be
made into a configurable param.



On Fri, Dec 21, 2012 at 10:42 AM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hi Steve,
> That was my take as well.   I've posted packet captures and logs here:
> http://jira.freeswitch.org/browse/FS-4957
>
> Is there any way to speed up the ReINVITE?
>
> Thanks,
> Spencer
>
>
> -----Original message-----
>
> *From: *Steve Underwood <steveu at coppice.org>*
> To: *freeswitch-users at lists.freeswitch.org*
> Sent: *Fri, Dec 21, 2012 09:57:07 GMT+00:00*
> Subject: *Re: [Freeswitch-users] t.38 Call Flow Information
>
> Hi,
>
> It looks like your reinvite occurred extremely late, and the FAX was
> already deep in progress as an audio exchange.
>
> Steve
>
>
> On 12/21/2012 03:14 AM, Spencer Thomason wrote:
> > Hello,
> > I've been analyzing some Wireshark traces to get a better grasp on the
> t.38 gateway process. I'm using t38_gateway to convert SIP t38 to TDM audio
> on the receiving end. I'm struggling to understand the following call flow.
> I've excluded the audio portion for brevity but t38 is negotiated correctly
> by both endpoints (FS on our end and Acme on the other). Note the absence
> of a DIS message and the PPS prior to CFR. Can anyone shed any light on
> this?
> >
> > Thanks in advance,
> > Spencer
> >
> >
> > Acme FreeSWITCH
> > |1716.691635| INVITE SDP (t38) | |SIP Request
> > | |(5060) <------------------ (5070) | |
> > |1716.719556| 100 trying -- your call is important to us | |SIP Status
> > | |(5060) ------------------> (5070) | |
> > |1716.830431| | no-signal | |t38:t30 Ind:no-signal
> > | | |(21440) <------------------ (15876) |
> > |1716.906132| 200 OK SDP (t38) | |SIP Status
> > | |(5060) ------------------> (5070) | |
> > |1716.912839| ACK | | |SIP Request
> > | |(5060) <------------------ (5070) | |
> > |1716.973011| | no-signal | |t38:t30 Ind:no-signal
> > | | |(21440) ------------------> (15876) |
> > |1717.450595| | v21-preamble |t38:t30 Ind:v21-preamble
> > | | |(21440) <------------------ (15876) |
> > |1717.730869| | PPS | |t38:v21:HDLC:Partial Page Signal
> > | | |(21440) <------------------ (15876) |
> > |1717.780811| | no-signal | |t38:t30 Ind:no-signal
> > | | |(21440) <------------------ (15876) |
> > |1718.130798| | v17-14400-long-training |t38:t30
> Ind:v17-14400-long-training
> > | | |(21440) <------------------ (15876) |
> > |1719.541095| | t4-non-ecm-data:v17-14400 |t38:t4-non-ecm-data:v17-14400
> Duration: 1.51s No packet lost
> > | | |(21440) <------------------ (15876) |
> > |1721.060850| | no-signal | |t38:t30 Ind:no-signal
> > | | |(21440) <------------------ (15876) |
> > |1722.573572| | v21-preamble |t38:t30 Ind:v21-preamble
> > | | |(21440) ------------------> (15876) |
> > |1723.593652| | CFR | |t38:v21:HDLC:Confirmation To Receive
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Anthony Minessale II

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
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From msc at freeswitch.org  Fri Dec 21 20:31:59 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 Dec 2012 09:31:59 -0800
Subject: [Freeswitch-users] Newbie question: Executing Lua scripts for
 incoming calls, how concurrency is to be handled?
In-Reply-To: <849EC2AD-D5F4-471E-B362-480DF549DB62@gmail.com>
References: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au>
	<50CE5D6E.2020401@quentustech.com>
	<965759A53E43FE439E43565A7715E5F058F4156D32@oa-exchange1.oa.com.au>
	<CAKzWOxWFh3RNYg9NY4Dqa-=f9KZOaXTvM8NGzHpvEvzV16VUWA@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156DD8@oa-exchange1.oa.com.au>
	<CAKzWOxUYyfZ_8dLgy_t-Ov2a+YXEDj04dDkdLSn14Q-tZk-8pQ@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156DF7@oa-exchange1.oa.com.au>
	<CAKzWOxUrdVNqqDJxQ8x4_OaE=oEOrGzdEb2D+XmvC-0tW_yfjA@mail.gmail.com>
	<965759A53E43FE439E43565A7715E5F058F4156E6C@oa-exchange1.oa.com.au>
	<849EC2AD-D5F4-471E-B362-480DF549DB62@gmail.com>
Message-ID: <CAKzWOxVW8REpO_7LX2pCV105aSR76M-Yc8-ann21vUCKMErn_g@mail.gmail.com>

Agreed, the hash/db options are safe and effective. You don't need global
variables. I highly recommend that you look at the default.xml dialplan
file and look at the places where "hash" occurs. In Local_Extension a lot
of items are written to the local database with hash "inserts" and
elsewhere you'll see hash "selects" and the occasional hash "delete". Use
the hash_dump command to see what's in the database. Make a few test calls
and do a hash_dump before, during, and after the call. You'll quickly see
what you can do with the hash command and storing information. Just store
information for each of your ports and use a Lua script to read and/or
write as needed.

-MC

On Thu, Dec 20, 2012 at 6:32 PM, Steven Ayre <steveayre at gmail.com> wrote:

> If I remember correctly when when anthm added the option by request, global_setvar
> is NOT safe for setting concurrently. Very occasionally should be ok for
> config changes without a restart, but it is not safe to use from call dial
> plans where calls are running alongside each other.
>
> You should look at using mod_hash or mod_db, which are much safer
>
> Sent from my iPad
>
> On 20 Dec 2012, at 23:23, Sirish Masur Mohan <Sirish.MasurMohan at oa.com.au>
> wrote:
>
> Hi Michael,****
>
> ** **
>
> Thanks for the reply, and the suggestions.****
>
> ** **
>
> My current implementation is a similar to what you have suggested, i.e.:**
> **
>
> 1.       I have defined 4 sets of global variables in vars.xml, where
> each set keeps a track of port?s status and last used timestamp****
>
> 2.       In the dialplan, I execute a simple Lua script which ?chooses?
> the line based on global variables status and timestamp (using
> "global_getvar") and update the status of the line to be used (using
> "global_setvar")****
>
> 3.       Bridge to the line returned by the above script****
>
> 4.       Execute another Lua script on hang-up, which updates the global
> variables (status and timestamp)****
>
> ** **
>
> If I were to implement the hash way, I would have to probably do the
> following:****
>
> 1.       In the dialplan, I execute a simple Lua script which ?chooses?
> the line by reading the hash values of the line status and timestamp and
> update the hash status of the line to be used ****
>
> 2.       Bridge to the line returned by the above script****
>
> 3.       Execute another Lua script on hang-up, which updates the hash
> values (status and timestamp)****
>
> ** **
>
> But I am still not clear on the concurrency question ? if FreeSWITCH has
> received 2 calls at the same time on the E1 line, should I be worried about
> protecting the script that gets executed to access the hash, decide on the
> line to be used, and then update the hash?****
>
> ** **
>
> Thanks again!****
>
> ** **
>
> With regards,****
>
> Sirish****
>
> ** **
>
> *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:* Friday, 21 December 2012 3:42 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Newbie question: Executing Lua scripts
> for incoming calls, how concurrency is to be handled?****
>
> ** **
>
> Sirish,
>
> Since you're using an E1 to connect to the PBX then really all you need to
> do is keep track of the last time each port was hung up and whether or not
> a given port is currently in use. I would use api_on_answer to launch a
> simple script to set a flag to say that a particular port is in use and
> then use the api_hangup_hook to launch another script when the call ends.
>
> The channel variables page on the wiki has some examples of how to use
> these. I recommend that you write simple Lua scripts that use the "hash"
> API to store information in the local database. Also, check out the
> "hash_dump" API as it is a useful way to quickly see what all is stored
> there.
>
> For an example of how to add, remove, and read information from the local
> database using the "hash" API please see conf/dialplan/default.xml. Search
> for "hash" and you'll see all sorts of examples of how the example dialplan
> uses the local database to store useful information that allows us to
> implement features like call return, call intercept, etc.
>
> -MC****
>
> On Tue, Dec 18, 2012 at 7:39 PM, Sirish Masur Mohan <
> Sirish.MasurMohan at oa.com.au> wrote:****
>
> Hi Michael,****
>
>  ****
>
> >> How are you physically connecting from FreeSWITCH to the PBX?****
>
>  ****
>
> I connect this via E1 link ? I have a Sangoma card installed on the
> FreeSWITCH machine.****
>
>  ****
>
> With regards,****
>
> Sirish****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Wednesday, 19 December 2012 1:31 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Newbie question: Executing Lua scripts
> for incoming calls, how concurrency is to be handled?****
>
>  ****
>
> To trigger SIP registrations you'd need the PBX to have a SIP client. I'm
> assuming this is possible, but maybe that's a false assumption. How are you
> physically connecting from FreeSWITCH to the PBX?
>
> -MC****
>
> On Tue, Dec 18, 2012 at 2:36 PM, Sirish Masur Mohan <
> Sirish.MasurMohan at oa.com.au> wrote:****
>
> Hey Guys,****
>
>  ****
>
> Would really appreciate if you could help me out here ? isn?t there a way
> to handle concurrent calls in the dial plan, especially when Lua scripts,
> accessing global variables, are executed on receiving calls? ****
>
>  ****
>
> Is mod_fifo the closest I could get to handle concurrency (as Michael has
> explained)? If yes, how do I trigger SIP registrations, especially working
> with a PBX which I don?t have full control of?****
>
>  ****
>
> With regards,****
>
> Sirish****
>
>  ****
>
> *From:* Sirish Masur Mohan
> *Sent:* Tuesday, 18 December 2012 1:02 PM
> *To:* FreeSWITCH Users Help
> *Subject:* RE: [Freeswitch-users] Newbie question: Executing Lua scripts
> for incoming calls, how concurrency is to be handled?****
>
>  ****
>
> Hi Michael,****
>
>  ****
>
> Thanks for the reply. ****
>
>  ****
>
> >> You would need a SIP registration from the PBX to FreeSWITCH for each
> of the modem extensions****
>
> I am seen SIP clients such as X-Lite sending out the SIP registrations,
> but could you please clarify as to how this can be achieved in the PBX? The
> final production environment would be out in the customer?s PBX, which I
> may not have complete control of.. ****
>
>  ****
>
> >> What application are you building?****
>
> I may not be able to provide the details because of the NDA with customer,
> but what I am trying to achieve is, to replace an existing IVR with
> FreeSWITCH in an old existing setup of the customer ? that?s the reason why
> we continue working with dialup modems!****
>
>  ****
>
> With regards,****
>
> Sirish****
>
>  ****
>
> *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:* Tuesday, 18 December 2012 6:23 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Newbie question: Executing Lua scripts
> for incoming calls, how concurrency is to be handled?****
>
>  ****
>
> You don't have to have actual human agents for mod_fifo. You could define
> a user for each modem and then manually "log in" those "agents" on the
> command line using the fifo_member API command. Something like this:
>
> fifo_member add fifo_name {fifo_member_wait=nowait}user/1234
>
> Where 1234 is the user id of one of the modems. You would need a SIP
> registration from the PBX to FreeSWITCH for each of the modem extensions.
>
> Having modems go through a VoIP system sounds a bit scary. What
> application are you building?
> -MC****
>
> On Sun, Dec 16, 2012 at 5:41 PM, Sirish Masur Mohan <
> Sirish.MasurMohan at oa.com.au> wrote:****
>
> Hi William,
>
> Thanks for the reply.
>
> My setup is as follows:
> Client(Caller)->dialup modem->PBX->FreeSWITCH->PBX-> 4 dialup
> modems->Server(Receiver)
>
> I basically need FreeSWITCH to bridge the incoming call to the best
> external destination (out of the 4 available), so that the modem training,
> connection etc can take place smoothly, before exchange of data. I am not
> sure if mod_fifo would help me in this scenario, as, I would require an
> agent to dial in and read the fifo. Could you please clarify?
>
> Thanks!
>
> With regards,
> Sirish****
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William King
> Sent: Monday, 17 December 2012 10:47 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for
> incoming calls, how concurrency is to be handled?
>
> Sounds like you want to take a look into mod_fifo.
>
> 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****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
> ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Fri Dec 21 20:34:15 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 Dec 2012 09:34:15 -0800
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <047501cddf87$58d11990$0a734cb0$@bizfocused.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<480f01cddbae$6d7d67e0$487837a0$@bizfocused.com>
	<047501cddf87$58d11990$0a734cb0$@bizfocused.com>
Message-ID: <CAKzWOxUqY-mOmXOaszVEFudcg+z=PxD62DV+AT9VQuZ1h7MnaQ@mail.gmail.com>

Can you add this information to the wiki:

http://wiki.freeswitch.org/wiki/Interop_List#Routers

You can just about copy and paste this email right in there. Ping me if you
have any questions.

Thanks,
MC

On Fri, Dec 21, 2012 at 6:27 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> I just wanted to add a note ?for the record?.  This is specific to my
> Polycom 335s and my FIOS router, but probably has much broader implications
> to NAT in general.****
>
> ** **
>
> I was searching for a way to eliminate SIP ALG on my router since it is
> disdained by many.  I was successful.  The main difference was that I had
> to set each phones local SIP signaling port to a unique number.  When any
> two used the same port (like 5060), the second one would not register.  It
> appears that some routers (I suspect many routers) do not map a single
> external port to multiple local destinations well.  My extensions were 101
> to 106, so I set the local SIP ports to 5101 to 5106.  They still all
> connect to the same SERVER port (5060).  This is just helping the NAT?ed
> Router determine the route of external packets back to the intended device
> (phone).  I still had the 2 NDLB settings on for these user profiles as
> mentioned below.****
>
> ** **
>
> Just FYI: On the polycom 335s, the setting through the web interface are
> SETTINGS=>SIP local port #  and SETTINGS=>NETWORK=>NAT sip signaling port.
> They should match.****
>
> ** **
>
> I cannot swear this, but I am pretty sure I had a similar issue with
> multiple lines on CISCO 504Gs.  Each line had to have a unique local port.
> ****
>
> ** **
>
> It is certainly easy enough to imagine how it would be much simpler to
> ?map? multiple ?one to one?  external ports to internal ports than to code
> for a single external port having multiple internal ?candidates? on the
> same port.  In actuality, the final IP address and PORT should be present
> and this should not be a problem ? but it is, A LOT of the time. ****
>
> ** **
>
> HTH,****
>
> Sean****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sean Devoy
> *Sent:* Sunday, December 16, 2012 11:57 AM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] NAT traversal - the final say..!****
>
> ** **
>
> I have spent many hours working on *NAT issues on client end*, my server
> has a public address.  ****
>
> ** **
>
> With CISCO brand phones I did not need any non-standards compliant
> settings, just turning on all the choices in the CISCO web setup NAT
> section.  However, with Polycom 335 phones (as of Dec 2012) I could not get
> registered or get audio without the following:****
>
> * NDLB-connectile-dysfunction****
>
> * NDLB-force-rport****
>
> * Enable SIP ALG on my FIOS router.****
>
> With those setting however, this has worked perfectly.  Also note that
> when I turned on SIP ALG, my Cisco phones quite working until I added the
> NDLB parameter/variable to the Cisco <user> in the directory.   They seem
> to be quite complimentary but seem be requirements for each other.****
>
> ** **
>
> I really tried to stay away from SIP ALG because so many posts were so
> negative about it.  Without the NDLB ?flags? I could never see any
> difference when enabling SIP ALG.  The combination for me has been
> fantastic.****
>
> ** **
>
> HOWEVER, since there are so many different versions of ?success? in the
> IRC and Wiki,  I am pretty sure that other router brands with different SIP
> ALG implementations and/or other phone brands or even firmware versions may
> need different configurations.  It is almost like we just need a checklist
> that says try these combinations until you find one that fits your site.**
> **
>
> ** **
>
> HTH,****
>
> sean****
>
> ** **
>
> *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 *Cal Leeming [Simplicity Media Ltd]
> *Sent:* Sunday, December 16, 2012 11:15 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] NAT traversal - the final say..!****
>
> ** **
>
> *Any and all feedback on this thread would be much welcomed.*****
>
> ** **
>
> Hello,****
>
> ** **
>
> There seems to be a large number of discussions surrounding NAT traversal,
> as well as lots of documentation, but with no concrete answers. ****
>
> ** **
>
> The NAT related wiki documentation is tedious, and depending on the
> outcome of this thread, I'd like to spend some time cleaning it up.****
>
> ** **
>
> The most common problem (the same as ours) was having a router with broken
> ALG and a softphone that does not seem to work with STUN.****
>
> ** **
>
> The following REGISTER is sent from a phone.****
>
> ** **
>
> REGISTER sip:1.2.3.4:5060 SIP/2.0****
>
> Via: SIP/2.0/UDP 192.168.1.102:57787
> ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport****
>
> Max-Forwards: 70****
>
> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>****
>
> To: "foxx"<sip:2000 at 1.2.3.4:5060>****
>
> From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448****
>
> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.****
>
> CSeq: 7 REGISTER****
>
> Expires: 120****
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
> REFER, INFO, MESSAGE****
>
> Supported: replaces****
>
> User-Agent: 3CXPhone 6.0.25732.0****
>
> Content-Length: 0****
>
> ** **
>
> As you can see, the client's public IP is not specified
> anywhere. FreeSWITCH offers several ways around this, the main ones being;
> ****
>
> ** **
>
> * NDLB-connectile-dysfunction****
>
> * NDLB-force-rport****
>
> * apply-nat-acl****
>
> * sip-force-contact****
>
> ** **
>
> The one that has worked in our case was "NDLB-connectile-dysfunction"
> (otherwise known as NAT HACK), however there seems to be a lot of negative
> comments about using this.****
>
> ** **
>
> From what I can tell, the general argument is that NAT HACK is considered
> a non RFC compliant hack, and the SIP phones should be doing a better job
> of keeping to the RFCs.****
>
> ** **
>
> In principle, this is a fair argument - but in practise, it's not a
> reasonable assumption that all phones are RFC compliant, and (imho) not a
> reasonable argument to have this functionality disabled by default.****
>
> ** **
>
> So, I'd like to present the following arguments;****
>
> ** **
>
> * Are there any other negative aspects about
> using NDLB-connectile-dysfunction, other than it is a non compliant RFC
> hack?****
>
> ** **
>
> * Why is NDLB-connectile-dysfunction not enabled by default when certain
> conditions are met? In the event that FreeSWITCH receives a REGISTER from a
> phone specifying a Contact/Via as 192.168.0.0/16, but received on a
> public IP, then it should be obvious that NAT is broken and automatically
> try to circumvent it.****
>
> ** **
>
> * People seem to get confused between server side and client side NAT
> problems, and that they both need to be resolved in a different way. The
> documentation doesn't seem to reflect this clearly.****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Fri Dec 21 20:35:18 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 Dec 2012 09:35:18 -0800
Subject: [Freeswitch-users] Tracing
In-Reply-To: <50D421F7.3080601@softnet.si>
References: <50D2C1E5.10708@softnet.si>
	<CAKzWOxWmLF05rzRCVjCCaQxqX2v6hahdve+PpuLtdts0C--xeg@mail.gmail.com>
	<50D421F7.3080601@softnet.si>
Message-ID: <CAKzWOxXKPCYjes6hDo0tJ+-QeZJCavjZBv66unQpDZ==grV6XQ@mail.gmail.com>

This is definitely a good idea if you are concerned about CPU load. Nice
work.
-MC

On Fri, Dec 21, 2012 at 12:46 AM, Miha <miha at softnet.si> wrote:

>  Michael, thanks!
>
> I installed pcapsipdump. In long term I thing that cpu load goes in piks
> much higher. I will do port mirroring on switch in trace this on different
> server:)
>
> Thanks!
>
> Dne 12/20/2012 10:01 PM, pi?e Michael Collins:
>
> Your best bet would be to use pcapsipdump and manually or via script clear
> out the old pcap files each day/week/month/whatever.
>
> Also, make sure that you are rotating your FreeSWITCH log files on a
> regular basis. Also, be sure to enable uuid logging in your
> logfile.conf.xml file. Having the uuid of the call in the fs log file is
> really helpful on a busy system.
>
> -MC
>
> On Wed, Dec 19, 2012 at 11:44 PM, Miha <miha at softnet.si> wrote:
>
>> Hi,
>>
>> I am experiencing some problems related with media with some users. It
>> is almost impossible to figure it out where is a problem because this
>> problem is not presented all day.
>> I need some sip trace dump for a specific user if it is possible due to
>> a lot of traffic. It would be nice that this would not cause to much CPU
>> load and disk consumption as this would need to be running and tracing
>> 24/7.
>>
>> Users are reporting the thy exeriacing noise in conversation and
>> sometime thy do not hear other side.
>>
>> What would be the best way?
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Fri Dec 21 20:41:38 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 Dec 2012 09:41:38 -0800
Subject: [Freeswitch-users] Freeswitch TLS and Yealink t26p
In-Reply-To: <1356090896.23186.170.camel@marces.madrid.commsmundi.com>
References: <1356082510.23186.154.camel@marces.madrid.commsmundi.com>
	<1356090896.23186.170.camel@marces.madrid.commsmundi.com>
Message-ID: <CAKzWOxXDUEJh4Kq183qFWsB_A933j+HtoShx9LKyLa42fT2LRA@mail.gmail.com>

Antonio,

It looks like we have a number of phones listed here:

http://wiki.freeswitch.org/wiki/Tls#SIP_TLS_Device_Interoperability

However, Yealink is not among them. I've created a spot for Yealinks here:

http://wiki.freeswitch.org/wiki/Interop_List#Yealink_TLS_Configuration

Could you put what you've learned into that spot on the wiki?

Thanks!
-MC


On Fri, Dec 21, 2012 at 3:54 AM, Antonio <asilva at wirelessmundi.com> wrote:

> **
> Answer to myself....
>
> In the yealink configuration, in the account parameters, the "transport"
> must be force to TLS.
>
> I don't know why it just works.... Before i was using DNS-SRV, that should
> be the first option, yealink should have some issue here... i will report
> to them.
>
>
> Thanks,
> Ant?nio
>
>
> On Fri, 2012-12-21 at 10:35 +0100, Antonio wrote:
>
> Hi,
>
> I'm trying to register a yealink with TLS, using my one certificates.
>
> I follow the wiki and In fs i have both agent.pem and cafile.pem . I
> install in the phone the root certificate.
>
> But when i try to register, i have (tport log):
>
>
> tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x808fb0) msg
> 0x7fe9d0aa8180 from (udp/192.168.10.1:5060) has 340 bytes, veclen = 1
> tport.c:3004 tport_deliver() tport_deliver(0x808fb0): msg 0x7fe9d0aa8180
> (340 bytes) from udp/192.168.10.23:5060/sip next=(nil)
> tport.c:4202 tport_release() tport_release(0x808fb0): 0x7fe9d01142f0 by
> 0x7fe9d025d920 with 0x7fe9d0aa8180
> tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fe9c802aad0): events IN
> tport.c:869 tport_alloc_secondary() tport_alloc_secondary(0x7fe9c802aad0):
> new secondary tport 0x7fe9c03e8450
> tport_type_tls.c:603 tport_tls_accept() tport_tls_accept(0x7fe9c03e8450):
> new connection from tls/192.168.10.36:48754/sips
> tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events
> NEGOTIATING
> tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events
> NEGOTIATING
> tport_tls.c:526 tls_post_connection_check()
> tls_post_connection_check(0x7fe9c03e8450): Peer did not provide X.509
> Certificate.
>
>
>
> I could make it work and have a register in the tls profile when i check
> on the phone the option in Security->Trusted Certificates: "Only Accept
> Trusted Certificates: DISABLED".
> Could it be some bug in the yealink, or I?m missing something in the
> conf...
>
> Another question, is there any problem if i choose to use this
> configuration... since is the phone that ignores the certificate and the
> validation is done by the server and not by the client.
>
> Can you help me?
>
> Thanks,
> Ant?nio
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
>
>   --
>
> Un cordial saludo / Best regards,
>
>  _________________________
>
> Ant?nio Silva
>
> E-mail:asilva at wirelessmundi.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From andrew at cassidywebservices.co.uk  Fri Dec 21 22:38:23 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Fri, 21 Dec 2012 19:38:23 +0000
Subject: [Freeswitch-users] Cisco-signed SSL Certificates/Alternative
 phone suggestions
In-Reply-To: <CAMbW9oKXfAXF4sdQovffXEK4UyvK63pcGK7gQjyj=QrbaEBMmw@mail.gmail.com>
References: <CAMbW9oL=AjAJCDn9D-tggWqPoy8pitDjJT0waZgH27HWJyOx0Q@mail.gmail.com>
	<CA+NjZ2uUmyw_eVAubva3vP1QgpsKKY0ymHFkEt4kkpgRnSYhsQ@mail.gmail.com>
	<CAMbW9oKXfAXF4sdQovffXEK4UyvK63pcGK7gQjyj=QrbaEBMmw@mail.gmail.com>
Message-ID: <CAMbW9oKVA8bthMYEROUfUSANaEtANEsCe_xBq2OebU90KsN2qg@mail.gmail.com>

Hi guys, another quick prod hoping I'm in time to catch some of you!

I have my Cisco signed SSL certificates! But the company I'm using is
having a hard time getting the CA certs for authentication out of them.
Does anyone have the CA certs for sipura and ciscosb they could pass on?

Thanks,

On 22 November 2012 16:21, Andrew Cassidy
<andrew at cassidywebservices.co.uk>wrote:

> Followed that. Also:
>
> *Note: A certificate will only be generated if a Cisco sales
> representative sends the CSR to the email alias.*
> *
> *
> The problem is I don't have a Cisco sales rep to deal with. I've tried
> emailing that address myself, it's just a black hole for us normal folk.
>
> On 22 November 2012 15:10, Nick Vines <jnvines at gmail.com> wrote:
>
>> You have to email them. This page has a walkthrough that should help.
>>
>> https://supportforums.cisco.com/docs/DOC-9852
>>
>>
>>
>>
>> On Thu, Nov 22, 2012 at 8:23 AM, Andrew Cassidy <
>> andrew at cassidywebservices.co.uk> wrote:
>>
>>> Hi guys, just looking for a little help. I've been using Cisco SPA50x
>>> phones for a little while now, they have some pretty nice features for the
>>> price, including the ability to write custom applications in XML. However,
>>> to run them over HTTPS instead of HTTP, you need a certificate signed by
>>> Cisco, which I'm having difficulty obtaining.
>>>
>>> So my questions are:
>>>
>>>    1. Can anyone help me obtain such a certificate? I have the CSR
>>>    already to go.
>>>    2. Are there any similarly priced and featured phones available that
>>>    you guys have experience with?
>>>
>>> Thanks guys,
>>>
>>> --
>>> *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
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> --
> *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 jpablolorenzetti at hotmail.com  Fri Dec 21 22:56:56 2012
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Fri, 21 Dec 2012 19:56:56 +0000
Subject: [Freeswitch-users] core dump when compiling master git
In-Reply-To: <DUB002-W1060B28EA70772D3F5DC19FD9360@phx.gbl>
References: <1FFF97C269757C458224B7C895F35F151F113D@cantor.std.visionutv.se>,
	<DUB002-W1060B28EA70772D3F5DC19FD9360@phx.gbl>
Message-ID: <DUB002-W1012C49E2F27D5CA7BCABB3D9360@phx.gbl>

Hi, well i reinstalled the whole system, the same version i had, again and downloaded freeswitch again and recompiled etc etc and i m not getting the core dump anymore .... so it was some environmental thing that i had before and i dont have now ..... in any case i m going to keep an eye on this and if i get the core dumps again in later recompilations i will let you know. thanks a lot for you prompt responses ...   

From: jpablolorenzetti at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 21 Dec 2012 16:10:11 +0000
Subject: Re: [Freeswitch-users] core dump when compiling master git




Hi, trying to install some debugging packages i messed up my OS, so i will re-install everything back and try again .. i will let you know asap. hopefully updating the OS will make the error go away. 
regards! 

From: peter.olsson at visionutveckling.se
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 21 Dec 2012 07:47:46 +0000
Subject: Re: [Freeswitch-users] core dump when compiling master git









One common issue is that you?ve set ?ulimit ?s 240? to start FS, and then it?s active in the running shell. Try to use login into new session and
 see if it helps.
 
/Peter
 
 
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]
F?r Anthony Minessale

Skickat: den 21 december 2012 01:34

Till: FreeSWITCH Users Help

?mne: Re: [Freeswitch-users] core dump when compiling master git
 


It looks like a seg in the helper app that generates part of the code in the spandsp build so I can see it being a real bug but really it may be a bug in a compliler or something too since we don't all suffer from this same problem.  It
 must be partially environmental.


 

+10 points for the form letter!!!!

 


w00t JIRA FTW!


 


Also we brought back "make sure" by popular demand that nukes the whole tree and recompiles.


 


 



 

On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice at freeswitch.org> wrote:

No its not even in spandsp... This happens when he is building spandsp... This means something like gcc is segfaulting...







On 12/20/12 5:22 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:



Technically, yes, SpanDSP but I thought Steve floats around JIRA?





On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice at freeswitch.org> wrote:



This is not a Segfault in FreeSWITCH you both fail....



This is probably a build system issue.... 



I would recommend pastebinning and a full trace from a clean tree of the bootstrap, configure and build process...










On 12/20/12 5:06 PM, "Michael Collins" <msc at freeswitch.org <http://msc at freeswitch.org>
 > wrote:



Brian, have you typed this one before? :D



Juan, yes, definitely file a Jira on this. Also, collect the backtrace on that segfault if you can and attach as a txt file to the Jira case.

-MC


On Thu, Dec 20, 2012 at 2:59 PM, Brian Foster <bdfoster at endigotech.com <http://bdfoster at endigotech.com>
 > wrote:



Dear OP,



We've detected that this is a potential bug in FreeSWITCH. Because this is a potential bug, the mailing list is really not the place to report this.



Before you file a bug report, and if you haven't done this already, please make sure that the symptoms you are experiencing do not concur with another previously submitted bug report. Search JIRA <http://jira.freeswitch.org>
  to ensure this. You might want to check the Wiki just in case this is a feature, not a bug.



You can make a difference. Bugs can be reported to the FreeSWITCH JIRA <http://jira.freeswitch.org> . By doing this one simple task, you can save the FreeSWITCH developers the blood, sweat, and
 tears that are needed in order to gather information related to your bug report and track the commits needed to fix the problem.



By filing a JIRA, you also help the developers gather relevant information so they can take the best course of action. Developers ask for several pieces of information when filing a report. Take a look at the Reporting Bugs <http://wiki.freeswitch.org/wiki/Reporting_Bugs>
  page for more details and instructions. The more information you can gather, the easier it will be to fix the issue and the faster a commit can be made.





Remember that FreeSWITCH is an open source project. FreeSWITCH Developers do not get paid to fix bugs, they simply do it for the sake of the community. They sacrifice sleep, family time, brain cells, and Advil to help YOU. Show them that you care.
Report bugs the correct way.








On Thu, Dec 20, 2012 at 5:41 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com <http://jpablolorenzetti at hotmail.com>
 > wrote:



Hi, i m trying to update my development server (VM) by making a clean installation from git master, 

but i m getting a core dump while compiling:



/bin/sh: line 1:  9386 Segmentation fault      (core dumped) ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.h

make[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139

make[6]: *** [all] Error 2

make[5]: *** [all-recursive] Error 1

make[4]: *** [/home/fs/freeswitch/libs/spandsp/src/libspandsp.la <http://libspandsp.la>
  <http://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





i followed the classic steps (bootstrap,configure and make) is anyone experimenting the same issues ???

my info is:

Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011 i686 Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux



thanks!

       



_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org <http://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://FreeSWITCH-users at lists.freeswitch.org>


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

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

http://www.freeswitch.org


 







-- 

Ken

http://www.FreeSWITCH.org

http://www.ClueCon.com

http://www.OSTAG.org

irc.freenode.net #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



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

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

http://www.freeswitch.org






 

-- 

Anthony Minessale II



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 

!DSPAM:50d3ac3432766009014104! 




_________________________________________________________________________
Professional FreeSWITCH Consulting 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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 Dec 21 22:59:03 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 21 Dec 2012 14:59:03 -0500
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAKzWOxUqY-mOmXOaszVEFudcg+z=PxD62DV+AT9VQuZ1h7MnaQ@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>	<480f01cddbae$6d7d67e0$487837a0$@bizfocused.com>	<047501cddf87$58d11990$0a734cb0$@bizfocused.com>
	<CAKzWOxUqY-mOmXOaszVEFudcg+z=PxD62DV+AT9VQuZ1h7MnaQ@mail.gmail.com>
Message-ID: <079801cddfb5$a07502b0$e15f0810$@bizfocused.com>

Hey MC,

 

Cal Leeming started this thread so he could gather info on NAT and add it to
the wiki in a structured way.  I hope this will be included in that NAT
section in some meaningful context.  I will certainly add it to the router
section as well.

Sean

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Friday, December 21, 2012 12:34 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] NAT traversal - the final say..!

 

Can you add this information to the wiki:

http://wiki.freeswitch.org/wiki/Interop_List#Routers

You can just about copy and paste this email right in there. Ping me if you
have any questions.

Thanks,
MC

On Fri, Dec 21, 2012 at 6:27 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

I just wanted to add a note "for the record".  This is specific to my
Polycom 335s and my FIOS router, but probably has much broader implications
to NAT in general.

 

I was searching for a way to eliminate SIP ALG on my router since it is
disdained by many.  I was successful.  The main difference was that I had to
set each phones local SIP signaling port to a unique number.  When any two
used the same port (like 5060), the second one would not register.  It
appears that some routers (I suspect many routers) do not map a single
external port to multiple local destinations well.  My extensions were 101
to 106, so I set the local SIP ports to 5101 to 5106.  They still all
connect to the same SERVER port (5060).  This is just helping the NAT'ed
Router determine the route of external packets back to the intended device
(phone).  I still had the 2 NDLB settings on for these user profiles as
mentioned below.

 

Just FYI: On the polycom 335s, the setting through the web interface are
SETTINGS=>SIP local port #  and SETTINGS=>NETWORK=>NAT sip signaling port.
They should match.

 

I cannot swear this, but I am pretty sure I had a similar issue with
multiple lines on CISCO 504Gs.  Each line had to have a unique local port.

 

It is certainly easy enough to imagine how it would be much simpler to "map"
multiple "one to one"  external ports to internal ports than to code for a
single external port having multiple internal "candidates" on the same port.
In actuality, the final IP address and PORT should be present and this
should not be a problem - but it is, A LOT of the time. 

 

HTH,

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Devoy
Sent: Sunday, December 16, 2012 11:57 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] NAT traversal - the final say..!

 

I have spent many hours working on NAT issues on client end, my server has a
public address.  

 

With CISCO brand phones I did not need any non-standards compliant settings,
just turning on all the choices in the CISCO web setup NAT section.
However, with Polycom 335 phones (as of Dec 2012) I could not get registered
or get audio without the following:

* NDLB-connectile-dysfunction

* NDLB-force-rport

* Enable SIP ALG on my FIOS router.

With those setting however, this has worked perfectly.  Also note that when
I turned on SIP ALG, my Cisco phones quite working until I added the NDLB
parameter/variable to the Cisco <user> in the directory.   They seem to be
quite complimentary but seem be requirements for each other.

 

I really tried to stay away from SIP ALG because so many posts were so
negative about it.  Without the NDLB "flags" I could never see any
difference when enabling SIP ALG.  The combination for me has been
fantastic.

 

HOWEVER, since there are so many different versions of "success" in the IRC
and Wiki,  I am pretty sure that other router brands with different SIP ALG
implementations and/or other phone brands or even firmware versions may need
different configurations.  It is almost like we just need a checklist that
says try these combinations until you find one that fits your site.

 

HTH,

sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cal
Leeming [Simplicity Media Ltd]
Sent: Sunday, December 16, 2012 11:15 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] NAT traversal - the final say..!

 

Any and all feedback on this thread would be much welcomed.

 

Hello,

 

There seems to be a large number of discussions surrounding NAT traversal,
as well as lots of documentation, but with no concrete answers. 

 

The NAT related wiki documentation is tedious, and depending on the outcome
of this thread, I'd like to spend some time cleaning it up.

 

The most common problem (the same as ours) was having a router with broken
ALG and a softphone that does not seem to work with STUN.

 

The following REGISTER is sent from a phone.

 

REGISTER sip:1.2.3.4:5060 SIP/2.0

Via: SIP/2.0/UDP
192.168.1.102:57787;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>

To: "foxx"<sip:2000 at 1.2.3.4:5060>

From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448

Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.

CSeq: 7 REGISTER

Expires: 120

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
REFER, INFO, MESSAGE

Supported: replaces

User-Agent: 3CXPhone 6.0.25732.0

Content-Length: 0

 

As you can see, the client's public IP is not specified anywhere. FreeSWITCH
offers several ways around this, the main ones being;

 

* NDLB-connectile-dysfunction

* NDLB-force-rport

* apply-nat-acl

* sip-force-contact

 

The one that has worked in our case was "NDLB-connectile-dysfunction"
(otherwise known as NAT HACK), however there seems to be a lot of negative
comments about using this.

 

>From what I can tell, the general argument is that NAT HACK is considered a
non RFC compliant hack, and the SIP phones should be doing a better job of
keeping to the RFCs.

 

In principle, this is a fair argument - but in practise, it's not a
reasonable assumption that all phones are RFC compliant, and (imho) not a
reasonable argument to have this functionality disabled by default.

 

So, I'd like to present the following arguments;

 

* Are there any other negative aspects about using
NDLB-connectile-dysfunction, other than it is a non compliant RFC hack?

 

* Why is NDLB-connectile-dysfunction not enabled by default when certain
conditions are met? In the event that FreeSWITCH receives a REGISTER from a
phone specifying a Contact/Via as 192.168.0.0/16, but received on a public
IP, then it should be obvious that NAT is broken and automatically try to
circumvent it.

 

* People seem to get confused between server side and client side NAT
problems, and that they both need to be resolved in a different way. The
documentation doesn't seem to reflect this clearly.


_________________________________________________________________________
Professional FreeSWITCH Consulting 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




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org



-------------- next part --------------
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From sdevoy at bizfocused.com  Fri Dec 21 23:53:03 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 21 Dec 2012 15:53:03 -0500
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAKzWOxUqY-mOmXOaszVEFudcg+z=PxD62DV+AT9VQuZ1h7MnaQ@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>	<480f01cddbae$6d7d67e0$487837a0$@bizfocused.com>	<047501cddf87$58d11990$0a734cb0$@bizfocused.com>
	<CAKzWOxUqY-mOmXOaszVEFudcg+z=PxD62DV+AT9VQuZ1h7MnaQ@mail.gmail.com>
Message-ID: <081301cddfbd$2bd191a0$8374b4e0$@bizfocused.com>

WIKI toll PAID!

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Friday, December 21, 2012 12:34 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] NAT traversal - the final say..!

 

Can you add this information to the wiki:

http://wiki.freeswitch.org/wiki/Interop_List#Routers

You can just about copy and paste this email right in there. Ping me if you
have any questions.

Thanks,
MC

On Fri, Dec 21, 2012 at 6:27 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

I just wanted to add a note "for the record".  This is specific to my
Polycom 335s and my FIOS router, but probably has much broader implications
to NAT in general.

 

I was searching for a way to eliminate SIP ALG on my router since it is
disdained by many.  I was successful.  The main difference was that I had to
set each phones local SIP signaling port to a unique number.  When any two
used the same port (like 5060), the second one would not register.  It
appears that some routers (I suspect many routers) do not map a single
external port to multiple local destinations well.  My extensions were 101
to 106, so I set the local SIP ports to 5101 to 5106.  They still all
connect to the same SERVER port (5060).  This is just helping the NAT'ed
Router determine the route of external packets back to the intended device
(phone).  I still had the 2 NDLB settings on for these user profiles as
mentioned below.

 

Just FYI: On the polycom 335s, the setting through the web interface are
SETTINGS=>SIP local port #  and SETTINGS=>NETWORK=>NAT sip signaling port.
They should match.

 

I cannot swear this, but I am pretty sure I had a similar issue with
multiple lines on CISCO 504Gs.  Each line had to have a unique local port.

 

It is certainly easy enough to imagine how it would be much simpler to "map"
multiple "one to one"  external ports to internal ports than to code for a
single external port having multiple internal "candidates" on the same port.
In actuality, the final IP address and PORT should be present and this
should not be a problem - but it is, A LOT of the time. 

 

HTH,

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Devoy
Sent: Sunday, December 16, 2012 11:57 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] NAT traversal - the final say..!

 

I have spent many hours working on NAT issues on client end, my server has a
public address.  

 

With CISCO brand phones I did not need any non-standards compliant settings,
just turning on all the choices in the CISCO web setup NAT section.
However, with Polycom 335 phones (as of Dec 2012) I could not get registered
or get audio without the following:

* NDLB-connectile-dysfunction

* NDLB-force-rport

* Enable SIP ALG on my FIOS router.

With those setting however, this has worked perfectly.  Also note that when
I turned on SIP ALG, my Cisco phones quite working until I added the NDLB
parameter/variable to the Cisco <user> in the directory.   They seem to be
quite complimentary but seem be requirements for each other.

 

I really tried to stay away from SIP ALG because so many posts were so
negative about it.  Without the NDLB "flags" I could never see any
difference when enabling SIP ALG.  The combination for me has been
fantastic.

 

HOWEVER, since there are so many different versions of "success" in the IRC
and Wiki,  I am pretty sure that other router brands with different SIP ALG
implementations and/or other phone brands or even firmware versions may need
different configurations.  It is almost like we just need a checklist that
says try these combinations until you find one that fits your site.

 

HTH,

sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cal
Leeming [Simplicity Media Ltd]
Sent: Sunday, December 16, 2012 11:15 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] NAT traversal - the final say..!

 

Any and all feedback on this thread would be much welcomed.

 

Hello,

 

There seems to be a large number of discussions surrounding NAT traversal,
as well as lots of documentation, but with no concrete answers. 

 

The NAT related wiki documentation is tedious, and depending on the outcome
of this thread, I'd like to spend some time cleaning it up.

 

The most common problem (the same as ours) was having a router with broken
ALG and a softphone that does not seem to work with STUN.

 

The following REGISTER is sent from a phone.

 

REGISTER sip:1.2.3.4:5060 SIP/2.0

Via: SIP/2.0/UDP
192.168.1.102:57787;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>

To: "foxx"<sip:2000 at 1.2.3.4:5060>

From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448

Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.

CSeq: 7 REGISTER

Expires: 120

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
REFER, INFO, MESSAGE

Supported: replaces

User-Agent: 3CXPhone 6.0.25732.0

Content-Length: 0

 

As you can see, the client's public IP is not specified anywhere. FreeSWITCH
offers several ways around this, the main ones being;

 

* NDLB-connectile-dysfunction

* NDLB-force-rport

* apply-nat-acl

* sip-force-contact

 

The one that has worked in our case was "NDLB-connectile-dysfunction"
(otherwise known as NAT HACK), however there seems to be a lot of negative
comments about using this.

 

>From what I can tell, the general argument is that NAT HACK is considered a
non RFC compliant hack, and the SIP phones should be doing a better job of
keeping to the RFCs.

 

In principle, this is a fair argument - but in practise, it's not a
reasonable assumption that all phones are RFC compliant, and (imho) not a
reasonable argument to have this functionality disabled by default.

 

So, I'd like to present the following arguments;

 

* Are there any other negative aspects about using
NDLB-connectile-dysfunction, other than it is a non compliant RFC hack?

 

* Why is NDLB-connectile-dysfunction not enabled by default when certain
conditions are met? In the event that FreeSWITCH receives a REGISTER from a
phone specifying a Contact/Via as 192.168.0.0/16, but received on a public
IP, then it should be obvious that NAT is broken and automatically try to
circumvent it.

 

* People seem to get confused between server side and client side NAT
problems, and that they both need to be resolved in a different way. The
documentation doesn't seem to reflect this clearly.


_________________________________________________________________________
Professional FreeSWITCH Consulting 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




-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org



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From spencer at 5ninesolutions.com  Sat Dec 22 01:22:23 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 21 Dec 2012 14:22:23 -0800
Subject: [Freeswitch-users] t.38 Call Flow Information
In-Reply-To: <CAKbxfG_6dJmuLu0EJb0fVy+Lx=G3UnjuS42MWuhdFNGhn-aDjg@mail.gmail.com>
References: <AAD54836-F121-4190-A473-B64D9A9E4612@5ninesolutions.com>
	<50D43159.2050108@coppice.org>
	<ca70e0ee-07f4-4dc4-9309-8e7d16ffde39@blur>
	<CAKbxfG_6dJmuLu0EJb0fVy+Lx=G3UnjuS42MWuhdFNGhn-aDjg@mail.gmail.com>
Message-ID: <4654C420-843B-4C8A-8D41-D72AB1436F85@5ninesolutions.com>

I tried tweaking this and it didn't seem to have any effect.  I'm calling t38_gateway like this:
<action application="set" data="execute_on_answer=t38_gateway self"/>
before the bridge.

Is there anything I should be doing to speed up the detection?  I should note that is on the receiving end.

The re-invite from rxfax worked without a hitch.  Would this suggest the default req_counter value is ok?


On Dec 21, 2012, at 9:12 AM, Anthony Minessale wrote:

> you can tweak mod_spandsp_fax.c:1391
> 
> req_counter is how many packets to wait before sending re-invite.
> 
> You can play with reducing this number and if its effective it could be made into a configurable param.
> 
> 
> 
> On Fri, Dec 21, 2012 at 10:42 AM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> Hi Steve,
> That was my take as well.   I've posted packet captures and logs here:
> http://jira.freeswitch.org/browse/FS-4957
> 
> Is there any way to speed up the ReINVITE?
> 
> Thanks,
> Spencer
> 
> 
> -----Original message-----
> From: Steve Underwood <steveu at coppice.org>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Fri, Dec 21, 2012 09:57:07 GMT+00:00
> Subject: Re: [Freeswitch-users] t.38 Call Flow Information
> 
> Hi,
> 
> It looks like your reinvite occurred extremely late, and the FAX was 
> already deep in progress as an audio exchange.
> 
> Steve
> 
> 
> On 12/21/2012 03:14 AM, Spencer Thomason wrote:
> > Hello,
> > I've been analyzing some Wireshark traces to get a better grasp on the t.38 gateway process. I'm using t38_gateway to convert SIP t38 to TDM audio on the receiving end. I'm struggling to understand the following call flow. I've excluded the audio portion for brevity but t38 is negotiated correctly by both endpoints (FS on our end and Acme on the other). Note the absence of a DIS message and the PPS prior to CFR.  Can anyone shed any light on this?
> >
> > Thanks in advance,
> > Spencer
> >
> >
> > Acme FreeSWITCH
> > |1716.691635| INVITE SDP (t38) | |SIP Request
> > | |(5060) <------------------ (5070) | |
> > |1716.719556| 100 trying -- your call is important to us | |SIP Status
> > | |(5060) ------------------> (5070) | |
> > |1716.830431| | no-signal | |t38:t30 Ind:no-signal
> > | | |(21440) <------------------ (15876) |
> > |1716.906132| 200 OK SDP (t38) | |SIP Status
> > | |(5060) ------------------> (5070) | |
> > |1716.912839| ACK | | |SIP Request
> > | |(5060) <------------------ (5070) | |
> > |1716.973011| | no-signal | |t38:t30 Ind:no-signal
> > | | |(21440) ------------------> (15876) |
> > |1717.450595| | v21-preamble |t38:t30 Ind:v21-preamble
> > | | |(21440) <------------------ (15876) |
> > |1717.730869| | PPS | |t38:v21:HDLC:Partial Page Signal
> > | | |(21440) <------------------ (15876) |
> > |1717.780811| | no-signal | |t38:t30 Ind:no-signal
> > | | |(21440) <------------------ (15876) |
> > |1718.130798| | v17-14400-long-training |t38:t30 Ind:v17-14400-long-training
> > | | |(21440) <------------------ (15876) |
> > |1719.541095| | t4-non-ecm-data:v17-14400 |t38:t4-non-ecm-data:v17-14400 Duration: 1.51s No packet lost
> > | | |(21440) <------------------ (15876) |
> > |1721.060850| | no-signal | |t38:t30 Ind:no-signal
> > | | |(21440) <------------------ (15876) |
> > |1722.573572| | v21-preamble |t38:t30 Ind:v21-preamble
> > | | |(21440) ------------------> (15876) |
> > |1723.593652| | CFR | |t38:v21:HDLC:Confirmation To Receive
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Dec 22 01:34:25 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Dec 2012 16:34:25 -0600
Subject: [Freeswitch-users] t.38 Call Flow Information
In-Reply-To: <4654C420-843B-4C8A-8D41-D72AB1436F85@5ninesolutions.com>
References: <AAD54836-F121-4190-A473-B64D9A9E4612@5ninesolutions.com>
	<50D43159.2050108@coppice.org>
	<ca70e0ee-07f4-4dc4-9309-8e7d16ffde39@blur>
	<CAKbxfG_6dJmuLu0EJb0fVy+Lx=G3UnjuS42MWuhdFNGhn-aDjg@mail.gmail.com>
	<4654C420-843B-4C8A-8D41-D72AB1436F85@5ninesolutions.com>
Message-ID: <CAKbxfG8Ai8K-kovORe+KPWPeviif3JgqNaaiMjKZ0O64VMwpJA@mail.gmail.com>

if you dont need actual detection you can skip it with nocng and jump right
into the processing,
You'll have to eyeball it but it should detect it right away, take note of
it by watching the logs.



On Fri, Dec 21, 2012 at 4:22 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> I tried tweaking this and it didn't seem to have any effect.  I'm calling
> t38_gateway like this:
> <action application="set" data="execute_on_answer=t38_gateway self"/>
> before the bridge.
>
> Is there anything I should be doing to speed up the detection?  I should
> note that is on the receiving end.
>
> The re-invite from rxfax worked without a hitch.  Would this suggest the
> default req_counter value is ok?
>
>
> On Dec 21, 2012, at 9:12 AM, Anthony Minessale wrote:
>
> you can tweak mod_spandsp_fax.c:1391
>
> req_counter is how many packets to wait before sending re-invite.
>
> You can play with reducing this number and if its effective it could be
> made into a configurable param.
>
>
>
> On Fri, Dec 21, 2012 at 10:42 AM, Spencer Thomason <
> spencer at 5ninesolutions.com> wrote:
>
>> Hi Steve,
>> That was my take as well.   I've posted packet captures and logs here:
>> http://jira.freeswitch.org/browse/FS-4957
>>
>> Is there any way to speed up the ReINVITE?
>>
>> Thanks,
>> Spencer
>>
>>
>> -----Original message-----
>>
>> *From: *Steve Underwood <steveu at coppice.org>*
>> To: *freeswitch-users at lists.freeswitch.org*
>> Sent: *Fri, Dec 21, 2012 09:57:07 GMT+00:00*
>> Subject: *Re: [Freeswitch-users] t.38 Call Flow Information
>>
>> Hi,
>>
>> It looks like your reinvite occurred extremely late, and the FAX was
>> already deep in progress as an audio exchange.
>>
>> Steve
>>
>>
>> On 12/21/2012 03:14 AM, Spencer Thomason wrote:
>> > Hello,
>> > I've been analyzing some Wireshark traces to get a better grasp on the
>> t.38 gateway process. I'm using t38_gateway to convert SIP t38 to TDM audio
>> on the receiving end. I'm struggling to understand the following call flow.
>> I've excluded the audio portion for brevity but t38 is negotiated correctly
>> by both endpoints (FS on our end and Acme on the other). Note the absence
>> of a DIS message and the PPS prior to CFR. Can anyone shed any light on
>> this?
>> >
>> > Thanks in advance,
>> > Spencer
>> >
>> >
>> > Acme FreeSWITCH
>> > |1716.691635| INVITE SDP (t38) | |SIP Request
>> > | |(5060) <------------------ (5070) | |
>> > |1716.719556| 100 trying -- your call is important to us | |SIP Status
>> > | |(5060) ------------------> (5070) | |
>> > |1716.830431| | no-signal | |t38:t30 Ind:no-signal
>> > | | |(21440) <------------------ (15876) |
>> > |1716.906132| 200 OK SDP (t38) | |SIP Status
>> > | |(5060) ------------------> (5070) | |
>> > |1716.912839| ACK | | |SIP Request
>> > | |(5060) <------------------ (5070) | |
>> > |1716.973011| | no-signal | |t38:t30 Ind:no-signal
>> > | | |(21440) ------------------> (15876) |
>> > |1717.450595| | v21-preamble |t38:t30 Ind:v21-preamble
>> > | | |(21440) <------------------ (15876) |
>> > |1717.730869| | PPS | |t38:v21:HDLC:Partial Page Signal
>> > | | |(21440) <------------------ (15876) |
>> > |1717.780811| | no-signal | |t38:t30 Ind:no-signal
>> > | | |(21440) <------------------ (15876) |
>> > |1718.130798| | v17-14400-long-training |t38:t30
>> Ind:v17-14400-long-training
>> > | | |(21440) <------------------ (15876) |
>> > |1719.541095| | t4-non-ecm-data:v17-14400
>> |t38:t4-non-ecm-data:v17-14400 Duration: 1.51s No packet lost
>> > | | |(21440) <------------------ (15876) |
>> > |1721.060850| | no-signal | |t38:t30 Ind:no-signal
>> > | | |(21440) <------------------ (15876) |
>> > |1722.573572| | v21-preamble |t38:t30 Ind:v21-preamble
>> > | | |(21440) ------------------> (15876) |
>> > |1723.593652| | CFR | |t38:v21:HDLC:Confirmation To Receive
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Anthony Minessale II

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
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From cal.leeming at simplicitymedialtd.co.uk  Sat Dec 22 03:06:06 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sat, 22 Dec 2012 00:06:06 +0000
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <079801cddfb5$a07502b0$e15f0810$@bizfocused.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<480f01cddbae$6d7d67e0$487837a0$@bizfocused.com>
	<047501cddf87$58d11990$0a734cb0$@bizfocused.com>
	<CAKzWOxUqY-mOmXOaszVEFudcg+z=PxD62DV+AT9VQuZ1h7MnaQ@mail.gmail.com>
	<079801cddfb5$a07502b0$e15f0810$@bizfocused.com>
Message-ID: <CAHKQagFfjkwrfneQXe8uXV+r7aEfsvLmq=kdAo0Ekd51CjrDrA@mail.gmail.com>

Absolutely, I'm sitting back and digesting all the info coming in, with a
view to rewriting the entire NAT section of the wiki - the feedback so far
has been fantastic, much better than I was expecting, so thank you everyone.

@Anthony, I am very curious though to hear your section on NAT from the
next book, as I feel it would really help when updating the wiki... would
you consider sending me a draft of just this section alone, before the book
is released?? I have some spare time coming up in the holidays, so would be
good to have all this to hand. If not, no worries, but thought I'd try!

Thanks

Cal

On Fri, Dec 21, 2012 at 7:59 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Hey MC,****
>
> ** **
>
> Cal Leeming started this thread so he could gather info on NAT and add it
> to the wiki in a structured way.  I hope this will be included in that NAT
> section in some meaningful context.  I will certainly add it to the router
> section as well.****
>
> ****
>
> Sean****
>
> ** **
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Friday, December 21, 2012 12:34 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] NAT traversal - the final say..!****
>
> ** **
>
> Can you add this information to the wiki:
>
> http://wiki.freeswitch.org/wiki/Interop_List#Routers
>
> You can just about copy and paste this email right in there. Ping me if
> you have any questions.
>
> Thanks,
> MC****
>
> On Fri, Dec 21, 2012 at 6:27 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> ****
>
> I just wanted to add a note ?for the record?.  This is specific to my
> Polycom 335s and my FIOS router, but probably has much broader implications
> to NAT in general.****
>
>  ****
>
> I was searching for a way to eliminate SIP ALG on my router since it is
> disdained by many.  I was successful.  The main difference was that I had
> to set each phones local SIP signaling port to a unique number.  When any
> two used the same port (like 5060), the second one would not register.  It
> appears that some routers (I suspect many routers) do not map a single
> external port to multiple local destinations well.  My extensions were 101
> to 106, so I set the local SIP ports to 5101 to 5106.  They still all
> connect to the same SERVER port (5060).  This is just helping the NAT?ed
> Router determine the route of external packets back to the intended device
> (phone).  I still had the 2 NDLB settings on for these user profiles as
> mentioned below.****
>
>  ****
>
> Just FYI: On the polycom 335s, the setting through the web interface are
> SETTINGS=>SIP local port #  and SETTINGS=>NETWORK=>NAT sip signaling port.
> They should match.****
>
>  ****
>
> I cannot swear this, but I am pretty sure I had a similar issue with
> multiple lines on CISCO 504Gs.  Each line had to have a unique local port.
> ****
>
>  ****
>
> It is certainly easy enough to imagine how it would be much simpler to
> ?map? multiple ?one to one?  external ports to internal ports than to code
> for a single external port having multiple internal ?candidates? on the
> same port.  In actuality, the final IP address and PORT should be present
> and this should not be a problem ? but it is, A LOT of the time. ****
>
>  ****
>
> HTH,****
>
> Sean****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sean Devoy
> *Sent:* Sunday, December 16, 2012 11:57 AM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] NAT traversal - the final say..!****
>
>  ****
>
> I have spent many hours working on *NAT issues on client end*, my server
> has a public address.  ****
>
>  ****
>
> With CISCO brand phones I did not need any non-standards compliant
> settings, just turning on all the choices in the CISCO web setup NAT
> section.  However, with Polycom 335 phones (as of Dec 2012) I could not get
> registered or get audio without the following:****
>
> * NDLB-connectile-dysfunction****
>
> * NDLB-force-rport****
>
> * Enable SIP ALG on my FIOS router.****
>
> With those setting however, this has worked perfectly.  Also note that
> when I turned on SIP ALG, my Cisco phones quite working until I added the
> NDLB parameter/variable to the Cisco <user> in the directory.   They seem
> to be quite complimentary but seem be requirements for each other.****
>
>  ****
>
> I really tried to stay away from SIP ALG because so many posts were so
> negative about it.  Without the NDLB ?flags? I could never see any
> difference when enabling SIP ALG.  The combination for me has been
> fantastic.****
>
>  ****
>
> HOWEVER, since there are so many different versions of ?success? in the
> IRC and Wiki,  I am pretty sure that other router brands with different SIP
> ALG implementations and/or other phone brands or even firmware versions may
> need different configurations.  It is almost like we just need a checklist
> that says try these combinations until you find one that fits your site.**
> **
>
>  ****
>
> HTH,****
>
> sean****
>
>  ****
>
> *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 *Cal Leeming [Simplicity Media Ltd]
> *Sent:* Sunday, December 16, 2012 11:15 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] NAT traversal - the final say..!****
>
>  ****
>
> *Any and all feedback on this thread would be much welcomed.*****
>
>  ****
>
> Hello,****
>
>  ****
>
> There seems to be a large number of discussions surrounding NAT traversal,
> as well as lots of documentation, but with no concrete answers. ****
>
>  ****
>
> The NAT related wiki documentation is tedious, and depending on the
> outcome of this thread, I'd like to spend some time cleaning it up.****
>
>  ****
>
> The most common problem (the same as ours) was having a router with broken
> ALG and a softphone that does not seem to work with STUN.****
>
>  ****
>
> The following REGISTER is sent from a phone.****
>
>  ****
>
> REGISTER sip:1.2.3.4:5060 SIP/2.0****
>
> Via: SIP/2.0/UDP 192.168.1.102:57787
> ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport****
>
> Max-Forwards: 70****
>
> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>****
>
> To: "foxx"<sip:2000 at 1.2.3.4:5060>****
>
> From: "foxx"<sip:2000 at 1.2.3.4:5060>;tag=83311448****
>
> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.****
>
> CSeq: 7 REGISTER****
>
> Expires: 120****
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
> REFER, INFO, MESSAGE****
>
> Supported: replaces****
>
> User-Agent: 3CXPhone 6.0.25732.0****
>
> Content-Length: 0****
>
>  ****
>
> As you can see, the client's public IP is not specified
> anywhere. FreeSWITCH offers several ways around this, the main ones being;
> ****
>
>  ****
>
> * NDLB-connectile-dysfunction****
>
> * NDLB-force-rport****
>
> * apply-nat-acl****
>
> * sip-force-contact****
>
>  ****
>
> The one that has worked in our case was "NDLB-connectile-dysfunction"
> (otherwise known as NAT HACK), however there seems to be a lot of negative
> comments about using this.****
>
>  ****
>
> From what I can tell, the general argument is that NAT HACK is considered
> a non RFC compliant hack, and the SIP phones should be doing a better job
> of keeping to the RFCs.****
>
>  ****
>
> In principle, this is a fair argument - but in practise, it's not a
> reasonable assumption that all phones are RFC compliant, and (imho) not a
> reasonable argument to have this functionality disabled by default.****
>
>  ****
>
> So, I'd like to present the following arguments;****
>
>  ****
>
> * Are there any other negative aspects about
> using NDLB-connectile-dysfunction, other than it is a non compliant RFC
> hack?****
>
>  ****
>
> * Why is NDLB-connectile-dysfunction not enabled by default when certain
> conditions are met? In the event that FreeSWITCH receives a REGISTER from a
> phone specifying a Contact/Via as 192.168.0.0/16, but received on a
> public IP, then it should be obvious that NAT is broken and automatically
> try to circumvent it.****
>
>  ****
>
> * People seem to get confused between server side and client side NAT
> problems, and that they both need to be resolved in a different way. The
> documentation doesn't seem to reflect this clearly.****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
> ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 mario_fs at mgtech.com  Sat Dec 22 03:39:41 2012
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 21 Dec 2012 16:39:41 -0800
Subject: [Freeswitch-users] Freeswitch TLS and Yealink t26p
In-Reply-To: <CAKzWOxXDUEJh4Kq183qFWsB_A933j+HtoShx9LKyLa42fT2LRA@mail.gmail.com>
References: <1356082510.23186.154.camel@marces.madrid.commsmundi.com>
	<1356090896.23186.170.camel@marces.madrid.commsmundi.com>
	<CAKzWOxXDUEJh4Kq183qFWsB_A933j+HtoShx9LKyLa42fT2LRA@mail.gmail.com>
Message-ID: <F773C6B8-0EFC-4131-B421-EFD7D05FE984@mgtech.com>

I have some of the new Yealink T32s I will add to the wiki. So far no issues with FreeSwitch. We really like them much better than the Cisco SPA962 they replaced.
Mario G

On Dec 21, 2012, at 9:41 AM, Michael Collins wrote:

> Antonio,
> 
> It looks like we have a number of phones listed here:
> 
> http://wiki.freeswitch.org/wiki/Tls#SIP_TLS_Device_Interoperability
> 
> However, Yealink is not among them. I've created a spot for Yealinks here:
> 
> http://wiki.freeswitch.org/wiki/Interop_List#Yealink_TLS_Configuration
> 
> Could you put what you've learned into that spot on the wiki?
> 
> Thanks!
> -MC
> 
> 
> On Fri, Dec 21, 2012 at 3:54 AM, Antonio <asilva at wirelessmundi.com> wrote:
> Answer to myself....
> 
> In the yealink configuration, in the account parameters, the "transport" must be force to TLS.  
> 
> I don't know why it just works.... Before i was using DNS-SRV, that should be the first option, yealink should have some issue here... i will report to them.
> 
> 
> Thanks,
> Ant?nio
> 
> 
> On Fri, 2012-12-21 at 10:35 +0100, Antonio wrote:
>> Hi,
>> 
>> I'm trying to register a yealink with TLS, using my one certificates.
>> 
>> I follow the wiki and In fs i have both agent.pem and cafile.pem . I install in the phone the root certificate.
>> 
>> But when i try to register, i have (tport log):
>> 
>> 
>> tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x808fb0) msg 0x7fe9d0aa8180 from (udp/192.168.10.1:5060) has 340 bytes, veclen = 1
>> tport.c:3004 tport_deliver() tport_deliver(0x808fb0): msg 0x7fe9d0aa8180 (340 bytes) from udp/192.168.10.23:5060/sip next=(nil)
>> tport.c:4202 tport_release() tport_release(0x808fb0): 0x7fe9d01142f0 by 0x7fe9d025d920 with 0x7fe9d0aa8180
>> tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fe9c802aad0): events IN
>> tport.c:869 tport_alloc_secondary() tport_alloc_secondary(0x7fe9c802aad0): new secondary tport 0x7fe9c03e8450
>> tport_type_tls.c:603 tport_tls_accept() tport_tls_accept(0x7fe9c03e8450): new connection from tls/192.168.10.36:48754/sips
>> tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events NEGOTIATING
>> tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events NEGOTIATING
>> tport_tls.c:526 tls_post_connection_check() tls_post_connection_check(0x7fe9c03e8450): Peer did not provide X.509 Certificate.
>> 
>> 
>> 
>> I could make it work and have a register in the tls profile when i check on the phone the option in Security->Trusted Certificates: "Only Accept Trusted Certificates: DISABLED".
>> Could it be some bug in the yealink, or I?m missing something in the conf...
>> 
>> Another question, is there any problem if i choose to use this configuration... since is the phone that ignores the certificate and the validation is done by the server and not by the client. 
>> 
>> Can you help me?
>> 
>> Thanks,
>> Ant?nio
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
> 
> -- 
> 
> Un cordial saludo / Best regards, 
> 
>  _________________________
> 
> Ant?nio Silva
> 
> E-mail:asilva at wirelessmundi.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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 sdevoy at bizfocused.com  Sat Dec 22 03:41:48 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 21 Dec 2012 19:41:48 -0500
Subject: [Freeswitch-users] Allow calls from Gateway based on GW IP Address
Message-ID: <097601cddfdd$20a62be0$61f283a0$@bizfocused.com>

HI All,

 

I am trying to setup Vitelity as a FS Gateway.  The wiki is 2 years out of
date, the menus @Vitelity have all changed and I have been unable to get the
FS setup to work.

 

Vitelity recommends IP Based Authentication.  I have read the wiki regarding
this and I am simply thrashing around.

 

So, here is the pastbin of the global siptrace:
http://pastebin.freeswitch.org/20354

Sorry about all the extra line feeds, they are not visible in notepad!

 

I just want to know how to setup FS to all accept all calls from a FQDN or
IP Address and route them to a specific context (like "from-trunk").

 

I will pay the WIKI TOLL when we get this working.

 

Thanks,

Sean

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From spencer at 5ninesolutions.com  Sat Dec 22 04:12:56 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 21 Dec 2012 17:12:56 -0800
Subject: [Freeswitch-users] t.38 Call Flow Information
In-Reply-To: <CAKbxfG8Ai8K-kovORe+KPWPeviif3JgqNaaiMjKZ0O64VMwpJA@mail.gmail.com>
References: <AAD54836-F121-4190-A473-B64D9A9E4612@5ninesolutions.com>
	<50D43159.2050108@coppice.org>
	<ca70e0ee-07f4-4dc4-9309-8e7d16ffde39@blur>
	<CAKbxfG_6dJmuLu0EJb0fVy+Lx=G3UnjuS42MWuhdFNGhn-aDjg@mail.gmail.com>
	<4654C420-843B-4C8A-8D41-D72AB1436F85@5ninesolutions.com>
	<CAKbxfG8Ai8K-kovORe+KPWPeviif3JgqNaaiMjKZ0O64VMwpJA@mail.gmail.com>
Message-ID: <4CFA050D-707D-4C6F-AEFB-38A1F2ACDC5B@5ninesolutions.com>

Hi Anthony,
I got successful transmission working.  
I'm testing like this:
Fax Machine --> Linksys SPA-3102 --t.38-- > FS1 --audio --> FS2 (rxfax)
In analyzing the logs its seems FS2 was reporting the ECM was negotiated where the t38 side from was negotiating ECM off because the Linksys adapter doesn't do ECM.  By explicitly disabling ECM on the receiving end (FS2) everything is working.  Should it not negotiate ECM off all the way through?

Also, I'm still seeing late re-invites so I tried to do:
<action application="set" data="execute_on_answer=t38_gateway self nocng"/>

That fails with:
2012-12-21 17:00:19.617866 [WARNING] mod_spandsp_fax.c:1578 sofia/default/1002 at 10.59.1.195 Cannot locate channel with uuid f4b7afd8-4bd2-11e2-b9db-2fac9796034a

I took a look at mod_spandsp.c and can find any reason why that wouldn't work.  Am I missing something or should I file a jira?  The strange thing is that I use nocng with sip_execute_on_image and it works fine.


On Dec 21, 2012, at 2:34 PM, Anthony Minessale wrote:

> if you dont need actual detection you can skip it with nocng and jump right into the processing,  
> You'll have to eyeball it but it should detect it right away, take note of it by watching the logs.
> 
> 
> 
> On Fri, Dec 21, 2012 at 4:22 PM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> I tried tweaking this and it didn't seem to have any effect.  I'm calling t38_gateway like this:
> <action application="set" data="execute_on_answer=t38_gateway self"/>
> before the bridge.
> 
> Is there anything I should be doing to speed up the detection?  I should note that is on the receiving end.
> 
> The re-invite from rxfax worked without a hitch.  Would this suggest the default req_counter value is ok?
> 
> 
> On Dec 21, 2012, at 9:12 AM, Anthony Minessale wrote:
> 
>> you can tweak mod_spandsp_fax.c:1391
>> 
>> req_counter is how many packets to wait before sending re-invite.
>> 
>> You can play with reducing this number and if its effective it could be made into a configurable param.
>> 
>> 
>> 
>> On Fri, Dec 21, 2012 at 10:42 AM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
>> Hi Steve,
>> That was my take as well.   I've posted packet captures and logs here:
>> http://jira.freeswitch.org/browse/FS-4957
>> 
>> Is there any way to speed up the ReINVITE?
>> 
>> Thanks,
>> Spencer
>> 
>> 
>> -----Original message-----
>> From: Steve Underwood <steveu at coppice.org>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Fri, Dec 21, 2012 09:57:07 GMT+00:00
>> Subject: Re: [Freeswitch-users] t.38 Call Flow Information
>> 
>> Hi,
>> 
>> It looks like your reinvite occurred extremely late, and the FAX was 
>> already deep in progress as an audio exchange.
>> 
>> Steve
>> 
>> 
>> On 12/21/2012 03:14 AM, Spencer Thomason wrote:
>> > Hello,
>> > I've been analyzing some Wireshark traces to get a better grasp on the t.38 gateway process. I'm using t38_gateway to convert SIP t38 to TDM audio on the receiving end. I'm struggling to understand the following call flow. I've excluded the audio portion for brevity but t38 is negotiated correctly by both endpoints (FS on our end and Acme on the other). Note the absence of a DIS message and the PPS prior to CFR. Can anyone shed any light on this?
>> >
>> > Thanks in advance,
>> > Spencer
>> >
>> >
>> > Acme FreeSWITCH
>> > |1716.691635| INVITE SDP (t38) | |SIP Request
>> > | |(5060) <------------------ (5070) | |
>> > |1716.719556| 100 trying -- your call is important to us | |SIP Status
>> > | |(5060) ------------------> (5070) | |
>> > |1716.830431| | no-signal | |t38:t30 Ind:no-signal
>> > | | |(21440) <------------------ (15876) |
>> > |1716.906132| 200 OK SDP (t38) | |SIP Status
>> > | |(5060) ------------------> (5070) | |
>> > |1716.912839| ACK | | |SIP Request
>> > | |(5060) <------------------ (5070) | |
>> > |1716.973011| | no-signal | |t38:t30 Ind:no-signal
>> > | | |(21440) ------------------> (15876) |
>> > |1717.450595| | v21-preamble |t38:t30 Ind:v21-preamble
>> > | | |(21440) <------------------ (15876) |
>> > |1717.730869| | PPS | |t38:v21:HDLC:Partial Page Signal
>> > | | |(21440) <------------------ (15876) |
>> > |1717.780811| | no-signal | |t38:t30 Ind:no-signal
>> > | | |(21440) <------------------ (15876) |
>> > |1718.130798| | v17-14400-long-training |t38:t30 Ind:v17-14400-long-training
>> > | | |(21440) <------------------ (15876) |
>> > |1719.541095| | t4-non-ecm-data:v17-14400 |t38:t4-non-ecm-data:v17-14400 Duration: 1.51s No packet lost
>> > | | |(21440) <------------------ (15876) |
>> > |1721.060850| | no-signal | |t38:t30 Ind:no-signal
>> > | | |(21440) <------------------ (15876) |
>> > |1722.573572| | v21-preamble |t38:t30 Ind:v21-preamble
>> > | | |(21440) ------------------> (15876) |
>> > |1723.593652| | CFR | |t38:v21:HDLC:Confirmation To Receive
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>> 
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> 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
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Dec 22 08:25:08 2012
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 22 Dec 2012 13:25:08 +0800
Subject: [Freeswitch-users] t.38 Call Flow Information
In-Reply-To: <4CFA050D-707D-4C6F-AEFB-38A1F2ACDC5B@5ninesolutions.com>
References: <AAD54836-F121-4190-A473-B64D9A9E4612@5ninesolutions.com>
	<50D43159.2050108@coppice.org>
	<ca70e0ee-07f4-4dc4-9309-8e7d16ffde39@blur>
	<CAKbxfG_6dJmuLu0EJb0fVy+Lx=G3UnjuS42MWuhdFNGhn-aDjg@mail.gmail.com>
	<4654C420-843B-4C8A-8D41-D72AB1436F85@5ninesolutions.com>
	<CAKbxfG8Ai8K-kovORe+KPWPeviif3JgqNaaiMjKZ0O64VMwpJA@mail.gmail.com>
	<4CFA050D-707D-4C6F-AEFB-38A1F2ACDC5B@5ninesolutions.com>
Message-ID: <50D54434.3030808@coppice.org>

Hi,

If a T.38 gateway or terminal is not prepared to handle ECM, it needs to 
edit the DIS messages to say ECM is not allowed. A problem occurs if you 
have late negotiation of T.38, because the T.38 message modifier is not 
in the path when it is needed, and doesn't edit the messages.

Steve

On 12/22/2012 09:12 AM, Spencer Thomason wrote:
> Hi Anthony,
> I got successful transmission working.
> I'm testing like this:
> Fax Machine --> Linksys SPA-3102 --t.38-- > FS1 --audio --> FS2 (rxfax)
> In analyzing the logs its seems FS2 was reporting the ECM was 
> negotiated where the t38 side from was negotiating ECM off because the 
> Linksys adapter doesn't do ECM.  By explicitly disabling ECM on the 
> receiving end (FS2) everything is working.  Should it not negotiate 
> ECM off all the way through?
>
> Also, I'm still seeing late re-invites so I tried to do:
> <action application="set" data="execute_on_answer=t38_gateway self 
> nocng"/>
>
> That fails with:
> 2012-12-21 17:00:19.617866 [WARNING] mod_spandsp_fax.c:1578 
> sofia/default/1002 at 10.59.1.195 Cannot locate channel with uuid 
> f4b7afd8-4bd2-11e2-b9db-2fac9796034a
>
> I took a look at mod_spandsp.c and can find any reason why that 
> wouldn't work.  Am I missing something or should I file a jira?  The 
> strange thing is that I use nocng with sip_execute_on_image and it 
> works fine.
>
>
> On Dec 21, 2012, at 2:34 PM, Anthony Minessale wrote:
>
>> if you dont need actual detection you can skip it with nocng and jump 
>> right into the processing,
>> You'll have to eyeball it but it should detect it right away, take 
>> note of it by watching the logs.
>>
>>
>>
>> On Fri, Dec 21, 2012 at 4:22 PM, Spencer Thomason 
>> <spencer at 5ninesolutions.com <mailto:spencer at 5ninesolutions.com>> wrote:
>>
>>     I tried tweaking this and it didn't seem to have any effect.  I'm
>>     calling t38_gateway like this:
>>     <action application="set" data="execute_on_answer=t38_gateway self"/>
>>     before the bridge.
>>
>>     Is there anything I should be doing to speed up the detection?  I
>>     should note that is on the receiving end.
>>
>>     The re-invite from rxfax worked without a hitch.  Would this
>>     suggest the default req_counter value is ok?
>>
>>
>>     On Dec 21, 2012, at 9:12 AM, Anthony Minessale wrote:
>>
>>>     you can tweak mod_spandsp_fax.c:1391
>>>
>>>     req_counter is how many packets to wait before sending re-invite.
>>>
>>>     You can play with reducing this number and if its effective it
>>>     could be made into a configurable param.
>>>
>>>
>>>
>>>     On Fri, Dec 21, 2012 at 10:42 AM, Spencer Thomason
>>>     <spencer at 5ninesolutions.com <mailto:spencer at 5ninesolutions.com>>
>>>     wrote:
>>>
>>>         Hi Steve,
>>>         That was my take as well.   I've posted packet captures and
>>>         logs here:
>>>         http://jira.freeswitch.org/browse/FS-4957
>>>
>>>         Is there any way to speed up the ReINVITE?
>>>
>>>         Thanks,
>>>         Spencer
>>>
>>>
>>>         -----Original message-----
>>>
>>>             *From: *Steve Underwood <steveu at coppice.org
>>>             <mailto:steveu at coppice.org>>*
>>>             To: *freeswitch-users at lists.freeswitch.org
>>>             <mailto:freeswitch-users at lists.freeswitch.org>*
>>>             Sent: *Fri, Dec 21, 2012 09:57:07 GMT+00:00*
>>>             Subject: *Re: [Freeswitch-users] t.38 Call Flow Information
>>>
>>>             Hi,
>>>
>>>             It looks like your reinvite occurred extremely late, and
>>>             the FAX was
>>>             already deep in progress as an audio exchange.
>>>
>>>             Steve
>>>
>>>
>>>             On 12/21/2012 03:14 AM, Spencer Thomason wrote:
>>>             > Hello,
>>>             > I've been analyzing some Wireshark traces to get a
>>>             better grasp on the t.38 gateway process. I'm using
>>>             t38_gateway to convert SIP t38 to TDM audio on the
>>>             receiving end. I'm struggling to understand the
>>>             following call flow. I've excluded the audio portion for
>>>             brevity but t38 is negotiated correctly by both
>>>             endpoints (FS on our end and Acme on the other). Note
>>>             the absence of a DIS message and the PPS prior to CFR.
>>>             Can anyone shed any light on this?
>>>             >
>>>             > Thanks in advance,
>>>             > Spencer
>>>             >
>>>             >
>>>             > Acme FreeSWITCH
>>>             > |1716.691635| INVITE SDP (t38) | |SIP Request
>>>             > | |(5060) <------------------ (5070) | |
>>>             > |1716.719556| 100 trying -- your call is important to
>>>             us | |SIP Status
>>>             > | |(5060) ------------------> (5070) | |
>>>             > |1716.830431| | no-signal | |t38:t30 Ind:no-signal
>>>             > | | |(21440) <------------------ (15876) |
>>>             > |1716.906132| 200 OK SDP (t38) | |SIP Status
>>>             > | |(5060) ------------------> (5070) | |
>>>             > |1716.912839| ACK | | |SIP Request
>>>             > | |(5060) <------------------ (5070) | |
>>>             > |1716.973011| | no-signal | |t38:t30 Ind:no-signal
>>>             > | | |(21440) ------------------> (15876) |
>>>             > |1717.450595| | v21-preamble |t38:t30 Ind:v21-preamble
>>>             > | | |(21440) <------------------ (15876) |
>>>             > |1717.730869| | PPS | |t38:v21:HDLC:Partial Page Signal
>>>             > | | |(21440) <------------------ (15876) |
>>>             > |1717.780811| | no-signal | |t38:t30 Ind:no-signal
>>>             > | | |(21440) <------------------ (15876) |
>>>             > |1718.130798| | v17-14400-long-training |t38:t30
>>>             Ind:v17-14400-long-training
>>>             > | | |(21440) <------------------ (15876) |
>>>             > |1719.541095| | t4-non-ecm-data:v17-14400
>>>             |t38:t4-non-ecm-data:v17-14400 Duration: 1.51s No packet
>>>             lost
>>>             > | | |(21440) <------------------ (15876) |
>>>             > |1721.060850| | no-signal | |t38:t30 Ind:no-signal
>>>             > | | |(21440) <------------------ (15876) |
>>>             > |1722.573572| | v21-preamble |t38:t30 Ind:v21-preamble
>>>             > | | |(21440) ------------------> (15876) |
>>>             > |1723.593652| | CFR | |t38:v21:HDLC:Confirmation To
>>>             Receive
>>>             >
>>>             _________________________________________________________________________
>>>             > Professional FreeSWITCH Consulting Services:
>>>             > consulting at freeswitch.org
>>>             <mailto:consulting at freeswitch.org>
>>>             > http://www.freeswitchsolutions.com
>>>             <http://www.freeswitchsolutions.com/>
>>>             >
>>>             > FreeSWITCH-powered IP PBX: The CudaTel Communication
>>>             Server
>>>             >  </>
>>>             >
>>>             > 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/>
>>>             >
>>>             > 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/>
>>>             >
>>>



From bdfoster at endigotech.com  Sat Dec 22 09:41:46 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 22 Dec 2012 01:41:46 -0500
Subject: [Freeswitch-users] Allow calls from Gateway based on GW IP
	Address
In-Reply-To: <097601cddfdd$20a62be0$61f283a0$@bizfocused.com>
References: <097601cddfdd$20a62be0$61f283a0$@bizfocused.com>
Message-ID: <34B4F06F-AD4D-486E-B466-8B519AE35B9F@endigotech.com>

ACL->specific context I.e public

Sent from my iPhone

On Dec 21, 2012, at 7:41 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

> HI All,
>  
> I am trying to setup Vitelity as a FS Gateway.  The wiki is 2 years out of date, the menus @Vitelity have all changed and I have been unable to get the FS setup to work.
>  
> Vitelity recommends IP Based Authentication.  I have read the wiki regarding this and I am simply thrashing around.
>  
> So, here is the pastbin of the global siptrace:  http://pastebin.freeswitch.org/20354
> Sorry about all the extra line feeds, they are not visible in notepad!
>  
> I just want to know how to setup FS to all accept all calls from a FQDN or IP Address and route them to a specific context (like ?from-trunk?).
>  
> I will pay the WIKI TOLL when we get this working.
>  
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From itr0508ganeshkumars at gmail.com  Sat Dec 22 10:42:29 2012
From: itr0508ganeshkumars at gmail.com (ganesh kumar)
Date: Sat, 22 Dec 2012 13:12:29 +0530
Subject: [Freeswitch-users] Problem in DTMF detection with PRI ( Duplicate
	DTMF / DTMF Removal )
Message-ID: <CAJ+8-+YvsHbfKgqFHQjPPtZt2rnF37jSrFmyM4VV-ADXALOU5w@mail.gmail.com>

Hi All,

        We are using the following version of FreeSWITCH.
        Version : FreeSWITCH Version 1.0.head (git-8f2ee97 2010-12-05
17-19-28 -0600)
        Linux Kernel Version : Linux My-Host 2.6.35-22-generic-pae
#35-Ubuntu SMP Sat Oct 16 22:16:51 UTC 2010 i686 GNU/Linux
        Sangoma PRI Card : AFT A102
        Wanpipe Version: WANPIPE Release: 3.5.19
        Libsng Version: libsng_isdn-5.3.0.i686

        I am having an application to receive DTMF and process based on the
received DTMF.
        The problem is, whenever receiving DTMF some of the digits got
duplicated and sometimes some digits are missing.

        I have searched lot about this problem. And I came to know that
many people faced this issue. But I couldn't get proper solution.
        I want to know whether this problem is solved in freeswitch or not.
        If it is solved, what is the solution or workaround for this
problem.

        Note:
               I have tried disabling the core DTMF detect and enabled the
spandsp DTMF detect but it too failed some times and it is not a reliable
solution
               for my problem.

       Kindly help me out.

      Thanks in advance.

--
Regards,
Ganeshkumar.S
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From freeswitch-list at puzzled.xs4all.nl  Sat Dec 22 14:27:38 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Sat, 22 Dec 2012 12:27:38 +0100
Subject: [Freeswitch-users] Problem in DTMF detection with PRI (
 Duplicate DTMF / DTMF Removal )
In-Reply-To: <CAJ+8-+YvsHbfKgqFHQjPPtZt2rnF37jSrFmyM4VV-ADXALOU5w@mail.gmail.com>
References: <CAJ+8-+YvsHbfKgqFHQjPPtZt2rnF37jSrFmyM4VV-ADXALOU5w@mail.gmail.com>
Message-ID: <50D5992A.20108@puzzled.xs4all.nl>

On 12/22/2012 08:42 AM, ganesh kumar wrote:
> Hi All,
>
>          We are using the following version of FreeSWITCH.
>          Version : FreeSWITCH Version 1.0.head (git-8f2ee97 2010-12-05
> 17-19-28 -0600)
>          Linux Kernel Version : Linux My-Host 2.6.35-22-generic-pae
> #35-Ubuntu SMP Sat Oct 16 22:16:51 UTC 2010 i686 GNU/Linux
>          Sangoma PRI Card : AFT A102
>          Wanpipe Version: WANPIPE Release: 3.5.19
>          Libsng Version: libsng_isdn-5.3.0.i686

All your software versions are quite old. Have you tried updating to the 
latest FreeSWITCH 1.2 version to see if the problem is still there?
While you are at it you might as well update your wanpipe release to 
3.5.28 and update libsng_isdn to 7.26.2. Off course don't do this on 
your production box but test it on another box first. If you have no 
choice but to do this on your production box then at least make sure you 
have a backup of the old configs and software so you can revert if needed.

Regards,
Patrick



From bhadrarao.kankatala at gmail.com  Sat Dec 22 12:11:34 2012
From: bhadrarao.kankatala at gmail.com (veerabhadrarao`)
Date: Sat, 22 Dec 2012 14:41:34 +0530
Subject: [Freeswitch-users] (no subject)
Message-ID: <50D57946.5040501@gmail.com>

hai

I am using Freeswitch 1.2.4

currently i am working on Attendent transfer can u please help me how to 
make attendant transfer.

let when i make a call A--->B  and B-->C then how can i connect to A--->C

please help me



From ratner2 at gmail.com  Sat Dec 22 16:44:57 2012
From: ratner2 at gmail.com (bratner bratner)
Date: Sat, 22 Dec 2012 15:44:57 +0200
Subject: [Freeswitch-users] mod_sofia gateway proxy definition,
	rtmp voice codec
Message-ID: <CAHA3fyWjpEX7D3_sYnb-oRezMdN4bXPrCD14hPT=RLtxF916tQ@mail.gmail.com>

Dear List,

I'm seeking an answer to the following questions:

1. If I define a gateway under a sip profile and I do it with
"outbound-proxy" setting then the INVITE message
to that proxy will also include the proxy in  its RURI. The expected
behavior is that the message will be sent to
to the specified proxy  BUT the RURI will point to the "domain" as defined
in the gateway config.

That behavior is also correct for REGISTER request and this is the way it
actually works if I define "register-proxy".
REGISTER request go to the ip:port of the defined "register-proxy" but the
domain part in RURI will be set to the domain from
gateway configuration or "register-domain".


2. mod_rtmp is hardcoded with SPEEX. So is of course the flex client. Is
there a reason for this?
According to flex docs, PCMU/A are also supported. I also heard that the
module is not really maintained.


Regards,
bratner.
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From stephen at picardogroup.com  Sat Dec 22 17:11:01 2012
From: stephen at picardogroup.com (stephen at picardogroup.com)
Date: Sat, 22 Dec 2012 07:11:01 -0700
Subject: [Freeswitch-users] FS opening ports
Message-ID: <20121222071101.0e1bd4d5c5064b420440751b21b10e46.81311ec1f9.wbe@email13.secureserver.net>

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From anthony.minessale at gmail.com  Sat Dec 22 20:14:12 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 22 Dec 2012 11:14:12 -0600
Subject: [Freeswitch-users] FS opening ports
In-Reply-To: <20121222071101.0e1bd4d5c5064b420440751b21b10e46.81311ec1f9.wbe@email13.secureserver.net>
References: <20121222071101.0e1bd4d5c5064b420440751b21b10e46.81311ec1f9.wbe@email13.secureserver.net>
Message-ID: <CAKbxfG98s8qAT3d+rO7yRVkLbUu1OEEOBZXnpXTk2sC-YrGZQQ@mail.gmail.com>

start FS with -nonat command line arg


On Sat, Dec 22, 2012 at 8:11 AM, <stephen at picardogroup.com> wrote:

> Why does FS open ports on my FIOS router by itself?
>
> I have been seeing SIP 503and telephone calls in ring busy
>
> Is there a way to stop or limit FS from opening too many ports?
>
> localhost
> xxx.x.x.x    Verizon FiOS Service
> Tcp Any -> 4567    All Broadband Devices    Active
>     new-host
> 192.168.1.xx
>     Destination Ports 8022
> TCP Any -> 8022    All Broadband Devices    Active
>
>     192.168.1.
>     SSH
> TCP Any -> 22    All Broadband Devices    Active
>
>     192.168.1.xx:5070
>     FreeSWITCH
> UDP Any -> 5070    All Broadband Devices    Active
>
>     192.168.1.xx:5080
>     FreeSWITCH
> UDP Any -> 5080    All Broadband Devices    Active
>
>     192.168.1.xx:5060
>     FreeSWITCH
> UDP Any -> 5060    All Broadband Devices    Active
>
>     192.168.1.xx:5070
>     FreeSWITCH
> TCP Any -> 5070    All Broadband Devices    Active
>
>     192.168.1.xx:5080
>     FreeSWITCH
> TCP Any -> 5080    All Broadband Devices    Active
>
>     192.168.1.xx:5060
>     FreeSWITCH
> TCP Any -> 5060    All Broadband Devices    Active
>
>     192.168.1.xx:15952
>     FreeSWITCH
> UDP Any -> 15952    All Broadband Devices    Active
>
>     192.168.1.xx:15953
>     FreeSWITCH
> UDP Any -> 15953    All Broadband Devices    Active
>
>     192.168.1.xx:11350
>     FreeSWITCH
> UDP Any -> 11350    All Broadband Devices    Active
>
>     192.168.1.xx:11351
>     FreeSWITCH
> UDP Any -> 11351    All Broadband Devices    Active
>
>     192.168.1.xx:16052
>     FreeSWITCH
> UDP Any -> 16052    All Broadband Devices    Active
>
>     192.168.1.xx:16053
>     FreeSWITCH
> UDP Any -> 16053    All Broadband Devices    Active
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Anthony Minessale II

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
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From miha at softnet.si  Sat Dec 22 20:35:50 2012
From: miha at softnet.si (Miha)
Date: Sat, 22 Dec 2012 18:35:50 +0100
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <50D57946.5040501@gmail.com>
References: <50D57946.5040501@gmail.com>
Message-ID: <web-88509966@softnet.si>

this will help you.

http://wiki.freeswitch.org/wiki/Dialplan_Handling_Incoming_Redirect

miha

On Sat, 22 Dec 2012 14:41:34 +0530
 veerabhadrarao` <bhadrarao.kankatala at gmail.com> wrote:
> hai
> 
> I am using Freeswitch 1.2.4
> 
> currently i am working on Attendent transfer can u please
> help me how to 
> make attendant transfer.
> 
> let when i make a call A--->B  and B-->C then how can i
> connect to A--->C
> 
> please help me
> 
> 
>
_________________________________________________________________________
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From lloyd.aloysius at gmail.com  Sat Dec 22 23:19:46 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Sat, 22 Dec 2012 15:19:46 -0500
Subject: [Freeswitch-users] MWI-Account - Issues
Message-ID: <CAGvuWYZwZeMwm8qryRj-Aq8k=H_U5AWqtorvC0XUkBemx4cTig@mail.gmail.com>

Hi All:

I am trying to send a user MWI to a different registered user



*Eg: user : amirt .. directory parameter is like below*

<param name="MWI-Account" value="amrit at mydomain.com"/>

Every thing works as expected.

============================================================

Now I change the MWI-Account to a different user

*user : amirt .. directory parameter is like below*
*
*
<param name="MWI-Account" value="eddie at mydomain.com"/>


Now user: amirt phone stop MWI . That is correct.


But the user eddie device not Light up for the user : amirt voice
mail notification

I have the following in the sip profile

<param name="send-message-query-on-register" value="true"/>

<param name="forward-unsolicited-mwi-notify" value="false"/>

Am I missing something.

Thanks
Lloyd
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From sdevoy at bizfocused.com  Sun Dec 23 03:36:34 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Sat, 22 Dec 2012 19:36:34 -0500
Subject: [Freeswitch-users] MWI-Account - Issues
In-Reply-To: <CAGvuWYZwZeMwm8qryRj-Aq8k=H_U5AWqtorvC0XUkBemx4cTig@mail.gmail.com>
References: <CAGvuWYZwZeMwm8qryRj-Aq8k=H_U5AWqtorvC0XUkBemx4cTig@mail.gmail.com>
Message-ID: <047701cde0a5$8f9dedd0$aed9c970$@bizfocused.com>

Hi Lloyd,

 

I am not 100% sure, but I will tell you what I believe is correct.

 

I think you are misinterpreting what the "MWI-Account" parameter means.
That parameter tells FreeSwitch what MWI info should go with THIS
REGISTRATION (aka phone/line).

 

If you have 

<user id="amirt">

                <variables>

                                . other values

                </variables>

                <params>

<param name="password" value="APasswordHere"/

<param name="MWI-Account" value="eddie at mydomain.com"/>

                </params>

</user>

 

That means that the MWI on the phone that registers a line with user=amirt
and a matching password will indicating that there are messages on the
eddie at mydomain.com account.  It does not make any changes to eddie's phone.

 

You said "Now user: amirt phone stop MWI . That is correct." - Wrong, amirt
phone should light up for eddie messages pnly now.

 

 

You said "But the user eddie device not Light up for the user : amirt voice
mail notification"

Correct, to do that add an MWI-Account to eddie user.

<user id="eddie">

                <variables>

                                . other values

                </variables>

                <params>

<param name="password" value="APasswordHere"/

<param name="MWI-Account" value="amirt at mydomain.com"/>

                </params>

</user>

THAT will make eddie's phone indicate messages for amirt account.

 

Hope that helps.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lloyd
Aloysius
Sent: Saturday, December 22, 2012 3:20 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] MWI-Account - Issues

 

Hi All:

 

I am trying to send a user MWI to a different registered user

 

 

 

Eg: user : amirt .. directory parameter is like below

 

<param name="MWI-Account" value="amrit at mydomain.com"/>

 

Every thing works as expected.

 

============================================================

 

Now I change the MWI-Account to a different user 

 

user : amirt .. directory parameter is like below

 

<param name="MWI-Account" value="eddie at mydomain.com"/>

 

 

Now user: amirt phone stop MWI . That is correct.

 

 

But the user eddie device not Light up for the user : amirt voice mail
notification

 

I have the following in the sip profile

 

<param name="send-message-query-on-register" value="true"/>

 

<param name="forward-unsolicited-mwi-notify" value="false"/>

 

Am I missing something. 

 

Thanks

Lloyd

 

 

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From fs-list at communicatefreely.net  Sun Dec 23 07:19:39 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Sat, 22 Dec 2012 23:19:39 -0500
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
Message-ID: <50D6865B.4010503@communicatefreely.net>

Just throwing this out there -

As it's clear how many problems NAT causes, why don't we all step up the pressure on phone 
manufacturers and ISPs to provide working IPv6 implementations so we can put NAT behind us 
  as soon as possible.

Just saying -

-Tim


Cal Leeming [Simplicity Media Ltd] wrote:
> *Any and all feedback on this thread would be much welcomed.*
> 
> Hello,
> *
> *
> There seems to be a large number of discussions surrounding NAT 
> traversal, as well as lots of documentation, but with no concrete answers. 
> 
> The NAT related wiki documentation is tedious, and depending on the 
> outcome of this thread, I'd like to spend some time cleaning it up.
> 
> The most common problem (the same as ours) was having a router with 
> broken ALG and a softphone that does not seem to work with STUN.
> 
> The following REGISTER is sent from a phone.
> 
> REGISTER sip:1.2.3.4:5060 <http://1.2.3.4:5060> SIP/2.0
> Via: SIP/2.0/UDP 
> 192.168.1.102:57787;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport
> Max-Forwards: 70
> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>
> To: "foxx"<sip:2000 at 1.2.3.4:5060 <http://sip:2000 at 1.2.3.4:5060>>
> From: "foxx"<sip:2000 at 1.2.3.4:5060 
> <http://sip:2000 at 1.2.3.4:5060>>;tag=83311448
> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.
> CSeq: 7 REGISTER
> Expires: 120
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, 
> REFER, INFO, MESSAGE
> Supported: replaces
> User-Agent: 3CXPhone 6.0.25732.0
> Content-Length: 0
> 
> As you can see, the client's public IP is not specified 
> anywhere. FreeSWITCH offers several ways around this, the main ones being;
> 
> * NDLB-connectile-dysfunction
> * NDLB-force-rport
> * apply-nat-acl
> * sip-force-contact
> 
> The one that has worked in our case was "NDLB-connectile-dysfunction" 
> (otherwise known as NAT HACK), however there seems to be a lot of 
> negative comments about using this.
> 
>  From what I can tell, the general argument is that NAT HACK is 
> considered a non RFC compliant hack, and the SIP phones should be doing 
> a better job of keeping to the RFCs.
> 
> In principle, this is a fair argument - but in practise, it's not a 
> reasonable assumption that all phones are RFC compliant, and (imho) not 
> a reasonable argument to have this functionality disabled by default.
> 
> So, I'd like to present the following arguments;
> 
> * Are there any other negative aspects about 
> using NDLB-connectile-dysfunction, other than it is a non compliant RFC 
> hack?
> 
> * Why is NDLB-connectile-dysfunction not enabled by default when certain 
> conditions are met? In the event that FreeSWITCH receives a REGISTER 
> from a phone specifying a Contact/Via as 192.168.0.0/16 
> <http://192.168.0.0/16>, but received on a public IP, then it should be 
> obvious that NAT is broken and automatically try to circumvent it.
> 
> * People seem to get confused between server side and client side NAT 
> problems, and that they both need to be resolved in a different way. The 
> documentation doesn't seem to reflect this clearly.
> 
> 
> ------------------------------------------------------------------------
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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



From nbhatti at gmail.com  Sun Dec 23 13:21:18 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Sun, 23 Dec 2012 13:21:18 +0300
Subject: [Freeswitch-users] stop a running lua script from luarun
Message-ID: <4F13F527-1F4A-41D5-81B6-4D7578D2426D@gmail.com>


Looks like simple one, but I can't figure it out if there a way to stop a lua script invoked via luarun which is stuck in an infinite loop and won't terminate itself? Of if we can come up with a way to monitor the thread and stop it?

Thanks,
--
Muhammad Naseer Bhatti



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From nbhatti at gmail.com  Sun Dec 23 15:27:38 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Sun, 23 Dec 2012 15:27:38 +0300
Subject: [Freeswitch-users] Performance hit originating calls via event
	socket
Message-ID: <D3FDC5F4-AAEE-4D29-86BC-AC2A5EDEC4C7@gmail.com>


WIth the latest development head running on a Debian box, I am trying to originate calls using event socket. Not sure why, but not able to generate more than 700 calls at any given time. Roughly around 40 CPS. For the sake of testing and simplicity, I just put the whole originate command multiple times in a text file and looping around it with fs_cli to execute. Dialplan sends the call to a lua script which randomly answers the call run the echo application and terminate with random seconds, something like http://wiki.freeswitch.org/wiki/Fakecall_responder

CPU remains low all the time. Looks like the opening and closing of ports to esl is the reason slowing originate rate?  I have tried from 8 to 24 CPU cores but the behavior remains the same. FS-4962 opened already in case we want to proceed from there.

Thanks,
--
Muhammad Naseer Bhatti



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From vijai.ganapathy at gmail.com  Sun Dec 23 15:51:16 2012
From: vijai.ganapathy at gmail.com (Vijai Ganapathy)
Date: Sun, 23 Dec 2012 18:21:16 +0530
Subject: [Freeswitch-users] New to Freeswitch
Message-ID: <CACahg4EKsXiRFkoYp-NbeRvtNsHLthxtu+k65pezCKw_wvhKbw@mail.gmail.com>

Dear all,

I am new to freeswitch and I have a question to ask.
I am planning to use gsm_open to have small call center setup.

>From the local market I could get Huawei E1732 model Dongle only and I am
wondering whether freeswitch / Linux Kernel
would support this model  for voice ?

Also would ATOM or Phenom would be enough to run 10 seat call center ?

Want to check with experts before jumping on to actual setup.

-- 
Vijai Ganapathy
Mailto: Vijai.Ganapathy at gmail.com
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From steveayre at gmail.com  Mon Dec 24 04:09:31 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 24 Dec 2012 01:09:31 +0000
Subject: [Freeswitch-users] Performance hit originating calls via event
	socket
In-Reply-To: <D3FDC5F4-AAEE-4D29-86BC-AC2A5EDEC4C7@gmail.com>
References: <D3FDC5F4-AAEE-4D29-86BC-AC2A5EDEC4C7@gmail.com>
Message-ID: <CAFiqYumQ-7wv9jstVqcZgds5Z7H3neHvk-nYq5peNFE=kyPjQA@mail.gmail.com>

Have you tried checking/tweaking the sessions-per-second limit?

You can adjust it at runtime:
http://wiki.freeswitch.org/wiki/Mod_commands#sps
or in switch.conf.xml



On 23 December 2012 12:27, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:

>
> WIth the latest development head running on a Debian box, I am trying to
> originate calls using event socket. Not sure why, but not able to generate
> more than 700 calls at any given time. Roughly around 40 CPS. For the sake
> of testing and simplicity, I just put the whole originate command multiple
> times in a text file and looping around it with fs_cli to execute. Dialplan
> sends the call to a lua script which randomly answers the call run the echo
> application and terminate with random seconds, something like
> http://wiki.freeswitch.org/wiki/Fakecall_responder
>
> CPU remains low all the time. Looks like the opening and closing of ports
> to esl is the reason slowing originate rate?  I have tried from 8 to 24 CPU
> cores but the behavior remains the same. FS-4962 opened already in case we
> want to proceed from there.
>
> Thanks,
> --
> Muhammad Naseer Bhatti
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 lloyd.aloysius at sunteltech.ca  Mon Dec 24 05:11:21 2012
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Sun, 23 Dec 2012 21:11:21 -0500
Subject: [Freeswitch-users] MWI-Account - Issues
In-Reply-To: <047701cde0a5$8f9dedd0$aed9c970$@bizfocused.com>
References: <CAGvuWYZwZeMwm8qryRj-Aq8k=H_U5AWqtorvC0XUkBemx4cTig@mail.gmail.com>
	<047701cde0a5$8f9dedd0$aed9c970$@bizfocused.com>
Message-ID: <CAGvuWYZzoGCyhS=mvAbgWom7M_KacypHMrEp98y8z+Du-+Y6AQ@mail.gmail.com>

Hi Sean:

Thanks for the reply. You are correct.

MWI-Account - SUBSCRIBE to monitor voicemail.

Thanks
Lloyd


*  <http://www.sunteltech.ca/blog/>*


On Sat, Dec 22, 2012 at 7:36 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Hi Lloyd,****
>
> ** **
>
> I am not 100% sure, but I will tell you what I believe is correct.****
>
> ** **
>
> I think you are misinterpreting what the ?MWI-Account? parameter means.
> That parameter tells FreeSwitch what MWI info should go with THIS
> REGISTRATION (aka phone/line).****
>
> ** **
>
> If you have ****
>
> <user id=?amirt?>****
>
>                 <variables>****
>
>                                 ? other values****
>
>                 </variables>****
>
>                 <params>****
>
> <param name="password" value="APasswordHere"/****
>
> <param name="MWI-Account" value="eddie at mydomain.com"/>****
>
>                 </params>****
>
> </user>****
>
> ** **
>
> That means that the MWI on the phone that registers a line with user=amirt
> and a matching password will indicating that there are messages on the
> eddie at mydomain.com account.  It does not make any changes to eddie?s
> phone.****
>
> ** **
>
> You said ?Now user: amirt phone stop MWI . That is correct.? ? Wrong,
> amirt phone should light up for eddie messages pnly now.****
>
> ** **
>
> ** **
>
> You said ?But the user eddie device not Light up for the user :
> amirt voice mail notification?****
>
> Correct, to do that add an MWI-Account to eddie user.****
>
> <user id=?eddie?>****
>
>                 <variables>****
>
>                                 ? other values****
>
>                 </variables>****
>
>                 <params>****
>
> <param name="password" value="APasswordHere"/****
>
> <param name="MWI-Account" value="amirt at mydomain.com"/>****
>
>                 </params>****
>
> </user>****
>
> THAT will make eddie?s phone indicate messages for amirt account.****
>
> ** **
>
> Hope that helps.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Lloyd
> Aloysius
> *Sent:* Saturday, December 22, 2012 3:20 PM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] MWI-Account - Issues****
>
> ** **
>
> Hi All:****
>
> ** **
>
> I am trying to send a user MWI to a different registered user****
>
> ** **
>
> ** **
>
> ** **
>
> *Eg: user : amirt .. directory parameter is like below*****
>
> ** **
>
> <param name="MWI-Account" value="amrit at mydomain.com"/>****
>
> ** **
>
> Every thing works as expected.****
>
> ** **
>
> ============================================================****
>
> ** **
>
> Now I change the MWI-Account to a different user ****
>
> ** **
>
> *user : amirt .. directory parameter is like below*****
>
> ** **
>
> <param name="MWI-Account" value="eddie at mydomain.com"/>****
>
> ** **
>
> ** **
>
> Now user: amirt phone stop MWI . That is correct.****
>
> ** **
>
> ** **
>
> But the user eddie device not Light up for the user : amirt voice
> mail notification****
>
> ** **
>
> I have the following in the sip profile****
>
> ** **
>
> <param name="send-message-query-on-register" value="true"/>****
>
> ** **
>
> <param name="forward-unsolicited-mwi-notify" value="false"/>****
>
> ** **
>
> Am I missing something. ****
>
> ** **
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nathan at nathansamson.be  Mon Dec 24 17:55:44 2012
From: nathan at nathansamson.be (Nathan Samson)
Date: Mon, 24 Dec 2012 15:55:44 +0100
Subject: [Freeswitch-users] Creating a call app based on freeswitch (and
	adhearsion)
Message-ID: <CABY4CiUgY2t83p6nYTiKqsKRnVRmfq2Ew8teP4q0LDw5g1eqkg@mail.gmail.com>

Hi all,

We have a lot of customers that should be able to call our information
center. Based on who (and when) he calls, the call should be forwarded to
one specific person (or stored as voice message outside the hours). Since
forwading the call to a mobile (or fixed) phone is costing money, we want
to forward it to SIP clients when this contact person is available via SIP
(so we first try SIP, if the contact person doesn't take the call we'll try
the phone number).

We also want to do outbound calls (as internal users), some via a normal
phone (if internet is not available), or also via SIP.
We also have a admin application ourselves to manage users, information
stored in this database should be used to specify the incoming calls
forwarding.

The main question is how we best manage our SIP users (we prefer to do
everything in our application).
Are there any APIs available to create/delete & manage SIP users for
freeswitch? Can we write (in a reasonably short amount of time) an
authentication plugin for freeswitch which will consult our internal
database.

Note that our application logic  (how things should be forwarded, what
application should run etc) will be written in the adhearsion framework, so
every call should be forwarded to adhearsion, which wil lthen handle all
logic (except for the SIP authentication).

Cheers,
Nathan
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From nbhatti at gmail.com  Mon Dec 24 19:22:48 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Mon, 24 Dec 2012 19:22:48 +0300
Subject: [Freeswitch-users] Performance hit originating calls via event
	socket
In-Reply-To: <CAFiqYumQ-7wv9jstVqcZgds5Z7H3neHvk-nYq5peNFE=kyPjQA@mail.gmail.com>
References: <D3FDC5F4-AAEE-4D29-86BC-AC2A5EDEC4C7@gmail.com>
	<CAFiqYumQ-7wv9jstVqcZgds5Z7H3neHvk-nYq5peNFE=kyPjQA@mail.gmail.com>
Message-ID: <546F9C9F-AFCD-4775-81A3-25F5CC8C51CA@gmail.com>


It's already been tuned. I think since opening and closing of sockets is an expensive operation which takes more CPU cycles, that could be one of the reason for this behavior. But I am not expert on this. Let's see what other have to say.

Thanks,
--
Muhammad Naseer Bhatti



On Dec 24, 2012, at 4:09 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Have you tried checking/tweaking the sessions-per-second limit?
> 
> You can adjust it at runtime:
> http://wiki.freeswitch.org/wiki/Mod_commands#sps
> or in switch.conf.xml
> 
> 
> 
> On 23 December 2012 12:27, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:
> 
> WIth the latest development head running on a Debian box, I am trying to originate calls using event socket. Not sure why, but not able to generate more than 700 calls at any given time. Roughly around 40 CPS. For the sake of testing and simplicity, I just put the whole originate command multiple times in a text file and looping around it with fs_cli to execute. Dialplan sends the call to a lua script which randomly answers the call run the echo application and terminate with random seconds, something like http://wiki.freeswitch.org/wiki/Fakecall_responder
> 
> CPU remains low all the time. Looks like the opening and closing of ports to esl is the reason slowing originate rate?  I have tried from 8 to 24 CPU cores but the behavior remains the same. FS-4962 opened already in case we want to proceed from there.
> 
> Thanks,
> --
> Muhammad Naseer Bhatti
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 24 20:21:35 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 24 Dec 2012 17:21:35 +0000
Subject: [Freeswitch-users] Performance hit originating calls via event
	socket
In-Reply-To: <546F9C9F-AFCD-4775-81A3-25F5CC8C51CA@gmail.com>
References: <D3FDC5F4-AAEE-4D29-86BC-AC2A5EDEC4C7@gmail.com>
	<CAFiqYumQ-7wv9jstVqcZgds5Z7H3neHvk-nYq5peNFE=kyPjQA@mail.gmail.com>
	<546F9C9F-AFCD-4775-81A3-25F5CC8C51CA@gmail.com>
Message-ID: <CAFiqYunJ1isiXhQvHE2ue6VX_uanOvJ67QY-Au=qupq0ae-RuQ@mail.gmail.com>

Are you opening a new ESL connection for each originate? It'd be far more
efficient to open a single persistent connection and reuse it.



On 24 December 2012 16:22, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:

>
> It's already been tuned. I think since opening and closing of sockets is
> an expensive operation which takes more CPU cycles, that could be one of
> the reason for this behavior. But I am not expert on this. Let's see what
> other have to say.
>
>
> Thanks,
> --
> Muhammad Naseer Bhatti
>
>
>
> On Dec 24, 2012, at 4:09 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
> Have you tried checking/tweaking the sessions-per-second limit?
>
> You can adjust it at runtime:
> http://wiki.freeswitch.org/wiki/Mod_commands#sps
> or in switch.conf.xml
>
>
>
> On 23 December 2012 12:27, Muhammad Naseer Bhatti <nbhatti at gmail.com>wrote:
>
>>
>> WIth the latest development head running on a Debian box, I am trying to
>> originate calls using event socket. Not sure why, but not able to generate
>> more than 700 calls at any given time. Roughly around 40 CPS. For the sake
>> of testing and simplicity, I just put the whole originate command multiple
>> times in a text file and looping around it with fs_cli to execute. Dialplan
>> sends the call to a lua script which randomly answers the call run the echo
>> application and terminate with random seconds, something like
>> http://wiki.freeswitch.org/wiki/Fakecall_responder
>>
>> CPU remains low all the time. Looks like the opening and closing of ports
>> to esl is the reason slowing originate rate?  I have tried from 8 to 24 CPU
>> cores but the behavior remains the same. FS-4962 opened already in case we
>> want to proceed from there.
>>
>> Thanks,
>> --
>> Muhammad Naseer Bhatti
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 sdevoy at bizfocused.com  Mon Dec 24 20:36:10 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 24 Dec 2012 12:36:10 -0500
Subject: [Freeswitch-users] XML DialPlan Gateay dialing redundancy
Message-ID: <0da201cde1fd$2a0b6de0$7e2249a0$@bizfocused.com>

 

I need a little help with a complex XML dialplan.  Also, I am on the Stable
release so I do not believe the much needed nested conditions exists for me
yet.

 

My problem is faulty gateway providers who periodically fail to connect the
call with [temp failure] because "our routing tables where not up to date."

 

So what I am after is an xml dialplan to do this

If (it is a 10 or 11 digit dial)

                Dial using gateway 1

                If no answer

                                Go to voice mail

                Else if gateway failure 

                                Dial using gateway 2

                If no answer

                                Go to voice mail

                Else if gateway failure 

                                Dial using gateway 3

                If no answer

                                Go to voice mail

                                Else if gateway failure

                                                Play a message that the
number is screwed!

 

Perhaps I need to breakdown and use a scripting language like LUA or
JavaScript for this, but I would rather keep everything in XML if possible.

 

Any thoughts?

 

Happy Holidays to all,

Sean

 

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From krice at freeswitch.org  Mon Dec 24 21:08:43 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 24 Dec 2012 12:08:43 -0600
Subject: [Freeswitch-users] Performance hit originating calls via event
 socket
In-Reply-To: <CAFiqYunJ1isiXhQvHE2ue6VX_uanOvJ67QY-Au=qupq0ae-RuQ@mail.gmail.com>
Message-ID: <CCFDF64B.626D4%krice@freeswitch.org>

Not to mention maybe using bgapi as opposed to api


On 12/24/12 11:21 AM, "Steven Ayre" <steveayre at gmail.com> wrote:

> Are you opening a new ESL connection for each originate? It'd be far more
> efficient to open a single persistent connection and reuse it.
> 
> 
> 
> On 24 December 2012 16:22, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:
>> 
>> It's already been tuned. I think since opening and closing of sockets is an
>> expensive operation which takes more CPU cycles, that could be one of the
>> reason for this behavior. But I am not expert on this. Let's see what other
>> have to say.
>> 
>> 
>> Thanks,
>> --
>> Muhammad Naseer Bhatti
>> 
>> 
>> 
>> On Dec 24, 2012, at 4:09 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> 
>>> Have you tried checking/tweaking the sessions-per-second limit?
>>> 
>>> You can adjust it at runtime:
>>> http://wiki.freeswitch.org/wiki/Mod_commands#sps
>>> or in switch.conf.xml
>>> 
>>> 
>>> 
>>> On 23 December 2012 12:27, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:
>>>> 
>>>> WIth the latest development head running on a Debian box, I am trying to
>>>> originate calls using event socket. Not sure why, but not able to generate
>>>> more than 700 calls at any given time. Roughly around 40 CPS. For the sake
>>>> of testing and simplicity, I just put the whole originate command multiple
>>>> times in a text file and looping around it with fs_cli to execute. Dialplan
>>>> sends the call to a lua script which randomly answers the call run the echo
>>>> application and terminate with random seconds, something
>>>> like?http://wiki.freeswitch.org/wiki/Fakecall_responder
>>>> 
>>>> CPU remains low all the time. Looks like the opening and closing of ports
>>>> to esl is the reason slowing originate rate? ?I have tried from 8 to 24 CPU
>>>> cores but the behavior remains the same.?FS-4962 opened already in case we
>>>> want to proceed from there.
>>>> 
>>>> Thanks,
>>>> --
>>>> Muhammad Naseer Bhatti
>>>> 
>>>> 
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> 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/>
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From Tim.Meade at Millicorp.com  Mon Dec 24 22:20:11 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Mon, 24 Dec 2012 19:20:11 +0000
Subject: [Freeswitch-users] xml_curl and ivr menus
Message-ID: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>

I am continually getting 'unable to find menu' while trying to retrieve an ivr_menu using xml_curl

Using xml_curl debug_on

My DialPlan

<document type="freeswitch/xml">
<section name="dialplan" description="example_curl_dialplan">
<context name="public">
<extension name="ws_origination">
<condition field="destination_number" expression="^\d+$" break="on-true">
<action inline="true" application="set" data="continue_on_fail=true"/>
<action inline="true" application="set" data="hangup_after_bridge=true"/>
<action application="answer"/>
<action application="ivr" data="tivr"/>
</condition>
</extension>
</context>
</section>

And the ivr configuration returned by my xml_curl

<include>
<!-- demo IVR setup -->
<!-- demo IVR, Main Menu -->
<menu name="tivr"
greet-long="phrase:demo_ivr_main_menu"
greet-short="phrase:demo_ivr_main_menu_short"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
confirm-macro=""
confirm-key=""
tts-engine="flite"
tts-voice="rms"
confirm-attempts="3"
timeout="10000"
inter-digit-timeout="2000"
max-failures="3"
max-timeouts="3"
digit-len="4">
<!-- The following are the definitions for the digits the user dials -->
<!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
<entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888 at conference.freeswitch.org"/>
<entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/> <!-- FS echo -->
<entry action="menu-exec-app" digits="3" param="transfer 9664 XML default"/> <!-- MOH -->
<entry action="menu-exec-app" digits="4" param="transfer 9191 XML default"/> <!-- ClueCon -->
<entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/> <!-- Screaming monkeys -->
<!-- Using a regex in the digits tag lets you define a dial pattern for the caller
You may define multiple regexes if you need a different pattern for some reason -->
<entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
<entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
</menu>
</include>

Any ideas on the 'unable to find menu' ????

Thanks

Tim



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From bdfoster at endigotech.com  Mon Dec 24 23:30:56 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 24 Dec 2012 15:30:56 -0500
Subject: [Freeswitch-users] XML DialPlan Gateay dialing redundancy
In-Reply-To: <0da201cde1fd$2a0b6de0$7e2249a0$@bizfocused.com>
References: <0da201cde1fd$2a0b6de0$7e2249a0$@bizfocused.com>
Message-ID: <F4777D15-B934-4B09-A620-E790715AD590@endigotech.com>

Take a look on the wiki for continue_on_fail. There's an example of this in the default.xml dialplan (example configuration) for the extension "Local_Extension".

Sent from my iPhone

On Dec 24, 2012, at 12:36 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:

>  
> I need a little help with a complex XML dialplan.  Also, I am on the Stable release so I do not believe the much needed nested conditions exists for me yet.
>  
> My problem is faulty gateway providers who periodically fail to connect the call with [temp failure] because ?our routing tables where not up to date.?
>  
> So what I am after is an xml dialplan to do this
> If (it is a 10 or 11 digit dial)
>                 Dial using gateway 1
>                 If no answer
>                                 Go to voice mail
>                 Else if gateway failure
>                                 Dial using gateway 2
>                 If no answer
>                                 Go to voice mail
>                 Else if gateway failure
>                                 Dial using gateway 3
>                 If no answer
>                                 Go to voice mail
>                                 Else if gateway failure
>                                                 Play a message that the number is screwed!
>  
> Perhaps I need to breakdown and use a scripting language like LUA or JavaScript for this, but I would rather keep everything in XML if possible.
>  
> Any thoughts?
>  
> Happy Holidays to all,
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From Dave.May at patlive.com  Mon Dec 24 23:31:43 2012
From: Dave.May at patlive.com (Dave May)
Date: Mon, 24 Dec 2012 20:31:43 -0000
Subject: [Freeswitch-users] xml_curl and ivr menus
Message-ID: <019c01cde215$af1f2d92$0400010a@patlive.local>




Sent from my HTC smartphone on the Now Network from Sprint!

----- Reply message -----
From: "Tim Meade" <Tim.Meade at Millicorp.com>
To: "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] xml_curl and ivr menus
Date: Mon, Dec 24, 2012 2:27 pm


I am continually getting 'unable to find menu' while trying to retrieve an ivr_menu using xml_curl

Using xml_curl debug_on

My DialPlan

<document type="freeswitch/xml">
<section name="dialplan" description="example_curl_dialplan">
<context name="public">
<extension name="ws_origination">
<condition field="destination_number" expression="^\d+$" break="on-true">
<action inline="true" application="set" data="continue_on_fail=true"/>
<action inline="true" application="set" data="hangup_after_bridge=true"/>
<action application="answer"/>
<action application="ivr" data="tivr"/>
</condition>
</extension>
</context>
</section>

And the ivr configuration returned by my xml_curl

<include>
<!-- demo IVR setup -->
<!-- demo IVR, Main Menu -->
<menu name="tivr"
greet-long="phrase:demo_ivr_main_menu"
greet-short="phrase:demo_ivr_main_menu_short"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
confirm-macro=""
confirm-key=""
tts-engine="flite"
tts-voice="rms"
confirm-attempts="3"
timeout="10000"
inter-digit-timeout="2000"
max-failures="3"
max-timeouts="3"
digit-len="4">
<!-- The following are the definitions for the digits the user dials -->
<!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
<entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888 at conference.freeswitch.org"/>
<entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/> <!-- FS echo -->
<entry action="menu-exec-app" digits="3" param="transfer 9664 XML default"/> <!-- MOH -->
<entry action="menu-exec-app" digits="4" param="transfer 9191 XML default"/> <!-- ClueCon -->
<entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/> <!-- Screaming monkeys -->
<!-- Using a regex in the digits tag lets you define a dial pattern for the caller
You may define multiple regexes if you need a different pattern for some reason -->
<entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
<entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
</menu>
</include>

Any ideas on the 'unable to find menu' ????

Thanks

Tim





From bdfoster at endigotech.com  Mon Dec 24 23:32:34 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 24 Dec 2012 15:32:34 -0500
Subject: [Freeswitch-users] Creating a call app based on freeswitch (and
	adhearsion)
In-Reply-To: <CABY4CiUgY2t83p6nYTiKqsKRnVRmfq2Ew8teP4q0LDw5g1eqkg@mail.gmail.com>
References: <CABY4CiUgY2t83p6nYTiKqsKRnVRmfq2Ew8teP4q0LDw5g1eqkg@mail.gmail.com>
Message-ID: <E3C1969D-A47A-46D5-975D-1264A19802AA@endigotech.com>

Check out the webapi (mod_xml_rpc) and you should probably think about using mod_xml_curl. 

Sent from my iPhone

On Dec 24, 2012, at 9:55 AM, Nathan Samson <nathan at nathansamson.be> wrote:

> Hi all,
> 
> We have a lot of customers that should be able to call our information center. Based on who (and when) he calls, the call should be forwarded to one specific person (or stored as voice message outside the hours). Since forwading the call to a mobile (or fixed) phone is costing money, we want to forward it to SIP clients when this contact person is available via SIP (so we first try SIP, if the contact person doesn't take the call we'll try the phone number).
> 
> We also want to do outbound calls (as internal users), some via a normal phone (if internet is not available), or also via SIP.
> We also have a admin application ourselves to manage users, information stored in this database should be used to specify the incoming calls forwarding.
> 
> The main question is how we best manage our SIP users (we prefer to do everything in our application).
> Are there any APIs available to create/delete & manage SIP users for  freeswitch? Can we write (in a reasonably short amount of time) an authentication plugin for freeswitch which will consult our internal database.
> 
> Note that our application logic  (how things should be forwarded, what application should run etc) will be written in the adhearsion framework, so every call should be forwarded to adhearsion, which wil lthen handle all logic (except for the SIP authentication).
> 
> Cheers,
> Nathan
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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


From bdfoster at endigotech.com  Mon Dec 24 23:34:10 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 24 Dec 2012 15:34:10 -0500
Subject: [Freeswitch-users] xml_curl and ivr menus
In-Reply-To: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>
References: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>
Message-ID: <01789F3B-22EB-4A71-AF35-C9F9CA18AABE@endigotech.com>

Pastebin a log of the call for others who are more familiar with this than I am.

Sent from my iPhone

On Dec 24, 2012, at 2:20 PM, Tim Meade <Tim.Meade at Millicorp.com> wrote:

> I am continually getting ?unable to find menu? while trying to retrieve an ivr_menu using xml_curl
>  
> Using xml_curl debug_on
>  
> My DialPlan
>  
> <document type="freeswitch/xml">
> <section name="dialplan" description="example_curl_dialplan">
> <context name="public">
> <extension name="ws_origination">
> <condition field="destination_number" expression="^\d+$" break="on-true">
> <action inline="true" application="set" data="continue_on_fail=true"/>
> <action inline="true" application="set" data="hangup_after_bridge=true"/>
> <action application="answer"/>
> <action application="ivr" data="tivr"/>
> </condition>
> </extension>
> </context>
> </section>
>  
> And the ivr configuration returned by my xml_curl
>  
> <include>
> <!-- demo IVR setup -->
> <!-- demo IVR, Main Menu -->
> <menu name="tivr"
> greet-long="phrase:demo_ivr_main_menu"
> greet-short="phrase:demo_ivr_main_menu_short"
> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
> exit-sound="voicemail/vm-goodbye.wav"
> confirm-macro=""
> confirm-key=""
> tts-engine="flite"
> tts-voice="rms"
> confirm-attempts="3"
> timeout="10000"
> inter-digit-timeout="2000"
> max-failures="3"
> max-timeouts="3"
> digit-len="4">
> <!-- The following are the definitions for the digits the user dials -->
> <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888 at conference.freeswitch.org"/>
> <entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/> <!-- FS echo -->
> <entry action="menu-exec-app" digits="3" param="transfer 9664 XML default"/> <!-- MOH -->
> <entry action="menu-exec-app" digits="4" param="transfer 9191 XML default"/> <!-- ClueCon -->
> <entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/> <!-- Screaming monkeys -->
> <!-- Using a regex in the digits tag lets you define a dial pattern for the caller
> You may define multiple regexes if you need a different pattern for some reason -->
> <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
> <entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
> </menu>
> </include>
>  
> Any ideas on the ?unable to find menu? ????
>  
> Thanks
>  
> Tim
>  
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 abaci64 at gmail.com  Mon Dec 24 23:42:17 2012
From: abaci64 at gmail.com (Abaci)
Date: Mon, 24 Dec 2012 15:42:17 -0500
Subject: [Freeswitch-users] xml_curl and ivr menus
In-Reply-To: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>
References: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>
Message-ID: <50D8BE29.7010404@gmail.com>

on freeswitch enable xml_curl debug and see if freeswitch is actually 
getting the correct ivr
'xml_curl debug_on'

On 12/24/2012 2:20 PM, Tim Meade wrote:
>
> I am continually getting 'unable to find menu' while trying to 
> retrieve an ivr_menu using xml_curl
>
> Using xml_curl debug_on
>
> My DialPlan
>
> <document type="freeswitch/xml">
>
> <section name="dialplan" description="example_curl_dialplan">
>
> <context name="public">
>
> <extension name="ws_origination">
>
> <condition field="destination_number" expression="^\d+$" break="on-true">
>
> <action inline="true" application="set" data="continue_on_fail=true"/>
>
> <action inline="true" application="set" data="hangup_after_bridge=true"/>
>
> <action application="answer"/>
>
> <action application="ivr" data="tivr"/>
>
> </condition>
>
> </extension>
>
> </context>
>
> </section>
>
> And the ivr configuration returned by my xml_curl
>
> <include>
>
> <!-- demo IVR setup -->
>
> <!-- demo IVR, Main Menu -->
>
> <menu name="tivr"
>
> greet-long="phrase:demo_ivr_main_menu"
>
> greet-short="phrase:demo_ivr_main_menu_short"
>
> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>
> exit-sound="voicemail/vm-goodbye.wav"
>
> confirm-macro=""
>
> confirm-key=""
>
> tts-engine="flite"
>
> tts-voice="rms"
>
> confirm-attempts="3"
>
> timeout="10000"
>
> inter-digit-timeout="2000"
>
> max-failures="3"
>
> max-timeouts="3"
>
> digit-len="4">
>
> <!-- The following are the definitions for the digits the user dials -->
>
> <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
>
> <entry action="menu-exec-app" digits="1" param="bridge 
> sofia/$${domain}/888 at conference.freeswitch.org"/>
>
> <entry action="menu-exec-app" digits="2" param="transfer 9196 XML 
> default"/> <!-- FS echo -->
>
> <entry action="menu-exec-app" digits="3" param="transfer 9664 XML 
> default"/> <!-- MOH -->
>
> <entry action="menu-exec-app" digits="4" param="transfer 9191 XML 
> default"/> <!-- ClueCon -->
>
> <entry action="menu-exec-app" digits="5" param="transfer 1234*256 
> enum"/> <!-- Screaming monkeys -->
>
> <!-- Using a regex in the digits tag lets you define a dial pattern 
> for the caller
>
> You may define multiple regexes if you need a different pattern for 
> some reason -->
>
> <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" 
> param="transfer $1 XML features"/>
>
> <entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
>
> </menu>
>
> </include>
>
> Any ideas on the 'unable to find menu' ????
>
> Thanks
>
> Tim
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 Tim.Meade at Millicorp.com  Tue Dec 25 00:22:34 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Mon, 24 Dec 2012 21:22:34 +0000
Subject: [Freeswitch-users] xml_curl and ivr menus
In-Reply-To: <50D8BE29.7010404@gmail.com>
References: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>
	<50D8BE29.7010404@gmail.com>
Message-ID: <804D48104511D4468F0D60DF9D3100350AF8A9E3@MAIL.millicorp.com>

The items I pasted below are from the cat of the xml_curl debug_on command.  It certainly appears to me that freeswitch is getting the correct name in this case 'tivr' which is what shows in both the dialplan and the curl'd ivr menu.

My concerns now are dealing with the xml format for the ivr as I have found several different versions on the wiki and in the prior postings.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abaci
Sent: Monday, December 24, 2012 3:42 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] xml_curl and ivr menus

on freeswitch enable xml_curl debug and see if freeswitch is actually getting the correct ivr
'xml_curl debug_on'

On 12/24/2012 2:20 PM, Tim Meade wrote:
I am continually getting 'unable to find menu' while trying to retrieve an ivr_menu using xml_curl

Using xml_curl debug_on

My DialPlan

<document type="freeswitch/xml">
<section name="dialplan" description="example_curl_dialplan">
<context name="public">
<extension name="ws_origination">
<condition field="destination_number" expression="^\d+$" break="on-true">
<action inline="true" application="set" data="continue_on_fail=true"/>
<action inline="true" application="set" data="hangup_after_bridge=true"/>
<action application="answer"/>
<action application="ivr" data="tivr"/>
</condition>
</extension>
</context>
</section>

And the ivr configuration returned by my xml_curl

<include>
<!-- demo IVR setup -->
<!-- demo IVR, Main Menu -->
<menu name="tivr"
greet-long="phrase:demo_ivr_main_menu"
greet-short="phrase:demo_ivr_main_menu_short"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
confirm-macro=""
confirm-key=""
tts-engine="flite"
tts-voice="rms"
confirm-attempts="3"
timeout="10000"
inter-digit-timeout="2000"
max-failures="3"
max-timeouts="3"
digit-len="4">
<!-- The following are the definitions for the digits the user dials -->
<!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
<entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888 at conference.freeswitch.org<mailto:domain%7d/888 at conference.freeswitch.org>"/>
<entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/> <!-- FS echo -->
<entry action="menu-exec-app" digits="3" param="transfer 9664 XML default"/> <!-- MOH -->
<entry action="menu-exec-app" digits="4" param="transfer 9191 XML default"/> <!-- ClueCon -->
<entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/> <!-- Screaming monkeys -->
<!-- Using a regex in the digits tag lets you define a dial pattern for the caller
You may define multiple regexes if you need a different pattern for some reason -->
<entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
<entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
</menu>
</include>

Any ideas on the 'unable to find menu' ????

Thanks

Tim







_________________________________________________________________________

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

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From abaci64 at gmail.com  Tue Dec 25 00:43:06 2012
From: abaci64 at gmail.com (Abaci)
Date: Mon, 24 Dec 2012 16:43:06 -0500
Subject: [Freeswitch-users] xml_curl and ivr menus
In-Reply-To: <804D48104511D4468F0D60DF9D3100350AF8A9E3@MAIL.millicorp.com>
References: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>
	<50D8BE29.7010404@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8A9E3@MAIL.millicorp.com>
Message-ID: <50D8CC6A.5000706@gmail.com>

Here is a sample from my server, I highlighted the missing stuff.

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
   <section name="configuration">
*    <configuration name="ivr.conf" description="IVR menus">**
**      <menus>*
         <menu name="ivrmenu1"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
               exit-sound="voicemail/vm-goodbye.wav"
               timeout ="3000"
               inter-digit-timeout="2000"
               max-failures="2"
               digit-len="3"
               phrase-lang="en">
           <entry action="menu-exec-app" digits="0" param="transfer 11 
XML voicemail " />
           <entry action="menu-exec-app" digits="1" param="transfer 
ivrmenu2 XML ivr " />
           <entry action="menu-exec-app" digits="2" param="transfer 
ivrmenu3 XML ivr " />
         </menu>
*      </menus>
     </configuration>*
   </section>
</document>

On 12/24/2012 4:22 PM, Tim Meade wrote:
>
> The items I pasted below are from the cat of the xml_curl debug_on 
> command.  It certainly appears to me that freeswitch is getting the 
> correct name in this case 'tivr' which is what shows in both the 
> dialplan and the curl'd ivr menu.
>
> My concerns now are dealing with the xml format for the ivr as I have 
> found several different versions on the wiki and in the prior postings.
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Abaci
> *Sent:* Monday, December 24, 2012 3:42 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus
>
> on freeswitch enable xml_curl debug and see if freeswitch is actually 
> getting the correct ivr
> 'xml_curl debug_on'
>
> On 12/24/2012 2:20 PM, Tim Meade wrote:
>
>     I am continually getting 'unable to find menu' while trying to
>     retrieve an ivr_menu using xml_curl
>
>     Using xml_curl debug_on
>
>     My DialPlan
>
>     <document type="freeswitch/xml">
>
>     <section name="dialplan" description="example_curl_dialplan">
>
>     <context name="public">
>
>     <extension name="ws_origination">
>
>     <condition field="destination_number" expression="^\d+$"
>     break="on-true">
>
>     <action inline="true" application="set" data="continue_on_fail=true"/>
>
>     <action inline="true" application="set"
>     data="hangup_after_bridge=true"/>
>
>     <action application="answer"/>
>
>     <action application="ivr" data="tivr"/>
>
>     </condition>
>
>     </extension>
>
>     </context>
>
>     </section>
>
>     And the ivr configuration returned by my xml_curl
>
>     <include>
>
>     <!-- demo IVR setup -->
>
>     <!-- demo IVR, Main Menu -->
>
>     <menu name="tivr"
>
>     greet-long="phrase:demo_ivr_main_menu"
>
>     greet-short="phrase:demo_ivr_main_menu_short"
>
>     invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>
>     exit-sound="voicemail/vm-goodbye.wav"
>
>     confirm-macro=""
>
>     confirm-key=""
>
>     tts-engine="flite"
>
>     tts-voice="rms"
>
>     confirm-attempts="3"
>
>     timeout="10000"
>
>     inter-digit-timeout="2000"
>
>     max-failures="3"
>
>     max-timeouts="3"
>
>     digit-len="4">
>
>     <!-- The following are the definitions for the digits the user
>     dials -->
>
>     <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
>
>     <entry action="menu-exec-app" digits="1" param="bridge
>     sofia/$${domain}/888 at conference.freeswitch.org
>     <mailto:domain%7d/888 at conference.freeswitch.org>"/>
>
>     <entry action="menu-exec-app" digits="2" param="transfer 9196 XML
>     default"/> <!-- FS echo -->
>
>     <entry action="menu-exec-app" digits="3" param="transfer 9664 XML
>     default"/> <!-- MOH -->
>
>     <entry action="menu-exec-app" digits="4" param="transfer 9191 XML
>     default"/> <!-- ClueCon -->
>
>     <entry action="menu-exec-app" digits="5" param="transfer 1234*256
>     enum"/> <!-- Screaming monkeys -->
>
>     <!-- Using a regex in the digits tag lets you define a dial
>     pattern for the caller
>
>     You may define multiple regexes if you need a different pattern
>     for some reason -->
>
>     <entry action="menu-exec-app" digits="/^(10[01][0-9])$/"
>     param="transfer $1 XML features"/>
>
>     <entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
>
>     </menu>
>
>     </include>
>
>     Any ideas on the 'unable to find menu' ????
>
>     Thanks
>
>     Tim
>
>
>
>
>     _________________________________________________________________________
>
>     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
> 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 abaci64 at gmail.com  Tue Dec 25 00:44:35 2012
From: abaci64 at gmail.com (Abaci)
Date: Mon, 24 Dec 2012 16:44:35 -0500
Subject: [Freeswitch-users] xml_curl and ivr menus
In-Reply-To: <50D8CC6A.5000706@gmail.com>
References: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>
	<50D8BE29.7010404@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8A9E3@MAIL.millicorp.com>
	<50D8CC6A.5000706@gmail.com>
Message-ID: <50D8CCC3.7090305@gmail.com>

you're also missing the "section" element.

On 12/24/2012 4:43 PM, Abaci wrote:
> Here is a sample from my server, I highlighted the missing stuff.
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="configuration">
> *    <configuration name="ivr.conf" description="IVR menus">**
> **      <menus>*
>         <menu name="ivrmenu1"
> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>               exit-sound="voicemail/vm-goodbye.wav"
>               timeout ="3000"
>               inter-digit-timeout="2000"
>               max-failures="2"
>               digit-len="3"
>               phrase-lang="en">
>           <entry action="menu-exec-app" digits="0" param="transfer 11 
> XML voicemail " />
>           <entry action="menu-exec-app" digits="1" param="transfer 
> ivrmenu2 XML ivr " />
>           <entry action="menu-exec-app" digits="2" param="transfer 
> ivrmenu3 XML ivr " />
>         </menu>
> *      </menus>
>     </configuration>*
>   </section>
> </document>
>
> On 12/24/2012 4:22 PM, Tim Meade wrote:
>>
>> The items I pasted below are from the cat of the xml_curl debug_on 
>> command.  It certainly appears to me that freeswitch is getting the 
>> correct name in this case 'tivr' which is what shows in both the 
>> dialplan and the curl'd ivr menu.
>>
>> My concerns now are dealing with the xml format for the ivr as I have 
>> found several different versions on the wiki and in the prior postings.
>>
>> *From:*freeswitch-users-bounces at lists.freeswitch.org 
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
>> *Abaci
>> *Sent:* Monday, December 24, 2012 3:42 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus
>>
>> on freeswitch enable xml_curl debug and see if freeswitch is actually 
>> getting the correct ivr
>> 'xml_curl debug_on'
>>
>> On 12/24/2012 2:20 PM, Tim Meade wrote:
>>
>>     I am continually getting 'unable to find menu' while trying to
>>     retrieve an ivr_menu using xml_curl
>>
>>     Using xml_curl debug_on
>>
>>     My DialPlan
>>
>>     <document type="freeswitch/xml">
>>
>>     <section name="dialplan" description="example_curl_dialplan">
>>
>>     <context name="public">
>>
>>     <extension name="ws_origination">
>>
>>     <condition field="destination_number" expression="^\d+$"
>>     break="on-true">
>>
>>     <action inline="true" application="set"
>>     data="continue_on_fail=true"/>
>>
>>     <action inline="true" application="set"
>>     data="hangup_after_bridge=true"/>
>>
>>     <action application="answer"/>
>>
>>     <action application="ivr" data="tivr"/>
>>
>>     </condition>
>>
>>     </extension>
>>
>>     </context>
>>
>>     </section>
>>
>>     And the ivr configuration returned by my xml_curl
>>
>>     <include>
>>
>>     <!-- demo IVR setup -->
>>
>>     <!-- demo IVR, Main Menu -->
>>
>>     <menu name="tivr"
>>
>>     greet-long="phrase:demo_ivr_main_menu"
>>
>>     greet-short="phrase:demo_ivr_main_menu_short"
>>
>>     invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>
>>     exit-sound="voicemail/vm-goodbye.wav"
>>
>>     confirm-macro=""
>>
>>     confirm-key=""
>>
>>     tts-engine="flite"
>>
>>     tts-voice="rms"
>>
>>     confirm-attempts="3"
>>
>>     timeout="10000"
>>
>>     inter-digit-timeout="2000"
>>
>>     max-failures="3"
>>
>>     max-timeouts="3"
>>
>>     digit-len="4">
>>
>>     <!-- The following are the definitions for the digits the user
>>     dials -->
>>
>>     <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
>>
>>     <entry action="menu-exec-app" digits="1" param="bridge
>>     sofia/$${domain}/888 at conference.freeswitch.org
>>     <mailto:domain%7d/888 at conference.freeswitch.org>"/>
>>
>>     <entry action="menu-exec-app" digits="2" param="transfer 9196 XML
>>     default"/> <!-- FS echo -->
>>
>>     <entry action="menu-exec-app" digits="3" param="transfer 9664 XML
>>     default"/> <!-- MOH -->
>>
>>     <entry action="menu-exec-app" digits="4" param="transfer 9191 XML
>>     default"/> <!-- ClueCon -->
>>
>>     <entry action="menu-exec-app" digits="5" param="transfer 1234*256
>>     enum"/> <!-- Screaming monkeys -->
>>
>>     <!-- Using a regex in the digits tag lets you define a dial
>>     pattern for the caller
>>
>>     You may define multiple regexes if you need a different pattern
>>     for some reason -->
>>
>>     <entry action="menu-exec-app" digits="/^(10[01][0-9])$/"
>>     param="transfer $1 XML features"/>
>>
>>     <entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
>>
>>     </menu>
>>
>>     </include>
>>
>>     Any ideas on the 'unable to find menu' ????
>>
>>     Thanks
>>
>>     Tim
>>
>>
>>
>>
>>     _________________________________________________________________________
>>
>>     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
>> 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 freeswitch at orresta.no-ip.com  Tue Dec 25 01:04:07 2012
From: freeswitch at orresta.no-ip.com (Jakob)
Date: Mon, 24 Dec 2012 23:04:07 +0100
Subject: [Freeswitch-users] anybody working on a swedish mod_say?
Message-ID: <50D8D157.7020409@orresta.no-ip.com>

Hi,

I started working on a project that would include a voicemail app. but I 
have trouble finding a swedish implementation of a mod_say so that I can 
populate it with swedish prompts.

I already have an working IVR, in swedish but there are still a few 
bugs, mainly getting english numbers in my swedish prompts.

Jakob


From Tim.Meade at Millicorp.com  Tue Dec 25 01:05:27 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Mon, 24 Dec 2012 22:05:27 +0000
Subject: [Freeswitch-users] xml_curl and ivr menus
In-Reply-To: <50D8CCC3.7090305@gmail.com>
References: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>
	<50D8BE29.7010404@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8A9E3@MAIL.millicorp.com>
	<50D8CC6A.5000706@gmail.com> <50D8CCC3.7090305@gmail.com>
Message-ID: <804D48104511D4468F0D60DF9D3100350AF8B121@MAIL.millicorp.com>

Thank you Abaci....   Just found that myself under the xml_curl section of the wiki.....  I hadn't noticed the xml_curl configuration parameters for the return xml are different.

Here is the completed working configuration from xml_curl debug_on.

<document type="freeswitch/xml">
<section name="configuration">
<configuration name="ivr.conf" description="IVR menus">
<menus>
<include>
<!-- demo IVR setup -->
<!-- demo IVR, Main Menu -->
<menu name="tivr"
greet-long="phrase:demo_ivr_main_menu"
greet-short="phrase:demo_ivr_main_menu_short"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
confirm-macro=""
confirm-key=""
tts-engine="flite"
tts-voice="rms"
confirm-attempts="3"
timeout="10000"
inter-digit-timeout="2000"
max-failures="3"
max-timeouts="3"
digit-len="4">
<!-- The following are the definitions for the digits the user dials -->
<!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
<entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888 at conference.freeswitch.org"/>
<entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/> <!-- FS echo -->
<entry action="menu-exec-app" digits="3" param="transfer 9664 XML default"/> <!-- MOH -->
<entry action="menu-exec-app" digits="4" param="transfer 9191 XML default"/> <!-- ClueCon -->
<entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/> <!-- Screaming monkeys -->
<!-- Using a regex in the digits tag lets you define a dial pattern for the caller
You may define multiple regexes if you need a different pattern for some reason -->
<entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
<entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
</menu>
</include>
</menus>
</configuration>
</section>
</document>

From: Abaci [mailto:abaci64 at gmail.com]
Sent: Monday, December 24, 2012 4:45 PM
To: FreeSWITCH Users Help
Cc: Tim Meade
Subject: Re: [Freeswitch-users] xml_curl and ivr menus

you're also missing the "section" element.

On 12/24/2012 4:43 PM, Abaci wrote:
Here is a sample from my server, I highlighted the missing stuff.

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="configuration">
    <configuration name="ivr.conf" description="IVR menus">
      <menus>
        <menu name="ivrmenu1"
              invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
              exit-sound="voicemail/vm-goodbye.wav"
              timeout ="3000"
              inter-digit-timeout="2000"
              max-failures="2"
              digit-len="3"
              phrase-lang="en">
          <entry action="menu-exec-app" digits="0" param="transfer 11 XML voicemail " />
          <entry action="menu-exec-app" digits="1" param="transfer ivrmenu2 XML ivr " />
          <entry action="menu-exec-app" digits="2" param="transfer ivrmenu3 XML ivr " />
        </menu>
      </menus>
    </configuration>
  </section>
</document>

On 12/24/2012 4:22 PM, Tim Meade wrote:
The items I pasted below are from the cat of the xml_curl debug_on command.  It certainly appears to me that freeswitch is getting the correct name in this case 'tivr' which is what shows in both the dialplan and the curl'd ivr menu.

My concerns now are dealing with the xml format for the ivr as I have found several different versions on the wiki and in the prior postings.

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 Abaci
Sent: Monday, December 24, 2012 3:42 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] xml_curl and ivr menus

on freeswitch enable xml_curl debug and see if freeswitch is actually getting the correct ivr
'xml_curl debug_on'

On 12/24/2012 2:20 PM, Tim Meade wrote:
I am continually getting 'unable to find menu' while trying to retrieve an ivr_menu using xml_curl

Using xml_curl debug_on

My DialPlan

<document type="freeswitch/xml">
<section name="dialplan" description="example_curl_dialplan">
<context name="public">
<extension name="ws_origination">
<condition field="destination_number" expression="^\d+$" break="on-true">
<action inline="true" application="set" data="continue_on_fail=true"/>
<action inline="true" application="set" data="hangup_after_bridge=true"/>
<action application="answer"/>
<action application="ivr" data="tivr"/>
</condition>
</extension>
</context>
</section>

And the ivr configuration returned by my xml_curl

<include>
<!-- demo IVR setup -->
<!-- demo IVR, Main Menu -->
<menu name="tivr"
greet-long="phrase:demo_ivr_main_menu"
greet-short="phrase:demo_ivr_main_menu_short"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
confirm-macro=""
confirm-key=""
tts-engine="flite"
tts-voice="rms"
confirm-attempts="3"
timeout="10000"
inter-digit-timeout="2000"
max-failures="3"
max-timeouts="3"
digit-len="4">
<!-- The following are the definitions for the digits the user dials -->
<!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
<entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888 at conference.freeswitch.org<mailto:domain%7d/888 at conference.freeswitch.org>"/>
<entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/> <!-- FS echo -->
<entry action="menu-exec-app" digits="3" param="transfer 9664 XML default"/> <!-- MOH -->
<entry action="menu-exec-app" digits="4" param="transfer 9191 XML default"/> <!-- ClueCon -->
<entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/> <!-- Screaming monkeys -->
<!-- Using a regex in the digits tag lets you define a dial pattern for the caller
You may define multiple regexes if you need a different pattern for some reason -->
<entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
<entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
</menu>
</include>

Any ideas on the 'unable to find menu' ????

Thanks

Tim








_________________________________________________________________________

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



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  Tue Dec 25 01:17:39 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 25 Dec 2012 00:17:39 +0200
Subject: [Freeswitch-users] anybody working on a swedish mod_say?
In-Reply-To: <50D8D157.7020409@orresta.no-ip.com>
References: <50D8D157.7020409@orresta.no-ip.com>
Message-ID: <CALNZuwnhLTjWbTA5rNpEGY_RcasMcYVa7QpJfn=jhiYWz89iAQ@mail.gmail.com>

I can't find a list on the wiki.. but there doesn't seem to be a mod_say_**
for swedish prompts.
see in git source: src/mod/say

If one of the languages is similar in making things singular/plural,
masculine/femanine you can use that as a template.

Or you can create speech phrases for numbers with the logic in there, but
that's a hack..


-Avi


On Tue, Dec 25, 2012 at 12:04 AM, Jakob <freeswitch at orresta.no-ip.com>wrote:

> Hi,
>
> I started working on a project that would include a voicemail app. but I
> have trouble finding a swedish implementation of a mod_say so that I can
> populate it with swedish prompts.
>
> I already have an working IVR, in swedish but there are still a few
> bugs, mainly getting english numbers in my swedish prompts.
>
> Jakob
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 8f27e956 at gmail.com  Tue Dec 25 03:14:03 2012
From: 8f27e956 at gmail.com (Scott)
Date: Mon, 24 Dec 2012 19:14:03 -0500
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <50D6865B.4010503@communicatefreely.net>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<50D6865B.4010503@communicatefreely.net>
Message-ID: <CA+Y7Lf4fG_eT_f6jAbpPhE0VhPd2=A9jdXt-OtJFnHd-jrtjeA@mail.gmail.com>

IPv6 is NOT a NAT cure all (it may look like it remedies IPv4
PORT-address-translations (PAT, not NAT) issues, but IPv6 brings its own
issues to the table.

The intrinsic structure of the IPv6 address favors carriers/providers.
Today, you get a global IPv4 WAN address and, typically, you then use a
10/8, 172/12, 192.168/16 LAN address schema.  WITH IPv6 and WITHOUT
IPv6-NAT (or NAT64), as an end user topology, the network portion of your
carrier/provider IPv6 WAN address is, all other things being equal,
propagated into LAN addresses schema on down to the deepest end points.  If
you subsequently change carrier/provider, then the network portion changes
and, in this use case, ALL your LAN addressing MUST change accordingly.
Things like statically addressed servers and router/switch ports, et al,
ALL have to be re-configured.

WITH IPv6 and WITH IPv6-NAT (and/or NAT64), you can utilize an internal
IPv6 LAN address schema that can then be 1:1 IPv6-NAT'ed to anything your
carrier/provider throws at you, regardless of whether you change C/P or the
C/P periodically changes your addresses.  This true
NETWORK-address-translation (NAT) affords a 1:1 address mapping, which
eliminates the shortfalls of the current IPv4 PORT-address-translations
(PAT, not NAT).  Other reasons for inside-outside translations include
topology hiding (security), et cetera.

Anyone with medium to large addressable end-points in their installations
really needs to look at implementing IPv6 WITH -- repeat WITH -- IPv6-NAT
(and/or NAT64) in the mix.

IMO, NAT will NOT BE DEAD in an IPv6 universe.  PAT may be n-stage, but not
NAT.



On 22 December 2012 23:19, Tim St. Pierre <fs-list at communicatefreely.net>wrote:

> Just throwing this out there -
>
> As it's clear how many problems NAT causes, why don't we all step up the
> pressure on phone
> manufacturers and ISPs to provide working IPv6 implementations so we can
> put NAT behind us
>   as soon as possible.
>
> Just saying -
>
> -Tim
>
>
> Cal Leeming [Simplicity Media Ltd] wrote:
> > *Any and all feedback on this thread would be much welcomed.*
> >
> > Hello,
> > *
> > *
> > There seems to be a large number of discussions surrounding NAT
> > traversal, as well as lots of documentation, but with no concrete
> answers.
> >
> > The NAT related wiki documentation is tedious, and depending on the
> > outcome of this thread, I'd like to spend some time cleaning it up.
> >
> > The most common problem (the same as ours) was having a router with
> > broken ALG and a softphone that does not seem to work with STUN.
> >
> > The following REGISTER is sent from a phone.
> >
> > REGISTER sip:1.2.3.4:5060 <http://1.2.3.4:5060> SIP/2.0
> > Via: SIP/2.0/UDP
> > 192.168.1.102:57787
> ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport
> > Max-Forwards: 70
> > Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>
> > To: "foxx"<sip:2000 at 1.2.3.4:5060 <http://sip:2000 at 1.2.3.4:5060>>
> > From: "foxx"<sip:2000 at 1.2.3.4:5060
> > <http://sip:2000 at 1.2.3.4:5060>>;tag=83311448
> > Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.
> > CSeq: 7 REGISTER
> > Expires: 120
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
> > REFER, INFO, MESSAGE
> > Supported: replaces
> > User-Agent: 3CXPhone 6.0.25732.0
> > Content-Length: 0
> >
> > As you can see, the client's public IP is not specified
> > anywhere. FreeSWITCH offers several ways around this, the main ones
> being;
> >
> > * NDLB-connectile-dysfunction
> > * NDLB-force-rport
> > * apply-nat-acl
> > * sip-force-contact
> >
> > The one that has worked in our case was "NDLB-connectile-dysfunction"
> > (otherwise known as NAT HACK), however there seems to be a lot of
> > negative comments about using this.
> >
> >  From what I can tell, the general argument is that NAT HACK is
> > considered a non RFC compliant hack, and the SIP phones should be doing
> > a better job of keeping to the RFCs.
> >
> > In principle, this is a fair argument - but in practise, it's not a
> > reasonable assumption that all phones are RFC compliant, and (imho) not
> > a reasonable argument to have this functionality disabled by default.
> >
> > So, I'd like to present the following arguments;
> >
> > * Are there any other negative aspects about
> > using NDLB-connectile-dysfunction, other than it is a non compliant RFC
> > hack?
> >
> > * Why is NDLB-connectile-dysfunction not enabled by default when certain
> > conditions are met? In the event that FreeSWITCH receives a REGISTER
> > from a phone specifying a Contact/Via as 192.168.0.0/16
> > <http://192.168.0.0/16>, but received on a public IP, then it should be
> > obvious that NAT is broken and automatically try to circumvent it.
> >
> > * People seem to get confused between server side and client side NAT
> > problems, and that they both need to be resolved in a different way. The
> > documentation doesn't seem to reflect this clearly.
> >
> >
> > ------------------------------------------------------------------------
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue Dec 25 05:06:05 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Mon, 24 Dec 2012 18:06:05 -0800
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CA+Y7Lf4fG_eT_f6jAbpPhE0VhPd2=A9jdXt-OtJFnHd-jrtjeA@mail.gmail.com>
Message-ID: <20121225020605.f01332d3@mail.tritonwest.net>

If you have a full IPv6 network, using all the protocols, like dns registration, DHCP-V6 for assigning networks to routers, Neighbor Discovery both directions, so that DNS delegations can be automated, then as long as you reference everything by host name, it should all just work. It's been working that way for me, with the only having to update nameserver helper records once and a while.  
   
If you happen have your own address space, and with to managed it downstream, all the services can also be implmented easy so that your down streams don't have to do anything to make all the auto address assignment, and DNS work.   
   
If you're going to use IPv6, set up the infrastructure FIRST, so managing it doesn't get out of hand. If done right at all layers it just works, when you use names.  
   
--Dave
      _____  

  From: Scott [mailto:8f27e956 at gmail.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Mon, 24 Dec 2012 16:14:03 -0800
Subject: Re: [Freeswitch-users] NAT traversal - the final say..!

IPv6 is NOT a NAT cure all (it may look like it remedies IPv4 PORT-address-translations (PAT, not NAT) issues, but IPv6 brings its own issues to the table.

The intrinsic structure of the IPv6 address favors carriers/providers.  Today, you get a global IPv4 WAN address and, typically, you then use a 10/8, 172/12, 192.168/16 LAN address schema.  WITH IPv6 and WITHOUT IPv6-NAT (or NAT64), as an end user topology, the network portion of your carrier/provider IPv6 WAN address is, all other things being equal, propagated into LAN addresses schema on down to the deepest end points.  If you subsequently change carrier/provider, then the network portion changes and, in this use case, ALL your LAN addressing MUST change accordingly.  Things like statically addressed servers and router/switch ports, et al, ALL have to be re-configured.

WITH IPv6 and WITH IPv6-NAT (and/or NAT64), you can utilize an internal IPv6 LAN address schema that can then be 1:1 IPv6-NAT'ed to anything your carrier/provider throws at you, regardless of whether you change C/P or the C/P periodically changes your addresses.  This true NETWORK-address-translation (NAT) affords a 1:1 address mapping, which eliminates the shortfalls of the current IPv4 PORT-address-translations (PAT, not NAT).  Other reasons for inside-outside translations include topology hiding (security), et cetera.

Anyone with medium to large addressable end-points in their installations really needs to look at implementing IPv6 WITH -- repeat WITH -- IPv6-NAT (and/or NAT64) in the mix. 

IMO, NAT will NOT BE DEAD in an IPv6 universe.  PAT may be n-stage, but not NAT.



  
On 22 December 2012 23:19, Tim St. Pierre <fs-list at communicatefreely.net> wrote:
  Just throwing this out there -

As it's clear how many problems NAT causes, why don't we all step up the pressure on phone
manufacturers and ISPs to provide working IPv6 implementations so we can put NAT behind us
  as soon as possible.

Just saying -

-Tim


Cal Leeming [Simplicity Media Ltd] wrote:
> *Any and all feedback on this thread would be much welcomed.*
>
> Hello,
> *
  
> *
> There seems to be a large number of discussions surrounding NAT
> traversal, as well as lots of documentation, but with no concrete answers.
>
> The NAT related wiki documentation is tedious, and depending on the
> outcome of this thread, I'd like to spend some time cleaning it up.
>
> The most common problem (the same as ours) was having a router with
> broken ALG and a softphone that does not seem to work with STUN.
>
> The following REGISTER is sent from a phone.
>
> REGISTER sip:1.2.3.4:5060 <http://1.2.3.4:5060> SIP/2.0
  
> Via: SIP/2.0/UDP
> 192.168.1.102:57787;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport
> Max-Forwards: 70
> Contact: <sip:2000 at 192.168.1.102:57787;rinstance=0c7190b115a36513>
> To: "foxx"<sip:2000 at 1.2.3.4:5060 <http://sip:2000 at 1.2.3.4:5060>>
  
> From: "foxx"<sip:2000 at 1.2.3.4:5060
> <http://sip:2000 at 1.2.3.4:5060>>;tag=83311448
  
  
> Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.
> CSeq: 7 REGISTER
> Expires: 120
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
> REFER, INFO, MESSAGE
> Supported: replaces
> User-Agent: 3CXPhone 6.0.25732.0
> Content-Length: 0
>
> As you can see, the client's public IP is not specified
> anywhere. FreeSWITCH offers several ways around this, the main ones being;
>
> * NDLB-connectile-dysfunction
> * NDLB-force-rport
> * apply-nat-acl
> * sip-force-contact
>
> The one that has worked in our case was "NDLB-connectile-dysfunction"
> (otherwise known as NAT HACK), however there seems to be a lot of
> negative comments about using this.
>
>  From what I can tell, the general argument is that NAT HACK is
> considered a non RFC compliant hack, and the SIP phones should be doing
> a better job of keeping to the RFCs.
>
> In principle, this is a fair argument - but in practise, it's not a
> reasonable assumption that all phones are RFC compliant, and (imho) not
> a reasonable argument to have this functionality disabled by default.
>
> So, I'd like to present the following arguments;
>
> * Are there any other negative aspects about
> using NDLB-connectile-dysfunction, other than it is a non compliant RFC
> hack?
>
> * Why is NDLB-connectile-dysfunction not enabled by default when certain
> conditions are met? In the event that FreeSWITCH receives a REGISTER
> from a phone specifying a Contact/Via as 192.168.0.0/16
> <http://192.168.0.0/16>, but received on a public IP, then it should be
  
> obvious that NAT is broken and automatically try to circumvent it.
>
> * People seem to get confused between server side and client side NAT
> problems, and that they both need to be resolved in a different way. The
> documentation doesn't seem to reflect this clearly.
>
>
> ------------------------------------------------------------------------
  
  
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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

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

      
   
 
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From 8f27e956 at gmail.com  Tue Dec 25 07:59:39 2012
From: 8f27e956 at gmail.com (S. Scott)
Date: Mon, 24 Dec 2012 23:59:39 -0500
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <20121225020605.f01332d3@mail.tritonwest.net>
References: <20121225020605.f01332d3@mail.tritonwest.net>
Message-ID: <9103480538928767893@unknownmsgid>

Dave, when you say, "...If you're going to use IPv6, set up the
infrastructure FIRST," we both agree 100%.  You appear to have a well
ordered DNS infrastructure and an admin discipline and overarch to
effect it, where the scope of the tool covers off your operations and
problem set.

:-)

The mental gymnastics -- and the cacoffiny -- will rise over what
"infrastructure" components work good/better/best for each admin once
the carriers/providers, data centers, hosting providers, wireless,
"guestnet" providers -- starbucks, hotels, corporate visitors nets,
etc. -- all start trying it their way.

And all i'm saying is that 1:1 NAT (not PAT) can (and will) play a
viable role in competent/robust IPv6 infrastructures; and, therefore,
IMO, NAT isn't going away through IPv6 pass 1, 2, or 3.

Bet 'ya a dollar and see you in five-seven years at Cluecon.

;-)

Cheers,

?????
iThing:  Big thumbs & little keys. Please excuse typo, spelling and
grammar errors ? Last night I played a blank CD at full blast. The
Mime next door went nuts.


On 2012-12-24, at 21:10, "Dave R. Kompel" <drk at drkngs.net> wrote:

> If you're going to use IPv6, set up the infrastructure FIRST,


From vbvbrj at gmail.com  Tue Dec 25 10:35:35 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Tue, 25 Dec 2012 09:35:35 +0200
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CA+Y7Lf4fG_eT_f6jAbpPhE0VhPd2=A9jdXt-OtJFnHd-jrtjeA@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<50D6865B.4010503@communicatefreely.net>
	<CA+Y7Lf4fG_eT_f6jAbpPhE0VhPd2=A9jdXt-OtJFnHd-jrtjeA@mail.gmail.com>
Message-ID: <50D95747.6090502@gmail.com>

On 25.12.2012 02:14, Scott wrote:

> Anyone with medium to large addressable end-points in their
> installations really needs to look at implementing IPv6 WITH -- repeat
> WITH -- IPv6-NAT (and/or NAT64) in the mix.
>
I will use this when moving to ipv6. At my company we have two internet 
and voip providers for fault tolerance and load balacing. Also for using 
lowest cost per call. The plans are to have third internet and voip 
provider.

So using internal ipv6 for addressing and dynamically mapping to those, 
provided by internet providers, is the only way to work.

-- 
Mimiko desu.


From Tim.Meade at Millicorp.com  Tue Dec 25 17:07:10 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Tue, 25 Dec 2012 14:07:10 +0000
Subject: [Freeswitch-users] xml_curl and ivr menus
In-Reply-To: <804D48104511D4468F0D60DF9D3100350AF8B121@MAIL.millicorp.com>
References: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>
	<50D8BE29.7010404@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8A9E3@MAIL.millicorp.com>
	<50D8CC6A.5000706@gmail.com> <50D8CCC3.7090305@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8B121@MAIL.millicorp.com>
Message-ID: <804D48104511D4468F0D60DF9D3100350AF8DCA3@MAIL.millicorp.com>

Here is an example of the xml_curl phrases section.  This is not in the WIKI.....  I'll try to update.  Notice the language section....

<document type="freeswitch/xml">
                  <section name="phrases">
                      <macros>
                           <include>
                                <language name="en" sound-path="$${sounds_dir}/en/us/callie" tts-engine="cepstral" tts-voice="callie">
                                     <macro name="demo_ivr_main_menu" pause="100"> <!-- See conf/autoload_config/ivr.conf.xml for an example on how to use this macro in an IVR -->
                                               <input pattern="(.*)">
                                                 <match>
                                                   <!-- string together several existing sound files to create one long greeting -->
                                                   <action function="play-file" data="ivr/ivr-welcome_to_freeswitch.wav"/>
                                                   <action function="play-file" data="ivr/ivr-this_ivr_will_let_you_test_features.wav"/>
                                                   <action function="play-file" data="ivr/ivr-you_may_exit_by_hanging_up.wav"/>
                                                   <!-- note that you can do more than just play files, e.g. have pauses and do TTS -->

                                                   <!-- Menu option 1: Call FreeSWITCH conference-->
                                                   <action function="play-file" data="ivr/ivr-enter_ext_pound.wav"/>
                                                   <action function="play-file" data="silence_stream://1500"/>
                                                   <action function="play-file" data="ivr/ivr-to_call_the_freeswitch_conference.wav"/>
                                                   <action function="play-file" data="ivr/ivr-please.wav"/>
                                                   <action function="play-file" data="voicemail/vm-press.wav"/>
                                                   <action function="play-file" data="digits/1.wav"/>

                                                   <!-- Menu option 2: Do FreeSWITCH echo test -->
                                                   <action function="play-file" data="ivr/ivr-to_do_a_freeswitch_echo_test.wav"/>
                                                   <action function="play-file" data="ivr/ivr-please.wav"/>
                                                   <action function="play-file" data="voicemail/vm-press.wav"/>
                                                   <action function="play-file" data="digits/2.wav"/>

                                                   <!-- Menu option 3: Listen to Music on Hold -->
                                                   <action function="play-file" data="ivr/ivr-to_listen_to_moh.wav"/>
                                                   <action function="play-file" data="ivr/ivr-please.wav"/>
                                                   <action function="play-file" data="voicemail/vm-press.wav"/>
                                                   <action function="play-file" data="digits/3.wav"/>

                                                   <!-- Menu option 4: Register for ClueCon -->
                                                   <action function="play-file" data="ivr/ivr-register_for_cluecon.wav"/>
                                                   <action function="play-file" data="digits/4.wav"/>

                                                   <!-- Menu option 5: Listen to screaming monkeys -->
                                                   <action function="play-file" data="ivr/ivr-to_hear_screaming_monkeys.wav"/>
                                                   <action function="play-file" data="ivr/ivr-please.wav"/>
                                                   <action function="play-file" data="voicemail/vm-press.wav"/>
                                                   <action function="play-file" data="digits/5.wav"/>

                                                   <!-- Menu option 6: Hear a sample submenu -->
                                                   <action function="play-file" data="ivr/ivr-to_hear_sample_submenu.wav"/>
                                                   <action function="play-file" data="ivr/ivr-please.wav"/>
                                                   <action function="play-file" data="voicemail/vm-press.wav"/>
                                                   <action function="play-file" data="digits/6.wav"/>

                                                   <!-- Menu option 9: Repeat these options -->
                                                   <action function="play-file" data="ivr/ivr-to_repeat_these_options.wav"/>
                                                   <action function="play-file" data="ivr/ivr-please.wav"/>
                                                   <action function="play-file" data="voicemail/vm-press.wav"/>
                                                   <action function="play-file" data="digits/9.wav"/>
                                                   <action function="play-file" data="silence_stream://2000"/>
                                                 </match>
                                               </input>
                                             </macro>
                                       </language>
                                </include>
                       </macros>
                  </section>
                </document>


Thanks All

Tim

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim Meade
Sent: Monday, December 24, 2012 5:05 PM
To: Abaci; FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] xml_curl and ivr menus

Thank you Abaci....   Just found that myself under the xml_curl section of the wiki.....  I hadn't noticed the xml_curl configuration parameters for the return xml are different.

Here is the completed working configuration from xml_curl debug_on.

<document type="freeswitch/xml">
<section name="configuration">
<configuration name="ivr.conf" description="IVR menus">
<menus>
<include>
<!-- demo IVR setup -->
<!-- demo IVR, Main Menu -->
<menu name="tivr"
greet-long="phrase:demo_ivr_main_menu"
greet-short="phrase:demo_ivr_main_menu_short"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
confirm-macro=""
confirm-key=""
tts-engine="flite"
tts-voice="rms"
confirm-attempts="3"
timeout="10000"
inter-digit-timeout="2000"
max-failures="3"
max-timeouts="3"
digit-len="4">
<!-- The following are the definitions for the digits the user dials -->
<!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
<entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888 at conference.freeswitch.org"/>
<entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/> <!-- FS echo -->
<entry action="menu-exec-app" digits="3" param="transfer 9664 XML default"/> <!-- MOH -->
<entry action="menu-exec-app" digits="4" param="transfer 9191 XML default"/> <!-- ClueCon -->
<entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/> <!-- Screaming monkeys -->
<!-- Using a regex in the digits tag lets you define a dial pattern for the caller
You may define multiple regexes if you need a different pattern for some reason -->
<entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
<entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
</menu>
</include>
</menus>
</configuration>
</section>
</document>

From: Abaci [mailto:abaci64 at gmail.com]
Sent: Monday, December 24, 2012 4:45 PM
To: FreeSWITCH Users Help
Cc: Tim Meade
Subject: Re: [Freeswitch-users] xml_curl and ivr menus

you're also missing the "section" element.

On 12/24/2012 4:43 PM, Abaci wrote:
Here is a sample from my server, I highlighted the missing stuff.

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="configuration">
    <configuration name="ivr.conf" description="IVR menus">
      <menus>
        <menu name="ivrmenu1"
              invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
              exit-sound="voicemail/vm-goodbye.wav"
              timeout ="3000"
              inter-digit-timeout="2000"
              max-failures="2"
              digit-len="3"
              phrase-lang="en">
          <entry action="menu-exec-app" digits="0" param="transfer 11 XML voicemail " />
          <entry action="menu-exec-app" digits="1" param="transfer ivrmenu2 XML ivr " />
          <entry action="menu-exec-app" digits="2" param="transfer ivrmenu3 XML ivr " />
        </menu>
      </menus>
    </configuration>
  </section>
</document>

On 12/24/2012 4:22 PM, Tim Meade wrote:
The items I pasted below are from the cat of the xml_curl debug_on command.  It certainly appears to me that freeswitch is getting the correct name in this case 'tivr' which is what shows in both the dialplan and the curl'd ivr menu.

My concerns now are dealing with the xml format for the ivr as I have found several different versions on the wiki and in the prior postings.

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 Abaci
Sent: Monday, December 24, 2012 3:42 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] xml_curl and ivr menus

on freeswitch enable xml_curl debug and see if freeswitch is actually getting the correct ivr
'xml_curl debug_on'

On 12/24/2012 2:20 PM, Tim Meade wrote:
I am continually getting 'unable to find menu' while trying to retrieve an ivr_menu using xml_curl

Using xml_curl debug_on

My DialPlan

<document type="freeswitch/xml">
<section name="dialplan" description="example_curl_dialplan">
<context name="public">
<extension name="ws_origination">
<condition field="destination_number" expression="^\d+$" break="on-true">
<action inline="true" application="set" data="continue_on_fail=true"/>
<action inline="true" application="set" data="hangup_after_bridge=true"/>
<action application="answer"/>
<action application="ivr" data="tivr"/>
</condition>
</extension>
</context>
</section>

And the ivr configuration returned by my xml_curl

<include>
<!-- demo IVR setup -->
<!-- demo IVR, Main Menu -->
<menu name="tivr"
greet-long="phrase:demo_ivr_main_menu"
greet-short="phrase:demo_ivr_main_menu_short"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
confirm-macro=""
confirm-key=""
tts-engine="flite"
tts-voice="rms"
confirm-attempts="3"
timeout="10000"
inter-digit-timeout="2000"
max-failures="3"
max-timeouts="3"
digit-len="4">
<!-- The following are the definitions for the digits the user dials -->
<!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
<entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888 at conference.freeswitch.org<mailto:domain%7d/888 at conference.freeswitch.org>"/>
<entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/> <!-- FS echo -->
<entry action="menu-exec-app" digits="3" param="transfer 9664 XML default"/> <!-- MOH -->
<entry action="menu-exec-app" digits="4" param="transfer 9191 XML default"/> <!-- ClueCon -->
<entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/> <!-- Screaming monkeys -->
<!-- Using a regex in the digits tag lets you define a dial pattern for the caller
You may define multiple regexes if you need a different pattern for some reason -->
<entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
<entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
</menu>
</include>

Any ideas on the 'unable to find menu' ????

Thanks

Tim







_________________________________________________________________________

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



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 tahir at ictinnovations.com  Wed Dec 26 12:05:13 2012
From: tahir at ictinnovations.com (Tahir Almas)
Date: Wed, 26 Dec 2012 14:05:13 +0500
Subject: [Freeswitch-users] ICTFAX Version 2.1 released
Message-ID: <CAOL0t+LcqXniMe0Z92G1P9VwoTxv1ZgZ6uBSa_hx2wayXkjh6g@mail.gmail.com>

Pleased to annouce the release of ICTFAX Ver 2.1  http://www.ictfax.org, a
freeswitch and plivo based Open Source Fax Over IP Solution


*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT
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From msc at freeswitch.org  Wed Dec 26 20:03:36 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Dec 2012 09:03:36 -0800
Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today
Message-ID: <CAKzWOxXFSYwXoCQSp6hY=RebbdqxoXH-6FEfqj5EMZO_nmJQGg@mail.gmail.com>

Hi gang!

I know a lot of folks are gone for the holidays but anyone who is around is
welcome to join us today. We have a light agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_26

However, I believe at least one person will be joining us to talk about a
potential job for a FreeSWITCH contractor. Hope to talk to you all soon!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec 26 20:08:16 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Dec 2012 09:08:16 -0800
Subject: [Freeswitch-users] FS opening ports
In-Reply-To: <CAKbxfG98s8qAT3d+rO7yRVkLbUu1OEEOBZXnpXTk2sC-YrGZQQ@mail.gmail.com>
References: <20121222071101.0e1bd4d5c5064b420440751b21b10e46.81311ec1f9.wbe@email13.secureserver.net>
	<CAKbxfG98s8qAT3d+rO7yRVkLbUu1OEEOBZXnpXTk2sC-YrGZQQ@mail.gmail.com>
Message-ID: <CAKzWOxWFZhhHKvr-kdjobw+wFW=QdUrAXd8K6vc3T5DtkEaw2w@mail.gmail.com>

For posterity's sake here's the wiki page that talks about the "autonat"
feature that is baked into FreeSWITCH:

http://wiki.freeswitch.org/wiki/Auto_NAT

The -nonat param disables the auto nat busting stuff like the UPnP/NAT-PMP
port openings.
-MC

On Sat, Dec 22, 2012 at 9:14 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> start FS with -nonat command line arg
>
>
> On Sat, Dec 22, 2012 at 8:11 AM, <stephen at picardogroup.com> wrote:
>
>> Why does FS open ports on my FIOS router by itself?
>>
>> I have been seeing SIP 503and telephone calls in ring busy
>>
>> Is there a way to stop or limit FS from opening too many ports?
>>
>> localhost
>> xxx.x.x.x    Verizon FiOS Service
>> Tcp Any -> 4567    All Broadband Devices    Active
>>     new-host
>> 192.168.1.xx
>>     Destination Ports 8022
>> TCP Any -> 8022    All Broadband Devices    Active
>>
>>     192.168.1.
>>     SSH
>> TCP Any -> 22    All Broadband Devices    Active
>>
>>     192.168.1.xx:5070
>>     FreeSWITCH
>> UDP Any -> 5070    All Broadband Devices    Active
>>
>>     192.168.1.xx:5080
>>     FreeSWITCH
>> UDP Any -> 5080    All Broadband Devices    Active
>>
>>     192.168.1.xx:5060
>>     FreeSWITCH
>> UDP Any -> 5060    All Broadband Devices    Active
>>
>>     192.168.1.xx:5070
>>     FreeSWITCH
>> TCP Any -> 5070    All Broadband Devices    Active
>>
>>     192.168.1.xx:5080
>>     FreeSWITCH
>> TCP Any -> 5080    All Broadband Devices    Active
>>
>>     192.168.1.xx:5060
>>     FreeSWITCH
>> TCP Any -> 5060    All Broadband Devices    Active
>>
>>     192.168.1.xx:15952
>>     FreeSWITCH
>> UDP Any -> 15952    All Broadband Devices    Active
>>
>>     192.168.1.xx:15953
>>     FreeSWITCH
>> UDP Any -> 15953    All Broadband Devices    Active
>>
>>     192.168.1.xx:11350
>>     FreeSWITCH
>> UDP Any -> 11350    All Broadband Devices    Active
>>
>>     192.168.1.xx:11351
>>     FreeSWITCH
>> UDP Any -> 11351    All Broadband Devices    Active
>>
>>     192.168.1.xx:16052
>>     FreeSWITCH
>> UDP Any -> 16052    All Broadband Devices    Active
>>
>>     192.168.1.xx:16053
>>     FreeSWITCH
>> UDP Any -> 16053    All Broadband Devices    Active
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> --
> Anthony Minessale II
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec 26 20:10:40 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Dec 2012 09:10:40 -0800
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <50D57946.5040501@gmail.com>
References: <50D57946.5040501@gmail.com>
Message-ID: <CAKzWOxXOZM8XdhtShN_Gmrpp3vf506yK3k2dO3ZyXKMmGffo2Q@mail.gmail.com>

Does B actually answer the call in this scenario? If so then you can use
the *4 DTMF combo to execute an attended transfer. Just have B dial *4 then
the target phone number for C. When C answers B just needs to hang up and
then A will be talking to C.

-MC

On Sat, Dec 22, 2012 at 1:11 AM, veerabhadrarao` <
bhadrarao.kankatala at gmail.com> wrote:

> hai
>
> I am using Freeswitch 1.2.4
>
> currently i am working on Attendent transfer can u please help me how to
> make attendant transfer.
>
> let when i make a call A--->B  and B-->C then how can i connect to A--->C
>
> please help me
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec 26 20:14:00 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Dec 2012 09:14:00 -0800
Subject: [Freeswitch-users] stop a running lua script from luarun
In-Reply-To: <4F13F527-1F4A-41D5-81B6-4D7578D2426D@gmail.com>
References: <4F13F527-1F4A-41D5-81B6-4D7578D2426D@gmail.com>
Message-ID: <CAKzWOxVsPdvHtF1Hf238qH=OjxjL=1LvRF_0bb=_D0Pzp=5k=A@mail.gmail.com>

Have you been able to determine why it's stuck in an infinite loop? That
would be a better long-term solution. In the meantime I think you can do
top -H to see individual threads but I don't know how easy/difficult it is
to pick out which one is running Lua.
-MC

On Sun, Dec 23, 2012 at 2:21 AM, Muhammad Naseer Bhatti
<nbhatti at gmail.com>wrote:

>
> Looks like simple one, but I can't figure it out if there a way to stop a
> lua script invoked via luarun which is stuck in an infinite loop and won't
> terminate itself? Of if we can come up with a way to monitor the thread and
> stop it?
>
> Thanks,
> --
> Muhammad Naseer Bhatti
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec 26 20:16:45 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Dec 2012 09:16:45 -0800
Subject: [Freeswitch-users] New to Freeswitch
In-Reply-To: <CACahg4EKsXiRFkoYp-NbeRvtNsHLthxtu+k65pezCKw_wvhKbw@mail.gmail.com>
References: <CACahg4EKsXiRFkoYp-NbeRvtNsHLthxtu+k65pezCKw_wvhKbw@mail.gmail.com>
Message-ID: <CAKzWOxUpdX9-VXzYYMpbtEvuB1hw11v0jQMEp3p-Boh7T1NYTA@mail.gmail.com>

This is a good question for those who've been using mod_gsmopen. According
to the wiki <http://wiki.freeswitch.org/wiki/GSMopen>the dongle needs to be
compatible with Huawei E1550. Do you know if the 1732 is compatible?

-MC

On Sun, Dec 23, 2012 at 4:51 AM, Vijai Ganapathy
<vijai.ganapathy at gmail.com>wrote:

> Dear all,
>
> I am new to freeswitch and I have a question to ask.
> I am planning to use gsm_open to have small call center setup.
>
> From the local market I could get Huawei E1732 model Dongle only and I am
> wondering whether freeswitch / Linux Kernel
> would support this model  for voice ?
>
> Also would ATOM or Phenom would be enough to run 10 seat call center ?
>
> Want to check with experts before jumping on to actual setup.
>
> --
> Vijai Ganapathy
> Mailto: Vijai.Ganapathy at gmail.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From gmaruzz at gmail.com  Wed Dec 26 20:27:27 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 26 Dec 2012 18:27:27 +0100
Subject: [Freeswitch-users] New to Freeswitch
In-Reply-To: <CAKzWOxUpdX9-VXzYYMpbtEvuB1hw11v0jQMEp3p-Boh7T1NYTA@mail.gmail.com>
References: <CACahg4EKsXiRFkoYp-NbeRvtNsHLthxtu+k65pezCKw_wvhKbw@mail.gmail.com>
	<CAKzWOxUpdX9-VXzYYMpbtEvuB1hw11v0jQMEp3p-Boh7T1NYTA@mail.gmail.com>
Message-ID: <CALXCt0pwZMYi0Ny9JdJvixwROQ+o9L6GTuoqKqBPpVZ8uHEO6Q@mail.gmail.com>

Vijai, if you are not sure if it's compatible, your best option is to
test it. Huawei makes a lot of different dongles, and most of them are
compatible. But I completely don't know about 1732, so try it at your
own risk. Also, you may try alibaba.com as source for dongles.

-giovanni

On Wed, Dec 26, 2012 at 6:16 PM, Michael Collins <msc at freeswitch.org> wrote:
> This is a good question for those who've been using mod_gsmopen. According
> to the wiki the dongle needs to be compatible with Huawei E1550. Do you know
> if the 1732 is compatible?
>
> -MC
>
> On Sun, Dec 23, 2012 at 4:51 AM, Vijai Ganapathy <vijai.ganapathy at gmail.com>
> wrote:
>>
>> Dear all,
>>
>> I am new to freeswitch and I have a question to ask.
>> I am planning to use gsm_open to have small call center setup.
>>
>> From the local market I could get Huawei E1732 model Dongle only and I am
>> wondering whether freeswitch / Linux Kernel
>> would support this model  for voice ?
>>
>> Also would ATOM or Phenom would be enough to run 10 seat call center ?
>>
>> Want to check with experts before jumping on to actual setup.
>>
>> --
>> Vijai Ganapathy
>> Mailto: Vijai.Ganapathy at gmail.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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From msc at freeswitch.org  Wed Dec 26 21:15:29 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Dec 2012 10:15:29 -0800
Subject: [Freeswitch-users] XML DialPlan Gateay dialing redundancy
In-Reply-To: <0da201cde1fd$2a0b6de0$7e2249a0$@bizfocused.com>
References: <0da201cde1fd$2a0b6de0$7e2249a0$@bizfocused.com>
Message-ID: <CAKzWOxV-qZ7_xbLNQY23XGX=+PQZn_m5toy9-RaaKJGd1h-=Tw@mail.gmail.com>

Yeah, that's a fun one. You'll probably need to concoct a configuration
making use of continue_on_fail and failure_causes chan vars. You'll also
need to figure out all the ways in which each gateway's carrier might fail
the call. Oh, and "no answer" is also a "failure" so you'll need to find a
way to handle that differently than the other "failure" conditions. Perhaps
after each bridge attempt you could all a lua script that checks the value
of bridge_hangup_cause and then decide what to do with the call, i.e. on a
no answer you could transfer the call to a voicemail extension but on any
other failure cause you could simply exit the script and let the dialplan
continue processing.

Just as a starter maybe something like this:

<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_brdige=false"/>
<action application="bridge" data="sofia/gateway/gw1/$1"/>
<action application="lua" data="check_cause.lua"/>
<action application="bridge" data="sofia/gateway/gw2/$1"/>
<action application="lua" data="check_cause.lua"/>
<action application="bridge" data="sofia/gateway/gw3/$1"/>
<action application="lua" data="check_cause.lua"/>

You'll need to make sure the lua script also checks to see if the previous
bridge was answered and if so then hangup the call, otherwise it will keep
trying the next bridge command.

Have fun with that one! :)

-Michael

On Mon, Dec 24, 2012 at 9:36 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> ** **
>
> I need a little help with a complex XML dialplan.  Also, I am on the
> Stable release so I do not believe the much needed nested conditions exists
> for me yet.****
>
> ** **
>
> My problem is faulty gateway providers who periodically fail to connect
> the call with [temp failure] because ?our routing tables where not up to
> date.?****
>
> ** **
>
> So what I am after is an xml dialplan to do this****
>
> If (it is a 10 or 11 digit dial)****
>
>                 Dial using gateway 1****
>
>                 If no answer****
>
>                                 Go to voice mail****
>
>                 Else if gateway failure ****
>
>                                 Dial using gateway 2****
>
>                 If no answer****
>
>                                 Go to voice mail****
>
>                 Else if gateway failure ****
>
>                                 Dial using gateway 3****
>
>                 If no answer****
>
>                                 Go to voice mail****
>
>                                 Else if gateway failure****
>
>                                                 Play a message that the
> number is screwed!****
>
> ** **
>
> Perhaps I need to breakdown and use a scripting language like LUA or
> JavaScript for this, but I would rather keep everything in XML if possible.
> ****
>
> ** **
>
> Any thoughts?****
>
> ** **
>
> Happy Holidays to all,****
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec 26 21:41:45 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Dec 2012 10:41:45 -0800
Subject: [Freeswitch-users] anybody working on a swedish mod_say?
In-Reply-To: <CALNZuwnhLTjWbTA5rNpEGY_RcasMcYVa7QpJfn=jhiYWz89iAQ@mail.gmail.com>
References: <50D8D157.7020409@orresta.no-ip.com>
	<CALNZuwnhLTjWbTA5rNpEGY_RcasMcYVa7QpJfn=jhiYWz89iAQ@mail.gmail.com>
Message-ID: <CAKzWOxVG0kwrNmGULCaoE43XvexZoigBFBbJnt1cxR_EUiAWXA@mail.gmail.com>

Also, if you get a set of Swedish recordings let us know and we'll get them
hosted on our files.freeswitch.org server.
-MC

On Mon, Dec 24, 2012 at 2:17 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I can't find a list on the wiki.. but there doesn't seem to be a
> mod_say_** for swedish prompts.
> see in git source: src/mod/say
>
> If one of the languages is similar in making things singular/plural,
> masculine/femanine you can use that as a template.
>
> Or you can create speech phrases for numbers with the logic in there, but
> that's a hack..
>
>
> -Avi
>
>
> On Tue, Dec 25, 2012 at 12:04 AM, Jakob <freeswitch at orresta.no-ip.com>wrote:
>
>> Hi,
>>
>> I started working on a project that would include a voicemail app. but I
>> have trouble finding a swedish implementation of a mod_say so that I can
>> populate it with swedish prompts.
>>
>> I already have an working IVR, in swedish but there are still a few
>> bugs, mainly getting english numbers in my swedish prompts.
>>
>> Jakob
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From nbhatti at gmail.com  Wed Dec 26 22:12:28 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Wed, 26 Dec 2012 22:12:28 +0300
Subject: [Freeswitch-users] stop a running lua script from luarun
In-Reply-To: <CAKzWOxVsPdvHtF1Hf238qH=OjxjL=1LvRF_0bb=_D0Pzp=5k=A@mail.gmail.com>
References: <4F13F527-1F4A-41D5-81B6-4D7578D2426D@gmail.com>
	<CAKzWOxVsPdvHtF1Hf238qH=OjxjL=1LvRF_0bb=_D0Pzp=5k=A@mail.gmail.com>
Message-ID: <A391089F-7CCF-4766-A151-73A9A7C2C4A7@gmail.com>


No, I am running this loop myself. But I found other way to terminate the script, like look for a condition  in each iteration of the loop and if condition is true terminate the loop. But the question remains the same, if there is a way to figure out which scripts are running  to stop them if possible. Since all process are spawn by freeswitch top -H is not a handy solution.

Thanks,
--
Muhammad Naseer Bhatti



On Dec 26, 2012, at 8:14 PM, Michael Collins <msc at freeswitch.org> wrote:

> Have you been able to determine why it's stuck in an infinite loop? That would be a better long-term solution. In the meantime I think you can do top -H to see individual threads but I don't know how easy/difficult it is to pick out which one is running Lua.
> -MC
> 
> On Sun, Dec 23, 2012 at 2:21 AM, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:
> 
> Looks like simple one, but I can't figure it out if there a way to stop a lua script invoked via luarun which is stuck in an infinite loop and won't terminate itself? Of if we can come up with a way to monitor the thread and stop it?
> 
> Thanks,
> --
> Muhammad Naseer Bhatti
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 abaci64 at gmail.com  Wed Dec 26 22:18:25 2012
From: abaci64 at gmail.com (Abaci)
Date: Wed, 26 Dec 2012 14:18:25 -0500
Subject: [Freeswitch-users] page application
Message-ID: <50DB4D81.9010000@gmail.com>

There is an application "page" (in mod_dptools), I didn't find any 
documentation for it on the wiki so I'm sending here if someone can 
provide details how to use it.
see 
http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/applications/mod_dptools/mod_dptools.c?r1=bc0912cd11efb3f4525b8150733035ef37fc2ee3&r2=0a0f5951ab616d5b136e79b7d4ad253510f9d0cc
Thanks


From vijai.ganapathy at gmail.com  Wed Dec 26 21:56:20 2012
From: vijai.ganapathy at gmail.com (Vijai Ganapathy)
Date: Thu, 27 Dec 2012 00:26:20 +0530
Subject: [Freeswitch-users] New to Freeswitch
In-Reply-To: <CALXCt0pwZMYi0Ny9JdJvixwROQ+o9L6GTuoqKqBPpVZ8uHEO6Q@mail.gmail.com>
References: <CACahg4EKsXiRFkoYp-NbeRvtNsHLthxtu+k65pezCKw_wvhKbw@mail.gmail.com>
	<CAKzWOxUpdX9-VXzYYMpbtEvuB1hw11v0jQMEp3p-Boh7T1NYTA@mail.gmail.com>
	<CALXCt0pwZMYi0Ny9JdJvixwROQ+o9L6GTuoqKqBPpVZ8uHEO6Q@mail.gmail.com>
Message-ID: <CACahg4Fb9=gVU_Mp4WTdsUcXixtsnKNRmfD2y1s_Nc3x_5MDMg@mail.gmail.com>

I will check and update here.

BTW. is there a quick way to test without all installing / configuring ?
like boot distro which load freeswitch ?


On Wed, Dec 26, 2012 at 10:57 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> Vijai, if you are not sure if it's compatible, your best option is to
> test it. Huawei makes a lot of different dongles, and most of them are
> compatible. But I completely don't know about 1732, so try it at your
> own risk. Also, you may try alibaba.com as source for dongles.
>
> -giovanni
>
> On Wed, Dec 26, 2012 at 6:16 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > This is a good question for those who've been using mod_gsmopen.
> According
> > to the wiki the dongle needs to be compatible with Huawei E1550. Do you
> know
> > if the 1732 is compatible?
> >
> > -MC
> >
> > On Sun, Dec 23, 2012 at 4:51 AM, Vijai Ganapathy <
> vijai.ganapathy at gmail.com>
> > wrote:
> >>
> >> Dear all,
> >>
> >> I am new to freeswitch and I have a question to ask.
> >> I am planning to use gsm_open to have small call center setup.
> >>
> >> From the local market I could get Huawei E1732 model Dongle only and I
> am
> >> wondering whether freeswitch / Linux Kernel
> >> would support this model  for voice ?
> >>
> >> Also would ATOM or Phenom would be enough to run 10 seat call center ?
> >>
> >> Want to check with experts before jumping on to actual setup.
> >>
> >> --
> >> Vijai Ganapathy
> >> Mailto: Vijai.Ganapathy at gmail.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
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Michael S Collins
> > Twitter: @mercutioviz
> > http://www.FreeSWITCH.org
> > http://www.ClueCon.com
> > http://www.OSTAG.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
> >
>
>
>
> --
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Vijai Ganapathy
Mailto: Vijai.Ganapathy at gmail.com
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From avi at avimarcus.net  Wed Dec 26 22:46:28 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 26 Dec 2012 21:46:28 +0200
Subject: [Freeswitch-users] XML DialPlan Gateay dialing redundancy
In-Reply-To: <CAKzWOxV-qZ7_xbLNQY23XGX=+PQZn_m5toy9-RaaKJGd1h-=Tw@mail.gmail.com>
References: <0da201cde1fd$2a0b6de0$7e2249a0$@bizfocused.com>
	<CAKzWOxV-qZ7_xbLNQY23XGX=+PQZn_m5toy9-RaaKJGd1h-=Tw@mail.gmail.com>
Message-ID: <CALNZuw=AWmp7oC7Tcn-UpLuVKWZi6byFfXJqXqEahR0pcS_mJQ@mail.gmail.com>

This seems like such a common case that it's sad we mostly have
individually built kludges for our bridges.
Personally, my routing continue on any error, which sometimes results in 3
different errors from the carriers.
Sometimes, it's 3 user_busy in a row -- and sometimes 2 busy and then the
call goes through, so I'm not sure I can even trust user_busy....

Perhaps we can figure out and wikify a "standard" for using attempting via
multiple carriers for connecting to PSTN...

-Avi

On Wed, Dec 26, 2012 at 8:15 PM, Michael Collins <msc at freeswitch.org> wrote:

> Yeah, that's a fun one. You'll probably need to concoct a configuration
> making use of continue_on_fail and failure_causes chan vars. You'll also
> need to figure out all the ways in which each gateway's carrier might fail
> the call. Oh, and "no answer" is also a "failure" so you'll need to find a
> way to handle that differently than the other "failure" conditions. Perhaps
> after each bridge attempt you could all a lua script that checks the value
> of bridge_hangup_cause and then decide what to do with the call, i.e. on a
> no answer you could transfer the call to a voicemail extension but on any
> other failure cause you could simply exit the script and let the dialplan
> continue processing.
>
> Just as a starter maybe something like this:
>
> <action application="set" data="continue_on_fail=true"/>
> <action application="set" data="hangup_after_brdige=false"/>
> <action application="bridge" data="sofia/gateway/gw1/$1"/>
> <action application="lua" data="check_cause.lua"/>
> <action application="bridge" data="sofia/gateway/gw2/$1"/>
> <action application="lua" data="check_cause.lua"/>
> <action application="bridge" data="sofia/gateway/gw3/$1"/>
> <action application="lua" data="check_cause.lua"/>
>
> You'll need to make sure the lua script also checks to see if the previous
> bridge was answered and if so then hangup the call, otherwise it will keep
> trying the next bridge command.
>
> Have fun with that one! :)
>
> -Michael
>
> On Mon, Dec 24, 2012 at 9:36 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>
>> ** **
>>
>> I need a little help with a complex XML dialplan.  Also, I am on the
>> Stable release so I do not believe the much needed nested conditions exists
>> for me yet.****
>>
>> ** **
>>
>> My problem is faulty gateway providers who periodically fail to connect
>> the call with [temp failure] because ?our routing tables where not up to
>> date.?****
>>
>> ** **
>>
>> So what I am after is an xml dialplan to do this****
>>
>> If (it is a 10 or 11 digit dial)****
>>
>>                 Dial using gateway 1****
>>
>>                 If no answer****
>>
>>                                 Go to voice mail****
>>
>>                 Else if gateway failure ****
>>
>>                                 Dial using gateway 2****
>>
>>                 If no answer****
>>
>>                                 Go to voice mail****
>>
>>                 Else if gateway failure ****
>>
>>                                 Dial using gateway 3****
>>
>>                 If no answer****
>>
>>                                 Go to voice mail****
>>
>>                                 Else if gateway failure****
>>
>>                                                 Play a message that the
>> number is screwed!****
>>
>> ** **
>>
>> Perhaps I need to breakdown and use a scripting language like LUA or
>> JavaScript for this, but I would rather keep everything in XML if possible.
>> ****
>>
>> ** **
>>
>> Any thoughts?****
>>
>> ** **
>>
>> Happy Holidays to all,****
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
> http://www.ClueCon.com <http://www.cluecon.com/>
> http://www.OSTAG.org <http://www.ostag.org/>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 lloyd.aloysius at gmail.com  Wed Dec 26 22:53:19 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Wed, 26 Dec 2012 14:53:19 -0500
Subject: [Freeswitch-users] Check user status before bridge
Message-ID: <CAGvuWYZ1PHjbUxK9bLXkNw0uU2L1OC4tWmSZBQiXWicP36DsYw@mail.gmail.com>

Hello,

Is there a way using lua to check the user status before bridge a call. I
want to check if the user is already on another the call or session?

Thanks
Lloyd
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From msc at freeswitch.org  Wed Dec 26 23:03:52 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Dec 2012 12:03:52 -0800
Subject: [Freeswitch-users] Check user status before bridge
In-Reply-To: <CAGvuWYZ1PHjbUxK9bLXkNw0uU2L1OC4tWmSZBQiXWicP36DsYw@mail.gmail.com>
References: <CAGvuWYZ1PHjbUxK9bLXkNw0uU2L1OC4tWmSZBQiXWicP36DsYw@mail.gmail.com>
Message-ID: <CAKzWOxUVSf=hHTSY+AOHM8edCr5O+VgLR7Uk6Tu2Za1HO7U7Kw@mail.gmail.com>

Yeah, that's not the easiest nut to crack. However, if you're willing to
make some assumptions then you could do a show channels and then grab all
the "presence_id" values. If you see <user>@<domain> then you could assume
that the user is on a call.

-MC

On Wed, Dec 26, 2012 at 11:53 AM, Lloyd Aloysius
<lloyd.aloysius at gmail.com>wrote:

> Hello,
>
> Is there a way using lua to check the user status before bridge a call. I
> want to check if the user is already on another the call or session?
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From vipkilla at gmail.com  Wed Dec 26 23:04:23 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Wed, 26 Dec 2012 15:04:23 -0500
Subject: [Freeswitch-users] page application
In-Reply-To: <50DB4D81.9010000@gmail.com>
References: <50DB4D81.9010000@gmail.com>
Message-ID: <CAC-LwPNh+iRp55Ev0F_jSrGHE_BSA+w-B227euLT4B1wBr3X8A@mail.gmail.com>

I'm not sure what you are trying to do, but if you are trying to page
a phone i dont believe that's what you want...
The "page" API is defined as "Send a file as a page" (not sure what that means)

to page a phone or group of phones use this:
http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom

On Wed, Dec 26, 2012 at 2:18 PM, Abaci <abaci64 at gmail.com> wrote:
> There is an application "page" (in mod_dptools), I didn't find any
> documentation for it on the wiki so I'm sending here if someone can
> provide details how to use it.
> see
> http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/applications/mod_dptools/mod_dptools.c?r1=bc0912cd11efb3f4525b8150733035ef37fc2ee3&r2=0a0f5951ab616d5b136e79b7d4ad253510f9d0cc
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Wed Dec 26 23:07:46 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Dec 2012 12:07:46 -0800
Subject: [Freeswitch-users] XML DialPlan Gateay dialing redundancy
In-Reply-To: <CALNZuw=AWmp7oC7Tcn-UpLuVKWZi6byFfXJqXqEahR0pcS_mJQ@mail.gmail.com>
References: <0da201cde1fd$2a0b6de0$7e2249a0$@bizfocused.com>
	<CAKzWOxV-qZ7_xbLNQY23XGX=+PQZn_m5toy9-RaaKJGd1h-=Tw@mail.gmail.com>
	<CALNZuw=AWmp7oC7Tcn-UpLuVKWZi6byFfXJqXqEahR0pcS_mJQ@mail.gmail.com>
Message-ID: <CAKzWOxV54LXUsSf9vdKQzyZMqCmh82cQbxr+GbzNVfPaRbWwbg@mail.gmail.com>

The only issue with "standard" is that everyone has his own definition of
what "standard" behavior should be. :) However, I'm not at all opposed to
having a cookbook-style set of recipes for how people have solved this
issue in various scenarios.

-MC

On Wed, Dec 26, 2012 at 11:46 AM, Avi Marcus <avi at avimarcus.net> wrote:

> This seems like such a common case that it's sad we mostly have
> individually built kludges for our bridges.
> Personally, my routing continue on any error, which sometimes results in 3
> different errors from the carriers.
> Sometimes, it's 3 user_busy in a row -- and sometimes 2 busy and then the
> call goes through, so I'm not sure I can even trust user_busy....
>
> Perhaps we can figure out and wikify a "standard" for using attempting via
> multiple carriers for connecting to PSTN...
>
> -Avi
>
> On Wed, Dec 26, 2012 at 8:15 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Yeah, that's a fun one. You'll probably need to concoct a configuration
>> making use of continue_on_fail and failure_causes chan vars. You'll also
>> need to figure out all the ways in which each gateway's carrier might fail
>> the call. Oh, and "no answer" is also a "failure" so you'll need to find a
>> way to handle that differently than the other "failure" conditions. Perhaps
>> after each bridge attempt you could all a lua script that checks the value
>> of bridge_hangup_cause and then decide what to do with the call, i.e. on a
>> no answer you could transfer the call to a voicemail extension but on any
>> other failure cause you could simply exit the script and let the dialplan
>> continue processing.
>>
>> Just as a starter maybe something like this:
>>
>> <action application="set" data="continue_on_fail=true"/>
>> <action application="set" data="hangup_after_brdige=false"/>
>> <action application="bridge" data="sofia/gateway/gw1/$1"/>
>> <action application="lua" data="check_cause.lua"/>
>> <action application="bridge" data="sofia/gateway/gw2/$1"/>
>> <action application="lua" data="check_cause.lua"/>
>> <action application="bridge" data="sofia/gateway/gw3/$1"/>
>> <action application="lua" data="check_cause.lua"/>
>>
>> You'll need to make sure the lua script also checks to see if the
>> previous bridge was answered and if so then hangup the call, otherwise it
>> will keep trying the next bridge command.
>>
>> Have fun with that one! :)
>>
>> -Michael
>>
>> On Mon, Dec 24, 2012 at 9:36 AM, Sean Devoy <sdevoy at bizfocused.com>wrote:
>>
>>> ** **
>>>
>>> I need a little help with a complex XML dialplan.  Also, I am on the
>>> Stable release so I do not believe the much needed nested conditions exists
>>> for me yet.****
>>>
>>> ** **
>>>
>>> My problem is faulty gateway providers who periodically fail to connect
>>> the call with [temp failure] because ?our routing tables where not up to
>>> date.?****
>>>
>>> ** **
>>>
>>> So what I am after is an xml dialplan to do this****
>>>
>>> If (it is a 10 or 11 digit dial)****
>>>
>>>                 Dial using gateway 1****
>>>
>>>                 If no answer****
>>>
>>>                                 Go to voice mail****
>>>
>>>                 Else if gateway failure ****
>>>
>>>                                 Dial using gateway 2****
>>>
>>>                 If no answer****
>>>
>>>                                 Go to voice mail****
>>>
>>>                 Else if gateway failure ****
>>>
>>>                                 Dial using gateway 3****
>>>
>>>                 If no answer****
>>>
>>>                                 Go to voice mail****
>>>
>>>                                 Else if gateway failure****
>>>
>>>                                                 Play a message that the
>>> number is screwed!****
>>>
>>> ** **
>>>
>>> Perhaps I need to breakdown and use a scripting language like LUA or
>>> JavaScript for this, but I would rather keep everything in XML if possible.
>>> ****
>>>
>>> ** **
>>>
>>> Any thoughts?****
>>>
>>> ** **
>>>
>>> Happy Holidays to all,****
>>>
>>> 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
>> http://www.ClueCon.com <http://www.cluecon.com/>
>> http://www.OSTAG.org <http://www.ostag.org/>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From gmaruzz at gmail.com  Wed Dec 26 23:08:23 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 26 Dec 2012 21:08:23 +0100
Subject: [Freeswitch-users] New to Freeswitch
In-Reply-To: <CACahg4Fb9=gVU_Mp4WTdsUcXixtsnKNRmfD2y1s_Nc3x_5MDMg@mail.gmail.com>
References: <CACahg4EKsXiRFkoYp-NbeRvtNsHLthxtu+k65pezCKw_wvhKbw@mail.gmail.com>
	<CAKzWOxUpdX9-VXzYYMpbtEvuB1hw11v0jQMEp3p-Boh7T1NYTA@mail.gmail.com>
	<CALXCt0pwZMYi0Ny9JdJvixwROQ+o9L6GTuoqKqBPpVZ8uHEO6Q@mail.gmail.com>
	<CACahg4Fb9=gVU_Mp4WTdsUcXixtsnKNRmfD2y1s_Nc3x_5MDMg@mail.gmail.com>
Message-ID: <CALXCt0q1bihDnh4KbRUapKht94=_jYu=BMcjJ2X1Tc+NCFPOmA@mail.gmail.com>

no



On 12/26/12, Vijai Ganapathy <vijai.ganapathy at gmail.com> wrote:
> I will check and update here.
>
> BTW. is there a quick way to test without all installing / configuring ?
> like boot distro which load freeswitch ?
>
>
> On Wed, Dec 26, 2012 at 10:57 PM, Giovanni Maruzzelli
> <gmaruzz at gmail.com>wrote:
>
>> Vijai, if you are not sure if it's compatible, your best option is to
>> test it. Huawei makes a lot of different dongles, and most of them are
>> compatible. But I completely don't know about 1732, so try it at your
>> own risk. Also, you may try alibaba.com as source for dongles.
>>
>> -giovanni
>>
>> On Wed, Dec 26, 2012 at 6:16 PM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > This is a good question for those who've been using mod_gsmopen.
>> According
>> > to the wiki the dongle needs to be compatible with Huawei E1550. Do you
>> know
>> > if the 1732 is compatible?
>> >
>> > -MC
>> >
>> > On Sun, Dec 23, 2012 at 4:51 AM, Vijai Ganapathy <
>> vijai.ganapathy at gmail.com>
>> > wrote:
>> >>
>> >> Dear all,
>> >>
>> >> I am new to freeswitch and I have a question to ask.
>> >> I am planning to use gsm_open to have small call center setup.
>> >>
>> >> From the local market I could get Huawei E1732 model Dongle only and I
>> am
>> >> wondering whether freeswitch / Linux Kernel
>> >> would support this model  for voice ?
>> >>
>> >> Also would ATOM or Phenom would be enough to run 10 seat call center ?
>> >>
>> >> Want to check with experts before jumping on to actual setup.
>> >>
>> >> --
>> >> Vijai Ganapathy
>> >> Mailto: Vijai.Ganapathy at gmail.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
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Michael S Collins
>> > Twitter: @mercutioviz
>> > http://www.FreeSWITCH.org
>> > http://www.ClueCon.com
>> > http://www.OSTAG.org
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting 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
>> >
>>
>>
>>
>> --
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Vijai Ganapathy
> Mailto: Vijai.Ganapathy at gmail.com
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From abaci64 at gmail.com  Wed Dec 26 23:37:40 2012
From: abaci64 at gmail.com (Abaci)
Date: Wed, 26 Dec 2012 15:37:40 -0500
Subject: [Freeswitch-users] page application
In-Reply-To: <CAC-LwPNh+iRp55Ev0F_jSrGHE_BSA+w-B227euLT4B1wBr3X8A@mail.gmail.com>
References: <50DB4D81.9010000@gmail.com>
	<CAC-LwPNh+iRp55Ev0F_jSrGHE_BSA+w-B227euLT4B1wBr3X8A@mail.gmail.com>
Message-ID: <50DB6014.6060800@gmail.com>

I was looking at it thinking it would something like page would do in 
asterisk, but notice the part where it says "Send a file as a page" 
(which is only in the api definition not the app definition), I would 
still like to know what it is and have it documented on the wiki.
Thanks

On 12/26/2012 3:04 PM, Vik Killa wrote:
> I'm not sure what you are trying to do, but if you are trying to page
> a phone i dont believe that's what you want...
> The "page" API is defined as "Send a file as a page" (not sure what that means)
>
> to page a phone or group of phones use this:
> http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom
>
> On Wed, Dec 26, 2012 at 2:18 PM, Abaci <abaci64 at gmail.com> wrote:
>> There is an application "page" (in mod_dptools), I didn't find any
>> documentation for it on the wiki so I'm sending here if someone can
>> provide details how to use it.
>> see
>> http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/applications/mod_dptools/mod_dptools.c?r1=bc0912cd11efb3f4525b8150733035ef37fc2ee3&r2=0a0f5951ab616d5b136e79b7d4ad253510f9d0cc
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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



From cal.leeming at simplicitymedialtd.co.uk  Wed Dec 26 23:57:00 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Wed, 26 Dec 2012 20:57:00 +0000
Subject: [Freeswitch-users] xml_curl and ivr menus
In-Reply-To: <804D48104511D4468F0D60DF9D3100350AF8DCA3@MAIL.millicorp.com>
References: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>
	<50D8BE29.7010404@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8A9E3@MAIL.millicorp.com>
	<50D8CC6A.5000706@gmail.com> <50D8CCC3.7090305@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8B121@MAIL.millicorp.com>
	<804D48104511D4468F0D60DF9D3100350AF8DCA3@MAIL.millicorp.com>
Message-ID: <CAHKQagE7MHcTO+AxXtMJPK+KS_P1L3vB+EETXhM1tVy3EvxgbA@mail.gmail.com>

Tim,

Chiming in on this, as my focus has been purely on mod_xml_curl for the
last 2 months.

My apologies that you struggled with finding a working examples, the
mod_xml_curl documentation is still a work in progress - and despite that
it has recently had a partial re-write, there are still many parts which
aren't clear enough.

I will be updating the docs soon (in the next week) to show example
request/response of as many different scenarios as possible, along with
context specific descriptions of each variable.

Cal

On Tue, Dec 25, 2012 at 2:07 PM, Tim Meade <Tim.Meade at millicorp.com> wrote:

>  Here is an example of the xml_curl phrases section.  This is not in the
> WIKI?..  I?ll try to update.  Notice the language section?.****
>
> ** **
>
> <document type="freeswitch/xml">****
>
>                   <section name="phrases">****
>
>                       <macros>****
>
>                            <include>****
>
>                                 <language name="en" sound-path="
> $${sounds_dir}/en/us/callie" tts-engine="cepstral" tts-voice="callie">****
>
>                                      <macro name="demo_ivr_main_menu"
> pause="100"> <!-- See conf/autoload_config/ivr.conf.xml for an example on
> how to use this macro in an IVR -->****
>
>                                                <input pattern="(.*)">****
>
>                                                  <match>****
>
>                                                    <!-- string together
> several existing sound files to create one long greeting -->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-welcome_to_freeswitch.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-this_ivr_will_let_you_test_features.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-you_may_exit_by_hanging_up.wav"/>****
>
>                                                    <!-- note that you can
> do more than just play files, e.g. have pauses and do TTS -->****
>
>                                            ****
>
>                                                    <!-- Menu option 1:
> Call FreeSWITCH conference-->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-enter_ext_pound.wav"/>****
>
>                                                    <action function="
> play-file" data="silence_stream://1500"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-to_call_the_freeswitch_conference.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-please.wav"/>****
>
>                                                    <action function="
> play-file" data="voicemail/vm-press.wav"/>****
>
>                                                    <action function="
> play-file" data="digits/1.wav"/>****
>
>                                            ****
>
>                                                    <!-- Menu option 2: Do
> FreeSWITCH echo test -->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-to_do_a_freeswitch_echo_test.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-please.wav"/>****
>
>                                                    <action function="
> play-file" data="voicemail/vm-press.wav"/>****
>
>                                                    <action function="
> play-file" data="digits/2.wav"/>****
>
>                                            ****
>
>                                                    <!-- Menu option 3:
> Listen to Music on Hold -->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-to_listen_to_moh.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-please.wav"/>****
>
>                                                    <action function="
> play-file" data="voicemail/vm-press.wav"/>****
>
>                                                    <action function="
> play-file" data="digits/3.wav"/>****
>
>                                            ****
>
>                                                    <!-- Menu option 4:
> Register for ClueCon -->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-register_for_cluecon.wav"/>****
>
>                                                    <action function="
> play-file" data="digits/4.wav"/>****
>
>                                            ****
>
>                                                    <!-- Menu option 5:
> Listen to screaming monkeys -->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-to_hear_screaming_monkeys.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-please.wav"/>****
>
>                                                    <action function="
> play-file" data="voicemail/vm-press.wav"/>****
>
>                                                    <action function="
> play-file" data="digits/5.wav"/>****
>
>                                            ****
>
>                                                    <!-- Menu option 6:
> Hear a sample submenu -->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-to_hear_sample_submenu.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-please.wav"/>****
>
>                                                    <action function="
> play-file" data="voicemail/vm-press.wav"/>****
>
>                                                    <action function="
> play-file" data="digits/6.wav"/>****
>
>                                            ****
>
>                                                    <!-- Menu option 9:
> Repeat these options -->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-to_repeat_these_options.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-please.wav"/>****
>
>                                                    <action function="
> play-file" data="voicemail/vm-press.wav"/>****
>
>                                                    <action function="
> play-file" data="digits/9.wav"/>****
>
>                                                    <action function="
> play-file" data="silence_stream://2000"/>****
>
>                                                  </match>****
>
>                                                </input>****
>
>                                              </macro>      ****
>
>                                        </language>
>       ****
>
>                                 </include>****
>
>                        </macros>
>                 ****
>
>                   </section>****
>
>                 </document>****
>
> ** **
>
> ** **
>
> Thanks All****
>
> ** **
>
> Tim****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tim Meade
> *Sent:* Monday, December 24, 2012 5:05 PM
> *To:* Abaci; FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>
> ** **
>
> Thank you Abaci?.   Just found that myself under the xml_curl section of
> the wiki?..  I hadn?t noticed the xml_curl configuration parameters for the
> return xml are different.    ****
>
> ** **
>
> Here is the completed working configuration from xml_curl debug_on.   ****
>
> ** **
>
> <document type="freeswitch/xml">****
>
> <section name="configuration">****
>
> <configuration name="ivr.conf" description="IVR menus">****
>
> <menus>****
>
> <include>****
>
> <!-- demo IVR setup -->****
>
> <!-- demo IVR, Main Menu -->****
>
> <menu name="tivr"****
>
> greet-long="phrase:demo_ivr_main_menu"****
>
> greet-short="phrase:demo_ivr_main_menu_short"****
>
> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"****
>
> exit-sound="voicemail/vm-goodbye.wav"****
>
> confirm-macro=""****
>
> confirm-key=""****
>
> tts-engine="flite"****
>
> tts-voice="rms"****
>
> confirm-attempts="3"****
>
> timeout="10000"****
>
> inter-digit-timeout="2000"****
>
> max-failures="3"****
>
> max-timeouts="3"****
>
> digit-len="4">****
>
> <!-- The following are the definitions for the digits the user dials -->**
> **
>
> <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->****
>
> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/
> 888 at conference.freeswitch.org"/>****
>
> <entry action="menu-exec-app" digits="2" param="transfer 9196 XML
> default"/> <!-- FS echo -->****
>
> <entry action="menu-exec-app" digits="3" param="transfer 9664 XML
> default"/> <!-- MOH -->****
>
> <entry action="menu-exec-app" digits="4" param="transfer 9191 XML
> default"/> <!-- ClueCon -->****
>
> <entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/>
> <!-- Screaming monkeys -->****
>
> <!-- Using a regex in the digits tag lets you define a dial pattern for
> the caller****
>
> You may define multiple regexes if you need a different pattern for some
> reason -->****
>
> <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer
> $1 XML features"/>****
>
> <entry action="menu-top" digits="9"/> <!-- Repeat this menu -->****
>
> </menu>****
>
> </include>****
>
> </menus>****
>
> </configuration>****
>
> </section>****
>
> </document>****
>
> ** **
>
> *From:* Abaci [mailto:abaci64 at gmail.com]
> *Sent:* Monday, December 24, 2012 4:45 PM
> *To:* FreeSWITCH Users Help
> *Cc:* Tim Meade
> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>
> ** **
>
> you're also missing the "section" element.
>
> On 12/24/2012 4:43 PM, Abaci wrote:****
>
>  Here is a sample from my server, I highlighted the missing stuff.
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="configuration">
> *    <configuration name="ivr.conf" description="IVR menus">
>       <menus>*
>         <menu name="ivrmenu1"
>               invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>               exit-sound="voicemail/vm-goodbye.wav"
>               timeout ="3000"
>               inter-digit-timeout="2000"
>               max-failures="2"
>               digit-len="3"
>               phrase-lang="en">
>           <entry action="menu-exec-app" digits="0" param="transfer 11 XML
> voicemail " />
>           <entry action="menu-exec-app" digits="1" param="transfer
> ivrmenu2 XML ivr " />
>           <entry action="menu-exec-app" digits="2" param="transfer
> ivrmenu3 XML ivr " />
>         </menu>
> *      </menus>
>     </configuration>*
>   </section>
> </document>
>
> On 12/24/2012 4:22 PM, Tim Meade wrote:****
>
> The items I pasted below are from the cat of the xml_curl debug_on
> command.  It certainly appears to me that freeswitch is getting the correct
> name in this case ?tivr? which is what shows in both the dialplan and the
> curl?d ivr menu.****
>
>  ****
>
> My concerns now are dealing with the xml format for the ivr as I have
> found several different versions on the wiki and in the prior postings.***
> *
>
>  ****
>
> *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 *Abaci
> *Sent:* Monday, December 24, 2012 3:42 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>
>  ****
>
> on freeswitch enable xml_curl debug and see if freeswitch is actually
> getting the correct ivr
> 'xml_curl debug_on'
>
> On 12/24/2012 2:20 PM, Tim Meade wrote:****
>
> I am continually getting ?unable to find menu? while trying to retrieve an
> ivr_menu using xml_curl****
>
>  ****
>
> Using xml_curl debug_on ****
>
>  ****
>
> My DialPlan****
>
>  ****
>
> <document type="freeswitch/xml">****
>
> <section name="dialplan" description="example_curl_dialplan">****
>
> <context name="public">****
>
> <extension name="ws_origination">****
>
> <condition field="destination_number" expression="^\d+$" break="on-true">*
> ***
>
> <action inline="true" application="set" data="continue_on_fail=true"/>****
>
> <action inline="true" application="set" data="hangup_after_bridge=true"/>
> ****
>
> <action application="answer"/>****
>
> <action application="ivr" data="tivr"/>****
>
> </condition>****
>
> </extension>****
>
> </context>****
>
> </section>****
>
>  ****
>
> And the ivr configuration returned by my xml_curl****
>
>  ****
>
> <include>****
>
> <!-- demo IVR setup -->****
>
> <!-- demo IVR, Main Menu -->****
>
> <menu name="tivr"****
>
> greet-long="phrase:demo_ivr_main_menu"****
>
> greet-short="phrase:demo_ivr_main_menu_short"****
>
> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"****
>
> exit-sound="voicemail/vm-goodbye.wav"****
>
> confirm-macro=""****
>
> confirm-key=""****
>
> tts-engine="flite"****
>
> tts-voice="rms"****
>
> confirm-attempts="3"****
>
> timeout="10000"****
>
> inter-digit-timeout="2000"****
>
> max-failures="3"****
>
> max-timeouts="3"****
>
> digit-len="4">****
>
> <!-- The following are the definitions for the digits the user dials -->**
> **
>
> <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->****
>
> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${
> domain}/888 at conference.freeswitch.org"/>****
>
> <entry action="menu-exec-app" digits="2" param="transfer 9196 XML
> default"/> <!-- FS echo -->****
>
> <entry action="menu-exec-app" digits="3" param="transfer 9664 XML
> default"/> <!-- MOH -->****
>
> <entry action="menu-exec-app" digits="4" param="transfer 9191 XML
> default"/> <!-- ClueCon -->****
>
> <entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/>
> <!-- Screaming monkeys -->****
>
> <!-- Using a regex in the digits tag lets you define a dial pattern for
> the caller****
>
> You may define multiple regexes if you need a different pattern for some
> reason -->****
>
> <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer
> $1 XML features"/>****
>
> <entry action="menu-top" digits="9"/> <!-- Repeat this menu -->****
>
> </menu>****
>
> </include>****
>
>  ****
>
> Any ideas on the ?unable to find menu? ????****
>
>  ****
>
> Thanks****
>
>  ****
>
> Tim****
>
>  ****
>
>  ****
>
>  ****
>
>
>
>
> ****
>
> _________________________________________________________________________****
>
> Professional FreeSWITCH Consulting 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****
>
> ** **
>
> FreeSWITCH-users mailing list****
>
> FreeSWITCH-users at lists.freeswitch.org****
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users****
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users****
>
> http://www.freeswitch.org****
>
>  ** **
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 msc at freeswitch.org  Thu Dec 27 02:57:39 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Dec 2012 15:57:39 -0800
Subject: [Freeswitch-users] page application
In-Reply-To: <50DB6014.6060800@gmail.com>
References: <50DB4D81.9010000@gmail.com>
	<CAC-LwPNh+iRp55Ev0F_jSrGHE_BSA+w-B227euLT4B1wBr3X8A@mail.gmail.com>
	<50DB6014.6060800@gmail.com>
Message-ID: <CAKzWOxWTA46Ft3bs+nAmVT7mub7naQxefqXPFKMBPvAcCh0yNg@mail.gmail.com>

It's a new item that hasn't been documented just yet. Both the app and api
are in the same commit as you saw. The app doesn't show up in the help list
atm but that's just a minor issue. We'll have to talk to Anthony about what
these things do unless you are able to decipher it just by look at the
code. ;)

Thanks,
MC

On Wed, Dec 26, 2012 at 12:37 PM, Abaci <abaci64 at gmail.com> wrote:

> I was looking at it thinking it would something like page would do in
> asterisk, but notice the part where it says "Send a file as a page"
> (which is only in the api definition not the app definition), I would
> still like to know what it is and have it documented on the wiki.
> Thanks
>
> On 12/26/2012 3:04 PM, Vik Killa wrote:
> > I'm not sure what you are trying to do, but if you are trying to page
> > a phone i dont believe that's what you want...
> > The "page" API is defined as "Send a file as a page" (not sure what that
> means)
> >
> > to page a phone or group of phones use this:
> > http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom
> >
> > On Wed, Dec 26, 2012 at 2:18 PM, Abaci <abaci64 at gmail.com> wrote:
> >> There is an application "page" (in mod_dptools), I didn't find any
> >> documentation for it on the wiki so I'm sending here if someone can
> >> provide details how to use it.
> >> see
> >>
> http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/applications/mod_dptools/mod_dptools.c?r1=bc0912cd11efb3f4525b8150733035ef37fc2ee3&r2=0a0f5951ab616d5b136e79b7d4ad253510f9d0cc
> >> 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
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From leshadorofeev at gmail.com  Thu Dec 27 11:12:37 2012
From: leshadorofeev at gmail.com (=?KOI8-R?B?4czFy9PFyiDkz9LPxsXF1w==?=)
Date: Thu, 27 Dec 2012 12:12:37 +0400
Subject: [Freeswitch-users] FS sends BYE after 32 seconds
Message-ID: <CAHAu2g4avq55KWacXOUypxhFnR1dQ_O8vBjEKVt7YxZZdbNuCg@mail.gmail.com>

Hello, all!
I need some help.
I have one problem with incoming calls - FS hung up after 32 seconds.
I think, FS does not accepted ACK because the field RequestURI is different
from the same field in INVITE...

Call trace:
------------------------------------------------------------------------
recv 1054 bytes from udp/[10.10.132.203]:5060 at 20:56:09.448547:
------------------------------------------------------------------------
INVITE sip:gw+voip.mtt.ru at 11.111.88.147:5080;transport=udp;gw=voip.mtt.ruSIP/2.0
Via: SIP/2.0/UDP 10.10.132.203:5060
;branch=z9hG4bK-d8754z-4438ff39749bbf13-1---d8754z-;rport
Via: SIP/2.0/UDP 10.10.132.203:5061
;branch=z9hG4bK-giy2ccrl24zvsgqk;rport=5061
Max-Forwards: 69
Record-Route: <sip:10.10.132.203:5060;lr;transport=UDP>
Contact: "Anonymous"<sip:10.10.132.203:5061>
To: <sip:7812BBBBBBB at 10.10.132.203>
From: "7812AAAAAAA"<sip:+7812AAAAAAA at 10.10.132.203>;tag=hmxjwhzuanhi5qq7.o
Call-ID: 1356-548022-325095~o
CSeq: 662 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
h323-conf-id: 1868605416-458121599-1280693483-4049122446
cisco-GUID: 1868605416-458121599-1280693483-4049122446
Content-Length: 283

------------------------------------------------------------------------
send 513 bytes to udp/[10.10.132.203]:5060 at 20:56:09.448869:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.132.203:5060
;branch=z9hG4bK-d8754z-4438ff39749bbf13-1---d8754z-;rport=5060
Via: SIP/2.0/UDP 10.10.132.203:5061
;branch=z9hG4bK-giy2ccrl24zvsgqk;rport=5061
Record-Route: <sip:10.10.132.203:5060;lr;transport=UDP>
From: "7812AAAAAAA"<sip:+7812AAAAAAA at 10.10.132.203>;tag=hmxjwhzuanhi5qq7.o
To: <sip:7812BBBBBBB at 10.10.132.203>
Call-ID: 1356-548022-325095~o
CSeq: 662 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.3.10b+git~20121209T055926Z~bfc3c17bcb
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.132.203:5060
;branch=z9hG4bK-d8754z-4438ff39749bbf13-1---d8754z-;rport=5060
Via: SIP/2.0/UDP 10.10.132.203:5061
;branch=z9hG4bK-giy2ccrl24zvsgqk;rport=5061
Record-Route: <sip:10.10.132.203:5060;lr;transport=UDP>
From: "7812AAAAAAA"<sip:+7812AAAAAAA at 10.10.132.203>;tag=hmxjwhzuanhi5qq7.o
To: <sip:7812BBBBBBB at 10.10.132.203>;tag=U98Z2BN38jemS
Call-ID: 1356-548022-325095~o
CSeq: 662 INVITE
Contact: <sip:7812BBBBBBB at 11.111.88.147:0;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.3.10b+git~20121209T055926Z~bfc3c17bcb
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
Remote-Party-ID: "7812BBBBBBB" <sip:7812BBBBBBB at 10.10.132.203
>;party=calling;privacy=off;screen=no

------------------------------------------------------------------------
recv 466 bytes from udp/[10.10.132.203]:5060 at 20:56:09.594850:
------------------------------------------------------------------------
ACK sip:7812BBBBBBB at 11.111.88.147;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.132.203:5060
;branch=z9hG4bK-d8754z-a05ecb4295a54946-1---d8754z-;rport
Via: SIP/2.0/UDP 10.10.132.203:5061
;rport=5061;branch=z9hG4bK-z4gyohqxmtxf5hg6
Max-Forwards: 69
To: <sip:7812BBBBBBB at 10.10.132.203>;tag=U98Z2BN38jemS
From: "7812AAAAAAA"<sip:+7812AAAAAAA at 10.10.132.203>;tag=hmxjwhzuanhi5qq7.o
Call-ID: 1356-548022-325095~o
CSeq: 662 ACK
User-Agent: Sippy
Content-Length: 0

nua_server.c:155 nua_stack_process_request() nua(0x1c7b6e0): strange ACK
from <sip:+7812AAAAAAA at 10.10.132.203>

<<<
*this exchange 200 OK - ACK repeats several times...*
*>>>
*

------------------------------------------------------------------------
send 691 bytes to udp/[10.10.132.203]:5060 at 20:56:41.566627:
------------------------------------------------------------------------
BYE sip:10.10.132.203:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5080;rport;branch=z9hG4bK0N7S27DDaFByF
Route: <sip:10.10.132.203:5060;lr;transport=UDP>
Max-Forwards: 70
From: <sip:7812BBBBBBB at 10.10.132.203>;tag=U98Z2BN38jemS
To: "7812AAAAAAA" <sip:+7812AAAAAAA at 10.10.132.203>;tag=hmxjwhzuanhi5qq7.o
Call-ID: 1356-548022-325095~o
CSeq: 37941636 BYE
Contact: <sip:7812BBBBBBB at 11.11.88.147:0;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.3.10b+git~20121209T055926Z~bfc3c17bcb
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Reason: SIP;cause=408;text="ACK Timeout"
Content-Length: 0

------------------------------------------------------------------------
recv 333 bytes from udp/[10.10.132.203]:5060 at 20:56:41.586770:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:5080
;received=11.11.88.147;rport=5080;branch=z9hG4bK0N7S27DDaFByF
To: "7812AAAAAAA"<sip:+7812AAAAAAA at 10.10.132.203>;tag=hmxjwhzuanhi5qq7.o
From: <sip:7812BBBBBBB at 10.10.132.203>;tag=U98Z2BN38jemS
Call-ID: 1356-548022-325095~o
CSeq: 37941636 BYE
Server: Sippy
Content-Length: 0

Thank you for you attention!
Alex.
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From a.venugopan at mundio.com  Thu Dec 27 15:40:45 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Thu, 27 Dec 2012 12:40:45 +0000
Subject: [Freeswitch-users] repeated registration messages in logs
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233D501@Mail-Kilo.squay.com>

Hi,

After registering the phones successfully I am getting the below message continuously every few seconds in my logs and this is filling up the logs and freeswitch goes down sometimes.
Can someone please tell me why I keep getting this message every few seconds? I have another server and there I don't see this issue.

2012-12-27 12:37:44.714558 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [Abarajit at vectonecloud.com] from ip 192.168.2.205
2012-12-27 12:41:58.744557 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [sana at vectonecloud.com] from ip 192.168.2.48

Regards,
Archana

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From ali.jawad at splendor.net  Thu Dec 27 15:51:50 2012
From: ali.jawad at splendor.net (Ali Jawad)
Date: Thu, 27 Dec 2012 14:51:50 +0200
Subject: [Freeswitch-users] Jingle updates since 2010 ?
In-Reply-To: <CAG2QdgaLXFA_0QP+oboYUCwi7xWEc+CN5=u_G5O3mqYE3r=NYQ@mail.gmail.com>
References: <CAG2QdgaLXFA_0QP+oboYUCwi7xWEc+CN5=u_G5O3mqYE3r=NYQ@mail.gmail.com>
Message-ID: <CAG2QdgaQ-kxyQE3BGKiOgexmq4xjSdbOkg6nMkX9=H4Zu8UhMQ@mail.gmail.com>

Any input please ?

On Fri, Dec 21, 2012 at 2:10 PM, Ali Jawad <ali.jawad at splendor.net> wrote:

> Hi
> Sometime in 2010, one of my employers wanted to use an xmpp server with
> jingle to be able to make pstn calls through freeswitch by sending the
> jingle call to freeswitch and freeswitch does convert to SIP. Anthony did
> help me at the time and the conclusion was
>
> Now I have to state the following. I have tried Freeswitch in client mode
> using the following two approaches
>
>
>    1.  With A Gtalk account and a gtalk client, this worked flawlessly
>    2. With server XMPP servers and jingle clients all register but FS was
>    not able to do Jingle to SIP conversion in this case. *The main reason
>    as per FS developer Anthony is that jingle is a point to point protocol and
>    FS was tested to work with Gtalk and telepathy. Note here that Freeswith
>    does not use libjingle it uses it's own special jingle implementation*
>    3. Another Approach I use was the component mode with my XMPP server,
>    everything worked right expect jingle to sip conversion !! :(.
>
> From
>
> http://www.alijawad.org/cms/index.php?option=com_content&task=view&id=21&Itemid=2
>
> Now, I do have a new employer requesting a similar scenario, and I would
> love to know if anything has changed since and if the scenario in question
> is now possible.
>
> Regards
>
>
> *
>
> *
>



-- 
*Ali Jawad
*
*Information Systems Manager
CISSP - PMP - ITIL V3 - RHCE - VCP - C|EH - CCNA - MCSA
*
*Splendor Telecom (www.splendor.net)
Beirut, Lebanon
Phone: +9611373725/ext 116
FAX: +9611375554

*
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From sanjay.sanjaysoni at gmail.com  Thu Dec 27 13:16:15 2012
From: sanjay.sanjaysoni at gmail.com (sanjay soni)
Date: Thu, 27 Dec 2012 15:46:15 +0530
Subject: [Freeswitch-users] caller-id-number not getting set
Message-ID: <CAJbRaS4aJJw9_XCw08hWY_s_3OtCqGVoezA7Q9ooo0npCy5gKA@mail.gmail.com>

In all the dialplan examples (Both in the book and wiki) most
places destination_number is being tested in the <condition>, BUT I want to
do something based on caller-id-name / caller-id-number channel variables.
But I see that all these variables (Caller-Caller-ID-Name,
sip_callee_id_name, sip_auth_username, caller-id-name, caller-id-number)
are empty.
however, i see these values properly in the event being fired But not in
the xml dialplan (Which is the place i want ot use them) ! What should I do
?

Thanks
Sanjay
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From luca.pradovera at gmail.com  Thu Dec 27 14:55:52 2012
From: luca.pradovera at gmail.com (Luca Pradovera)
Date: Thu, 27 Dec 2012 12:55:52 +0100
Subject: [Freeswitch-users] uuid_record and RECORD_STEREO
Message-ID: <89F15D50-E189-4D86-9320-102F086DF18F@gmail.com>

Hello,
is there a way to invoke uuid_record through bgapi, setting RECORD_STEREO to true as a local parameter instead of using uuid_setvar, over inbound Event Socket?
I would prefer to not set a global state on the channel if possible.
Thanks!

--
Luca Pradovera
luca.pradovera at gmail.com





From bhadrarao.kankatala at gmail.com  Thu Dec 27 15:52:12 2012
From: bhadrarao.kankatala at gmail.com (veerabhadrarao`)
Date: Thu, 27 Dec 2012 18:22:12 +0530
Subject: [Freeswitch-users] regarding attendent transfer
Message-ID: <50DC447C.1080703@gmail.com>

hai

Can anyone plaese tell me how to make attendent transfer. I am able to 
working on Blind transfer using default.xml and features.xml but not 
attendenet transfer successfully. I am using Freeswitch 1.2.4 version.

thank you


From bhadrarao.kankatala at gmail.com  Thu Dec 27 16:00:09 2012
From: bhadrarao.kankatala at gmail.com (veerabhadrarao`)
Date: Thu, 27 Dec 2012 18:30:09 +0530
Subject: [Freeswitch-users] Error while making Attendent transfer
Message-ID: <50DC4659.9080405@gmail.com>

hai,

     i am working on Freeswitch. I am getting error which i displayed 
below while i making attendent transfer using default and features xml 
files.
And also tell me how to make successful attendent transfer in freeswitch

2012-12-27 18:29:16.662856 [CRIT] mod_local_stream.c:297 Leaking stream 
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1237]

I am using twinkle and SJphone as soft phones.

please help me

thank you


From a.venugopan at mundio.com  Thu Dec 27 17:09:27 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Thu, 27 Dec 2012 14:09:27 +0000
Subject: [Freeswitch-users] repeated registration messages in logs
In-Reply-To: <CAJwc9xECzG84Oc4LwnHSKYrRGFYdzxyjyyarofG1=X=XjMmG+g@mail.gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF233D501@Mail-Kilo.squay.com>
	<CAJwc9xECzG84Oc4LwnHSKYrRGFYdzxyjyyarofG1=X=XjMmG+g@mail.gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233D56B@Mail-Kilo.squay.com>

Hi,
Thanks. If there is no activity for a while then calls drops out and I need to re-start the FS. I also see these messages frequently after each call and after registration and while FS is idle.
Am not sure why is this happening.

Regards,
Archana

From: Mick Stevens [mailto:mick.stevens at smartipx.com]
Sent: 27 December 2012 12:49
To: Archana Venugopan
Subject: Re: [Freeswitch-users] repeated registration messages in logs

Hi Archana,

This is just your SIP client (softphone etc) sending keep alive/SIP re-register messages. You can normally adjust this in the settings of your client.

I find setting "reregister every 300 seconds" normally works ok for me.

Rgds, Mick


On 27 December 2012 12:40, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

After registering the phones successfully I am getting the below message continuously every few seconds in my logs and this is filling up the logs and freeswitch goes down sometimes.
Can someone please tell me why I keep getting this message every few seconds? I have another server and there I don?t see this issue.

2012-12-27 12:37:44.714558 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [Abarajit at vectonecloud.com<mailto:Abarajit at vectonecloud.com>] from ip 192.168.2.205
2012-12-27 12:41:58.744557 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [sana at vectonecloud.com<mailto:sana at vectonecloud.com>] from ip 192.168.2.48

Regards,
Archana


_________________________________________________________________________
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



--
Regards

Mick Stevens
Communications Technologist
Smart IPX Ltd
Tel/Fax. +44(0)20 7001 6869
Email. mick.stevens at smartipx.com<mailto:mick.stevens at smartipx.com>
Skype. mick.smartipx.com<http://mick.gno247.com/>
www.smartipx.com<http://www.smartipx.com/>
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From andrew at cassidywebservices.co.uk  Thu Dec 27 17:25:47 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 27 Dec 2012 14:25:47 +0000
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <50D95747.6090502@gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<50D6865B.4010503@communicatefreely.net>
	<CA+Y7Lf4fG_eT_f6jAbpPhE0VhPd2=A9jdXt-OtJFnHd-jrtjeA@mail.gmail.com>
	<50D95747.6090502@gmail.com>
Message-ID: <CAMbW9oLonp2S7WG7KzXEtXHH8bYzo_EdpAnGdEVKvGOzYD_ekg@mail.gmail.com>

if my subnet changes i only need to change ONE setting on my entire
network.

On 25 December 2012 07:35, Mimiko <vbvbrj at gmail.com> wrote:

> On 25.12.2012 02:14, Scott wrote:
>
> > Anyone with medium to large addressable end-points in their
> > installations really needs to look at implementing IPv6 WITH -- repeat
> > WITH -- IPv6-NAT (and/or NAT64) in the mix.
> >
> I will use this when moving to ipv6. At my company we have two internet
> and voip providers for fault tolerance and load balacing. Also for using
> lowest cost per call. The plans are to have third internet and voip
> provider.
>
> So using internal ipv6 for addressing and dynamically mapping to those,
> provided by internet providers, is the only way to work.
>
> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
*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 a.venugopan at mundio.com  Thu Dec 27 17:44:05 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Thu, 27 Dec 2012 14:44:05 +0000
Subject: [Freeswitch-users] repeated registration messages in logs
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233D56B@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233D501@Mail-Kilo.squay.com>
	<CAJwc9xECzG84Oc4LwnHSKYrRGFYdzxyjyyarofG1=X=XjMmG+g@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233D56B@Mail-Kilo.squay.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233D592@Mail-Kilo.squay.com>

Hi,
Thanks. If there is no activity for a while then calls drops out and I need to re-start the FS. I also see these messages frequently after each call and after registration and while FS is idle.
Am not sure why is this happening.

I keep getting those messages every 5 mins.

Regards,
Archana

From: Mick Stevens [mailto:mick.stevens at smartipx.com]
Sent: 27 December 2012 12:49
To: Archana Venugopan
Subject: Re: [Freeswitch-users] repeated registration messages in logs

Hi Archana,

This is just your SIP client (softphone etc) sending keep alive/SIP re-register messages. You can normally adjust this in the settings of your client.

I find setting "reregister every 300 seconds" normally works ok for me.

Rgds, Mick


On 27 December 2012 12:40, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

After registering the phones successfully I am getting the below message continuously every few seconds in my logs and this is filling up the logs and freeswitch goes down sometimes.
Can someone please tell me why I keep getting this message every few seconds? I have another server and there I don?t see this issue.

2012-12-27 12:37:44.714558 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [Abarajit at vectonecloud.com<mailto:Abarajit at vectonecloud.com>] from ip 192.168.2.205
2012-12-27 12:41:58.744557 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [sana at vectonecloud.com<mailto:sana at vectonecloud.com>] from ip 192.168.2.48

Regards,
Archana


_________________________________________________________________________
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



--
Regards

Mick Stevens
Communications Technologist
Smart IPX Ltd
Tel/Fax. +44(0)20 7001 6869
Email. mick.stevens at smartipx.com<mailto:mick.stevens at smartipx.com>
Skype. mick.smartipx.com<http://mick.gno247.com/>
www.smartipx.com<http://www.smartipx.com/>
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From andrew at cassidywebservices.co.uk  Thu Dec 27 17:45:10 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 27 Dec 2012 14:45:10 +0000
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAMbW9oLonp2S7WG7KzXEtXHH8bYzo_EdpAnGdEVKvGOzYD_ekg@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<50D6865B.4010503@communicatefreely.net>
	<CA+Y7Lf4fG_eT_f6jAbpPhE0VhPd2=A9jdXt-OtJFnHd-jrtjeA@mail.gmail.com>
	<50D95747.6090502@gmail.com>
	<CAMbW9oLonp2S7WG7KzXEtXHH8bYzo_EdpAnGdEVKvGOzYD_ekg@mail.gmail.com>
Message-ID: <CAMbW9o+25tTOkdz410apxVJkb4t-iOLx8hL=CwsVPkMG+MuKGQ@mail.gmail.com>

Also if you're even thinking about using NAT for IPv6, you're doing it
wrong and are still thinking of NAT as a useful mechanism which suggests
you're not looking at IPv6 with a completely fresh mindset. Yes there are
v4/v6 similarities, but they are few and far between. Any problems caused
by a missing feature in v6 are because that feature in v4 is BAD.

For example, people use NAT as a security mechanism. This is
bad, because it's not. NAPT even violates the OSI model.

If you want services to only be accessible privately, assign both private
and public address and use your host and edge firewalls and listen
parameters on the service so that only connections from the local subnets
are allowed.

IPv6 specifies that all hosts generate a non-routeable link-local address
anyway. This makes perfect sense, as it brings ip-related layer 2 services
u to layer 3, making them link-type independent. One such example is DHCPv6.

There are also specifications for non-externally-routeable site-local
addresses, which are only routeable within the LAN but can
span multiple subnets. That's larger deployments taken care of.

The thing to remember is that IPv6 is designed for each host to have
multiple addresses, meaning you can have a link-local, site-local and
globally routeable IP address per host which can each be
secured independently. Manage it all correctly and there are no problems. A
subnet change should just mean that you need to update DNS records for your
external services and edge routers with the new subnet mask.

Just my 2 cents.

On 27 December 2012 14:25, Andrew Cassidy
<andrew at cassidywebservices.co.uk>wrote:

> if my subnet changes i only need to change ONE setting on my entire
> network.
>
>
> On 25 December 2012 07:35, Mimiko <vbvbrj at gmail.com> wrote:
>
>> On 25.12.2012 02:14, Scott wrote:
>>
>> > Anyone with medium to large addressable end-points in their
>> > installations really needs to look at implementing IPv6 WITH -- repeat
>> > WITH -- IPv6-NAT (and/or NAT64) in the mix.
>> >
>> I will use this when moving to ipv6. At my company we have two internet
>> and voip providers for fault tolerance and load balacing. Also for using
>> lowest cost per call. The plans are to have third internet and voip
>> provider.
>>
>> So using internal ipv6 for addressing and dynamically mapping to those,
>> provided by internet providers, is the only way to work.
>>
>> --
>> Mimiko desu.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>
>
>
> --
> *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 Tim.Meade at Millicorp.com  Thu Dec 27 19:15:39 2012
From: Tim.Meade at Millicorp.com (Tim Meade)
Date: Thu, 27 Dec 2012 16:15:39 +0000
Subject: [Freeswitch-users] xml_curl and ivr menus
In-Reply-To: <CAHKQagE7MHcTO+AxXtMJPK+KS_P1L3vB+EETXhM1tVy3EvxgbA@mail.gmail.com>
References: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>
	<50D8BE29.7010404@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8A9E3@MAIL.millicorp.com>
	<50D8CC6A.5000706@gmail.com> <50D8CCC3.7090305@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8B121@MAIL.millicorp.com>
	<804D48104511D4468F0D60DF9D3100350AF8DCA3@MAIL.millicorp.com>
	<CAHKQagE7MHcTO+AxXtMJPK+KS_P1L3vB+EETXhM1tVy3EvxgbA@mail.gmail.com>
Message-ID: <804D48104511D4468F0D60DF9D3100350AF9F06B@MAIL.millicorp.com>

That's great Cal!  I was hoping to find the time to add some examples myself.    Big one is that there is not a section for phrases.   The example below is working for me.

Let me know if there is anything I can do to help.  We use xml_curl for just about everything here.

Tim

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cal Leeming [Simplicity Media Ltd]
Sent: Wednesday, December 26, 2012 3:57 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] xml_curl and ivr menus

Tim,

Chiming in on this, as my focus has been purely on mod_xml_curl for the last 2 months.

My apologies that you struggled with finding a working examples, the mod_xml_curl documentation is still a work in progress - and despite that it has recently had a partial re-write, there are still many parts which aren't clear enough.

I will be updating the docs soon (in the next week) to show example request/response of as many different scenarios as possible, along with context specific descriptions of each variable.

Cal

On Tue, Dec 25, 2012 at 2:07 PM, Tim Meade <Tim.Meade at millicorp.com<mailto:Tim.Meade at millicorp.com>> wrote:
Here is an example of the xml_curl phrases section.  This is not in the WIKI.....  I'll try to update.  Notice the language section....

<document type="freeswitch/xml">
                  <section name="phrases">
                      <macros>
                           <include>
                                <language name="en" sound-path="$${sounds_dir}/en/us/callie" tts-engine="cepstral" tts-voice="callie">
                                     <macro name="demo_ivr_main_menu" pause="100"> <!-- See conf/autoload_config/ivr.conf.xml for an example on how to use this macro in an IVR -->
                                               <input pattern="(.*)">
                                                 <match>
                                                   <!-- string together several existing sound files to create one long greeting -->
                                                   <action function="play-file" data="ivr/ivr-welcome_to_freeswitch.wav"/>
                                                   <action function="play-file" data="ivr/ivr-this_ivr_will_let_you_test_features.wav"/>
                                                   <action function="play-file" data="ivr/ivr-you_may_exit_by_hanging_up.wav"/>
                                                   <!-- note that you can do more than just play files, e.g. have pauses and do TTS -->

                                                   <!-- Menu option 1: Call FreeSWITCH conference-->
                                                   <action function="play-file" data="ivr/ivr-enter_ext_pound.wav"/>
                                                   <action function="play-file" data="silence_stream://1500"/>
                                                   <action function="play-file" data="ivr/ivr-to_call_the_freeswitch_conference.wav"/>
                                                   <action function="play-file" data="ivr/ivr-please.wav"/>
                                                   <action function="play-file" data="voicemail/vm-press.wav"/>
                                                   <action function="play-file" data="digits/1.wav"/>

                                                   <!-- Menu option 2: Do FreeSWITCH echo test -->
                                                   <action function="play-file" data="ivr/ivr-to_do_a_freeswitch_echo_test.wav"/>
                                                   <action function="play-file" data="ivr/ivr-please.wav"/>
                                                   <action function="play-file" data="voicemail/vm-press.wav"/>
                                                   <action function="play-file" data="digits/2.wav"/>

                                                   <!-- Menu option 3: Listen to Music on Hold -->
                                                   <action function="play-file" data="ivr/ivr-to_listen_to_moh.wav"/>
                                                   <action function="play-file" data="ivr/ivr-please.wav"/>
                                                   <action function="play-file" data="voicemail/vm-press.wav"/>
                                                   <action function="play-file" data="digits/3.wav"/>

                                                   <!-- Menu option 4: Register for ClueCon -->
                                                   <action function="play-file" data="ivr/ivr-register_for_cluecon.wav"/>
                                                   <action function="play-file" data="digits/4.wav"/>

                                                   <!-- Menu option 5: Listen to screaming monkeys -->
                                                   <action function="play-file" data="ivr/ivr-to_hear_screaming_monkeys.wav"/>
                                                   <action function="play-file" data="ivr/ivr-please.wav"/>
                                                   <action function="play-file" data="voicemail/vm-press.wav"/>
                                                   <action function="play-file" data="digits/5.wav"/>

                                                   <!-- Menu option 6: Hear a sample submenu -->
                                                   <action function="play-file" data="ivr/ivr-to_hear_sample_submenu.wav"/>
                                                   <action function="play-file" data="ivr/ivr-please.wav"/>
                                                   <action function="play-file" data="voicemail/vm-press.wav"/>
                                                   <action function="play-file" data="digits/6.wav"/>

                                                   <!-- Menu option 9: Repeat these options -->
                                                   <action function="play-file" data="ivr/ivr-to_repeat_these_options.wav"/>
                                                   <action function="play-file" data="ivr/ivr-please.wav"/>
                                                   <action function="play-file" data="voicemail/vm-press.wav"/>
                                                   <action function="play-file" data="digits/9.wav"/>
                                                   <action function="play-file" data="silence_stream://2000"/>
                                                 </match>
                                               </input>
                                             </macro>
                                       </language>
                                </include>
                       </macros>
                  </section>
                </document>


Thanks All

Tim

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 Tim Meade
Sent: Monday, December 24, 2012 5:05 PM
To: Abaci; FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] xml_curl and ivr menus

Thank you Abaci....   Just found that myself under the xml_curl section of the wiki.....  I hadn't noticed the xml_curl configuration parameters for the return xml are different.

Here is the completed working configuration from xml_curl debug_on.

<document type="freeswitch/xml">
<section name="configuration">
<configuration name="ivr.conf" description="IVR menus">
<menus>
<include>
<!-- demo IVR setup -->
<!-- demo IVR, Main Menu -->
<menu name="tivr"
greet-long="phrase:demo_ivr_main_menu"
greet-short="phrase:demo_ivr_main_menu_short"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
confirm-macro=""
confirm-key=""
tts-engine="flite"
tts-voice="rms"
confirm-attempts="3"
timeout="10000"
inter-digit-timeout="2000"
max-failures="3"
max-timeouts="3"
digit-len="4">
<!-- The following are the definitions for the digits the user dials -->
<!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
<entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888 at conference.freeswitch.org<mailto:888 at conference.freeswitch.org>"/>
<entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/> <!-- FS echo -->
<entry action="menu-exec-app" digits="3" param="transfer 9664 XML default"/> <!-- MOH -->
<entry action="menu-exec-app" digits="4" param="transfer 9191 XML default"/> <!-- ClueCon -->
<entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/> <!-- Screaming monkeys -->
<!-- Using a regex in the digits tag lets you define a dial pattern for the caller
You may define multiple regexes if you need a different pattern for some reason -->
<entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
<entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
</menu>
</include>
</menus>
</configuration>
</section>
</document>

From: Abaci [mailto:abaci64 at gmail.com<mailto:abaci64 at gmail.com>]
Sent: Monday, December 24, 2012 4:45 PM
To: FreeSWITCH Users Help
Cc: Tim Meade
Subject: Re: [Freeswitch-users] xml_curl and ivr menus

you're also missing the "section" element.

On 12/24/2012 4:43 PM, Abaci wrote:
Here is a sample from my server, I highlighted the missing stuff.

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="configuration">
    <configuration name="ivr.conf" description="IVR menus">
      <menus>
        <menu name="ivrmenu1"
              invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
              exit-sound="voicemail/vm-goodbye.wav"
              timeout ="3000"
              inter-digit-timeout="2000"
              max-failures="2"
              digit-len="3"
              phrase-lang="en">
          <entry action="menu-exec-app" digits="0" param="transfer 11 XML voicemail " />
          <entry action="menu-exec-app" digits="1" param="transfer ivrmenu2 XML ivr " />
          <entry action="menu-exec-app" digits="2" param="transfer ivrmenu3 XML ivr " />
        </menu>
      </menus>
    </configuration>
  </section>
</document>

On 12/24/2012 4:22 PM, Tim Meade wrote:
The items I pasted below are from the cat of the xml_curl debug_on command.  It certainly appears to me that freeswitch is getting the correct name in this case 'tivr' which is what shows in both the dialplan and the curl'd ivr menu.

My concerns now are dealing with the xml format for the ivr as I have found several different versions on the wiki and in the prior postings.

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 Abaci
Sent: Monday, December 24, 2012 3:42 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] xml_curl and ivr menus

on freeswitch enable xml_curl debug and see if freeswitch is actually getting the correct ivr
'xml_curl debug_on'

On 12/24/2012 2:20 PM, Tim Meade wrote:
I am continually getting 'unable to find menu' while trying to retrieve an ivr_menu using xml_curl

Using xml_curl debug_on

My DialPlan

<document type="freeswitch/xml">
<section name="dialplan" description="example_curl_dialplan">
<context name="public">
<extension name="ws_origination">
<condition field="destination_number" expression="^\d+$" break="on-true">
<action inline="true" application="set" data="continue_on_fail=true"/>
<action inline="true" application="set" data="hangup_after_bridge=true"/>
<action application="answer"/>
<action application="ivr" data="tivr"/>
</condition>
</extension>
</context>
</section>

And the ivr configuration returned by my xml_curl

<include>
<!-- demo IVR setup -->
<!-- demo IVR, Main Menu -->
<menu name="tivr"
greet-long="phrase:demo_ivr_main_menu"
greet-short="phrase:demo_ivr_main_menu_short"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
confirm-macro=""
confirm-key=""
tts-engine="flite"
tts-voice="rms"
confirm-attempts="3"
timeout="10000"
inter-digit-timeout="2000"
max-failures="3"
max-timeouts="3"
digit-len="4">
<!-- The following are the definitions for the digits the user dials -->
<!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
<entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888 at conference.freeswitch.org<mailto:domain%7d/888 at conference.freeswitch.org>"/>
<entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/> <!-- FS echo -->
<entry action="menu-exec-app" digits="3" param="transfer 9664 XML default"/> <!-- MOH -->
<entry action="menu-exec-app" digits="4" param="transfer 9191 XML default"/> <!-- ClueCon -->
<entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/> <!-- Screaming monkeys -->
<!-- Using a regex in the digits tag lets you define a dial pattern for the caller
You may define multiple regexes if you need a different pattern for some reason -->
<entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
<entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
</menu>
</include>

Any ideas on the 'unable to find menu' ????

Thanks

Tim






_________________________________________________________________________

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



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

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 msc at freeswitch.org  Thu Dec 27 20:33:50 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Dec 2012 09:33:50 -0800
Subject: [Freeswitch-users] xml_curl and ivr menus
In-Reply-To: <804D48104511D4468F0D60DF9D3100350AF9F06B@MAIL.millicorp.com>
References: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>
	<50D8BE29.7010404@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8A9E3@MAIL.millicorp.com>
	<50D8CC6A.5000706@gmail.com> <50D8CCC3.7090305@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8B121@MAIL.millicorp.com>
	<804D48104511D4468F0D60DF9D3100350AF8DCA3@MAIL.millicorp.com>
	<CAHKQagE7MHcTO+AxXtMJPK+KS_P1L3vB+EETXhM1tVy3EvxgbA@mail.gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF9F06B@MAIL.millicorp.com>
Message-ID: <CAKzWOxXmk_M6ZXs79h5_D+QSjjTuQtC6gq7cYA+TYYw5+4F5xw@mail.gmail.com>

Gentlemen,

Many thanks for stepping up to help get this documented. If anyone needs
assistance w/ wiki stuff or has any documentation questions please let me
know and I will be glad to assist!

Thanks,
MC

On Thu, Dec 27, 2012 at 8:15 AM, Tim Meade <Tim.Meade at millicorp.com> wrote:

>  That?s great Cal!  I was hoping to find the time to add some examples
> myself.    Big one is that there is not a section for phrases.   The
> example below is working for me.****
>
> ** **
>
> Let me know if there is anything I can do to help.  We use xml_curl for
> just about everything here.****
>
> ** **
>
> Tim****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal Leeming
> [Simplicity Media Ltd]
> *Sent:* Wednesday, December 26, 2012 3:57 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>
> ** **
>
> Tim,****
>
> ** **
>
> Chiming in on this, as my focus has been purely on mod_xml_curl for the
> last 2 months.****
>
> ** **
>
> My apologies that you struggled with finding a working examples, the
> mod_xml_curl documentation is still a work in progress - and despite that
> it has recently had a partial re-write, there are still many parts which
> aren't clear enough.****
>
> ** **
>
> I will be updating the docs soon (in the next week) to show example
> request/response of as many different scenarios as possible, along with
> context specific descriptions of each variable.****
>
> ** **
>
> Cal****
>
> ** **
>
> On Tue, Dec 25, 2012 at 2:07 PM, Tim Meade <Tim.Meade at millicorp.com>
> wrote:****
>
> Here is an example of the xml_curl phrases section.  This is not in the
> WIKI?..  I?ll try to update.  Notice the language section?.****
>
>  ****
>
> <document type="freeswitch/xml">****
>
>                   <section name="phrases">****
>
>                       <macros>****
>
>                            <include>****
>
>                                 <language name="en" sound-path="
> $${sounds_dir}/en/us/callie" tts-engine="cepstral" tts-voice="callie">****
>
>                                      <macro name="demo_ivr_main_menu"
> pause="100"> <!-- See conf/autoload_config/ivr.conf.xml for an example on
> how to use this macro in an IVR -->****
>
>                                                <input pattern="(.*)">****
>
>                                                  <match>****
>
>                                                    <!-- string together
> several existing sound files to create one long greeting -->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-welcome_to_freeswitch.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-this_ivr_will_let_you_test_features.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-you_may_exit_by_hanging_up.wav"/>****
>
>                                                    <!-- note that you can
> do more than just play files, e.g. have pauses and do TTS -->****
>
>                                            ****
>
>                                                    <!-- Menu option 1:
> Call FreeSWITCH conference-->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-enter_ext_pound.wav"/>****
>
>                                                    <action function="
> play-file" data="silence_stream://1500"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-to_call_the_freeswitch_conference.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-please.wav"/>****
>
>                                                    <action function="
> play-file" data="voicemail/vm-press.wav"/>****
>
>                                                    <action function="
> play-file" data="digits/1.wav"/>****
>
>                                            ****
>
>                                                    <!-- Menu option 2: Do
> FreeSWITCH echo test -->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-to_do_a_freeswitch_echo_test.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-please.wav"/>****
>
>                                                    <action function="
> play-file" data="voicemail/vm-press.wav"/>****
>
>                                                    <action function="
> play-file" data="digits/2.wav"/>****
>
>                                            ****
>
>                                                    <!-- Menu option 3:
> Listen to Music on Hold -->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-to_listen_to_moh.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-please.wav"/>****
>
>                                                    <action function="
> play-file" data="voicemail/vm-press.wav"/>****
>
>                                                    <action function="
> play-file" data="digits/3.wav"/>****
>
>                                            ****
>
>                                                    <!-- Menu option 4:
> Register for ClueCon -->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-register_for_cluecon.wav"/>****
>
>                                                    <action function="
> play-file" data="digits/4.wav"/>****
>
>                                            ****
>
>                                                    <!-- Menu option 5:
> Listen to screaming monkeys -->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-to_hear_screaming_monkeys.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-please.wav"/>****
>
>                                                    <action function="
> play-file" data="voicemail/vm-press.wav"/>****
>
>                                                    <action function="
> play-file" data="digits/5.wav"/>****
>
>                                            ****
>
>                                                    <!-- Menu option 6:
> Hear a sample submenu -->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-to_hear_sample_submenu.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-please.wav"/>****
>
>                                                    <action function="
> play-file" data="voicemail/vm-press.wav"/>****
>
>                                                    <action function="
> play-file" data="digits/6.wav"/>****
>
>                                            ****
>
>                                                    <!-- Menu option 9:
> Repeat these options -->****
>
>                                                    <action function="
> play-file" data="ivr/ivr-to_repeat_these_options.wav"/>****
>
>                                                    <action function="
> play-file" data="ivr/ivr-please.wav"/>****
>
>                                                    <action function="
> play-file" data="voicemail/vm-press.wav"/>****
>
>                                                    <action function="
> play-file" data="digits/9.wav"/>****
>
>                                                    <action function="
> play-file" data="silence_stream://2000"/>****
>
>                                                  </match>****
>
>                                                </input>****
>
>                                              </macro>      ****
>
>                                        </language>
>       ****
>
>                                 </include>****
>
>                        </macros>
>                 ****
>
>                   </section>****
>
>                 </document>****
>
>  ****
>
>  ****
>
> Thanks All****
>
>  ****
>
> Tim****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tim Meade
> *Sent:* Monday, December 24, 2012 5:05 PM
> *To:* Abaci; FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>
>  ****
>
> Thank you Abaci?.   Just found that myself under the xml_curl section of
> the wiki?..  I hadn?t noticed the xml_curl configuration parameters for the
> return xml are different.    ****
>
>  ****
>
> Here is the completed working configuration from xml_curl debug_on.   ****
>
>  ****
>
> <document type="freeswitch/xml">****
>
> <section name="configuration">****
>
> <configuration name="ivr.conf" description="IVR menus">****
>
> <menus>****
>
> <include>****
>
> <!-- demo IVR setup -->****
>
> <!-- demo IVR, Main Menu -->****
>
> <menu name="tivr"****
>
> greet-long="phrase:demo_ivr_main_menu"****
>
> greet-short="phrase:demo_ivr_main_menu_short"****
>
> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"****
>
> exit-sound="voicemail/vm-goodbye.wav"****
>
> confirm-macro=""****
>
> confirm-key=""****
>
> tts-engine="flite"****
>
> tts-voice="rms"****
>
> confirm-attempts="3"****
>
> timeout="10000"****
>
> inter-digit-timeout="2000"****
>
> max-failures="3"****
>
> max-timeouts="3"****
>
> digit-len="4">****
>
> <!-- The following are the definitions for the digits the user dials -->**
> **
>
> <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->****
>
> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/
> 888 at conference.freeswitch.org"/>****
>
> <entry action="menu-exec-app" digits="2" param="transfer 9196 XML
> default"/> <!-- FS echo -->****
>
> <entry action="menu-exec-app" digits="3" param="transfer 9664 XML
> default"/> <!-- MOH -->****
>
> <entry action="menu-exec-app" digits="4" param="transfer 9191 XML
> default"/> <!-- ClueCon -->****
>
> <entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/>
> <!-- Screaming monkeys -->****
>
> <!-- Using a regex in the digits tag lets you define a dial pattern for
> the caller****
>
> You may define multiple regexes if you need a different pattern for some
> reason -->****
>
> <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer
> $1 XML features"/>****
>
> <entry action="menu-top" digits="9"/> <!-- Repeat this menu -->****
>
> </menu>****
>
> </include>****
>
> </menus>****
>
> </configuration>****
>
> </section>****
>
> </document>****
>
>  ****
>
> *From:* Abaci [mailto:abaci64 at gmail.com]
> *Sent:* Monday, December 24, 2012 4:45 PM
> *To:* FreeSWITCH Users Help
> *Cc:* Tim Meade
> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>
>  ****
>
> you're also missing the "section" element.
>
> On 12/24/2012 4:43 PM, Abaci wrote:****
>
>  Here is a sample from my server, I highlighted the missing stuff.
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="configuration">
> *    <configuration name="ivr.conf" description="IVR menus">
>       <menus>*
>         <menu name="ivrmenu1"
>               invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>               exit-sound="voicemail/vm-goodbye.wav"
>               timeout ="3000"
>               inter-digit-timeout="2000"
>               max-failures="2"
>               digit-len="3"
>               phrase-lang="en">
>           <entry action="menu-exec-app" digits="0" param="transfer 11 XML
> voicemail " />
>           <entry action="menu-exec-app" digits="1" param="transfer
> ivrmenu2 XML ivr " />
>           <entry action="menu-exec-app" digits="2" param="transfer
> ivrmenu3 XML ivr " />
>         </menu>
> *      </menus>
>     </configuration>*
>   </section>
> </document>
>
> On 12/24/2012 4:22 PM, Tim Meade wrote:****
>
> The items I pasted below are from the cat of the xml_curl debug_on
> command.  It certainly appears to me that freeswitch is getting the correct
> name in this case ?tivr? which is what shows in both the dialplan and the
> curl?d ivr menu.****
>
>  ****
>
> My concerns now are dealing with the xml format for the ivr as I have
> found several different versions on the wiki and in the prior postings.***
> *
>
>  ****
>
> *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 *Abaci
> *Sent:* Monday, December 24, 2012 3:42 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>
>  ****
>
> on freeswitch enable xml_curl debug and see if freeswitch is actually
> getting the correct ivr
> 'xml_curl debug_on'
>
> On 12/24/2012 2:20 PM, Tim Meade wrote:****
>
> I am continually getting ?unable to find menu? while trying to retrieve an
> ivr_menu using xml_curl****
>
>  ****
>
> Using xml_curl debug_on ****
>
>  ****
>
> My DialPlan****
>
>  ****
>
> <document type="freeswitch/xml">****
>
> <section name="dialplan" description="example_curl_dialplan">****
>
> <context name="public">****
>
> <extension name="ws_origination">****
>
> <condition field="destination_number" expression="^\d+$" break="on-true">*
> ***
>
> <action inline="true" application="set" data="continue_on_fail=true"/>****
>
> <action inline="true" application="set" data="hangup_after_bridge=true"/>
> ****
>
> <action application="answer"/>****
>
> <action application="ivr" data="tivr"/>****
>
> </condition>****
>
> </extension>****
>
> </context>****
>
> </section>****
>
>  ****
>
> And the ivr configuration returned by my xml_curl****
>
>  ****
>
> <include>****
>
> <!-- demo IVR setup -->****
>
> <!-- demo IVR, Main Menu -->****
>
> <menu name="tivr"****
>
> greet-long="phrase:demo_ivr_main_menu"****
>
> greet-short="phrase:demo_ivr_main_menu_short"****
>
> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"****
>
> exit-sound="voicemail/vm-goodbye.wav"****
>
> confirm-macro=""****
>
> confirm-key=""****
>
> tts-engine="flite"****
>
> tts-voice="rms"****
>
> confirm-attempts="3"****
>
> timeout="10000"****
>
> inter-digit-timeout="2000"****
>
> max-failures="3"****
>
> max-timeouts="3"****
>
> digit-len="4">****
>
> <!-- The following are the definitions for the digits the user dials -->**
> **
>
> <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->****
>
> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${
> domain}/888 at conference.freeswitch.org"/>****
>
> <entry action="menu-exec-app" digits="2" param="transfer 9196 XML
> default"/> <!-- FS echo -->****
>
> <entry action="menu-exec-app" digits="3" param="transfer 9664 XML
> default"/> <!-- MOH -->****
>
> <entry action="menu-exec-app" digits="4" param="transfer 9191 XML
> default"/> <!-- ClueCon -->****
>
> <entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/>
> <!-- Screaming monkeys -->****
>
> <!-- Using a regex in the digits tag lets you define a dial pattern for
> the caller****
>
> You may define multiple regexes if you need a different pattern for some
> reason -->****
>
> <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer
> $1 XML features"/>****
>
> <entry action="menu-top" digits="9"/> <!-- Repeat this menu -->****
>
> </menu>****
>
> </include>****
>
>  ****
>
> Any ideas on the ?unable to find menu? ????****
>
>  ****
>
> Thanks****
>
>  ****
>
> Tim****
>
>  ****
>
>  ****
>
>  ****
>
>
>
> ****
>
> _________________________________________________________________________****
>
> Professional FreeSWITCH Consulting 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****
>
>  ****
>
> FreeSWITCH-users mailing list****
>
> FreeSWITCH-users at lists.freeswitch.org****
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users****
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users****
>
> http://www.freeswitch.org****
>
>   ****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Dec 27 20:43:51 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Dec 2012 09:43:51 -0800
Subject: [Freeswitch-users] caller-id-number not getting set
In-Reply-To: <CAJbRaS4aJJw9_XCw08hWY_s_3OtCqGVoezA7Q9ooo0npCy5gKA@mail.gmail.com>
References: <CAJbRaS4aJJw9_XCw08hWY_s_3OtCqGVoezA7Q9ooo0npCy5gKA@mail.gmail.com>
Message-ID: <CAKzWOxXAa+XGGzV8tN2vS6z6_oJurU0qvK6qHiXgfNr8b2kF6A@mail.gmail.com>

If you want to do a condition based on the caller id name or number then
use the "special" variables like this:

<condition field="caller_id_name" field="^(.*)">
or
<condition field="caller_id_number" field="^(.*)">

If that doesn't work then you'll need to pastebin your dialplan and the
console debug output of a call traversing the dialplan so that we can have
a look.
-MC

On Thu, Dec 27, 2012 at 2:16 AM, sanjay soni <sanjay.sanjaysoni at gmail.com>wrote:

> In all the dialplan examples (Both in the book and wiki) most
> places destination_number is being tested in the <condition>, BUT I want to
> do something based on caller-id-name / caller-id-number channel variables.
> But I see that all these variables (Caller-Caller-ID-Name,
> sip_callee_id_name, sip_auth_username, caller-id-name, caller-id-number)
> are empty.
> however, i see these values properly in the event being fired But not in
> the xml dialplan (Which is the place i want ot use them) ! What should I do
> ?
>
> Thanks
> Sanjay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Dec 27 20:45:01 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Dec 2012 09:45:01 -0800
Subject: [Freeswitch-users] uuid_record and RECORD_STEREO
In-Reply-To: <89F15D50-E189-4D86-9320-102F086DF18F@gmail.com>
References: <89F15D50-E189-4D86-9320-102F086DF18F@gmail.com>
Message-ID: <CAKzWOxX+167M=k7acjY45H=WA2_eprwbs3S=L-9VZyTDCbQ4FQ@mail.gmail.com>

I'm not sure I understand the question. Could you elaborate? What do you
mean by "set a global state on the channel"?

-MC

On Thu, Dec 27, 2012 at 3:55 AM, Luca Pradovera <luca.pradovera at gmail.com>wrote:

> Hello,
> is there a way to invoke uuid_record through bgapi, setting RECORD_STEREO
> to true as a local parameter instead of using uuid_setvar, over inbound
> Event Socket?
> I would prefer to not set a global state on the channel if possible.
> Thanks!
>
> --
> Luca Pradovera
> luca.pradovera at gmail.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Dec 27 20:49:35 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Dec 2012 09:49:35 -0800
Subject: [Freeswitch-users] Error while making Attendent transfer
In-Reply-To: <50DC4659.9080405@gmail.com>
References: <50DC4659.9080405@gmail.com>
Message-ID: <CAKzWOxV_LaXNoqrZk8N-9ROAe3hYAwGNOBcp=J+Mh2QtrdUxqg@mail.gmail.com>

How often does this log line appear? If it only occurs a few times for a
given call then you should be okay, but if it keeps appearing over and over
again then there's definitely something not right with your configuration.

I recommend that you pastebin a complete console debug log with the call
from start to finish. Also, be sure to specify what IP addresses belong to
each device so that it's easier for us to decipher what we see in the logs.
Lastly, be sure to set syntax highlighting to "FreeSWITCH Log".

-MC

On Thu, Dec 27, 2012 at 5:00 AM, veerabhadrarao` <
bhadrarao.kankatala at gmail.com> wrote:

> hai,
>
>      i am working on Freeswitch. I am getting error which i displayed
> below while i making attendent transfer using default and features xml
> files.
> And also tell me how to make successful attendent transfer in freeswitch
>
> 2012-12-27 18:29:16.662856 [CRIT] mod_local_stream.c:297 Leaking stream
> handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1237]
>
> I am using twinkle and SJphone as soft phones.
>
> please help me
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Dec 27 20:54:47 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Dec 2012 09:54:47 -0800
Subject: [Freeswitch-users] regarding attendent transfer
In-Reply-To: <50DC447C.1080703@gmail.com>
References: <50DC447C.1080703@gmail.com>
Message-ID: <CAKzWOxUiRAWA1z=yZLM-mWELRYG7XpPVZcYR-XaQeEyFiA40OA@mail.gmail.com>

Are you using the *4 + extension number to do an attended transfer? Also,
keep in mind that the *4 attended transfer will *only* transfer to a
registered user on the system, that is, it transfers to "user/xxxx" where
xxxx is the destination number that you dial after pressing *4.

-MC

On Thu, Dec 27, 2012 at 4:52 AM, veerabhadrarao` <
bhadrarao.kankatala at gmail.com> wrote:

> hai
>
> Can anyone plaese tell me how to make attendent transfer. I am able to
> working on Blind transfer using default.xml and features.xml but not
> attendenet transfer successfully. I am using Freeswitch 1.2.4 version.
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Dec 27 20:57:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Dec 2012 09:57:27 -0800
Subject: [Freeswitch-users] repeated registration messages in logs
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233D501@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233D501@Mail-Kilo.squay.com>
Message-ID: <CAKzWOxW9qscVSDDizwQsxOrJOjiroKUd3KoZTXOiRLrJ1G2p-Q@mail.gmail.com>

In addition to what Nick mentioned in his reply...

These "warnings" are also used by fail2ban. If I understand correctly you
can avoid these log lines by using mod_fail2ban but I haven't actually
tried that yet.

-MC

On Thu, Dec 27, 2012 at 4:40 AM, Archana Venugopan
<a.venugopan at mundio.com>wrote:

>  Hi,****
>
> ** **
>
> After registering the phones successfully I am getting the below message
> continuously every few seconds in my logs and this is filling up the logs
> and freeswitch goes down sometimes.****
>
> Can someone please tell me why I keep getting this message every few
> seconds? I have another server and there I don?t see this issue. ****
>
> ** **
>
> 2012-12-27 12:37:44.714558 [WARNING] sofia_reg.c:1484 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [Abarajit at vectonecloud.com]
> from ip 192.168.2.205****
>
> 2012-12-27 12:41:58.744557 [WARNING] sofia_reg.c:1484 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [sana at vectonecloud.com] from
> ip 192.168.2.48****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jayesh.voip at gmail.com  Thu Dec 27 22:05:58 2012
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Fri, 28 Dec 2012 00:35:58 +0530
Subject: [Freeswitch-users] failover when storing recordings
Message-ID: <CALvF6vAUirzxuqxzryzGQ0BrP_+EEQVHy2WO2rbT-hNFDj_rig@mail.gmail.com>

Hi All,
Is there a way in freeswitch wherein if the call recording is not
successfully stored, it can store it in some separate path?
Basically the idea is, I am planning to use NFS for data storage. So
freeswitch is actually storing the recordings on an NFS mounted directory.
For some reason, if the NFS mount fails, or if the disc on the storage
server is full or whatever may be the reason, could there be a way that
freeswitch can store it on a separate directory which is local to the
server. Does this sound like a feature or a requirement?? Thanks.

--- Jayesh
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From steveayre at gmail.com  Thu Dec 27 22:20:39 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 27 Dec 2012 19:20:39 +0000
Subject: [Freeswitch-users] repeated registration messages in logs
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233D592@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233D501@Mail-Kilo.squay.com>
	<CAJwc9xECzG84Oc4LwnHSKYrRGFYdzxyjyyarofG1=X=XjMmG+g@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233D56B@Mail-Kilo.squay.com>
	<592A9CF93E12394E8472A6CC66E66BF233D592@Mail-Kilo.squay.com>
Message-ID: <36376C0D-2F14-4C83-97DE-DD8DE4B1C3A6@gmail.com>

Registrations work by the client sending a REGISTER to the server, which returns 401 to challenge and client resends REGISTER with the authentication details.

The register has an expiry time. To remain registered the client must reREGISTER within this time, and each REGISTER has its own 401 authentication (to prevent replay attacks).

What you're seeing is a log message each time the registration is renewed. That's why you still see it even though FS is apparently idle.

The log message really is only informative. It logs at WARNING level so that you can use fail2ban while only logging serious messages so you don't get massive log files.

FS will log both the initial REGISTER/401 then at the 2nd REGISTER whether it succeeds or fails. Both are required to detect bots looking to break into an insecure or DDOS a SIP server, since the client might never send the 2nd REGISTER.

If you are not using fail2ban and want quieter logs, set the log-auth-failures param to false on the relevant Sofia profile. Add it if it doesn't exist.

mod_fail2ban provides native integration without having to log these messages for fail2ban to detect. If you use it you'll probably still need to set the above param. (Disclaimer: I haven't used the module).

Steve




On 27 Dec 2012, at 14:44, Archana Venugopan <a.venugopan at mundio.com> wrote:

> Hi,
> Thanks. If there is no activity for a while then calls drops out and I need to re-start the FS. I also see these messages frequently after each call and after registration and while FS is idle.
> Am not sure why is this happening.
>  
> I keep getting those messages every 5 mins.
>  
> Regards,
> Archana
>  
> From: Mick Stevens [mailto:mick.stevens at smartipx.com] 
> Sent: 27 December 2012 12:49
> To: Archana Venugopan
> Subject: Re: [Freeswitch-users] repeated registration messages in logs
>  
> Hi Archana,
>  
> This is just your SIP client (softphone etc) sending keep alive/SIP re-register messages. You can normally adjust this in the settings of your client.
>  
> I find setting "reregister every 300 seconds" normally works ok for me.
>  
> Rgds, Mick
>  
>  
> 
> On 27 December 2012 12:40, Archana Venugopan <a.venugopan at mundio.com> wrote:
> Hi,
>  
> After registering the phones successfully I am getting the below message continuously every few seconds in my logs and this is filling up the logs and freeswitch goes down sometimes.
> Can someone please tell me why I keep getting this message every few seconds? I have another server and there I don?t see this issue.
>  
> 2012-12-27 12:37:44.714558 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [Abarajit at vectonecloud.com] from ip 192.168.2.205
> 2012-12-27 12:41:58.744557 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [sana at vectonecloud.com] from ip 192.168.2.48
>  
> Regards,
> Archana
>  
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> 
> 
>  
> --
> Regards
> 
> Mick Stevens
> Communications Technologist
> Smart IPX Ltd
> Tel/Fax. +44(0)20 7001 6869
> Email. mick.stevens at smartipx.com
> Skype. mick.smartipx.com
> www.smartipx.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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Dec 27 23:02:03 2012
From: josefu at gmail.com (=?ISO-8859-1?Q?Jose_Fco=2E_Irles_Dur=E1?=)
Date: Thu, 27 Dec 2012 21:02:03 +0100
Subject: [Freeswitch-users] Avoid loopback channels
Message-ID: <CAJOPU6U2x7E8VSYLvkDJMTNZTZsD_eRkA_4seQVTfsqqM+yQrQ@mail.gmail.com>

Hello everybody,

I want to avoid using loopback channels in freeswitch, but I don't
know the correct way to build some behaviors.

For examle, a click2call (origination) to a "virtual destination":
I have a logic in my dialplan that when somebody dials XXX, it sends
the call through a round robin set of gateways (saving de last gw used
with the "hash/db" app and dialing sequentially from the least used to
most used if one of it fails), what is the freeswitch way for this
without loopback channels?

At this moment, i do this originate:
In the example, first ring in 256987455 (the call goes through
dialplan and fs bridge with diferent gws if some fails) and when it
answers, I transfer the call to the same dialplan with a lua script
(the scripts uses "transfer" app).

The originate data is:

{origination_uuid=uuid,bridge_generate_comfort_noise=true,origination_caller_id_name='12547895',origination_caller_id_number=12547895,effective_caller_id_number=12547895,effective_caller_id_name='12547895'}loopback/256987455/default
&lua('originate.lua uuid 12547895)

I think that with a "calculated" dial-string with the gateways to dial
and with variables "execute_on_XXX" it's a possibily, but it's the
only way?

Another case is dial to one of this "virtual destinations" in a ring
all group or dial to a sip device with call forwarding if no answer
set at the pbx (not at the phone).

Thanks and sorry if i don't write properly.
--
Jose Fco. Irles Dur?


From cal.leeming at simplicitymedialtd.co.uk  Thu Dec 27 23:48:35 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Thu, 27 Dec 2012 20:48:35 +0000
Subject: [Freeswitch-users] xml_curl and ivr menus
In-Reply-To: <CAKzWOxXmk_M6ZXs79h5_D+QSjjTuQtC6gq7cYA+TYYw5+4F5xw@mail.gmail.com>
References: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>
	<50D8BE29.7010404@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8A9E3@MAIL.millicorp.com>
	<50D8CC6A.5000706@gmail.com> <50D8CCC3.7090305@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8B121@MAIL.millicorp.com>
	<804D48104511D4468F0D60DF9D3100350AF8DCA3@MAIL.millicorp.com>
	<CAHKQagE7MHcTO+AxXtMJPK+KS_P1L3vB+EETXhM1tVy3EvxgbA@mail.gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF9F06B@MAIL.millicorp.com>
	<CAKzWOxXmk_M6ZXs79h5_D+QSjjTuQtC6gq7cYA+TYYw5+4F5xw@mail.gmail.com>
Message-ID: <CAHKQagGNgNcp2p=bEGSpsz8vghZQxuL+E9ogd0CYYSrE-Uae-w@mail.gmail.com>

Anthony has kindly sent across the NAT section of the new book, and the
feedback so far has been great.

I'm going to start on this tomorrow, will paste the URL once ready -
contributions/amendments are obviously welcome.

Cal

On Thu, Dec 27, 2012 at 5:33 PM, Michael Collins <msc at freeswitch.org> wrote:

> Gentlemen,
>
> Many thanks for stepping up to help get this documented. If anyone needs
> assistance w/ wiki stuff or has any documentation questions please let me
> know and I will be glad to assist!
>
> Thanks,
> MC
>
>
> On Thu, Dec 27, 2012 at 8:15 AM, Tim Meade <Tim.Meade at millicorp.com>wrote:
>
>>  That?s great Cal!  I was hoping to find the time to add some examples
>> myself.    Big one is that there is not a section for phrases.   The
>> example below is working for me.****
>>
>> ** **
>>
>> Let me know if there is anything I can do to help.  We use xml_curl for
>> just about everything here.****
>>
>> ** **
>>
>> Tim****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal
>> Leeming [Simplicity Media Ltd]
>> *Sent:* Wednesday, December 26, 2012 3:57 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>>
>> ** **
>>
>> Tim,****
>>
>> ** **
>>
>> Chiming in on this, as my focus has been purely on mod_xml_curl for the
>> last 2 months.****
>>
>> ** **
>>
>> My apologies that you struggled with finding a working examples, the
>> mod_xml_curl documentation is still a work in progress - and despite that
>> it has recently had a partial re-write, there are still many parts which
>> aren't clear enough.****
>>
>> ** **
>>
>> I will be updating the docs soon (in the next week) to show example
>> request/response of as many different scenarios as possible, along with
>> context specific descriptions of each variable.****
>>
>> ** **
>>
>> Cal****
>>
>> ** **
>>
>> On Tue, Dec 25, 2012 at 2:07 PM, Tim Meade <Tim.Meade at millicorp.com>
>> wrote:****
>>
>> Here is an example of the xml_curl phrases section.  This is not in the
>> WIKI?..  I?ll try to update.  Notice the language section?.****
>>
>>  ****
>>
>> <document type="freeswitch/xml">****
>>
>>                   <section name="phrases">****
>>
>>                       <macros>****
>>
>>                            <include>****
>>
>>                                 <language name="en" sound-path="
>> $${sounds_dir}/en/us/callie" tts-engine="cepstral" tts-voice="callie">***
>> *
>>
>>                                      <macro name="demo_ivr_main_menu"
>> pause="100"> <!-- See conf/autoload_config/ivr.conf.xml for an example
>> on how to use this macro in an IVR -->****
>>
>>                                                <input pattern="(.*)">****
>>
>>                                                  <match>****
>>
>>                                                    <!-- string together
>> several existing sound files to create one long greeting -->****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-welcome_to_freeswitch.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-this_ivr_will_let_you_test_features.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-you_may_exit_by_hanging_up.wav"/>****
>>
>>                                                    <!-- note that you
>> can do more than just play files, e.g. have pauses and do TTS -->****
>>
>>                                            ****
>>
>>                                                    <!-- Menu option 1:
>> Call FreeSWITCH conference-->****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-enter_ext_pound.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="silence_stream://1500"/>****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-to_call_the_freeswitch_conference.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-please.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="voicemail/vm-press.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="digits/1.wav"/>****
>>
>>                                            ****
>>
>>                                                    <!-- Menu option 2:
>> Do FreeSWITCH echo test -->****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-to_do_a_freeswitch_echo_test.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-please.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="voicemail/vm-press.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="digits/2.wav"/>****
>>
>>                                            ****
>>
>>                                                    <!-- Menu option 3:
>> Listen to Music on Hold -->****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-to_listen_to_moh.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-please.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="voicemail/vm-press.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="digits/3.wav"/>****
>>
>>                                            ****
>>
>>                                                    <!-- Menu option 4:
>> Register for ClueCon -->****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-register_for_cluecon.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="digits/4.wav"/>****
>>
>>                                            ****
>>
>>                                                    <!-- Menu option 5:
>> Listen to screaming monkeys -->****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-to_hear_screaming_monkeys.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-please.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="voicemail/vm-press.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="digits/5.wav"/>****
>>
>>                                            ****
>>
>>                                                    <!-- Menu option 6:
>> Hear a sample submenu -->****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-to_hear_sample_submenu.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-please.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="voicemail/vm-press.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="digits/6.wav"/>****
>>
>>                                            ****
>>
>>                                                    <!-- Menu option 9:
>> Repeat these options -->****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-to_repeat_these_options.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="ivr/ivr-please.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="voicemail/vm-press.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="digits/9.wav"/>****
>>
>>                                                    <action function="
>> play-file" data="silence_stream://2000"/>****
>>
>>                                                  </match>****
>>
>>                                                </input>****
>>
>>                                              </macro>      ****
>>
>>                                        </language>
>>       ****
>>
>>                                 </include>****
>>
>>                        </macros>
>>                 ****
>>
>>                   </section>****
>>
>>                 </document>****
>>
>>  ****
>>
>>  ****
>>
>> Thanks All****
>>
>>  ****
>>
>> Tim****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tim Meade
>> *Sent:* Monday, December 24, 2012 5:05 PM
>> *To:* Abaci; FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>>
>>  ****
>>
>> Thank you Abaci?.   Just found that myself under the xml_curl section of
>> the wiki?..  I hadn?t noticed the xml_curl configuration parameters for the
>> return xml are different.    ****
>>
>>  ****
>>
>> Here is the completed working configuration from xml_curl debug_on.   ***
>> *
>>
>>  ****
>>
>> <document type="freeswitch/xml">****
>>
>> <section name="configuration">****
>>
>> <configuration name="ivr.conf" description="IVR menus">****
>>
>> <menus>****
>>
>> <include>****
>>
>> <!-- demo IVR setup -->****
>>
>> <!-- demo IVR, Main Menu -->****
>>
>> <menu name="tivr"****
>>
>> greet-long="phrase:demo_ivr_main_menu"****
>>
>> greet-short="phrase:demo_ivr_main_menu_short"****
>>
>> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"****
>>
>> exit-sound="voicemail/vm-goodbye.wav"****
>>
>> confirm-macro=""****
>>
>> confirm-key=""****
>>
>> tts-engine="flite"****
>>
>> tts-voice="rms"****
>>
>> confirm-attempts="3"****
>>
>> timeout="10000"****
>>
>> inter-digit-timeout="2000"****
>>
>> max-failures="3"****
>>
>> max-timeouts="3"****
>>
>> digit-len="4">****
>>
>> <!-- The following are the definitions for the digits the user dials -->*
>> ***
>>
>> <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->****
>>
>> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/
>> 888 at conference.freeswitch.org"/>****
>>
>> <entry action="menu-exec-app" digits="2" param="transfer 9196 XML
>> default"/> <!-- FS echo -->****
>>
>> <entry action="menu-exec-app" digits="3" param="transfer 9664 XML
>> default"/> <!-- MOH -->****
>>
>> <entry action="menu-exec-app" digits="4" param="transfer 9191 XML
>> default"/> <!-- ClueCon -->****
>>
>> <entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/>
>> <!-- Screaming monkeys -->****
>>
>> <!-- Using a regex in the digits tag lets you define a dial pattern for
>> the caller****
>>
>> You may define multiple regexes if you need a different pattern for some
>> reason -->****
>>
>> <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer
>> $1 XML features"/>****
>>
>> <entry action="menu-top" digits="9"/> <!-- Repeat this menu -->****
>>
>> </menu>****
>>
>> </include>****
>>
>> </menus>****
>>
>> </configuration>****
>>
>> </section>****
>>
>> </document>****
>>
>>  ****
>>
>> *From:* Abaci [mailto:abaci64 at gmail.com]
>> *Sent:* Monday, December 24, 2012 4:45 PM
>> *To:* FreeSWITCH Users Help
>> *Cc:* Tim Meade
>> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>>
>>  ****
>>
>> you're also missing the "section" element.
>>
>> On 12/24/2012 4:43 PM, Abaci wrote:****
>>
>>  Here is a sample from my server, I highlighted the missing stuff.
>>
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>>   <section name="configuration">
>> *    <configuration name="ivr.conf" description="IVR menus">
>>       <menus>*
>>         <menu name="ivrmenu1"
>>               invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>               exit-sound="voicemail/vm-goodbye.wav"
>>               timeout ="3000"
>>               inter-digit-timeout="2000"
>>               max-failures="2"
>>               digit-len="3"
>>               phrase-lang="en">
>>           <entry action="menu-exec-app" digits="0" param="transfer 11 XML
>> voicemail " />
>>           <entry action="menu-exec-app" digits="1" param="transfer
>> ivrmenu2 XML ivr " />
>>           <entry action="menu-exec-app" digits="2" param="transfer
>> ivrmenu3 XML ivr " />
>>         </menu>
>> *      </menus>
>>     </configuration>*
>>   </section>
>> </document>
>>
>> On 12/24/2012 4:22 PM, Tim Meade wrote:****
>>
>> The items I pasted below are from the cat of the xml_curl debug_on
>> command.  It certainly appears to me that freeswitch is getting the correct
>> name in this case ?tivr? which is what shows in both the dialplan and the
>> curl?d ivr menu.****
>>
>>  ****
>>
>> My concerns now are dealing with the xml format for the ivr as I have
>> found several different versions on the wiki and in the prior postings.**
>> **
>>
>>  ****
>>
>> *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 *Abaci
>> *Sent:* Monday, December 24, 2012 3:42 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>>
>>  ****
>>
>> on freeswitch enable xml_curl debug and see if freeswitch is actually
>> getting the correct ivr
>> 'xml_curl debug_on'
>>
>> On 12/24/2012 2:20 PM, Tim Meade wrote:****
>>
>> I am continually getting ?unable to find menu? while trying to retrieve
>> an ivr_menu using xml_curl****
>>
>>  ****
>>
>> Using xml_curl debug_on ****
>>
>>  ****
>>
>> My DialPlan****
>>
>>  ****
>>
>> <document type="freeswitch/xml">****
>>
>> <section name="dialplan" description="example_curl_dialplan">****
>>
>> <context name="public">****
>>
>> <extension name="ws_origination">****
>>
>> <condition field="destination_number" expression="^\d+$" break="on-true">
>> ****
>>
>> <action inline="true" application="set" data="continue_on_fail=true"/>***
>> *
>>
>> <action inline="true" application="set" data="hangup_after_bridge=true"/>
>> ****
>>
>> <action application="answer"/>****
>>
>> <action application="ivr" data="tivr"/>****
>>
>> </condition>****
>>
>> </extension>****
>>
>> </context>****
>>
>> </section>****
>>
>>  ****
>>
>> And the ivr configuration returned by my xml_curl****
>>
>>  ****
>>
>> <include>****
>>
>> <!-- demo IVR setup -->****
>>
>> <!-- demo IVR, Main Menu -->****
>>
>> <menu name="tivr"****
>>
>> greet-long="phrase:demo_ivr_main_menu"****
>>
>> greet-short="phrase:demo_ivr_main_menu_short"****
>>
>> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"****
>>
>> exit-sound="voicemail/vm-goodbye.wav"****
>>
>> confirm-macro=""****
>>
>> confirm-key=""****
>>
>> tts-engine="flite"****
>>
>> tts-voice="rms"****
>>
>> confirm-attempts="3"****
>>
>> timeout="10000"****
>>
>> inter-digit-timeout="2000"****
>>
>> max-failures="3"****
>>
>> max-timeouts="3"****
>>
>> digit-len="4">****
>>
>> <!-- The following are the definitions for the digits the user dials -->*
>> ***
>>
>> <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->****
>>
>> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${
>> domain}/888 at conference.freeswitch.org"/>****
>>
>> <entry action="menu-exec-app" digits="2" param="transfer 9196 XML
>> default"/> <!-- FS echo -->****
>>
>> <entry action="menu-exec-app" digits="3" param="transfer 9664 XML
>> default"/> <!-- MOH -->****
>>
>> <entry action="menu-exec-app" digits="4" param="transfer 9191 XML
>> default"/> <!-- ClueCon -->****
>>
>> <entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/>
>> <!-- Screaming monkeys -->****
>>
>> <!-- Using a regex in the digits tag lets you define a dial pattern for
>> the caller****
>>
>> You may define multiple regexes if you need a different pattern for some
>> reason -->****
>>
>> <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer
>> $1 XML features"/>****
>>
>> <entry action="menu-top" digits="9"/> <!-- Repeat this menu -->****
>>
>> </menu>****
>>
>> </include>****
>>
>>  ****
>>
>> Any ideas on the ?unable to find menu? ????****
>>
>>  ****
>>
>> Thanks****
>>
>>  ****
>>
>> Tim****
>>
>>  ****
>>
>>  ****
>>
>>  ****
>>
>>
>>
>> ****
>>
>> _________________________________________________________________________****
>>
>> Professional FreeSWITCH Consulting 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****
>>
>>  ****
>>
>> FreeSWITCH-users mailing list****
>>
>> FreeSWITCH-users at lists.freeswitch.org****
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users****
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users****
>>
>> http://www.freeswitch.org****
>>
>>   ****
>>
>>  ****
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 cal.leeming at simplicitymedialtd.co.uk  Thu Dec 27 23:50:58 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Thu, 27 Dec 2012 20:50:58 +0000
Subject: [Freeswitch-users] xml_curl and ivr menus
In-Reply-To: <CAHKQagGNgNcp2p=bEGSpsz8vghZQxuL+E9ogd0CYYSrE-Uae-w@mail.gmail.com>
References: <804D48104511D4468F0D60DF9D3100350AF8A46F@MAIL.millicorp.com>
	<50D8BE29.7010404@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8A9E3@MAIL.millicorp.com>
	<50D8CC6A.5000706@gmail.com> <50D8CCC3.7090305@gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF8B121@MAIL.millicorp.com>
	<804D48104511D4468F0D60DF9D3100350AF8DCA3@MAIL.millicorp.com>
	<CAHKQagE7MHcTO+AxXtMJPK+KS_P1L3vB+EETXhM1tVy3EvxgbA@mail.gmail.com>
	<804D48104511D4468F0D60DF9D3100350AF9F06B@MAIL.millicorp.com>
	<CAKzWOxXmk_M6ZXs79h5_D+QSjjTuQtC6gq7cYA+TYYw5+4F5xw@mail.gmail.com>
	<CAHKQagGNgNcp2p=bEGSpsz8vghZQxuL+E9ogd0CYYSrE-Uae-w@mail.gmail.com>
Message-ID: <CAHKQagEf1tDO7gsKVukXJFvztoA5YrRrMhE4JFkmQ_Hu2UjK1A@mail.gmail.com>

Sorry, ignore the first sentence about NAT/new book, I got two different
threads confused in my head there for a moment!

Cal

On Thu, Dec 27, 2012 at 8:48 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Anthony has kindly sent across the NAT section of the new book, and the
> feedback so far has been great.
>
> I'm going to start on this tomorrow, will paste the URL once ready -
> contributions/amendments are obviously welcome.
>
> Cal
>
>
> On Thu, Dec 27, 2012 at 5:33 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Gentlemen,
>>
>> Many thanks for stepping up to help get this documented. If anyone needs
>> assistance w/ wiki stuff or has any documentation questions please let me
>> know and I will be glad to assist!
>>
>> Thanks,
>> MC
>>
>>
>> On Thu, Dec 27, 2012 at 8:15 AM, Tim Meade <Tim.Meade at millicorp.com>wrote:
>>
>>>  That?s great Cal!  I was hoping to find the time to add some examples
>>> myself.    Big one is that there is not a section for phrases.   The
>>> example below is working for me.****
>>>
>>> ** **
>>>
>>> Let me know if there is anything I can do to help.  We use xml_curl for
>>> just about everything here.****
>>>
>>> ** **
>>>
>>> Tim****
>>>
>>> ** **
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal
>>> Leeming [Simplicity Media Ltd]
>>> *Sent:* Wednesday, December 26, 2012 3:57 PM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>>>
>>> ** **
>>>
>>> Tim,****
>>>
>>> ** **
>>>
>>> Chiming in on this, as my focus has been purely on mod_xml_curl for the
>>> last 2 months.****
>>>
>>> ** **
>>>
>>> My apologies that you struggled with finding a working examples, the
>>> mod_xml_curl documentation is still a work in progress - and despite that
>>> it has recently had a partial re-write, there are still many parts which
>>> aren't clear enough.****
>>>
>>> ** **
>>>
>>> I will be updating the docs soon (in the next week) to show example
>>> request/response of as many different scenarios as possible, along with
>>> context specific descriptions of each variable.****
>>>
>>> ** **
>>>
>>> Cal****
>>>
>>> ** **
>>>
>>> On Tue, Dec 25, 2012 at 2:07 PM, Tim Meade <Tim.Meade at millicorp.com>
>>> wrote:****
>>>
>>> Here is an example of the xml_curl phrases section.  This is not in the
>>> WIKI?..  I?ll try to update.  Notice the language section?.****
>>>
>>>  ****
>>>
>>> <document type="freeswitch/xml">****
>>>
>>>                   <section name="phrases">****
>>>
>>>                       <macros>****
>>>
>>>                            <include>****
>>>
>>>                                 <language name="en" sound-path="
>>> $${sounds_dir}/en/us/callie" tts-engine="cepstral" tts-voice="callie">**
>>> **
>>>
>>>                                      <macro name="demo_ivr_main_menu"
>>> pause="100"> <!-- See conf/autoload_config/ivr.conf.xml for an example
>>> on how to use this macro in an IVR -->****
>>>
>>>                                                <input pattern="(.*)">***
>>> *
>>>
>>>                                                  <match>****
>>>
>>>                                                    <!-- string together
>>> several existing sound files to create one long greeting -->****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-welcome_to_freeswitch.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-this_ivr_will_let_you_test_features.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-you_may_exit_by_hanging_up.wav"/>****
>>>
>>>                                                    <!-- note that you
>>> can do more than just play files, e.g. have pauses and do TTS -->****
>>>
>>>                                            ****
>>>
>>>                                                    <!-- Menu option 1:
>>> Call FreeSWITCH conference-->****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-enter_ext_pound.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="silence_stream://1500"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-to_call_the_freeswitch_conference.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-please.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="voicemail/vm-press.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="digits/1.wav"/>****
>>>
>>>                                            ****
>>>
>>>                                                    <!-- Menu option 2:
>>> Do FreeSWITCH echo test -->****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-to_do_a_freeswitch_echo_test.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-please.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="voicemail/vm-press.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="digits/2.wav"/>****
>>>
>>>                                            ****
>>>
>>>                                                    <!-- Menu option 3:
>>> Listen to Music on Hold -->****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-to_listen_to_moh.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-please.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="voicemail/vm-press.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="digits/3.wav"/>****
>>>
>>>                                            ****
>>>
>>>                                                    <!-- Menu option 4:
>>> Register for ClueCon -->****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-register_for_cluecon.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="digits/4.wav"/>****
>>>
>>>                                            ****
>>>
>>>                                                    <!-- Menu option 5:
>>> Listen to screaming monkeys -->****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-to_hear_screaming_monkeys.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-please.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="voicemail/vm-press.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="digits/5.wav"/>****
>>>
>>>                                            ****
>>>
>>>                                                    <!-- Menu option 6:
>>> Hear a sample submenu -->****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-to_hear_sample_submenu.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-please.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="voicemail/vm-press.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="digits/6.wav"/>****
>>>
>>>                                            ****
>>>
>>>                                                    <!-- Menu option 9:
>>> Repeat these options -->****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-to_repeat_these_options.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="ivr/ivr-please.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="voicemail/vm-press.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="digits/9.wav"/>****
>>>
>>>                                                    <action function="
>>> play-file" data="silence_stream://2000"/>****
>>>
>>>                                                  </match>****
>>>
>>>                                                </input>****
>>>
>>>                                              </macro>      ****
>>>
>>>                                        </language>
>>>       ****
>>>
>>>                                 </include>****
>>>
>>>                        </macros>
>>>                 ****
>>>
>>>                   </section>****
>>>
>>>                 </document>****
>>>
>>>  ****
>>>
>>>  ****
>>>
>>> Thanks All****
>>>
>>>  ****
>>>
>>> Tim****
>>>
>>>  ****
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tim Meade
>>> *Sent:* Monday, December 24, 2012 5:05 PM
>>> *To:* Abaci; FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>>>
>>>  ****
>>>
>>> Thank you Abaci?.   Just found that myself under the xml_curl section of
>>> the wiki?..  I hadn?t noticed the xml_curl configuration parameters for the
>>> return xml are different.    ****
>>>
>>>  ****
>>>
>>> Here is the completed working configuration from xml_curl debug_on.   **
>>> **
>>>
>>>  ****
>>>
>>> <document type="freeswitch/xml">****
>>>
>>> <section name="configuration">****
>>>
>>> <configuration name="ivr.conf" description="IVR menus">****
>>>
>>> <menus>****
>>>
>>> <include>****
>>>
>>> <!-- demo IVR setup -->****
>>>
>>> <!-- demo IVR, Main Menu -->****
>>>
>>> <menu name="tivr"****
>>>
>>> greet-long="phrase:demo_ivr_main_menu"****
>>>
>>> greet-short="phrase:demo_ivr_main_menu_short"****
>>>
>>> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"****
>>>
>>> exit-sound="voicemail/vm-goodbye.wav"****
>>>
>>> confirm-macro=""****
>>>
>>> confirm-key=""****
>>>
>>> tts-engine="flite"****
>>>
>>> tts-voice="rms"****
>>>
>>> confirm-attempts="3"****
>>>
>>> timeout="10000"****
>>>
>>> inter-digit-timeout="2000"****
>>>
>>> max-failures="3"****
>>>
>>> max-timeouts="3"****
>>>
>>> digit-len="4">****
>>>
>>> <!-- The following are the definitions for the digits the user dials -->
>>> ****
>>>
>>> <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->****
>>>
>>> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/
>>> 888 at conference.freeswitch.org"/>****
>>>
>>> <entry action="menu-exec-app" digits="2" param="transfer 9196 XML
>>> default"/> <!-- FS echo -->****
>>>
>>> <entry action="menu-exec-app" digits="3" param="transfer 9664 XML
>>> default"/> <!-- MOH -->****
>>>
>>> <entry action="menu-exec-app" digits="4" param="transfer 9191 XML
>>> default"/> <!-- ClueCon -->****
>>>
>>> <entry action="menu-exec-app" digits="5" param="transfer 1234*256
>>> enum"/> <!-- Screaming monkeys -->****
>>>
>>> <!-- Using a regex in the digits tag lets you define a dial pattern for
>>> the caller****
>>>
>>> You may define multiple regexes if you need a different pattern for some
>>> reason -->****
>>>
>>> <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer
>>> $1 XML features"/>****
>>>
>>> <entry action="menu-top" digits="9"/> <!-- Repeat this menu -->****
>>>
>>> </menu>****
>>>
>>> </include>****
>>>
>>> </menus>****
>>>
>>> </configuration>****
>>>
>>> </section>****
>>>
>>> </document>****
>>>
>>>  ****
>>>
>>> *From:* Abaci [mailto:abaci64 at gmail.com]
>>> *Sent:* Monday, December 24, 2012 4:45 PM
>>> *To:* FreeSWITCH Users Help
>>> *Cc:* Tim Meade
>>> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>>>
>>>  ****
>>>
>>> you're also missing the "section" element.
>>>
>>> On 12/24/2012 4:43 PM, Abaci wrote:****
>>>
>>>  Here is a sample from my server, I highlighted the missing stuff.
>>>
>>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>>> <document type="freeswitch/xml">
>>>   <section name="configuration">
>>> *    <configuration name="ivr.conf" description="IVR menus">
>>>       <menus>*
>>>         <menu name="ivrmenu1"
>>>               invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>>               exit-sound="voicemail/vm-goodbye.wav"
>>>               timeout ="3000"
>>>               inter-digit-timeout="2000"
>>>               max-failures="2"
>>>               digit-len="3"
>>>               phrase-lang="en">
>>>           <entry action="menu-exec-app" digits="0" param="transfer 11
>>> XML voicemail " />
>>>           <entry action="menu-exec-app" digits="1" param="transfer
>>> ivrmenu2 XML ivr " />
>>>           <entry action="menu-exec-app" digits="2" param="transfer
>>> ivrmenu3 XML ivr " />
>>>         </menu>
>>> *      </menus>
>>>     </configuration>*
>>>   </section>
>>> </document>
>>>
>>> On 12/24/2012 4:22 PM, Tim Meade wrote:****
>>>
>>> The items I pasted below are from the cat of the xml_curl debug_on
>>> command.  It certainly appears to me that freeswitch is getting the correct
>>> name in this case ?tivr? which is what shows in both the dialplan and the
>>> curl?d ivr menu.****
>>>
>>>  ****
>>>
>>> My concerns now are dealing with the xml format for the ivr as I have
>>> found several different versions on the wiki and in the prior postings.*
>>> ***
>>>
>>>  ****
>>>
>>> *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 *Abaci
>>> *Sent:* Monday, December 24, 2012 3:42 PM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] xml_curl and ivr menus****
>>>
>>>  ****
>>>
>>> on freeswitch enable xml_curl debug and see if freeswitch is actually
>>> getting the correct ivr
>>> 'xml_curl debug_on'
>>>
>>> On 12/24/2012 2:20 PM, Tim Meade wrote:****
>>>
>>> I am continually getting ?unable to find menu? while trying to retrieve
>>> an ivr_menu using xml_curl****
>>>
>>>  ****
>>>
>>> Using xml_curl debug_on ****
>>>
>>>  ****
>>>
>>> My DialPlan****
>>>
>>>  ****
>>>
>>> <document type="freeswitch/xml">****
>>>
>>> <section name="dialplan" description="example_curl_dialplan">****
>>>
>>> <context name="public">****
>>>
>>> <extension name="ws_origination">****
>>>
>>> <condition field="destination_number" expression="^\d+$" break="on-true">
>>> ****
>>>
>>> <action inline="true" application="set" data="continue_on_fail=true"/>**
>>> **
>>>
>>> <action inline="true" application="set"
>>> data="hangup_after_bridge=true"/> ****
>>>
>>> <action application="answer"/>****
>>>
>>> <action application="ivr" data="tivr"/>****
>>>
>>> </condition>****
>>>
>>> </extension>****
>>>
>>> </context>****
>>>
>>> </section>****
>>>
>>>  ****
>>>
>>> And the ivr configuration returned by my xml_curl****
>>>
>>>  ****
>>>
>>> <include>****
>>>
>>> <!-- demo IVR setup -->****
>>>
>>> <!-- demo IVR, Main Menu -->****
>>>
>>> <menu name="tivr"****
>>>
>>> greet-long="phrase:demo_ivr_main_menu"****
>>>
>>> greet-short="phrase:demo_ivr_main_menu_short"****
>>>
>>> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"****
>>>
>>> exit-sound="voicemail/vm-goodbye.wav"****
>>>
>>> confirm-macro=""****
>>>
>>> confirm-key=""****
>>>
>>> tts-engine="flite"****
>>>
>>> tts-voice="rms"****
>>>
>>> confirm-attempts="3"****
>>>
>>> timeout="10000"****
>>>
>>> inter-digit-timeout="2000"****
>>>
>>> max-failures="3"****
>>>
>>> max-timeouts="3"****
>>>
>>> digit-len="4">****
>>>
>>> <!-- The following are the definitions for the digits the user dials -->
>>> ****
>>>
>>> <!-- Digit 1 transfer caller to the public FreeSWITCH conference -->****
>>>
>>> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${
>>> domain}/888 at conference.freeswitch.org"/>****
>>>
>>> <entry action="menu-exec-app" digits="2" param="transfer 9196 XML
>>> default"/> <!-- FS echo -->****
>>>
>>> <entry action="menu-exec-app" digits="3" param="transfer 9664 XML
>>> default"/> <!-- MOH -->****
>>>
>>> <entry action="menu-exec-app" digits="4" param="transfer 9191 XML
>>> default"/> <!-- ClueCon -->****
>>>
>>> <entry action="menu-exec-app" digits="5" param="transfer 1234*256
>>> enum"/> <!-- Screaming monkeys -->****
>>>
>>> <!-- Using a regex in the digits tag lets you define a dial pattern for
>>> the caller****
>>>
>>> You may define multiple regexes if you need a different pattern for some
>>> reason -->****
>>>
>>> <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer
>>> $1 XML features"/>****
>>>
>>> <entry action="menu-top" digits="9"/> <!-- Repeat this menu -->****
>>>
>>> </menu>****
>>>
>>> </include>****
>>>
>>>  ****
>>>
>>> Any ideas on the ?unable to find menu? ????****
>>>
>>>  ****
>>>
>>> Thanks****
>>>
>>>  ****
>>>
>>> Tim****
>>>
>>>  ****
>>>
>>>  ****
>>>
>>>  ****
>>>
>>>
>>>
>>> ****
>>>
>>> _________________________________________________________________________****
>>>
>>> Professional FreeSWITCH Consulting 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****
>>>
>>>  ****
>>>
>>> FreeSWITCH-users mailing list****
>>>
>>> FreeSWITCH-users at lists.freeswitch.org****
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users****
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users****
>>>
>>> http://www.freeswitch.org****
>>>
>>>   ****
>>>
>>>  ****
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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 lists at kavun.ch  Fri Dec 28 00:44:39 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 27 Dec 2012 16:44:39 -0500
Subject: [Freeswitch-users] Speex@32000h@20i = no audio
Message-ID: <0BAFF781-476A-4A9F-973D-3C64C4CD45C6@kavun.ch>

Hi all,

I am not getting any audio (in or out) when using Speex at 32000h@20i.
I tried multiple softphones with same results.
A show channels gives me the right codecs.
SPEEX,32000,44000,SPEEX,32000,44000

Using Speex at 16000h@20i works fine.

Any idea?
Is this broken?

E
Version: FreeSWITCH Version 1.2.5.3+git~20121206T171012Z~d74bef3f2f (git d74bef3 2012-12-06 17:10:12Z)

From dchs at abv.bg  Fri Dec 28 00:25:08 2012
From: dchs at abv.bg (dchs dchs)
Date: Thu, 27 Dec 2012 23:25:08 +0200 (EET)
Subject: [Freeswitch-users] CLIR gsmopen
Message-ID: <1488148659.96831.1356643508085.JavaMail.apache@nm22.abv.bg>

Hello, Is there any way to set up CLIR on outgoing calls placed over gsmopen?Do Huawei modems even support such an AT command?  Any pointers towards any info on how to accomplish this would be much appreciated! 
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From lists at kavun.ch  Fri Dec 28 01:11:28 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 27 Dec 2012 17:11:28 -0500
Subject: [Freeswitch-users] FreeSwitch Video Calls Bria iPhone Client
In-Reply-To: <9D3AEDA9-4587-4A01-9A53-966BECD7EA60@mgtech.com>
References: <2B4CD419-FE1B-443C-86AB-2E5A98C63ED2@me.com>
	<603F099C-8BDB-4EB5-9375-6DEF87CF06EE@mgtech.com>
	<CAO47HpeOng3U4TE8TPLY6JcAyTvJYiufYpfUT30qz3OgNqhLNA@mail.gmail.com>
	<7DE0AD3C-2A16-49B6-8533-3F0834D54768@me.com>
	<9D3AEDA9-4587-4A01-9A53-966BECD7EA60@mgtech.com>
Message-ID: <932B8FDE-0134-47F1-99D1-6A4901A761A5@kavun.ch>

Hi there,

I somewhat managed to have it working but am experiencing a few annoying issues.

1. I had to go in my iOS Bria settings -> accounts -> account -> account specific features.
2. I checked always offer video and start sending video automatically.

This seems to work when making calls. There may be issues on incoming calls though.

It's a little more complicated on the Desktop side of Bria.
There is no setting to always offer video during the session setup and it seems like FS does not support the update once the call is established.

Is this a limitation?
Any idea on how to optimize this so that video can be used selectively?

Cheers
Emrah
On Sep 24, 2012, at 12:47 PM, Mario G <mario_fs at mgtech.com> wrote:

> Thanks for the response!  I have tested without ALG months ago to no avail. No problems for over a year, it started when I upgraded FreeSwitch from early 2011 to early 2012 version. However, at the same time added the iPads/Bria and switched the Linksys SPA962s from SLA to individual user IDs. I am suspecting the Linksys SPA962s doing something wonky (and affecting Bria?) which is why I am replacing them. Wanted to go all iPads, especially since Apple may come out with a 7 incher soon. Otherwise looking at SNOM 870s and Polycoms. This is driving me nuts since some days no problem and some days it's bad with no pattern. Lots of traces.
> 
> FreeSwitch is connected to the Zyxel using multiple WANs (static + dynamic) which balance and backup each other. Works great, can pull the plug on either at any time and FreeSwitch goes merrily along and recovers (when I get this problem sorted out I will add a wiki about the setup). Using the router FreeSwitch knows nothing about the IPs, NAT, etc. no messy setup. SIP ALG is always suspected first since it gets such a bad rap but so far no issues here. The problems below don't seem to be related to it. If I ever get this sorted out I will post in case some other poor soul runs into this. Thanks again! Wish I could have helped with your issue.
> Mario G
> 
> On Sep 23, 2012, at 6:59 PM, Mike Burlingame wrote:
> 
>> I agree I am not seeing any issues on call setup / take down or audio RTP or SMS only issues with the video setting up correctly
>> 
>> Have you tried to remove the ALG and reproduced the same issue?
>> 
>> Sent from my iPhone 4S
>> 
>> On Sep 23, 2012, at 5:39 PM, jay binks <jaybinks at gmail.com> wrote:
>> 
>>> You know your zyxel  could be the issue  for lots of that .  Sip aware Nat router = sip ALG = black magic going on that neither end can predict.
>>> 
>>> Remove or disable the alg  and try again .
>>> 
>>> On Sep 24, 2012 4:10 AM, "Mario G" <mario_fs at mgtech.com> wrote:
>>> FWIW, I was trying to replace SPA962s with Bria on iPads in our office. Had several issues, at least 6 open problems with Counterpath, they were great at trying to help but pointed the finger at FreeSwitch. I posted here for months to no avail so I assume no one here using Bria to this extent. FYI, all my issues are ringing/answering related (listed below). If you find anything please post here as I plan t do the same to keep someone else from pulling their hair out.
>>> 
>>> 1. Bria does not always ring the iPad so you can't answer. I am using TCP to avoid battery drain although UDP did not seem to help.
>>> 2. Bria rings, but when you hit answer there is nothing there and the other extensions keep ringing. FS trace shows FS disconnecting.
>>> 3. If Bria is not in the foreground and a ringing alert is displayed, pressing it sometimes results in the last issue of not answering (FS says hangup) but other phones ring.
>>> 4. Not a Bria or FS issue but a pain: Apparently Apple limits the alert to 25 seconds, if you don't press it in that time it closed and you can't answer. This is an IOS issue that Apple needs to allow more flexibility with.
>>> 
>>> Once a call is connected there are no issues with the recent Bria iPad updates, there were some issues a month ago. I highly recommend opening an issue with Counterpath as they seemed to know SIP and understood some of the FS trace stuff. They are very quick to update the Bria if an issue is found. But in my case, after 6+ months of traces we can't find the culprit.
>>> 
>>> I have no Nat issues at all using Zyxel routers with SIP support. Good luck to you, I am writing the president of CounterPath to see if they can get a FreeSwitch to test with since it would be great for both to work together.
>>> Mario G
>>> 
>>> On Sep 22, 2012, at 7:46 PM, Mike Burlingame wrote:
>>> 
>>> > I have been playing around trying to get my iPad and iPhone using Bria to work with video it *seems* that FreeSwitch is passing the signaling to both devices and FreeSwitch is holding the media due to the NAT issue of the devices - I just wanted to know before I spent too much time on this (not a high priority) if anyone else has gotten FreeSwitch to work with Bria for iPhone or Android?
>>> >
>>> > 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
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From errotan at elder.hu  Fri Dec 28 01:20:45 2012
From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=)
Date: Thu, 27 Dec 2012 23:20:45 +0100
Subject: [Freeswitch-users] failover when storing recordings
In-Reply-To: <CALvF6vAUirzxuqxzryzGQ0BrP_+EEQVHy2WO2rbT-hNFDj_rig@mail.gmail.com>
References: <CALvF6vAUirzxuqxzryzGQ0BrP_+EEQVHy2WO2rbT-hNFDj_rig@mail.gmail.com>
Message-ID: <50DCC9BD.3020401@elder.hu>


Would be simpler and more reliable to store the file during the
recording on the local machine (hard or RAM disk) than copy the file
after hangup.


2012-12-27 20:05 keltez?ssel, Jayesh Nambiar ?rta:
> Hi All,
> Is there a way in freeswitch wherein if the call recording is not
> successfully stored, it can store it in some separate path?
> Basically the idea is, I am planning to use NFS for data storage. So
> freeswitch is actually storing the recordings on an NFS mounted
> directory. For some reason, if the NFS mount fails, or if the disc on
> the storage server is full or whatever may be the reason, could there
> be a way that freeswitch can store it on a separate directory which is
> local to the server. Does this sound like a feature or a requirement??
> Thanks.
>
> --- Jayesh
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 msc at freeswitch.org  Fri Dec 28 01:40:59 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Dec 2012 14:40:59 -0800
Subject: [Freeswitch-users] Need community input: hangup_hook recipes
Message-ID: <CAKzWOxWBcGOuQBxkCNMCxqpPPBEF__GwEs4qmjuuJN14BSU7dA@mail.gmail.com>

Hey all,

I was working on a feature for a friend of mine and it turns out that it
might be useful to put on the wiki. I noticed that we don't have a place
explicitly for hangup hook recipes. I was thinking about making a page just
for "interesting things you can do when the call ends" and then linking to
it from the dialplan recipes page.

Two questions: does that sound like a good place to put these recipes? does
anyone have any recipes they'd like to contribute?

Thanks!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From avi at avimarcus.net  Fri Dec 28 02:08:30 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 28 Dec 2012 01:08:30 +0200
Subject: [Freeswitch-users] failover when storing recordings
In-Reply-To: <50DCC9BD.3020401@elder.hu>
References: <CALvF6vAUirzxuqxzryzGQ0BrP_+EEQVHy2WO2rbT-hNFDj_rig@mail.gmail.com>
	<50DCC9BD.3020401@elder.hu>
Message-ID: <CALNZuwkZ3a81EaoHMEDhFSSB5-1cAvpCw3Ec9tqiFu9NUYiGeg@mail.gmail.com>

Are there any particular suggestions for how to trigger that copy script
afterwards, preferably with the path/name of the file that was just written?
Also, the same thing when a file gets deleted. I didn't see an event being
fired in those cases last time I checked.

-Avi

On Fri, Dec 28, 2012 at 12:20 AM, Pusk?s Zsolt <errotan at elder.hu> wrote:

>
> Would be simpler and more reliable to store the file during the recording
> on the local machine (hard or RAM disk) than copy the file after hangup.
>
>
> 2012-12-27 20:05 keltez?ssel, Jayesh Nambiar ?rta:
>
> Hi All,
> Is there a way in freeswitch wherein if the call recording is not
> successfully stored, it can store it in some separate path?
> Basically the idea is, I am planning to use NFS for data storage. So
> freeswitch is actually storing the recordings on an NFS mounted directory..
> For some reason, if the NFS mount fails, or if the disc on the storage
> server is full or whatever may be the reason, could there be a way that
> freeswitch can store it on a separate directory which is local to the
> server. Does this sound like a feature or a requirement?? Thanks.
>
>  --- Jayesh
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dchs at abv.bg  Fri Dec 28 02:22:45 2012
From: dchs at abv.bg (dchs dchs)
Date: Fri, 28 Dec 2012 01:22:45 +0200 (EET)
Subject: [Freeswitch-users] CLIR gsmopen
Message-ID: <13369548.98491.1356650565174.JavaMail.apache@nm22.abv.bg>

To answer my own question-CLIR can be set by AT+CLIR=1.It does work with mobile partner software.However,CLIR is disabled
as soon as gsmopen is loaded. I can't figure out how to configure gsmopen to invoke CLIR.








 -------- ?????????? ????? --------

??: dchs dchs dchs at abv.bg

???????: [Freeswitch-users] CLIR gsmopen

??: FreeSWITCH-users at lists.freeswitch.org

????????? ??: ?????????, 2012, ???????? 27 23:25:08 EET



 
 
   Hello,
   
   Is there any way to set up CLIR on outgoing calls placed over gsmopen?Do Huawei modems even support such an AT command?
   
   
   Any pointers towards any info on how to accomplish this would be much appreciated!
    
 
 
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From schoch+freeswitch.org at xwin32.com  Fri Dec 28 02:50:40 2012
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Thu, 27 Dec 2012 15:50:40 -0800
Subject: [Freeswitch-users] Newbie question about Cisco phones
Message-ID: <CAOkuKdcf3O2d=Q-QMr04me3htgARdib9Ug9Zhr_22_4osXSaZQ@mail.gmail.com>

I have FreeSwitch installed, and now I'm trying to test it.

I'm attempting to use a Cisco UC phone, model CP-3805.
I have it talking to the TFTP server and getting a configuration file that
I got from the FreeSwitch wiki.  It contains the IP address of the switch,
and a couple of login/password lines:

<authName>1001</authName>
<authPassword>654987</authPassword>

I have this same login/password in conf/directory/1001.xml:

  <user id="1001">
    <params>
      <param name="password" value="654987"/>


Using tcpwatch, I see the switch keeps sending "SIP/2.0 401 Unauthorized"
and "SIP/2.0 403 Forbidden" messages to the phone, so I obviously have
something wrong.

I'm thinking I have to put something in conf/sip_profiles, but that's just
a guess at this point.

I'm assuming this is a FAQ, but I found very little in the wiki about
configuring phones.

-- 
Steve
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From msc at freeswitch.org  Fri Dec 28 03:36:41 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Dec 2012 16:36:41 -0800
Subject: [Freeswitch-users] failover when storing recordings
In-Reply-To: <50DCC9BD.3020401@elder.hu>
References: <CALvF6vAUirzxuqxzryzGQ0BrP_+EEQVHy2WO2rbT-hNFDj_rig@mail.gmail.com>
	<50DCC9BD.3020401@elder.hu>
Message-ID: <CAKzWOxVHJQ5PmKSvyx8XWJdO9gxrE_=_4zoVfmgCq=c3-m15Xg@mail.gmail.com>

On Thu, Dec 27, 2012 at 2:20 PM, Pusk?s Zsolt <errotan at elder.hu> wrote:

>
> Would be simpler and more reliable to store the file during the recording
> on the local machine (hard or RAM disk) than copy the file after hangup.
>
+1

You're better off doing a cron job that copies the files to the target
host, confirms they made it, then deletes the files from the local host
(i.e. the FreeSWITCH box). You could even write a daemon that runs
constantly and waits for a lull in system activity and moves the files when
things are quiet. You could get really fancy if you wanted to but I think
most sysadmins around here would say: keep it simple, try a cron job that
runs every 10 minutes or so and uses "nice" or something similar to make
sure the copy process doesn't spike your disk, network, or cpu usage.

-MC
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From msc at freeswitch.org  Fri Dec 28 03:50:05 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Dec 2012 16:50:05 -0800
Subject: [Freeswitch-users] Speex@32000h@20i = no audio
In-Reply-To: <0BAFF781-476A-4A9F-973D-3C64C4CD45C6@kavun.ch>
References: <0BAFF781-476A-4A9F-973D-3C64C4CD45C6@kavun.ch>
Message-ID: <CAKzWOxX6QvxftaUxPn4k=Ym1iWRV8O34xqbVkU7_VxRCPYFRyA@mail.gmail.com>

I was able to use FSClient set for Speex32k with no issues. The only thing
I did was add "speex at 32000@20i" to the beginning of my global_codec_prefs
variable in vars.xml.

The only difference is that I'm on a slightly newer release than you are.
Also, I had only one leg that supports speex32k so the other leg was g722.
Still, the x-coding all worked well. Maybe you could try just a single leg
with speex32k and see what happens.

-MC

On Thu, Dec 27, 2012 at 1:44 PM, Emrah <lists at kavun.ch> wrote:

> Hi all,
>
> I am not getting any audio (in or out) when using Speex at 32000h@20i.
> I tried multiple softphones with same results.
> A show channels gives me the right codecs.
> SPEEX,32000,44000,SPEEX,32000,44000
>
> Using Speex at 16000h@20i works fine.
>
> Any idea?
> Is this broken?
>
> E
> Version: FreeSWITCH Version 1.2.5.3+git~20121206T171012Z~d74bef3f2f (git
> d74bef3 2012-12-06 17:10:12Z)
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From abaci64 at gmail.com  Fri Dec 28 03:50:15 2012
From: abaci64 at gmail.com (Abaci)
Date: Thu, 27 Dec 2012 19:50:15 -0500
Subject: [Freeswitch-users] failover when storing recordings
In-Reply-To: <CAKzWOxVHJQ5PmKSvyx8XWJdO9gxrE_=_4zoVfmgCq=c3-m15Xg@mail.gmail.com>
References: <CALvF6vAUirzxuqxzryzGQ0BrP_+EEQVHy2WO2rbT-hNFDj_rig@mail.gmail.com>
	<50DCC9BD.3020401@elder.hu>
	<CAKzWOxVHJQ5PmKSvyx8XWJdO9gxrE_=_4zoVfmgCq=c3-m15Xg@mail.gmail.com>
Message-ID: <50DCECC7.8090502@gmail.com>

http://wiki.freeswitch.org/wiki/Channel_Variables#record_post_process_exec_api

On 12/27/2012 7:36 PM, Michael Collins wrote:
>
>
> On Thu, Dec 27, 2012 at 2:20 PM, Pusk?s Zsolt <errotan at elder.hu 
> <mailto:errotan at elder.hu>> wrote:
>
>
>     Would be simpler and more reliable to store the file during the
>     recording on the local machine (hard or RAM disk) than copy the
>     file after hangup.
>
> +1
>
> You're better off doing a cron job that copies the files to the target 
> host, confirms they made it, then deletes the files from the local 
> host (i.e. the FreeSWITCH box). You could even write a daemon that 
> runs constantly and waits for a lull in system activity and moves the 
> files when things are quiet. You could get really fancy if you wanted 
> to but I think most sysadmins around here would say: keep it simple, 
> try a cron job that runs every 10 minutes or so and uses "nice" or 
> something similar to make sure the copy process doesn't spike your 
> disk, network, or cpu usage.
>
> -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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 28 03:53:19 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Dec 2012 16:53:19 -0800
Subject: [Freeswitch-users] Newbie question about Cisco phones
In-Reply-To: <CAOkuKdcf3O2d=Q-QMr04me3htgARdib9Ug9Zhr_22_4osXSaZQ@mail.gmail.com>
References: <CAOkuKdcf3O2d=Q-QMr04me3htgARdib9Ug9Zhr_22_4osXSaZQ@mail.gmail.com>
Message-ID: <CAKzWOxU05dUYyG+mM65-r0T=B=5o8Xoy+YLYK+80UgUmtQ+abg@mail.gmail.com>

Can you capture the SIP traffic between the phone and the server and drop
it on pastebin.freeswitch.org? It might be good to review the SIP dialog,
just in case there are some clues there. You can use the fs_cli if you wish:
sofia profile internal siptrace on
(if you're using the example configs that come with FreeSWITCH. Use the
correct profile name if you have one other than "internal" that you're
using.)


On Thu, Dec 27, 2012 at 3:50 PM, Steven Schoch <
schoch+freeswitch.org at xwin32.com> wrote:

> I have FreeSwitch installed, and now I'm trying to test it.
>
> I'm attempting to use a Cisco UC phone, model CP-3805.
> I have it talking to the TFTP server and getting a configuration file that
> I got from the FreeSwitch wiki.  It contains the IP address of the switch,
> and a couple of login/password lines:
>
> <authName>1001</authName>
> <authPassword>654987</authPassword>
>
> I have this same login/password in conf/directory/1001.xml:
>
>   <user id="1001">
>     <params>
>       <param name="password" value="654987"/>
>
>
> Using tcpwatch, I see the switch keeps sending "SIP/2.0 401 Unauthorized"
> and "SIP/2.0 403 Forbidden" messages to the phone, so I obviously have
> something wrong.
>
> I'm thinking I have to put something in conf/sip_profiles, but that's just
> a guess at this point.
>
> I'm assuming this is a FAQ, but I found very little in the wiki about
> configuring phones.
>
> --
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From avi at avimarcus.net  Fri Dec 28 04:08:13 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 28 Dec 2012 03:08:13 +0200
Subject: [Freeswitch-users] failover when storing recordings
In-Reply-To: <50DCECC7.8090502@gmail.com>
References: <CALvF6vAUirzxuqxzryzGQ0BrP_+EEQVHy2WO2rbT-hNFDj_rig@mail.gmail.com>
	<50DCC9BD.3020401@elder.hu>
	<CAKzWOxVHJQ5PmKSvyx8XWJdO9gxrE_=_4zoVfmgCq=c3-m15Xg@mail.gmail.com>
	<50DCECC7.8090502@gmail.com>
Message-ID: <CALNZuwnAn9WUZVGSzL5Sm84yG-jbQLim1hjUjCAFsC8v2-s9KA@mail.gmail.com>

That gets triggered for VM too? How do you pass in the name of the file
that was recorded?

-Avi

On Fri, Dec 28, 2012 at 2:50 AM, Abaci <abaci64 at gmail.com> wrote:

>
> http://wiki.freeswitch.org/wiki/Channel_Variables#record_post_process_exec_api
>
>
> On 12/27/2012 7:36 PM, Michael Collins wrote:
>
>
>
> On Thu, Dec 27, 2012 at 2:20 PM, Pusk?s Zsolt <errotan at elder.hu> wrote:
>
>>
>> Would be simpler and more reliable to store the file during the recording
>> on the local machine (hard or RAM disk) than copy the file after hangup.
>>
> +1
>
> You're better off doing a cron job that copies the files to the target
> host, confirms they made it, then deletes the files from the local host
> (i.e. the FreeSWITCH box). You could even write a daemon that runs
> constantly and waits for a lull in system activity and moves the files when
> things are quiet. You could get really fancy if you wanted to but I think
> most sysadmins around here would say: keep it simple, try a cron job that
> runs every 10 minutes or so and uses "nice" or something similar to make
> sure the copy process doesn't spike your disk, network, or cpu usage.
>
> -MC
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 28 04:23:37 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Dec 2012 17:23:37 -0800
Subject: [Freeswitch-users] failover when storing recordings
In-Reply-To: <CALNZuwnAn9WUZVGSzL5Sm84yG-jbQLim1hjUjCAFsC8v2-s9KA@mail.gmail.com>
References: <CALvF6vAUirzxuqxzryzGQ0BrP_+EEQVHy2WO2rbT-hNFDj_rig@mail.gmail.com>
	<50DCC9BD.3020401@elder.hu>
	<CAKzWOxVHJQ5PmKSvyx8XWJdO9gxrE_=_4zoVfmgCq=c3-m15Xg@mail.gmail.com>
	<50DCECC7.8090502@gmail.com>
	<CALNZuwnAn9WUZVGSzL5Sm84yG-jbQLim1hjUjCAFsC8v2-s9KA@mail.gmail.com>
Message-ID: <CAKzWOxVUPV3uHtnaktZbb=FY6vd=p0VtQV5cTJmYmSFvr6MpbA@mail.gmail.com>

I've never actually used that. I prefer to do all that stuff after the fact
or in a hangup hook.
-MC

On Thu, Dec 27, 2012 at 5:08 PM, Avi Marcus <avi at avimarcus.net> wrote:

> That gets triggered for VM too? How do you pass in the name of the file
> that was recorded?
>
> -Avi
>
> On Fri, Dec 28, 2012 at 2:50 AM, Abaci <abaci64 at gmail.com> wrote:
>
>>
>> http://wiki.freeswitch.org/wiki/Channel_Variables#record_post_process_exec_api
>>
>>
>> On 12/27/2012 7:36 PM, Michael Collins wrote:
>>
>>
>>
>> On Thu, Dec 27, 2012 at 2:20 PM, Pusk?s Zsolt <errotan at elder.hu> wrote:
>>
>>>
>>> Would be simpler and more reliable to store the file during the
>>> recording on the local machine (hard or RAM disk) than copy the file after
>>> hangup.
>>>
>> +1
>>
>> You're better off doing a cron job that copies the files to the target
>> host, confirms they made it, then deletes the files from the local host
>> (i.e. the FreeSWITCH box). You could even write a daemon that runs
>> constantly and waits for a lull in system activity and moves the files when
>> things are quiet. You could get really fancy if you wanted to but I think
>> most sysadmins around here would say: keep it simple, try a cron job that
>> runs every 10 minutes or so and uses "nice" or something similar to make
>> sure the copy process doesn't spike your disk, network, or cpu usage.
>>
>> -MC
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From abaci64 at gmail.com  Fri Dec 28 04:29:10 2012
From: abaci64 at gmail.com (Abaci)
Date: Thu, 27 Dec 2012 20:29:10 -0500
Subject: [Freeswitch-users] failover when storing recordings
In-Reply-To: <CALNZuwnAn9WUZVGSzL5Sm84yG-jbQLim1hjUjCAFsC8v2-s9KA@mail.gmail.com>
References: <CALvF6vAUirzxuqxzryzGQ0BrP_+EEQVHy2WO2rbT-hNFDj_rig@mail.gmail.com>
	<50DCC9BD.3020401@elder.hu>
	<CAKzWOxVHJQ5PmKSvyx8XWJdO9gxrE_=_4zoVfmgCq=c3-m15Xg@mail.gmail.com>
	<50DCECC7.8090502@gmail.com>
	<CALNZuwnAn9WUZVGSzL5Sm84yG-jbQLim1hjUjCAFsC8v2-s9KA@mail.gmail.com>
Message-ID: <50DCF5E6.4050901@gmail.com>

His question was about call recording not voicemail, not sure if that's 
triggered for voicemail (I think it is), you can probably add a var in 
mod_voicemail.c to the file_path and use that var in the post_process 
api/app.

On 12/27/2012 8:08 PM, Avi Marcus wrote:
> That gets triggered for VM too? How do you pass in the name of the 
> file that was recorded?
>
> -Avi
>
> On Fri, Dec 28, 2012 at 2:50 AM, Abaci <abaci64 at gmail.com 
> <mailto:abaci64 at gmail.com>> wrote:
>
>     http://wiki.freeswitch.org/wiki/Channel_Variables#record_post_process_exec_api
>
>
>
>     On 12/27/2012 7:36 PM, Michael Collins wrote:
>>
>>
>>     On Thu, Dec 27, 2012 at 2:20 PM, Pusk?s Zsolt <errotan at elder.hu
>>     <mailto:errotan at elder.hu>> wrote:
>>
>>
>>         Would be simpler and more reliable to store the file during
>>         the recording on the local machine (hard or RAM disk) than
>>         copy the file after hangup.
>>
>>     +1
>>
>>     You're better off doing a cron job that copies the files to the
>>     target host, confirms they made it, then deletes the files from
>>     the local host (i.e. the FreeSWITCH box). You could even write a
>>     daemon that runs constantly and waits for a lull in system
>>     activity and moves the files when things are quiet. You could get
>>     really fancy if you wanted to but I think most sysadmins around
>>     here would say: keep it simple, try a cron job that runs every 10
>>     minutes or so and uses "nice" or something similar to make sure
>>     the copy process doesn't spike your disk, network, or cpu usage.
>>
>>     -MC
>>
>>
>>
>>     _________________________________________________________________________
>>     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/>
>>
>>     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/>
>
>     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
>
> FreeSWITCH-users mailing 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 schoch+freeswitch.org at xwin32.com  Fri Dec 28 04:38:36 2012
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Thu, 27 Dec 2012 17:38:36 -0800
Subject: [Freeswitch-users] Newbie question about Cisco phones
In-Reply-To: <CAKzWOxU05dUYyG+mM65-r0T=B=5o8Xoy+YLYK+80UgUmtQ+abg@mail.gmail.com>
References: <CAOkuKdcf3O2d=Q-QMr04me3htgARdib9Ug9Zhr_22_4osXSaZQ@mail.gmail.com>
	<CAKzWOxU05dUYyG+mM65-r0T=B=5o8Xoy+YLYK+80UgUmtQ+abg@mail.gmail.com>
Message-ID: <CAOkuKdfjMgf-fDiT4zEyCmVu7QZ=FwgTT4Xz6n2Yu7-ON5pVoA@mail.gmail.com>

Here are the important lines from the trace:  (I don't think I need to
paste the whole thing.)

recv 1521 bytes from tcp/[192.168.4.254]:4025 at 01:28:45.696774:
   ------------------------------------------------------------------------
   REGISTER sip:192.168.4.1:5060;transport=tcp SIP/2.0
   Via: SIP/2.0/TCP 192.168.4.254:4025
;rport;branch=z9hG4bKPjZiQxS2mheTBbTXalnSTcX-9-vSvvYydR
   Max-Forwards: 70
   From: "110" <sip:1001 at 192.168.4.1>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl
   To: "110" <sip:1001 at 192.168.4.1>
   Call-ID: 6Y7zlZiZJekTAYupYa4SU.v2.8vpto97
   CSeq: 4019 REGISTER
   User-Agent: Cisco-CP3905/9.2.1
[other stuff...]

   ------------------------------------------------------------------------
send 684 bytes to tcp/[192.168.4.254]:4025 at 01:28:45.856104:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/TCP 192.168.4.254:4025
;rport=4025;branch=z9hG4bKPjZiQxS2mheTBbTXalnSTcX-9-vSvvYydR
   From: "110" <sip:1001 at 192.168.4.1>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl
   To: "110" <sip:1001 at 192.168.4.1>;tag=UrQB9NZ278Hpc
   Call-ID: 6Y7zlZiZJekTAYupYa4SU.v2.8vpto97
[I don't think the other stuff is pertinent.]

And later:
   ------------------------------------------------------------------------
send 562 bytes to tcp/[192.168.4.254]:4025 at 01:28:46.032084:
   ------------------------------------------------------------------------
   SIP/2.0 403 Forbidden
   Via: SIP/2.0/TCP 192.168.4.254:4025
;rport=4025;branch=z9hG4bKPjbdyYmV-sH2X-.Cs-LZUG9P36AHF5dn4E
   From: "110" <sip:1001 at 192.168.4.1>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl
   To: "110" <sip:1001 at 192.168.4.1>;tag=v1g4aHg64H88Q
   Call-ID: 6Y7zlZiZJekTAYupYa4SU.v2.8vpto97
   CSeq: 4020 REGISTER


I have put a name/password in the Cisco config file, and the same
name/password in conf/directory/1001.xml.  Should this go in the
conf/sip_profiles/internal section instead?

-- 
Steve

On Thu, Dec 27, 2012 at 4:53 PM, Michael Collins <msc at freeswitch.org> wrote:

> Can you capture the SIP traffic between the phone and the server and drop
> it on pastebin.freeswitch.org? It might be good to review the SIP dialog,
> just in case there are some clues there. You can use the fs_cli if you wish:
> sofia profile internal siptrace on
> (if you're using the example configs that come with FreeSWITCH. Use the
> correct profile name if you have one other than "internal" that you're
> using.)
>
>
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From red.rain.seven at gmail.com  Fri Dec 28 06:15:41 2012
From: red.rain.seven at gmail.com (Henry Huang)
Date: Thu, 27 Dec 2012 19:15:41 -0800
Subject: [Freeswitch-users] Need community input: hangup_hook recipes
In-Reply-To: <CAKzWOxWBcGOuQBxkCNMCxqpPPBEF__GwEs4qmjuuJN14BSU7dA@mail.gmail.com>
References: <CAKzWOxWBcGOuQBxkCNMCxqpPPBEF__GwEs4qmjuuJN14BSU7dA@mail.gmail.com>
Message-ID: <CAP_G397=XFKma3x8FKjoQR=Gg31Bhf+X33aHUV3xPyMJ6e07hQ@mail.gmail.com>

I think it's a good idea and good place to start.


On Thu, Dec 27, 2012 at 2:40 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hey all,
>
> I was working on a feature for a friend of mine and it turns out that it
> might be useful to put on the wiki. I noticed that we don't have a place
> explicitly for hangup hook recipes. I was thinking about making a page just
> for "interesting things you can do when the call ends" and then linking to
> it from the dialplan recipes page.
>
> Two questions: does that sound like a good place to put these recipes?
> does anyone have any recipes they'd like to contribute?
>
> Thanks!
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 jaybinks at gmail.com  Fri Dec 28 09:08:41 2012
From: jaybinks at gmail.com (jay binks)
Date: Fri, 28 Dec 2012 16:08:41 +1000
Subject: [Freeswitch-users] failover when storing recordings
In-Reply-To: <50DCF5E6.4050901@gmail.com>
References: <CALvF6vAUirzxuqxzryzGQ0BrP_+EEQVHy2WO2rbT-hNFDj_rig@mail.gmail.com>
	<50DCC9BD.3020401@elder.hu>
	<CAKzWOxVHJQ5PmKSvyx8XWJdO9gxrE_=_4zoVfmgCq=c3-m15Xg@mail.gmail.com>
	<50DCECC7.8090502@gmail.com>
	<CALNZuwnAn9WUZVGSzL5Sm84yG-jbQLim1hjUjCAFsC8v2-s9KA@mail.gmail.com>
	<50DCF5E6.4050901@gmail.com>
Message-ID: <CAO47HpcJ0M-6-NdFBpor_arA1WVYrAa2d+7CHQSi_oD8=xYCuw@mail.gmail.com>

maybe look at inotifywait (
http://stackoverflow.com/questions/10533200/processing-data-with-inotify-tools-as-a-daemon)
or http://code.google.com/p/lsyncd/

or use inotify in your own daemon if you want.


On 28 December 2012 11:29, Abaci <abaci64 at gmail.com> wrote:

>  His question was about call recording not voicemail, not sure if that's
> triggered for voicemail (I think it is), you can probably add a var in
> mod_voicemail.c to the file_path and use that var in the post_process
> api/app.
>
>
> On 12/27/2012 8:08 PM, Avi Marcus wrote:
>
> That gets triggered for VM too? How do you pass in the name of the file
> that was recorded?
>
>  -Avi
>
> On Fri, Dec 28, 2012 at 2:50 AM, Abaci <abaci64 at gmail.com> wrote:
>
>>
>> http://wiki.freeswitch.org/wiki/Channel_Variables#record_post_process_exec_api
>>
>>
>> On 12/27/2012 7:36 PM, Michael Collins wrote:
>>
>>
>>
>> On Thu, Dec 27, 2012 at 2:20 PM, Pusk?s Zsolt <errotan at elder.hu> wrote:
>>
>>>
>>> Would be simpler and more reliable to store the file during the
>>> recording on the local machine (hard or RAM disk) than copy the file after
>>> hangup.
>>>
>> +1
>>
>> You're better off doing a cron job that copies the files to the target
>> host, confirms they made it, then deletes the files from the local host
>> (i.e. the FreeSWITCH box). You could even write a daemon that runs
>> constantly and waits for a lull in system activity and moves the files when
>> things are quiet. You could get really fancy if you wanted to but I think
>> most sysadmins around here would say: keep it simple, try a cron job that
>> runs every 10 minutes or so and uses "nice" or something similar to make
>> sure the copy process doesn't spike your disk, network, or cpu usage.
>>
>> -MC
>>
>>
>>
>>   _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/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
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 erik.dekkers at certhon.com  Fri Dec 28 10:24:27 2012
From: erik.dekkers at certhon.com (Erik Dekkers)
Date: Fri, 28 Dec 2012 08:24:27 +0100
Subject: [Freeswitch-users] mod limit on inbound calls
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE5158FDF63E@FS1.wvds.local>

Hi List,

I do have this setup:
User has one deskphone and one portable (IP DECT) phone. Both phones have the same extention number and multiple registrations is set in the sip config

This is the situation:
If a user is on one of the phones and a second call comes in both phones will ring.

This is the wanted situation:
If a user is on one of the phones and a second call comes in a USER BUSY signal has to be send back


I know you can use limit for this, but im not sure how to configure it on incoming calls when a user has 2 phones registered on the same extention number.

Thank you

Regards,

Erik Dekkers
(wvds-nl on irc)


[cid:image7432f4.PNG at de4c91c0.4b889d03]
ABC Westland 555        Tel:    +31 174 22 50 80
P.O. Box 90     Fax:    +31 174 22 50 81
        Mob:    +31 624 423 009
2685 ZH Poeldijk        erik.dekkers at certhon.com<mailto:erik.dekkers at certhon.com>
The Netherlands www.certhon.com<http://www.certhon.com>

DISCLAIMER
All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS.
Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing.

ISO9001 CERTIFIED

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From avi at avimarcus.net  Fri Dec 28 10:34:49 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 28 Dec 2012 09:34:49 +0200
Subject: [Freeswitch-users] mod limit on inbound calls
In-Reply-To: <CDF5E00590C9BA4AB96BE7CDD28C86AE5158FDF63E@FS1.wvds.local>
References: <CDF5E00590C9BA4AB96BE7CDD28C86AE5158FDF63E@FS1.wvds.local>
Message-ID: <CALNZuw=RHw1=KsYTvts92Uwo9VrTYEYUi4zpp6y2RozSpRhyvQ@mail.gmail.com>

I don't think how many phones they have registered is an issue-- it sounds
like you want at most 1 inbound (and outbound?) call to each user.
So set the limit for that user with max 1 for the inbound route (and on
outbound count it, too), and if that fails, send user_busy.

-Avi

On Fri, Dec 28, 2012 at 9:24 AM, Erik Dekkers <erik.dekkers at certhon.com>wrote:

> Hi List,****
>
> ** **
>
> I do have this setup:****
>
> User has one deskphone and one portable (IP DECT) phone. Both phones have
> the same extention number and multiple registrations is set in the sip
> config****
>
> ** **
>
> This is the situation:****
>
> If a user is on one of the phones and a second call comes in both phones
> will ring.****
>
> ** **
>
> This is the wanted situation:****
>
> If a user is on one of the phones and a second call comes in a USER BUSY
> signal has to be send back****
>
> ** **
>
> ** **
>
> I know you can use limit for this, but im not sure how to configure it on
> incoming calls when a user has 2 phones registered on the same extention
> number.****
>
> ** **
>
> Thank you****
>
> ** **
>
> Regards,****
>
> ** **
>
> Erik Dekkers****
>
> (wvds-nl on irc)****
>
> ** **
>
>
>   ABC Westland 555 Tel: +31 174 22 50 80 P.O. Box 90 Fax: +31 174 22 50 81
> Mob: +31 624 423 009 2685 ZH Poeldijk erik.dekkers at certhon.com The
> Netherlands www.certhon.com
> DISCLAIMER
> All our quotations, all orders and all contracts are subject to the
> AVAG-CONDITIONS.
> Op alle offertes, opdrachten en overeenkomsten zijn de
> AVAG-verkoopvoorwaarden van toepassing.
>
> ISO9001 CERTIFIED
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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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>
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From bryan at bsdjournal.net  Fri Dec 28 10:36:39 2012
From: bryan at bsdjournal.net (Bryan Vyhmeister)
Date: Thu, 27 Dec 2012 23:36:39 -0800
Subject: [Freeswitch-users] Conference Multi Channel Recording
Message-ID: <960C7CA2-884C-40E0-BBF6-DC59C74911ED@bsdjournal.net>

I'm toying with the idea of having two or more callers in a conference and having the conference session recorded with a separate channel for each caller. This would likely only potentially work with wav and similar formats. My goal is to use SILK or another wideband codec and have a great multi channel recording. I expect this isn't currently possible. Is that true?

I found RECORD_STEREO but I think that only applies to recording a regular call with uuid_record and not to a conference? Perhaps I could record each incoming leg to the conference and end up with one channel as that caller and the second channel as everyone else in the conference?

Any ideas? Thank you. 

Bryan

---
Bryan Vyhmeister
Sent from my iPhone

From lists at kavun.ch  Fri Dec 28 10:45:18 2012
From: lists at kavun.ch (Emrah)
Date: Fri, 28 Dec 2012 02:45:18 -0500
Subject: [Freeswitch-users] Speex@32000h@20i = no audio
In-Reply-To: <CAKzWOxX6QvxftaUxPn4k=Ym1iWRV8O34xqbVkU7_VxRCPYFRyA@mail.gmail.com>
References: <0BAFF781-476A-4A9F-973D-3C64C4CD45C6@kavun.ch>
	<CAKzWOxX6QvxftaUxPn4k=Ym1iWRV8O34xqbVkU7_VxRCPYFRyA@mail.gmail.com>
Message-ID: <72E46BC3-5102-43C1-948C-A9BE9E773FD5@kavun.ch>

Hey Michael,

Thanks for this.
I actually tried calling some FS extensions and there were all one legged calls.
I'm all set in my codec prefs and Speex was enabled on my clients.

I tried both Telephone (www.tlphn.com) and Bria on Mac OS X.

I'm trying a make current now. Please share any further suggestions you may have.

E
On Dec 27, 2012, at 7:50 PM, Michael Collins <msc at freeswitch.org> wrote:

> I was able to use FSClient set for Speex32k with no issues. The only thing I did was add "speex at 32000@20i" to the beginning of my global_codec_prefs variable in vars.xml.
> 
> The only difference is that I'm on a slightly newer release than you are. Also, I had only one leg that supports speex32k so the other leg was g722. Still, the x-coding all worked well. Maybe you could try just a single leg with speex32k and see what happens.
> 
> -MC
> 
> On Thu, Dec 27, 2012 at 1:44 PM, Emrah <lists at kavun.ch> wrote:
> Hi all,
> 
> I am not getting any audio (in or out) when using Speex at 32000h@20i.
> I tried multiple softphones with same results.
> A show channels gives me the right codecs.
> SPEEX,32000,44000,SPEEX,32000,44000
> 
> Using Speex at 16000h@20i works fine.
> 
> Any idea?
> Is this broken?
> 
> E
> Version: FreeSWITCH Version 1.2.5.3+git~20121206T171012Z~d74bef3f2f (git d74bef3 2012-12-06 17:10:12Z)
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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



From avi at avimarcus.net  Fri Dec 28 10:57:24 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 28 Dec 2012 09:57:24 +0200
Subject: [Freeswitch-users] Voicemail in a Cluster
Message-ID: <CALNZuwkzkrO4GQ3BD9jm8npQ=u3jhd2s0vfK_ymvvz5tT=j39w@mail.gmail.com>

Instead of further hijacking a specific question about call recording, I'll
start a new thread...

It's common to have more than one FS media server to handle calls. So then,
you'll have them handling voicemail too.
How do you ensure that the voicemail will be available for the user? Surely
this is a common thing!

Some options present themselves:
1) Ensure that each user always connects to the same server (this make
maintenance impossible if the VM is only on one machine!)
2) Mount the VM directory to point to Amazon S3 (this doesn't sound ideal.
Perhaps for long-term archival -- if you allow such a thing -- and
seemingly the network connection could be a problem)
3) Some sort of NFS - but then this become a single point of failure?
4) Some sort of network drive that has multiple copies of the file, and it
could even be local -- suggestions? Gluster? I've never used any. This
would be transparent to FS.
5) couchdb, riak, (rethinkdb)  (effectively) have  multi-master
replication/multiple sharding, but FS doesn't know how to access that. (In
couchdb it's not a shard, it would be a full copy. The others configure a
multiple of copies that are available)
5) Abaci suggested possibly record_post_process_exec_api or if it's not in
VM that you can probably add a var in mod_voicemail.c to the file_path and
use that var in the post_process api/app. Then, you trigger whatever "Copy"
script to the NFS.
6) Jay Binks suggest http://code.google.com/p/lsyncd/ or using inotify
directly, but it sounds like you need to establish a watch per folder, and
simply - it's a hack. This should be something that can be triggered (if
not fully handled) in FS itself.


What are people currently doing to "share" voicemail? Can we work out some
sort of recipe to easily implement #4 (or #5)?


-Avi Marcus
BestFone
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From lists at kavun.ch  Fri Dec 28 11:11:52 2012
From: lists at kavun.ch (Emrah)
Date: Fri, 28 Dec 2012 03:11:52 -0500
Subject: [Freeswitch-users] Speex@32000h@20i = no audio
In-Reply-To: <72E46BC3-5102-43C1-948C-A9BE9E773FD5@kavun.ch>
References: <0BAFF781-476A-4A9F-973D-3C64C4CD45C6@kavun.ch>
	<CAKzWOxX6QvxftaUxPn4k=Ym1iWRV8O34xqbVkU7_VxRCPYFRyA@mail.gmail.com>
	<72E46BC3-5102-43C1-948C-A9BE9E773FD5@kavun.ch>
Message-ID: <63D2FD30-1A59-4B15-B76B-2439E7AC6B45@kavun.ch>

Well, my bad? It looks like Bria doesn't offer Speex at 32000hz, see below?

2012-12-28 02:57:42.478165 [INFO] switch_core_session.c:2526 Sending early media
2012-12-28 02:57:42.478165 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [SPEEX:100:16000:20:0]/[G7221:115:32000:20:48000]
2012-12-28 02:57:42.478165 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [G722:9:8000:20:64000]/[G7221:115:32000:20:48000]
2012-12-28 02:57:42.478165 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [DVI4:6:16000:20:0]/[G7221:115:32000:20:48000]
2012-12-28 02:57:42.478165 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000]
2012-12-28 02:57:42.478165 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000]
2012-12-28 02:57:42.478165 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000]
2012-12-28 02:57:42.478165 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
2012-12-28 02:57:42.478165 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf send/recv payload to 101
2012-12-28 02:57:42.478165 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [SPEEX:100:16000:20:0]/[SPEEX:99:32000:20:44000]
2012-12-28 02:57:42.478165 [DEBUG] sofia_glue.c:5156 Bah HUMBUG! Sticking with SPEEX at 32000h@20i
2012-12-28 02:57:42.478165 [DEBUG] sofia_glue.c:3093 Set Codec sofia/internal/10000 at privatepbx.domain.com SPEEX/32000 20 ms 640 samples 44000 bits

Do you know a softphone on Mac OS that supports Speex at 32000h?

Cheers
On Dec 28, 2012, at 2:45 AM, Emrah <lists at kavun.ch> wrote:

> Hey Michael,
> 
> Thanks for this.
> I actually tried calling some FS extensions and there were all one legged calls.
> I'm all set in my codec prefs and Speex was enabled on my clients.
> 
> I tried both Telephone (www.tlphn.com) and Bria on Mac OS X.
> 
> I'm trying a make current now. Please share any further suggestions you may have.
> 
> E
> On Dec 27, 2012, at 7:50 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
>> I was able to use FSClient set for Speex32k with no issues. The only thing I did was add "speex at 32000@20i" to the beginning of my global_codec_prefs variable in vars.xml.
>> 
>> The only difference is that I'm on a slightly newer release than you are. Also, I had only one leg that supports speex32k so the other leg was g722. Still, the x-coding all worked well. Maybe you could try just a single leg with speex32k and see what happens.
>> 
>> -MC
>> 
>> On Thu, Dec 27, 2012 at 1:44 PM, Emrah <lists at kavun.ch> wrote:
>> Hi all,
>> 
>> I am not getting any audio (in or out) when using Speex at 32000h@20i.
>> I tried multiple softphones with same results.
>> A show channels gives me the right codecs.
>> SPEEX,32000,44000,SPEEX,32000,44000
>> 
>> Using Speex at 16000h@20i works fine.
>> 
>> Any idea?
>> Is this broken?
>> 
>> E
>> Version: FreeSWITCH Version 1.2.5.3+git~20121206T171012Z~d74bef3f2f (git d74bef3 2012-12-06 17:10:12Z)
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>> 
>> 
>> 
>> -- 
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
> 



From gabe at gundy.org  Fri Dec 28 11:13:41 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 28 Dec 2012 01:13:41 -0700
Subject: [Freeswitch-users] Voicemail in a Cluster
In-Reply-To: <CALNZuwkzkrO4GQ3BD9jm8npQ=u3jhd2s0vfK_ymvvz5tT=j39w@mail.gmail.com>
References: <CALNZuwkzkrO4GQ3BD9jm8npQ=u3jhd2s0vfK_ymvvz5tT=j39w@mail.gmail.com>
Message-ID: <CAMwZ-tJO7uVmLiCDwU6TR44jL+KMaz26yG0U-2KgU8iJv=9MvQ@mail.gmail.com>

On Fri, Dec 28, 2012 at 12:57 AM, Avi Marcus <avi at avimarcus.net> wrote:
> 3) Some sort of NFS - but then this become a single point of failure?

ODBC and NFS is a great way to go. Yes, they are each single points of
failure, but each can be made redundant.

That's where I would start before spending too much time tooling around.


Best,
Gabe


From avi at avimarcus.net  Fri Dec 28 11:26:41 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 28 Dec 2012 10:26:41 +0200
Subject: [Freeswitch-users] Voicemail in a Cluster
In-Reply-To: <CAMwZ-tJO7uVmLiCDwU6TR44jL+KMaz26yG0U-2KgU8iJv=9MvQ@mail.gmail.com>
References: <CALNZuwkzkrO4GQ3BD9jm8npQ=u3jhd2s0vfK_ymvvz5tT=j39w@mail.gmail.com>
	<CAMwZ-tJO7uVmLiCDwU6TR44jL+KMaz26yG0U-2KgU8iJv=9MvQ@mail.gmail.com>
Message-ID: <CALNZuwkhvyi0rW4CGaewFE8x-vn6cSTKjfm8NXtEWdshxO9CmQ@mail.gmail.com>

Ah, I missed it that part... the DB listing of the VM files needs to be
shared across the machines, too.
-Avi

On Fri, Dec 28, 2012 at 10:13 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Fri, Dec 28, 2012 at 12:57 AM, Avi Marcus <avi at avimarcus.net> wrote:
> > 3) Some sort of NFS - but then this become a single point of failure?
>
> ODBC and NFS is a great way to go. Yes, they are each single points of
> failure, but each can be made redundant.
>
> That's where I would start before spending too much time tooling around.
>
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Dec 28 13:13:59 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 28 Dec 2012 10:13:59 +0000
Subject: [Freeswitch-users] Newbie question about Cisco phones
In-Reply-To: <CAOkuKdfjMgf-fDiT4zEyCmVu7QZ=FwgTT4Xz6n2Yu7-ON5pVoA@mail.gmail.com>
References: <CAOkuKdcf3O2d=Q-QMr04me3htgARdib9Ug9Zhr_22_4osXSaZQ@mail.gmail.com>
	<CAKzWOxU05dUYyG+mM65-r0T=B=5o8Xoy+YLYK+80UgUmtQ+abg@mail.gmail.com>
	<CAOkuKdfjMgf-fDiT4zEyCmVu7QZ=FwgTT4Xz6n2Yu7-ON5pVoA@mail.gmail.com>
Message-ID: <3C11F92D-11A2-4514-B1AD-27F04EE5E286@gmail.com>

This is incomplete. The normal flow is
-> REGISTER
<- 401
-> REGISTER
<- 200 or 403

If you look at the CSeq in the 403 you'll see its a reply to the register after the initial REGISTER.

401 contains a challenge that's used to generate the auth data to put in the 2nd register - that allows digest authentication which avoids sending the password in plaintext and uses a nonce to prevent replay attacks (an attacker can't capture the register and resend it later to auth themselves).

The digest includes the domain, if the user and password match perhaps the problem lies there...

Sent from my iPad



On 28 Dec 2012, at 01:38, Steven Schoch <schoch+freeswitch.org at xwin32.com> wrote:

> Here are the important lines from the trace:  (I don't think I need to paste the whole thing.)
> 
> recv 1521 bytes from tcp/[192.168.4.254]:4025 at 01:28:45.696774:
>    ------------------------------------------------------------------------
>    REGISTER sip:192.168.4.1:5060;transport=tcp SIP/2.0
>    Via: SIP/2.0/TCP 192.168.4.254:4025;rport;branch=z9hG4bKPjZiQxS2mheTBbTXalnSTcX-9-vSvvYydR
>    Max-Forwards: 70
>    From: "110" <sip:1001 at 192.168.4.1>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl
>    To: "110" <sip:1001 at 192.168.4.1>
>    Call-ID: 6Y7zlZiZJekTAYupYa4SU.v2.8vpto97
>    CSeq: 4019 REGISTER
>    User-Agent: Cisco-CP3905/9.2.1
> [other stuff...]
> 
>    ------------------------------------------------------------------------
> send 684 bytes to tcp/[192.168.4.254]:4025 at 01:28:45.856104:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/TCP 192.168.4.254:4025;rport=4025;branch=z9hG4bKPjZiQxS2mheTBbTXalnSTcX-9-vSvvYydR
>    From: "110" <sip:1001 at 192.168.4.1>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl
>    To: "110" <sip:1001 at 192.168.4.1>;tag=UrQB9NZ278Hpc
>    Call-ID: 6Y7zlZiZJekTAYupYa4SU.v2.8vpto97
> [I don't think the other stuff is pertinent.]
> 
> And later:
>    ------------------------------------------------------------------------
> send 562 bytes to tcp/[192.168.4.254]:4025 at 01:28:46.032084:
>    ------------------------------------------------------------------------
>    SIP/2.0 403 Forbidden
>    Via: SIP/2.0/TCP 192.168.4.254:4025;rport=4025;branch=z9hG4bKPjbdyYmV-sH2X-.Cs-LZUG9P36AHF5dn4E
>    From: "110" <sip:1001 at 192.168.4.1>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl
>    To: "110" <sip:1001 at 192.168.4.1>;tag=v1g4aHg64H88Q
>    Call-ID: 6Y7zlZiZJekTAYupYa4SU.v2.8vpto97
>    CSeq: 4020 REGISTER
> 
> 
> I have put a name/password in the Cisco config file, and the same name/password in conf/directory/1001.xml.  Should this go in the conf/sip_profiles/internal section instead?
> 
> -- 
> Steve
> 
> On Thu, Dec 27, 2012 at 4:53 PM, Michael Collins <msc at freeswitch.org> wrote:
>> Can you capture the SIP traffic between the phone and the server and drop it on pastebin.freeswitch.org? It might be good to review the SIP dialog, just in case there are some clues there. You can use the fs_cli if you wish:
>> sofia profile internal siptrace on
>> (if you're using the example configs that come with FreeSWITCH. Use the correct profile name if you have one other than "internal" that you're using.)
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 bhadrarao.kankatala at gmail.com  Fri Dec 28 07:20:18 2012
From: bhadrarao.kankatala at gmail.com (veerabhadrarao`)
Date: Fri, 28 Dec 2012 09:50:18 +0530
Subject: [Freeswitch-users] regarding attendent transfer
In-Reply-To: <CAKzWOxUiRAWA1z=yZLM-mWELRYG7XpPVZcYR-XaQeEyFiA40OA@mail.gmail.com>
References: <50DC447C.1080703@gmail.com>
	<CAKzWOxUiRAWA1z=yZLM-mWELRYG7XpPVZcYR-XaQeEyFiA40OA@mail.gmail.com>
Message-ID: <50DD1E02.7090503@gmail.com>

On 12/27/2012 11:24 PM, Michael Collins wrote:
> Are you using the *4 + extension number to do an attended transfer? 
> Also, keep in mind that the *4 attended transfer will *only* transfer 
> to a registered user on the system, that is, it transfers to 
> "user/xxxx" where xxxx is the destination number that you dial after 
> pressing *4.
>
> -MC
>
> On Thu, Dec 27, 2012 at 4:52 AM, veerabhadrarao` 
> <bhadrarao.kankatala at gmail.com <mailto:bhadrarao.kankatala at gmail.com>> 
> wrote:
>
>     hai
>
>     Can anyone plaese tell me how to make attendent transfer. I am able to
>     working on Blind transfer using default.xml and features.xml but not
>     attendenet transfer successfully. I am using Freeswitch 1.2.4 version.
>
>     thank you
>
>     _________________________________________________________________________
>     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
>
>
>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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

Hai thank you for your Response

     I am dialling *4 and Extension then the call is Establishing From 
B-->C after that how can i connect A--->C.
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From bhadrarao.kankatala at gmail.com  Fri Dec 28 11:52:23 2012
From: bhadrarao.kankatala at gmail.com (veerabhadrarao`)
Date: Fri, 28 Dec 2012 14:22:23 +0530
Subject: [Freeswitch-users] Regarding Attendent transfer
Message-ID: <50DD5DC7.8050400@gmail.com>

hai

     I am facing problem while making  attendent transfer.
first i establish call from A-->B and then i dial *4 + extension then 
call is establishig from B-->C then after that what i have to do to 
connect A-->C.
I am usind default.xml file.

please help me

Thanks


From erik.dekkers at certhon.com  Fri Dec 28 13:52:09 2012
From: erik.dekkers at certhon.com (Erik Dekkers)
Date: Fri, 28 Dec 2012 11:52:09 +0100
Subject: [Freeswitch-users] Newbie question about Cisco phones
In-Reply-To: <3C11F92D-11A2-4514-B1AD-27F04EE5E286@gmail.com>
References: <CAOkuKdcf3O2d=Q-QMr04me3htgARdib9Ug9Zhr_22_4osXSaZQ@mail.gmail.com>
	<CAKzWOxU05dUYyG+mM65-r0T=B=5o8Xoy+YLYK+80UgUmtQ+abg@mail.gmail.com>
	<CAOkuKdfjMgf-fDiT4zEyCmVu7QZ=FwgTT4Xz6n2Yu7-ON5pVoA@mail.gmail.com>
	<3C11F92D-11A2-4514-B1AD-27F04EE5E286@gmail.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE5158FDF641@FS1.wvds.local>

Hi,

Seeing the firmware version in the User Agent header this phone is probably build on the same platform as the 79XX series phones and having the same sip-stack.
From experience I can tell you that sip-stack is really really horrible. I don?t know if you have bought a lot of these phone or just testing one.

If you?re just testing one you?re better of with the Cisco SPA500 series phones instead of these ones. Especially if you?re new to ip telephony, these phones will give you bad headaches.

Erik



[cid:imagedf2f15.PNG at 6f1c3921.4486eef7]
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Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Steven Ayre
Verzonden: vrijdag 28 december 2012 11:14
Aan: FreeSWITCH Users Help
Onderwerp: Re: [Freeswitch-users] Newbie question about Cisco phones

This is incomplete. The normal flow is
-> REGISTER
<- 401
-> REGISTER
<- 200 or 403

If you look at the CSeq in the 403 you'll see its a reply to the register after the initial REGISTER.

401 contains a challenge that's used to generate the auth data to put in the 2nd register - that allows digest authentication which avoids sending the password in plaintext and uses a nonce to prevent replay attacks (an attacker can't capture the register and resend it later to auth themselves).

The digest includes the domain, if the user and password match perhaps the problem lies there...

Sent from my iPad



On 28 Dec 2012, at 01:38, Steven Schoch <schoch+freeswitch.org at xwin32.com<mailto:schoch+freeswitch.org at xwin32.com>> wrote:
Here are the important lines from the trace:  (I don't think I need to paste the whole thing.)

recv 1521 bytes from tcp/[192.168.4.254]:4025 at 01:28:45.696774:
   ------------------------------------------------------------------------
   REGISTER sip:192.168.4.1:5060;transport=tcp SIP/2.0
   Via: SIP/2.0/TCP 192.168.4.254:4025;rport;branch=z9hG4bKPjZiQxS2mheTBbTXalnSTcX-9-vSvvYydR
   Max-Forwards: 70
   From: "110" <sip:1001 at 192.168.4.1<mailto:sip%3A1001 at 192.168.4.1>>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl
   To: "110" <sip:1001 at 192.168.4.1<mailto:sip%3A1001 at 192.168.4.1>>
   Call-ID: 6Y7zlZiZJekTAYupYa4SU.v2.8vpto97
   CSeq: 4019 REGISTER
   User-Agent: Cisco-CP3905/9.2.1
[other stuff...]

   ------------------------------------------------------------------------
send 684 bytes to tcp/[192.168.4.254]:4025 at 01:28:45.856104:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/TCP 192.168.4.254:4025;rport=4025;branch=z9hG4bKPjZiQxS2mheTBbTXalnSTcX-9-vSvvYydR
   From: "110" <sip:1001 at 192.168.4.1<mailto:sip%3A1001 at 192.168.4.1>>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl
   To: "110" <sip:1001 at 192.168.4.1<mailto:sip%3A1001 at 192.168.4.1>>;tag=UrQB9NZ278Hpc
   Call-ID: 6Y7zlZiZJekTAYupYa4SU.v2.8vpto97
[I don't think the other stuff is pertinent.]

And later:
   ------------------------------------------------------------------------
send 562 bytes to tcp/[192.168.4.254]:4025 at 01:28:46.032084:
   ------------------------------------------------------------------------
   SIP/2.0 403 Forbidden
   Via: SIP/2.0/TCP 192.168.4.254:4025;rport=4025;branch=z9hG4bKPjbdyYmV-sH2X-.Cs-LZUG9P36AHF5dn4E
   From: "110" <sip:1001 at 192.168.4.1<mailto:sip%3A1001 at 192.168.4.1>>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl
   To: "110" <sip:1001 at 192.168.4.1<mailto:sip%3A1001 at 192.168.4.1>>;tag=v1g4aHg64H88Q
   Call-ID: 6Y7zlZiZJekTAYupYa4SU.v2.8vpto97
   CSeq: 4020 REGISTER


I have put a name/password in the Cisco config file, and the same name/password in conf/directory/1001.xml.  Should this go in the conf/sip_profiles/internal section instead?

--
Steve

On Thu, Dec 27, 2012 at 4:53 PM, Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>> wrote:
Can you capture the SIP traffic between the phone and the server and drop it on pastebin.freeswitch.org<http://pastebin.freeswitch.org>? It might be good to review the SIP dialog, just in case there are some clues there. You can use the fs_cli if you wish:
sofia profile internal siptrace on
(if you're using the example configs that come with FreeSWITCH. Use the correct profile name if you have one other than "internal" that you're using.)
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
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http://wiki.freeswitch.org
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From steveayre at gmail.com  Fri Dec 28 17:20:35 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 28 Dec 2012 14:20:35 +0000
Subject: [Freeswitch-users] Voicemail in a Cluster
In-Reply-To: <CAMwZ-tJO7uVmLiCDwU6TR44jL+KMaz26yG0U-2KgU8iJv=9MvQ@mail.gmail.com>
References: <CALNZuwkzkrO4GQ3BD9jm8npQ=u3jhd2s0vfK_ymvvz5tT=j39w@mail.gmail.com>
	<CAMwZ-tJO7uVmLiCDwU6TR44jL+KMaz26yG0U-2KgU8iJv=9MvQ@mail.gmail.com>
Message-ID: <CAFiqYumfM=UXbKPn1_=wpx5Pdj+QAevXX95QdZeBN1A-kdYKrA@mail.gmail.com>

ODBC isn't necessarily a single point of failure - you can connect to a
MySQL Cluster database or a multi-master circular replicated database for
example.


On 28 December 2012 08:13, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Fri, Dec 28, 2012 at 12:57 AM, Avi Marcus <avi at avimarcus.net> wrote:
> > 3) Some sort of NFS - but then this become a single point of failure?
>
> ODBC and NFS is a great way to go. Yes, they are each single points of
> failure, but each can be made redundant.
>
> That's where I would start before spending too much time tooling around.
>
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Dec 28 17:21:41 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 28 Dec 2012 14:21:41 +0000
Subject: [Freeswitch-users] Voicemail in a Cluster
In-Reply-To: <CAFiqYumfM=UXbKPn1_=wpx5Pdj+QAevXX95QdZeBN1A-kdYKrA@mail.gmail.com>
References: <CALNZuwkzkrO4GQ3BD9jm8npQ=u3jhd2s0vfK_ymvvz5tT=j39w@mail.gmail.com>
	<CAMwZ-tJO7uVmLiCDwU6TR44jL+KMaz26yG0U-2KgU8iJv=9MvQ@mail.gmail.com>
	<CAFiqYumfM=UXbKPn1_=wpx5Pdj+QAevXX95QdZeBN1A-kdYKrA@mail.gmail.com>
Message-ID: <CAFiqYunqy-abJ-y+4-VOKGV_=O8oe4mBX_7kc01wTekPvSx3mQ@mail.gmail.com>

As for NFS... using a cluster filesystem on top of DRBD is one way to avoid
the filesystem becoming a SPOF.



On 28 December 2012 14:20, Steven Ayre <steveayre at gmail.com> wrote:

> ODBC isn't necessarily a single point of failure - you can connect to a
> MySQL Cluster database or a multi-master circular replicated database for
> example.
>
>
> On 28 December 2012 08:13, Gabriel Gunderson <gabe at gundy.org> wrote:
>
>> On Fri, Dec 28, 2012 at 12:57 AM, Avi Marcus <avi at avimarcus.net> wrote:
>> > 3) Some sort of NFS - but then this become a single point of failure?
>>
>> ODBC and NFS is a great way to go. Yes, they are each single points of
>> failure, but each can be made redundant.
>>
>> That's where I would start before spending too much time tooling around.
>>
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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 Dec 28 17:49:22 2012
From: miha at softnet.si (Miha)
Date: Fri, 28 Dec 2012 15:49:22 +0100
Subject: [Freeswitch-users] Need a help with topology
Message-ID: <50DDB172.60205@softnet.si>

Hi,

Now I have only on FS server which works very very well. As I would try 
to migrate more user to FS server I am implementing opensips before FS 
for load_balancing and also registration. I will use opensips also for 
cdr (if it will work ok, like fs) and for enum.

On FS I have now few groups which are using group pickup in that kind of 
things.
Is it possible to connect FS boxes together so that it will be possible 
to do group call pickup and still call from other FS box or would would 
you suggest or how to deal with this issue?

The same with CFWD and etc.

thanks!

Miha


From a.venugopan at mundio.com  Fri Dec 28 18:32:18 2012
From: a.venugopan at mundio.com (Archana Venugopan)
Date: Fri, 28 Dec 2012 15:32:18 +0000
Subject: [Freeswitch-users] repeated registration messages in logs
In-Reply-To: <36376C0D-2F14-4C83-97DE-DD8DE4B1C3A6@gmail.com>
References: <592A9CF93E12394E8472A6CC66E66BF233D501@Mail-Kilo.squay.com>
	<CAJwc9xECzG84Oc4LwnHSKYrRGFYdzxyjyyarofG1=X=XjMmG+g@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233D56B@Mail-Kilo.squay.com>
	<592A9CF93E12394E8472A6CC66E66BF233D592@Mail-Kilo.squay.com>
	<36376C0D-2F14-4C83-97DE-DD8DE4B1C3A6@gmail.com>
Message-ID: <592A9CF93E12394E8472A6CC66E66BF233D7D8@Mail-Kilo.squay.com>

Hi,

Thanks. I initially thought these logs might have caused time out of freeswitch. Now I am not sure why freeswitch is timing out.

I have no entries in logs for any errors on this time out. But after few hours am not able to make calls and when I open fs_cli, and give show registrations its just hanging and I need to re-start FS to bring back to normal.
I have 1 other production machine where FS(1.0.head) does not hang or time out and I have set everything similar to that except for the FS version(1.2.3) difference.

Can you please let me know why FS behaves like this?

Regards,
Archana

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: 27 December 2012 19:21
To: FreeSWITCH Users Help
Cc: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] repeated registration messages in logs

Registrations work by the client sending a REGISTER to the server, which returns 401 to challenge and client resends REGISTER with the authentication details.

The register has an expiry time. To remain registered the client must reREGISTER within this time, and each REGISTER has its own 401 authentication (to prevent replay attacks).

What you're seeing is a log message each time the registration is renewed. That's why you still see it even though FS is apparently idle.

The log message really is only informative. It logs at WARNING level so that you can use fail2ban while only logging serious messages so you don't get massive log files.

FS will log both the initial REGISTER/401 then at the 2nd REGISTER whether it succeeds or fails. Both are required to detect bots looking to break into an insecure or DDOS a SIP server, since the client might never send the 2nd REGISTER.

If you are not using fail2ban and want quieter logs, set the log-auth-failures param to false on the relevant Sofia profile. Add it if it doesn't exist.


mod_fail2ban provides native integration without having to log these messages for fail2ban to detect. If you use it you'll probably still need to set the above param. (Disclaimer: I haven't used the module).

Steve




On 27 Dec 2012, at 14:44, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,
Thanks. If there is no activity for a while then calls drops out and I need to re-start the FS. I also see these messages frequently after each call and after registration and while FS is idle.
Am not sure why is this happening.

I keep getting those messages every 5 mins.

Regards,
Archana

From: Mick Stevens [mailto:mick.stevens at smartipx.com]
Sent: 27 December 2012 12:49
To: Archana Venugopan
Subject: Re: [Freeswitch-users] repeated registration messages in logs

Hi Archana,

This is just your SIP client (softphone etc) sending keep alive/SIP re-register messages. You can normally adjust this in the settings of your client.

I find setting "reregister every 300 seconds" normally works ok for me.

Rgds, Mick


On 27 December 2012 12:40, Archana Venugopan <a.venugopan at mundio.com<mailto:a.venugopan at mundio.com>> wrote:
Hi,

After registering the phones successfully I am getting the below message continuously every few seconds in my logs and this is filling up the logs and freeswitch goes down sometimes.
Can someone please tell me why I keep getting this message every few seconds? I have another server and there I don?t see this issue.

2012-12-27 12:37:44.714558 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [Abarajit at vectonecloud.com<mailto:Abarajit at vectonecloud.com>] from ip 192.168.2.205
2012-12-27 12:41:58.744557 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [sana at vectonecloud.com<mailto:sana at vectonecloud.com>] from ip 192.168.2.48

Regards,
Archana


_________________________________________________________________________
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



--
Regards

Mick Stevens
Communications Technologist
Smart IPX Ltd
Tel/Fax. +44(0)20 7001 6869
Email. mick.stevens at smartipx.com<mailto:mick.stevens at smartipx.com>
Skype. mick.smartipx.com<http://mick.gno247.com/>
www.smartipx.com<http://www.smartipx.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

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://lists.freeswitch.org/mailman/options/freeswitch-users>
http://www.freeswitch.org
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From bhomia at gmail.com  Fri Dec 28 16:31:41 2012
From: bhomia at gmail.com (ashutosh bhomia)
Date: Fri, 28 Dec 2012 19:01:41 +0530
Subject: [Freeswitch-users] Java esl client
Message-ID: <CAJsS1pY32J+gQxMZJaTUb2EhotTuuM4=4Np=xj62CzGHkp9Oqw@mail.gmail.com>

Hi,
we had developed a java application. We are using jar eslclient0.9.3 to
connect with freeswitch.Parser we had used is netty. Now when we start the
application the thread count continually keep on increasing till they reach
32000. when we use jstack to analyse the behavior we are getting following
output.
Threads in state Blocked by Locks
3054 threads running in
org.freeswitch.esl.client.internal.AbstractEslClientHandler.messageReceived(AbstractEslClientHandler.java:81)
515 threads running in
org.freeswitch.esl.client.internal.AbstractEslClientHandler$SyncCallback.get(AbstractEslClientHandler.java:234)
515 threads running in
org.jboss.netty.channel.SimpleChannelUpstreamHandler.handleUpstream(SimpleChannelUpstreamHandler.java:112)
268 threads running in System Thread
8 threads running in
org.apache.tomcat.util.threads.TaskQueue.poll(TaskQueue.java:86)
Total Blocked by Locks Threads: 4360

Any help on this is greatly appreciated.


-- 
Regards
Ashutosh Bhomia
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From steveayre at gmail.com  Fri Dec 28 20:41:40 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 28 Dec 2012 17:41:40 +0000
Subject: [Freeswitch-users] repeated registration messages in logs
In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233D7D8@Mail-Kilo.squay.com>
References: <592A9CF93E12394E8472A6CC66E66BF233D501@Mail-Kilo.squay.com>
	<CAJwc9xECzG84Oc4LwnHSKYrRGFYdzxyjyyarofG1=X=XjMmG+g@mail.gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233D56B@Mail-Kilo.squay.com>
	<592A9CF93E12394E8472A6CC66E66BF233D592@Mail-Kilo.squay.com>
	<36376C0D-2F14-4C83-97DE-DD8DE4B1C3A6@gmail.com>
	<592A9CF93E12394E8472A6CC66E66BF233D7D8@Mail-Kilo.squay.com>
Message-ID: <CAFiqYumy_XMZiEgtcQhEi8PLc7aY1mptzJELZrH+=v=w=hbJHQ@mail.gmail.com>

You could collect a coredump when FS is hanging using using gcore.
http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29

First I'd try updating to either the 1.2.5.3 release, or git head of the
v1.2.stable or master branches. It could be a bug that's already been
discovered and resolved.

-Steve



On 28 December 2012 15:32, Archana Venugopan <a.venugopan at mundio.com> wrote:

>  Hi,****
>
> ** **
>
> Thanks. I initially thought these logs might have caused time out of
> freeswitch. Now I am not sure why freeswitch is timing out.****
>
> ** **
>
> I have no entries in logs for any errors on this time out. But after few
> hours am not able to make calls and when I open fs_cli, and give show
> registrations its just hanging and I need to re-start FS to bring back to
> normal. ****
>
> I have 1 other production machine where FS(1.0.head) does not hang or time
> out and I have set everything similar to that except for the FS
> version(1.2.3) difference. ****
>
> ** **
>
> Can you please let me know why FS behaves like this?****
>
> ** **
>
> Regards,****
>
> Archana****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* 27 December 2012 19:21
> *To:* FreeSWITCH Users Help
> *Cc:* FreeSWITCH Users Help
>
> *Subject:* Re: [Freeswitch-users] repeated registration messages in logs**
> **
>
>  ** **
>
> Registrations work by the client sending a REGISTER to the server, which
> returns 401 to challenge and client resends REGISTER with the
> authentication details.****
>
> ** **
>
> The register has an expiry time. To remain registered the client must
> reREGISTER within this time, and each REGISTER has its own 401
> authentication (to prevent replay attacks).****
>
> ** **
>
> What you're seeing is a log message each time the registration is renewed.
> That's why you still see it even though FS is apparently idle.****
>
> ** **
>
> The log message really is only informative. It logs at WARNING level so
> that you can use fail2ban while only logging serious messages so you don't
> get massive log files.****
>
> ** **
>
> FS will log both the initial REGISTER/401 then at the 2nd REGISTER whether
> it succeeds or fails. Both are required to detect bots looking to break
> into an insecure or DDOS a SIP server, since the client might never send
> the 2nd REGISTER.****
>
> ** **
>
> If you are not using fail2ban and want quieter logs, set the
> log-auth-failures param to false on the relevant Sofia profile. Add it if
> it doesn't exist.****
>
>
>
> ****
>
> mod_fail2ban provides native integration without having to log these
> messages for fail2ban to detect. If you use it you'll probably still need
> to set the above param. (Disclaimer: I haven't used the module).****
>
> ** **
>
> Steve****
>
> ** **
>
> ** **
>
> ** **
>
>
> On 27 Dec 2012, at 14:44, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
>  Hi,****
>
> Thanks. If there is no activity for a while then calls drops out and I
> need to re-start the FS. I also see these messages frequently after each
> call and after registration and while FS is idle.****
>
> Am not sure why is this happening. ****
>
>  ****
>
> I keep getting those messages every 5 mins.****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
> *From:* Mick Stevens [mailto:mick.stevens at smartipx.com<mick.stevens at smartipx.com>]
>
> *Sent:* 27 December 2012 12:49
> *To:* Archana Venugopan
> *Subject:* Re: [Freeswitch-users] repeated registration messages in logs**
> **
>
>  ****
>
> Hi Archana,****
>
>  ****
>
> This is just your SIP client (softphone etc) sending keep alive/SIP
> re-register messages. You can normally adjust this in the settings of your
> client.****
>
>  ****
>
> I find setting "reregister every 300 seconds" normally works ok for me.***
> *
>
>  ****
>
> Rgds, Mick****
>
>  ****
>
>  ****
>
> On 27 December 2012 12:40, Archana Venugopan <a.venugopan at mundio.com>
> wrote:****
>
> Hi,****
>
>  ****
>
> After registering the phones successfully I am getting the below message
> continuously every few seconds in my logs and this is filling up the logs
> and freeswitch goes down sometimes.****
>
> Can someone please tell me why I keep getting this message every few
> seconds? I have another server and there I don?t see this issue. ****
>
>  ****
>
> 2012-12-27 12:37:44.714558 [WARNING] sofia_reg.c:1484 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [Abarajit at vectonecloud.com]
> from ip 192.168.2.205****
>
> 2012-12-27 12:41:58.744557 [WARNING] sofia_reg.c:1484 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [sana at vectonecloud.com] from
> ip 192.168.2.48****
>
>  ****
>
> Regards,****
>
> Archana****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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****
>
>
>
> ****
>
>  ****
>
> -- ****
>
> Regards
>
> Mick Stevens
> *Communications Technologist**
> Smart IPX Ltd
> *Tel/Fax. +44(0)20 7001 6869****
>
> Email. mick.stevens at smartipx.com
> Skype. mick.smartipx.com <http://mick.gno247.com/>****
>
> www.smartipx.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 schoch+freeswitch.org at xwin32.com  Fri Dec 28 21:25:12 2012
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Fri, 28 Dec 2012 10:25:12 -0800
Subject: [Freeswitch-users] Newbie question about Cisco phones
In-Reply-To: <3C11F92D-11A2-4514-B1AD-27F04EE5E286@gmail.com>
References: <CAOkuKdcf3O2d=Q-QMr04me3htgARdib9Ug9Zhr_22_4osXSaZQ@mail.gmail.com>
	<CAKzWOxU05dUYyG+mM65-r0T=B=5o8Xoy+YLYK+80UgUmtQ+abg@mail.gmail.com>
	<CAOkuKdfjMgf-fDiT4zEyCmVu7QZ=FwgTT4Xz6n2Yu7-ON5pVoA@mail.gmail.com>
	<3C11F92D-11A2-4514-B1AD-27F04EE5E286@gmail.com>
Message-ID: <CAOkuKddkYstyLUc44QvAsFOCvZ94waFo2pMA6NEdrt34X08xdw@mail.gmail.com>

On Fri, Dec 28, 2012 at 2:13 AM, Steven Ayre <steveayre at gmail.com> wrote:

> This is incomplete. The normal flow is
> -> REGISTER
> <- 401
> -> REGISTER
> <- 200 or 403
>

I get it!  It's like an HTTP basic auth sequence.  The client does a GET,
the server sends a 401 Authorization Required, and then the client prompts
the user for login/password.


The digest includes the domain, if the user and password match perhaps the
> problem lies there...
>

Here is the entire header from the 2nd request:
   REGISTER sip:192.168.4.1:5060;transport=tcp SIP/2.0
   Via: SIP/2.0/TCP 192.168.4.254:4025
;rport;branch=z9hG4bKPj3ewRQ6PZNlIanuwbxr2CmdFx1fK2BF0A
   Max-Forwards: 70
   From: "110" <sip:1001 at 192.168.4.1>;tag=fP-z0E14AtiuJz99qM1-4639EYKwlVjD
   To: "110" <sip:1001 at 192.168.4.1>
   Call-ID: TJDE1DQJNq40N64qIBpmll0KBNhezSyR
   CSeq: 22408 REGISTER
   User-Agent: Cisco-CP3905/9.2.1
   Contact: <sip:110 at 192.168.4.254:4025
;transport=TCP>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-f47f353ce879>";+u.sip!
devicename.ccm.cisco.com="SEPF47F353CE879";+u.sip!model.ccm.cisco.com="592"
   Expires: 3600
   Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
   Content-Disposition: session;handling=optional
   Content-Type: application/x-cisco-remotecc-request+xml
   Content-Length: 413
   Reason: SIP;cause=200;text="cisco-alarm:25 Name=SEPF47F353CE879
Load=3905.9-2-1-0 Last=initialized"
   Supported:
replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-monrec,X-cisco-config,X-cisco-sis-4.0.0,X-cisco-xsi-7.0.1
   Authorization: Digest username="1001", realm="192.168.4.1",
nonce="ba548054-7dac-4cae-8939-076a2b438ee6",
uri="sip:192.168.4.1:5060;transport=tcp",
response="d9057b4ea439ce750e6b4e751cfc5e69", algorithm=MD5,
cnonce="G4bay98LUnpnFmf.907akOTr60NEaeup", qop=auth, nc=00000001

The configuration file from /tftpboot includes:
<name>1001</name>
...
<authName>1001</authName>
<authPassword>654987</authPassword>

...so I assume it's sending that password in the digest.

I think my problem is that I don't yet fully understand the FreeSwitch
configuration.  The only place that password occurs is in
conf/directory/1001.xml.  It it supposed to go somewhere else?

On Fri, Dec 28, 2012 at 2:52 AM, Erik Dekkers <erik.dekkers at certhon.com>
 wrote:

> Seeing the firmware version in the User Agent header this phone is
> probably build on the same platform as the 79XX series phones and having
> the same sip-stack.
>
> From experience I can tell you that sip-stack is really really horrible. I
> don?t know if you have bought a lot of these phone or just testing one.
>

Fortunately, I only purchased 2 for testing.

I also purchased 3 Polycom SoundPoint IP 320 SIP phones, also for testing.
 I haven't gotten to them yet, because they need to boot with FTP instead
of TFTP, and I need to figure out how to get vsftpd to work.

I also bought an ultra-cheap Zultys (their headquarters is 1 mile from my
house) phone, but I haven't even powered it up yet, and I probably won't
use many of them because it doesn't have PoE.

I'll try your suggestion on the SPA500 series.

-- 
Steve
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From andrew at cassidywebservices.co.uk  Fri Dec 28 22:24:12 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Fri, 28 Dec 2012 19:24:12 +0000
Subject: [Freeswitch-users] Voicemail in a Cluster
In-Reply-To: <CAFiqYunqy-abJ-y+4-VOKGV_=O8oe4mBX_7kc01wTekPvSx3mQ@mail.gmail.com>
References: <CALNZuwkzkrO4GQ3BD9jm8npQ=u3jhd2s0vfK_ymvvz5tT=j39w@mail.gmail.com>
	<CAMwZ-tJO7uVmLiCDwU6TR44jL+KMaz26yG0U-2KgU8iJv=9MvQ@mail.gmail.com>
	<CAFiqYumfM=UXbKPn1_=wpx5Pdj+QAevXX95QdZeBN1A-kdYKrA@mail.gmail.com>
	<CAFiqYunqy-abJ-y+4-VOKGV_=O8oe4mBX_7kc01wTekPvSx3mQ@mail.gmail.com>
Message-ID: <CAMbW9o+ZChwqXZWiGVJOFEz5EgrTqPG0WV=8MAAxjUVHzqw1hQ@mail.gmail.com>

Or even glusterfs, I've had goo results with that.

On 28 December 2012 14:21, Steven Ayre <steveayre at gmail.com> wrote:

> As for NFS... using a cluster filesystem on top of DRBD is one way to
> avoid the filesystem becoming a SPOF.
>
>
>
> On 28 December 2012 14:20, Steven Ayre <steveayre at gmail.com> wrote:
>
>> ODBC isn't necessarily a single point of failure - you can connect to a
>> MySQL Cluster database or a multi-master circular replicated database for
>> example.
>>
>>
>> On 28 December 2012 08:13, Gabriel Gunderson <gabe at gundy.org> wrote:
>>
>>> On Fri, Dec 28, 2012 at 12:57 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>> > 3) Some sort of NFS - but then this become a single point of failure?
>>>
>>> ODBC and NFS is a great way to go. Yes, they are each single points of
>>> failure, but each can be made redundant.
>>>
>>> That's where I would start before spending too much time tooling around.
>>>
>>>
>>> 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
>


-- 
*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 schoch+freeswitch.org at xwin32.com  Fri Dec 28 22:44:12 2012
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Fri, 28 Dec 2012 11:44:12 -0800
Subject: [Freeswitch-users] Best practices question about SIP registration
Message-ID: <CAOkuKdfpX7BHNqQWpj0RmoEF9-+mzq2hJW06_p-WE5N=C_Ey4g@mail.gmail.com>

Please excuse me, I'm a VoIP newbie.

I found the very helpful comment in conf/directory/default.xml.  It says
that the default domain is the IP address of the Freeswitch server.  It
seems to me that this could cause problems if the IP changed for some
reason, or if the server has multiple NICs (which one does it choose in
that case?)

Is it better do just define a well known text string for the default
domain, or even use a real DNS domain name?

2nd question is about SIP usernames.  At least one phone I have found uses
the MAC address as the default SIP user.  Would it be better to avoid
having a separate TFTP file for each phone, and just use the already unique
MAC address as the user?

-- 
Steve
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From schoch+freeswitch.org at xwin32.com  Sat Dec 29 02:29:59 2012
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Fri, 28 Dec 2012 15:29:59 -0800
Subject: [Freeswitch-users] You must define a domain called 'xx.com' in your
	directory
Message-ID: <CAOkuKdeZRrn5Jq-Lhaca5s+AtTK14N2pBEvDC4+Oy6Lc-O+hOg@mail.gmail.com>

I must be getting closer to getting this to work, because now I get this
message:

2012-12-28 15:18:23.596375 [WARNING] sofia_reg.c:2485 Can't find user [
103 at xx.com]
You must define a domain called 'xx.com' in your directory and add a user
with the id="103" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.

I have a file named conf/directory/xx.com.xml.  It's a copy of
conf/directory/default.xml, except the include line specifies "xx.com/*.xml"
instead of "default/*.xml".  I have the conf/directory/xx.com/103.xml file.

sofia status says:

                     Name          Type
  Data      State
=================================================================================================
                   xx.com         alias
internal      ALIASED
                 internal       profile
sip:mod_sofia at 192.168.4.1:5060      RUNNING (0)
            internal-ipv6       profile
sip:mod_sofia@[::1]:5060
     RUNNING (0)
                 external       profile
sip:mod_sofia at 192.168.4.1:5080      RUNNING (0)
    external::example.com       gateway
sip:joeuser at example.com      NOREG
      external::flowroute       gateway
sip:xxxxxxxxxx at sip.flowroute.com      REGED
=================================================================================================
3 profiles 1 alias

I checked the file log/freeswitch.xml.fsxml, and it has the lines from the
"103.xml" file, which has the password defined.

I can't figure out what to do next.  Help?

-- 
Steve
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From cal.leeming at simplicitymedialtd.co.uk  Sat Dec 29 03:50:10 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sat, 29 Dec 2012 00:50:10 +0000
Subject: [Freeswitch-users] mod_xml_curl + variables documentation
Message-ID: <CAHKQagGWH7vt1AXtQMvpwcWU_ZkTiWxn00xVZfjbbs8m=RtagA@mail.gmail.com>

Spent several hours working on further re-structuring the mod_xml_curl
docs, but I'm still not happy with it :/

http://wiki.freeswitch.org/wiki/Mod_xml_curl

As mod_xml_curl request variables can be very difficult to figure out, I
have started to document which request variables apply where, which are
common, which are section specific etc.

However - there is also a huge lack of variable documentation, and although
this isn't strictly a mod_xml_curl thing, it would benefit many others
areas too if we had all these documented.* *

Although the mod_xml_curl docs page isn't the proper home for variable
descriptions, it will suffice for now until we have them all collected.

For example:
http://wiki.freeswitch.org/wiki/Mod_xml_curl#sip_user_agent
http://wiki.freeswitch.org/wiki/Mod_xml_curl#sip_request_port

Compared to:
http://wiki.freeswitch.org/wiki/Variable_sip_user_agent
http://wiki.freeswitch.org/wiki/Variable_sip_request_port

To some extent, I think that the concept of having individual pages for
each variable is not very user friendly - would anyone care to speculate on
whether this should be phased out in favour of a single page with all the
variables on?

Either way, if anyone has maybe a spare 10 minutes to contribute towards
getting these variables documented, it would really help reduce the work
load (some are a simple Google job, others require trawling through source
code).

Cal
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From steveayre at gmail.com  Sat Dec 29 07:24:47 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 29 Dec 2012 04:24:47 +0000
Subject: [Freeswitch-users] mod_xml_curl + variables documentation
In-Reply-To: <CAHKQagGWH7vt1AXtQMvpwcWU_ZkTiWxn00xVZfjbbs8m=RtagA@mail.gmail.com>
References: <CAHKQagGWH7vt1AXtQMvpwcWU_ZkTiWxn00xVZfjbbs8m=RtagA@mail.gmail.com>
Message-ID: <CDA1C659-889E-428C-B713-0BDA11AA7193@gmail.com>

Please read the wiki source of the http://wiki.freeswitch.org/wiki/Channel_Variables and http://wiki.freeswitch.org/wiki/Variable_sip_auth_username wiki pages.

The variables pages contain <onlyinclude> markup while the channel variables page liberally includes the descriptions from the Variable_ pages. That means the descriptions can be maintained in a single location and reused across the wiki, and changes automatically update everywhere.

I notice you have written a new sip_auth_username section instead of including the existing one (and possibly modifying it), so I'm not sure whether you're aware of this feature.

Regards,
Steve



On 29 Dec 2012, at 00:50, "Cal Leeming [Simplicity Media Ltd]" <cal.leeming at simplicitymedialtd.co.uk> wrote:

> Spent several hours working on further re-structuring the mod_xml_curl docs, but I'm still not happy with it :/
> 
> http://wiki.freeswitch.org/wiki/Mod_xml_curl
> 
> As mod_xml_curl request variables can be very difficult to figure out, I have started to document which request variables apply where, which are common, which are section specific etc.
> 
> However - there is also a huge lack of variable documentation, and although this isn't strictly a mod_xml_curl thing, it would benefit many others areas too if we had all these documented. 
> 
> Although the mod_xml_curl docs page isn't the proper home for variable descriptions, it will suffice for now until we have them all collected.
> 
> For example:
> http://wiki.freeswitch.org/wiki/Mod_xml_curl#sip_user_agent
> http://wiki.freeswitch.org/wiki/Mod_xml_curl#sip_request_port
> 
> Compared to:
> http://wiki.freeswitch.org/wiki/Variable_sip_user_agent
> http://wiki.freeswitch.org/wiki/Variable_sip_request_port
> 
> To some extent, I think that the concept of having individual pages for each variable is not very user friendly - would anyone care to speculate on whether this should be phased out in favour of a single page with all the variables on?
> 
> Either way, if anyone has maybe a spare 10 minutes to contribute towards getting these variables documented, it would really help reduce the work load (some are a simple Google job, others require trawling through source code).
> 
> Cal
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 fs-list at communicatefreely.net  Sat Dec 29 07:47:00 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 28 Dec 2012 23:47:00 -0500
Subject: [Freeswitch-users] Best practices question about SIP
	registration
In-Reply-To: <CAOkuKdfpX7BHNqQWpj0RmoEF9-+mzq2hJW06_p-WE5N=C_Ey4g@mail.gmail.com>
References: <CAOkuKdfpX7BHNqQWpj0RmoEF9-+mzq2hJW06_p-WE5N=C_Ey4g@mail.gmail.com>
Message-ID: <50DE75C4.1030500@communicatefreely.net>

Hi Steven,

I would recommend using a proper domain name as much as possible.  For one, it looks 
nicer!  A SIP URI is supposed to be user at domain like an e-mail address is, and I hope that 
one day URI dialing will be common place, so we might as well do it right the first time. 
  It also helps to manage things.  If you set up SRV DNS records, many devices only need 
the user and domain, as the address and ports are looked up in the SRV record.  This also 
allows for load share or failover right in the DNS record!

For the second question, the MAC address probably doesn't make a lot of sense to me.  The 
user part is supposed to be a user name, not a hardware address.  What if you replace the 
phone?  This is normally an extension number, a telephone number, or perhaps a username 
like the user part of an e-mail address.  Any could be correct, but they should always be 
meaningful, and abstracted from the hardware.  The user part is often presented as caller 
ID, so it should mean something if it comes up on someone else's display as the 
originating number.

If you have a lot of phones, you should look into a better provisioning mechanism than 
individual TFTP files.  It makes sense to have a master config file that contains common 
settings, but it in a larger system, some sort of automation is good.

If you are running out of a database, you can use a server-side script to auto-generate 
the config as the phone asks for it.  This requires the phone to be able to use HTTP to 
fetch config.  Most phones can.

Failing that, many other phones will let you include multiple files, so you can have a 
master config file, then a per-phone file that just has overrides.  You may also be able 
to name the individual files based on the username, then create a symbolic link to the 
file named as the MAC address.  If you want to re-assign a phone to a different user, just 
update the symlink.

I recommend the HTTP script if you are up for it, as you can automate everything and save 
yourself a lot of time in the long run.  Create one database that you manage your users in 
and have Freeswitch config (via XML Curl) as well as the phone config fetch the 
appropriate settings from it.  That way, they will always match.

-Tim

Steven Schoch wrote:
> Please excuse me, I'm a VoIP newbie.
> 
> I found the very helpful comment in conf/directory/default.xml.  It says 
> that the default domain is the IP address of the Freeswitch server.  It 
> seems to me that this could cause problems if the IP changed for some 
> reason, or if the server has multiple NICs (which one does it choose in 
> that case?)
> 
> Is it better do just define a well known text string for the default 
> domain, or even use a real DNS domain name?
> 
> 2nd question is about SIP usernames.  At least one phone I have found 
> uses the MAC address as the default SIP user.  Would it be better to 
> avoid having a separate TFTP file for each phone, and just use the 
> already unique MAC address as the user?
> 
> -- 
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anton.jugatsu at gmail.com  Sat Dec 29 08:28:34 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Sat, 29 Dec 2012 09:28:34 +0400
Subject: [Freeswitch-users] You must define a domain called 'xx.com' in
 your directory
In-Reply-To: <CAOkuKdeZRrn5Jq-Lhaca5s+AtTK14N2pBEvDC4+Oy6Lc-O+hOg@mail.gmail.com>
References: <CAOkuKdeZRrn5Jq-Lhaca5s+AtTK14N2pBEvDC4+Oy6Lc-O+hOg@mail.gmail.com>
Message-ID: <CANBhyPD5ShRr52nG+A+D6g_mjhQdTQyHKMRZQi3kk1iCafU0WA@mail.gmail.com>

<domain name="xx.com">
</domain>

F6


2012/12/29 Steven Schoch <schoch+freeswitch.org at xwin32.com>

> I must be getting closer to getting this to work, because now I get this
> message:
>
> 2012-12-28 15:18:23.596375 [WARNING] sofia_reg.c:2485 Can't find user [
> 103 at xx.com]
> You must define a domain called 'xx.com' in your directory and add a user
> with the id="103" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
>
> I have a file named conf/directory/xx.com.xml.  It's a copy of
> conf/directory/default.xml, except the include line specifies "
> xx.com/*.xml" instead of "default/*.xml".  I have the conf/directory/
> xx.com/103.xml file.
>
> sofia status says:
>
>                      Name          Type
>     Data      State
>
> =================================================================================================
>                    xx.com         alias
> internal      ALIASED
>                  internal       profile
> sip:mod_sofia at 192.168.4.1:5060      RUNNING (0)
>             internal-ipv6       profile                   sip:mod_sofia@[::1]:5060
>      RUNNING (0)
>                  external       profile
> sip:mod_sofia at 192.168.4.1:5080      RUNNING (0)
>     external::example.com       gateway
> sip:joeuser at example.com      NOREG
>       external::flowroute       gateway
> sip:xxxxxxxxxx at sip.flowroute.com      REGED
>
> =================================================================================================
> 3 profiles 1 alias
>
> I checked the file log/freeswitch.xml.fsxml, and it has the lines from the
> "103.xml" file, which has the password defined.
>
> I can't figure out what to do next.  Help?
>
> --
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sertys at gmail.com  Sat Dec 29 08:45:58 2012
From: sertys at gmail.com (Daniel Ivanov)
Date: Sat, 29 Dec 2012 06:45:58 +0100
Subject: [Freeswitch-users] Need a help with topology
In-Reply-To: <50DDB172.60205@softnet.si>
References: <50DDB172.60205@softnet.si>
Message-ID: <CAMeZp612x+6wpudbjZeLQ=yX_aOb3rVx2s-CcBki_DpdH+6WTA@mail.gmail.com>

But of course you can manage pickup groups on different servers. It is a
matter of dialstring management. You just need a directory.to keep track of
endpoints across servers and construct the dialstrings. Maybe use mod_lua
or other scripting and a simple db.
On Dec 28, 2012 4:52 PM, "Miha" <miha at softnet.si> wrote:

> Hi,
>
> Now I have only on FS server which works very very well. As I would try
> to migrate more user to FS server I am implementing opensips before FS
> for load_balancing and also registration. I will use opensips also for
> cdr (if it will work ok, like fs) and for enum.
>
> On FS I have now few groups which are using group pickup in that kind of
> things.
> Is it possible to connect FS boxes together so that it will be possible
> to do group call pickup and still call from other FS box or would would
> you suggest or how to deal with this issue?
>
> The same with CFWD and etc.
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Dec 29 11:32:16 2012
From: miha at softnet.si (Miha)
Date: Sat, 29 Dec 2012 09:32:16 +0100
Subject: [Freeswitch-users] Need a help with topology
In-Reply-To: <CAMeZp612x+6wpudbjZeLQ=yX_aOb3rVx2s-CcBki_DpdH+6WTA@mail.gmail.com>
References: <50DDB172.60205@softnet.si>
	<CAMeZp612x+6wpudbjZeLQ=yX_aOb3rVx2s-CcBki_DpdH+6WTA@mail.gmail.com>
Message-ID: <web-88567440@softnet.si>

Hi Daniel,

thank for your answer. Could you be pls more precise or
give me an example how to do that because I cant truly
imagine how to do that?

How to pick up call from differnet FS.

Thanks!

Miha
On Sat, 29 Dec 2012 06:45:58 +0100
 Daniel Ivanov <sertys at gmail.com> wrote:
> But of course you can manage pickup groups on different
> servers. It is a
> matter of dialstring management. You just need a
> directory.to keep track of
> endpoints across servers and construct the dialstrings.
> Maybe use mod_lua
> or other scripting and a simple db.
> On Dec 28, 2012 4:52 PM, "Miha" <miha at softnet.si> wrote:
> 
> > Hi,
> >
> > Now I have only on FS server which works very very
> well. As I would try
> > to migrate more user to FS server I am implementing
> opensips before FS
> > for load_balancing and also registration. I will use
> opensips also for
> > cdr (if it will work ok, like fs) and for enum.
> >
> > On FS I have now few groups which are using group
> pickup in that kind of
> > things.
> > Is it possible to connect FS boxes together so that it
> will be possible
> > to do group call pickup and still call from other FS
> box or would would
> > you suggest or how to deal with this issue?
> >
> > The same with CFWD and etc.
> >
> > thanks!
> >
> > Miha
> >
> >
>
_________________________________________________________________________
> > 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
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> >
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >


From gabe at gundy.org  Sat Dec 29 12:32:27 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 29 Dec 2012 02:32:27 -0700
Subject: [Freeswitch-users] failover when storing recordings
In-Reply-To: <CAKzWOxVHJQ5PmKSvyx8XWJdO9gxrE_=_4zoVfmgCq=c3-m15Xg@mail.gmail.com>
References: <CALvF6vAUirzxuqxzryzGQ0BrP_+EEQVHy2WO2rbT-hNFDj_rig@mail.gmail.com>
	<50DCC9BD.3020401@elder.hu>
	<CAKzWOxVHJQ5PmKSvyx8XWJdO9gxrE_=_4zoVfmgCq=c3-m15Xg@mail.gmail.com>
Message-ID: <CAMwZ-t+v47S6DHW=cy9dQ2VAJjt5RtsXq6War-PPqN+HcNFrQg@mail.gmail.com>

On Thu, Dec 27, 2012 at 5:36 PM, Michael Collins <msc at freeswitch.org> wrote:
> You're better off doing a cron job that copies the files to the target host,
> confirms they made it, then deletes the files from the local host (i.e. the
> FreeSWITCH box).

I'm a fan of inotify's incron:
http://inotify.aiken.cz/?section=incron&page=doc&lang=en

Basically you configure incron to watch the files and dirs that you
care about, and when it sees the type of activity that you're looking
for, it will call the scripts that you'd like run on them.

This will save me some typing, but give you a feel for how it works;
from the docs:
####################################################################

The user table rows have the following syntax (use one or more spaces
between elements):
<path> <mask> <command>

Where:
<path> is a filesystem path (each whitespace must be prepended by a backslash)
<mask> is a symbolic (see inotify.h; use commas for separating
symbols) or numeric mask for events
<command> is an application or script to run on the events

The command may contain these wildcards:
$$ - a dollar sign
$@ - the watched filesystem path (see above)
$# - the event-related file name
$% - the event flags (textually)
$& - the event flags (numerically)

The mask may additionaly contain a special symbol IN_NO_LOOP which
disables events occurred during processing the event (to avoid loops).

Example: You need to run program 'abc' with the full file path as an
argument every time a file is changed in /var/mail. One of the
solutions follows:

/var/mail IN_CLOSE_WRITE abc $@/$#
####################################################################

Hope that helps.


Best,
Gabe


From gabe at gundy.org  Sat Dec 29 12:50:56 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 29 Dec 2012 02:50:56 -0700
Subject: [Freeswitch-users] You must define a domain called 'xx.com' in
 your directory
In-Reply-To: <CAOkuKdeZRrn5Jq-Lhaca5s+AtTK14N2pBEvDC4+Oy6Lc-O+hOg@mail.gmail.com>
References: <CAOkuKdeZRrn5Jq-Lhaca5s+AtTK14N2pBEvDC4+Oy6Lc-O+hOg@mail.gmail.com>
Message-ID: <CAMwZ-tL68sap0rT8thDa96KJRBOf4KzjAWZES+oRaSyb1Dj92g@mail.gmail.com>

On Fri, Dec 28, 2012 at 4:29 PM, Steven Schoch
<schoch+freeswitch.org at xwin32.com> wrote:
> I have a file named conf/directory/xx.com.xml.  It's a copy of
> conf/directory/default.xml, except the include line specifies "xx.com/*.xml"
> instead of "default/*.xml".  I have the conf/directory/xx.com/103.xml file.

Play around with find_user_xml and user_exists to trouble shoot your
directory independent of any specific SUA:

http://wiki.freeswitch.org/wiki/Mod_commands#find_user_xml
http://wiki.freeswitch.org/wiki/Mod_commands#user_exists

Once you're satisfied that your directory is setup properly, move on
to the sofia part of the puzzle.

I'm going from memory here but I think you'll find that the default
configs (internal.xml) set this paramater:

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#force-register-domain

to $${domain}. And $${domain} is set in vars.xml to $${local_ip_v4}.
That is magically set to some non-routable IPv4 address on your
server. So, that means that when you register your phone (with any
domain), FreeSWITCH ends up looking for 103 at 192.168.0.1 (or whatever
local IP your server you have) and *not* 103 at xx.com.

The reason they do that is to make it easier to get FreeSWITCH working
out of the box for a first time user. More advance users will figure
out what's going on and be able to move on from that simple setup to
something that's domain based.

Now, it's late and I didn't read your question very closely, so take
this all with a grain of salt ;) Good luck and good night!


Best,
Gabe


From gabe at gundy.org  Sat Dec 29 12:56:43 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 29 Dec 2012 02:56:43 -0700
Subject: [Freeswitch-users] Regarding Attendent transfer
In-Reply-To: <50DD5DC7.8050400@gmail.com>
References: <50DD5DC7.8050400@gmail.com>
Message-ID: <CAMwZ-t+0Yt2Vx6gfxkVd4usez0Y8HkBCpgLOy+DFAxYqzitUjQ@mail.gmail.com>

On Fri, Dec 28, 2012 at 1:52 AM, veerabhadrarao`
<bhadrarao.kankatala at gmail.com> wrote:
> I am facing problem while making  attendent transfer.

Please don't open a new thread for an existing question. Continue with
the one you posted earlier. Keeping the list manageable for 1000s of
users is more important than 'bumping' your topic.


Kind regards,
Gabe


From gabe at gundy.org  Sat Dec 29 13:04:42 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 29 Dec 2012 03:04:42 -0700
Subject: [Freeswitch-users] caller-id-number not getting set
In-Reply-To: <CAJbRaS4aJJw9_XCw08hWY_s_3OtCqGVoezA7Q9ooo0npCy5gKA@mail.gmail.com>
References: <CAJbRaS4aJJw9_XCw08hWY_s_3OtCqGVoezA7Q9ooo0npCy5gKA@mail.gmail.com>
Message-ID: <CAMwZ-tLCByn7UtBodbAUeMXNAu7bTG63SZqmebKmNXz2AujUbQ@mail.gmail.com>

On Thu, Dec 27, 2012 at 3:16 AM, sanjay soni
<sanjay.sanjaysoni at gmail.com> wrote:
> However, i see these values properly in the event being fired But not in the
> xml dialplan (Which is the place i want ot use them) ! What should I do ?

I suspect they *are* there, but you're calling them by the wrong name
or looking for them in the wrong place. I don't know, that just a
guess.

Start by putting a info app in your dialplan (somewhere it will always
get run) and have a look at what's there.

<action application="info"/>

Hope that helps. Let us know what you find.


Best,
Gabe


From gabe at gundy.org  Sat Dec 29 13:06:52 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 29 Dec 2012 03:06:52 -0700
Subject: [Freeswitch-users] Jingle updates since 2010 ?
In-Reply-To: <CAG2QdgaQ-kxyQE3BGKiOgexmq4xjSdbOkg6nMkX9=H4Zu8UhMQ@mail.gmail.com>
References: <CAG2QdgaLXFA_0QP+oboYUCwi7xWEc+CN5=u_G5O3mqYE3r=NYQ@mail.gmail.com>
	<CAG2QdgaQ-kxyQE3BGKiOgexmq4xjSdbOkg6nMkX9=H4Zu8UhMQ@mail.gmail.com>
Message-ID: <CAMwZ-tKn3q+FfWr6D_EtuRXyhDrNxhGGXzW5o-3iTnkTFDm9Lw@mail.gmail.com>

On Thu, Dec 27, 2012 at 5:51 AM, Ali Jawad <ali.jawad at splendor.net> wrote:
> Any input please ?

It's open source :) Find the lib that does the stuff you want and
check git to see how much activity it has seen in the last 24 months.



Best,
Gabe


From gabe at gundy.org  Sat Dec 29 13:08:14 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 29 Dec 2012 03:08:14 -0700
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
Message-ID: <CAMwZ-tKNbRAcNMT-upT3csbL=CuAg+u2J_Ntq7oQJMfuvkG7DQ@mail.gmail.com>

On Sun, Dec 16, 2012 at 9:15 AM, Cal Leeming [Simplicity Media Ltd]
<cal.leeming at simplicitymedialtd.co.uk> wrote:
> Any and all feedback on this thread would be much welcomed.

Sorry to be a wet blanket... but when it comes to NAT, there will be
no "final say" --only endless suffering ;)


Gabe


From gabe at gundy.org  Sat Dec 29 13:10:57 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 29 Dec 2012 03:10:57 -0700
Subject: [Freeswitch-users] anybody working on a swedish mod_say?
In-Reply-To: <CAKzWOxVG0kwrNmGULCaoE43XvexZoigBFBbJnt1cxR_EUiAWXA@mail.gmail.com>
References: <50D8D157.7020409@orresta.no-ip.com>
	<CALNZuwnhLTjWbTA5rNpEGY_RcasMcYVa7QpJfn=jhiYWz89iAQ@mail.gmail.com>
	<CAKzWOxVG0kwrNmGULCaoE43XvexZoigBFBbJnt1cxR_EUiAWXA@mail.gmail.com>
Message-ID: <CAMwZ-tLEqFVyW8jx6tg0ie9KY7tbY+OoxxnCN=KhvpkWuyp32A@mail.gmail.com>

On Wed, Dec 26, 2012 at 11:41 AM, Michael Collins <msc at freeswitch.org> wrote:
> Also, if you get a set of Swedish recordings let us know and we'll get them
> hosted on our files.freeswitch.org server.


If you do Swedish *chef*, I can help with recordings. I'm pretty sure
I can get a few people around the office to help too :)


Gabe


From andrew at cassidywebservices.co.uk  Sat Dec 29 13:59:45 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sat, 29 Dec 2012 10:59:45 +0000
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAMwZ-tKNbRAcNMT-upT3csbL=CuAg+u2J_Ntq7oQJMfuvkG7DQ@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<CAMwZ-tKNbRAcNMT-upT3csbL=CuAg+u2J_Ntq7oQJMfuvkG7DQ@mail.gmail.com>
Message-ID: <CAMbW9o+hPxQ2WzNesaYgvthEHwHs3MdLakfuRwMBZ_JZmj=sfw@mail.gmail.com>

Just wait til the ISPs who cab to set up IPv6 start double-natting.

On 29 December 2012 10:08, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sun, Dec 16, 2012 at 9:15 AM, Cal Leeming [Simplicity Media Ltd]
> <cal.leeming at simplicitymedialtd.co.uk> wrote:
> > Any and all feedback on this thread would be much welcomed.
>
> Sorry to be a wet blanket... but when it comes to NAT, there will be
> no "final say" --only endless suffering ;)
>
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 freeswitch-list at puzzled.xs4all.nl  Sat Dec 29 17:50:21 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Sat, 29 Dec 2012 15:50:21 +0100
Subject: [Freeswitch-users] anybody working on a swedish mod_say?
In-Reply-To: <CAMwZ-tLEqFVyW8jx6tg0ie9KY7tbY+OoxxnCN=KhvpkWuyp32A@mail.gmail.com>
References: <50D8D157.7020409@orresta.no-ip.com>
	<CALNZuwnhLTjWbTA5rNpEGY_RcasMcYVa7QpJfn=jhiYWz89iAQ@mail.gmail.com>
	<CAKzWOxVG0kwrNmGULCaoE43XvexZoigBFBbJnt1cxR_EUiAWXA@mail.gmail.com>
	<CAMwZ-tLEqFVyW8jx6tg0ie9KY7tbY+OoxxnCN=KhvpkWuyp32A@mail.gmail.com>
Message-ID: <50DF032D.8090209@puzzled.xs4all.nl>

On 12/29/2012 11:10 AM, Gabriel Gunderson wrote:
> On Wed, Dec 26, 2012 at 11:41 AM, Michael Collins <msc at freeswitch.org> wrote:
>> Also, if you get a set of Swedish recordings let us know and we'll get them
>> hosted on our files.freeswitch.org server.
>
>
> If you do Swedish *chef*, I can help with recordings. I'm pretty sure
> I can get a few people around the office to help too :)

Some practice material should anyone want to pitch in:

Swedish Chef - Meatballs
http://www.youtube.com/watch?v=sY_Yf4zz-yo

Swedish Chef - Chicken in the basket:
http://www.youtube.com/watch?v=Adfipy3lp9o

Regards,
Patrick



From lloyd.aloysius at gmail.com  Sat Dec 29 19:58:33 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Sat, 29 Dec 2012 11:58:33 -0500
Subject: [Freeswitch-users] xml_curl - switch.conf
Message-ID: <CAGvuWYYyBv=ywC39WQ4MkG8C6rknsoXQP9XcrV7+srk+PQ2Uuw@mail.gmail.com>

Hi,

Can we provide the switch.conf through xml_curl. I see freeswitch not
requesting the switch.conf?

I would like to eliminate all the .conf files except the xml_curl.conf.
Does this possible? Which .conf files cannot use through the xml_curl?


Thanks
Lloyd
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From cal.leeming at simplicitymedialtd.co.uk  Sat Dec 29 20:38:39 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sat, 29 Dec 2012 17:38:39 +0000
Subject: [Freeswitch-users] mod_xml_curl + variables documentation
In-Reply-To: <CDA1C659-889E-428C-B713-0BDA11AA7193@gmail.com>
References: <CAHKQagGWH7vt1AXtQMvpwcWU_ZkTiWxn00xVZfjbbs8m=RtagA@mail.gmail.com>
	<CDA1C659-889E-428C-B713-0BDA11AA7193@gmail.com>
Message-ID: <CAHKQagF4jK5pTBReedetuetnoZEk0Oqfb7mO8WyNgpWPhF=dRw@mail.gmail.com>

Ah, I was not aware of these pages, no idea how I managed to miss it!!

Thanks for letting me know :)

Cal

On Sat, Dec 29, 2012 at 4:24 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Please read the wiki source of the
> http://wiki.freeswitch.org/wiki/Channel_Variables and
> http://wiki.freeswitch.org/wiki/Variable_sip_auth_username wiki pages.
>
> The variables pages contain <onlyinclude> markup while the channel
> variables page liberally includes the descriptions from the Variable_
> pages. That means the descriptions can be maintained in a single location
> and reused across the wiki, and changes automatically update everywhere.
>
> I notice you have written a new sip_auth_username section instead of
> including the existing one (and possibly modifying it), so I'm not sure
> whether you're aware of this feature.
>
> Regards,
> Steve
>
>
>
> On 29 Dec 2012, at 00:50, "Cal Leeming [Simplicity Media Ltd]" <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
> Spent several hours working on further re-structuring the mod_xml_curl
> docs, but I'm still not happy with it :/
>
> http://wiki.freeswitch.org/wiki/Mod_xml_curl
>
> As mod_xml_curl request variables can be very difficult to figure out, I
> have started to document which request variables apply where, which are
> common, which are section specific etc.
>
> However - there is also a huge lack of variable documentation, and
> although this isn't strictly a mod_xml_curl thing, it would benefit many
> others areas too if we had all these documented.* *
>
> Although the mod_xml_curl docs page isn't the proper home for variable
> descriptions, it will suffice for now until we have them all collected.
>
> For example:
> http://wiki.freeswitch.org/wiki/Mod_xml_curl#sip_user_agent
> http://wiki.freeswitch.org/wiki/Mod_xml_curl#sip_request_port
>
> Compared to:
> http://wiki.freeswitch.org/wiki/Variable_sip_user_agent
> http://wiki.freeswitch.org/wiki/Variable_sip_request_port
>
> To some extent, I think that the concept of having individual pages for
> each variable is not very user friendly - would anyone care to speculate on
> whether this should be phased out in favour of a single page with all the
> variables on?
>
> Either way, if anyone has maybe a spare 10 minutes to contribute towards
> getting these variables documented, it would really help reduce the work
> load (some are a simple Google job, others require trawling through source
> code).
>
> Cal
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 29 20:43:58 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 29 Dec 2012 17:43:58 +0000
Subject: [Freeswitch-users] xml_curl - switch.conf
In-Reply-To: <CAGvuWYYyBv=ywC39WQ4MkG8C6rknsoXQP9XcrV7+srk+PQ2Uuw@mail.gmail.com>
References: <CAGvuWYYyBv=ywC39WQ4MkG8C6rknsoXQP9XcrV7+srk+PQ2Uuw@mail.gmail.com>
Message-ID: <0A23212E-FDE4-4421-A6AB-A06E52CD9026@gmail.com>

switch.conf.xml is read before modules (including mod_xml_curl) are loaded.

Sent from my iPad




On 29 Dec 2012, at 16:58, Lloyd Aloysius <lloyd.aloysius at gmail.com> wrote:

> Hi,
> 
> Can we provide the switch.conf through xml_curl. I see freeswitch not requesting the switch.conf?
> 
> I would like to eliminate all the .conf files except the xml_curl.conf. Does this possible? Which .conf files cannot use through the xml_curl?
> 
> 
> 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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From cal.leeming at simplicitymedialtd.co.uk  Sat Dec 29 21:03:13 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sat, 29 Dec 2012 18:03:13 +0000
Subject: [Freeswitch-users] xml_curl - switch.conf
In-Reply-To: <CAGvuWYYyBv=ywC39WQ4MkG8C6rknsoXQP9XcrV7+srk+PQ2Uuw@mail.gmail.com>
References: <CAGvuWYYyBv=ywC39WQ4MkG8C6rknsoXQP9XcrV7+srk+PQ2Uuw@mail.gmail.com>
Message-ID: <CAHKQagFe7qUJ7r3mcKJpFcai1cw=tzeiAVOAeCe1RE-4sEj7QQ@mail.gmail.com>

Yes you can send back switch.conf, but you need to do it via post_load.

Take a look at "post_load configuration" under;
http://wiki.freeswitch.org/wiki/Mod_xml_curl

Although it does seem to take note of the variables sent back, I'm not 100%
if all of them are applied in post_load (i.e. there may be some that don't
get applied via this method).

You'll have to use some trial and error to see (update the docs if there is
anything specific that doesn't get applied using post_load).

Let me know if this doesn't make any sense

Cal


On Sat, Dec 29, 2012 at 4:58 PM, Lloyd Aloysius <lloyd.aloysius at gmail.com>wrote:

> Hi,
>
> Can we provide the switch.conf through xml_curl. I see freeswitch not
> requesting the switch.conf?
>
> I would like to eliminate all the .conf files except the xml_curl.conf.
> Does this possible? Which .conf files cannot use through the xml_curl?
>
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cal.leeming at simplicitymedialtd.co.uk  Sat Dec 29 21:21:57 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sat, 29 Dec 2012 18:21:57 +0000
Subject: [Freeswitch-users] mod_xml_curl + variables documentation
In-Reply-To: <CAHKQagF4jK5pTBReedetuetnoZEk0Oqfb7mO8WyNgpWPhF=dRw@mail.gmail.com>
References: <CAHKQagGWH7vt1AXtQMvpwcWU_ZkTiWxn00xVZfjbbs8m=RtagA@mail.gmail.com>
	<CDA1C659-889E-428C-B713-0BDA11AA7193@gmail.com>
	<CAHKQagF4jK5pTBReedetuetnoZEk0Oqfb7mO8WyNgpWPhF=dRw@mail.gmail.com>
Message-ID: <CAHKQagG_-cN6wPza6_n5G1H14MUCeBR2-xGzTydH0yENreQOgg@mail.gmail.com>

I've made some further improvements.

The channel variables page now contains placeholders for all the variables
that were missing (approx 119 added).

http://wiki.freeswitch.org/wiki/Channel_Variables

I have also created/updated the following variable documentation pages with
the info collected from mod_xml_curl docs

http://wiki.freeswitch.org/wiki/Variable_sip_auth_realm
http://wiki.freeswitch.org/wiki/Variable_sip_auth_username
http://wiki.freeswitch.org/wiki/Variable_sip_auth_method
http://wiki.freeswitch.org/wiki/Variable_sip_user_agent
http://wiki.freeswitch.org/wiki/Variable_sip_request_host
http://wiki.freeswitch.org/wiki/Variable_sip_request_port
http://wiki.freeswitch.org/wiki/Variable_sip_profile
http://wiki.freeswitch.org/wiki/Variable_sip_from_port
http://wiki.freeswitch.org/wiki/Variable_sip_from_user
http://wiki.freeswitch.org/wiki/Variable_sip_from_host
http://wiki.freeswitch.org/wiki/Variable_sip_to_host
http://wiki.freeswitch.org/wiki/Variable_sip_to_port
http://wiki.freeswitch.org/wiki/Variable_sip_to_user
http://wiki.freeswitch.org/wiki/Variable_sip_contact_host
http://wiki.freeswitch.org/wiki/Variable_sip_contact_port
http://wiki.freeswitch.org/wiki/Variable_sip_contact_user

As such, I have also removed the field descriptions from mod_xml_curl and
placed the necessary references back to the correct documentation.

Could someone please review this and make sure it's in line with
quality/expectations etc.

Cal

On Sat, Dec 29, 2012 at 5:38 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Ah, I was not aware of these pages, no idea how I managed to miss it!!
>
> Thanks for letting me know :)
>
> Cal
>
>
> On Sat, Dec 29, 2012 at 4:24 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Please read the wiki source of the
>> http://wiki.freeswitch.org/wiki/Channel_Variables and
>> http://wiki.freeswitch.org/wiki/Variable_sip_auth_username wiki pages.
>>
>> The variables pages contain <onlyinclude> markup while the channel
>> variables page liberally includes the descriptions from the Variable_
>> pages. That means the descriptions can be maintained in a single location
>> and reused across the wiki, and changes automatically update everywhere.
>>
>> I notice you have written a new sip_auth_username section instead of
>> including the existing one (and possibly modifying it), so I'm not sure
>> whether you're aware of this feature.
>>
>> Regards,
>> Steve
>>
>>
>>
>> On 29 Dec 2012, at 00:50, "Cal Leeming [Simplicity Media Ltd]" <
>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>
>> Spent several hours working on further re-structuring the mod_xml_curl
>> docs, but I'm still not happy with it :/
>>
>> http://wiki.freeswitch.org/wiki/Mod_xml_curl
>>
>> As mod_xml_curl request variables can be very difficult to figure out, I
>> have started to document which request variables apply where, which are
>> common, which are section specific etc.
>>
>> However - there is also a huge lack of variable documentation, and
>> although this isn't strictly a mod_xml_curl thing, it would benefit many
>> others areas too if we had all these documented.* *
>>
>> Although the mod_xml_curl docs page isn't the proper home for variable
>> descriptions, it will suffice for now until we have them all collected.
>>
>> For example:
>> http://wiki.freeswitch.org/wiki/Mod_xml_curl#sip_user_agent
>> http://wiki.freeswitch.org/wiki/Mod_xml_curl#sip_request_port
>>
>> Compared to:
>> http://wiki.freeswitch.org/wiki/Variable_sip_user_agent
>> http://wiki.freeswitch.org/wiki/Variable_sip_request_port
>>
>> To some extent, I think that the concept of having individual pages for
>> each variable is not very user friendly - would anyone care to speculate on
>> whether this should be phased out in favour of a single page with all the
>> variables on?
>>
>> Either way, if anyone has maybe a spare 10 minutes to contribute towards
>> getting these variables documented, it would really help reduce the work
>> load (some are a simple Google job, others require trawling through source
>> code).
>>
>> Cal
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From cal.leeming at simplicitymedialtd.co.uk  Sat Dec 29 21:26:36 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sat, 29 Dec 2012 18:26:36 +0000
Subject: [Freeswitch-users] NAT traversal - the final say..!
In-Reply-To: <CAMwZ-tKNbRAcNMT-upT3csbL=CuAg+u2J_Ntq7oQJMfuvkG7DQ@mail.gmail.com>
References: <CAHKQagGJrOaLhwEUiWNNY-uLB1cmCNPCebJYWBe1Wg8mXf50Ug@mail.gmail.com>
	<CAMwZ-tKNbRAcNMT-upT3csbL=CuAg+u2J_Ntq7oQJMfuvkG7DQ@mail.gmail.com>
Message-ID: <CAHKQagHJ2f1WrtTaYbkHaJG=aSh6Gcs=fJbV=LymvGZF_GtCgA@mail.gmail.com>

Lol yes I'm starting to get that feeling.

With all this info I'll be able to put something together, but it'll need a
lot of proof reading I think

Cal

On Sat, Dec 29, 2012 at 10:08 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sun, Dec 16, 2012 at 9:15 AM, Cal Leeming [Simplicity Media Ltd]
> <cal.leeming at simplicitymedialtd.co.uk> wrote:
> > Any and all feedback on this thread would be much welcomed.
>
> Sorry to be a wet blanket... but when it comes to NAT, there will be
> no "final say" --only endless suffering ;)
>
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Dec 29 21:37:05 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 29 Dec 2012 20:37:05 +0200
Subject: [Freeswitch-users] mod_xml_curl + variables documentation
In-Reply-To: <CAHKQagG_-cN6wPza6_n5G1H14MUCeBR2-xGzTydH0yENreQOgg@mail.gmail.com>
References: <CAHKQagGWH7vt1AXtQMvpwcWU_ZkTiWxn00xVZfjbbs8m=RtagA@mail.gmail.com>
	<CDA1C659-889E-428C-B713-0BDA11AA7193@gmail.com>
	<CAHKQagF4jK5pTBReedetuetnoZEk0Oqfb7mO8WyNgpWPhF=dRw@mail.gmail.com>
	<CAHKQagG_-cN6wPza6_n5G1H14MUCeBR2-xGzTydH0yENreQOgg@mail.gmail.com>
Message-ID: <CALNZuwmK19xpW0txsb7ToqDtQF_9gZKjguFf7ehH4VkRkRTVHg@mail.gmail.com>

The only thing is.. it looks like this list is read-only variable.

Most channel variable can be changed to get a result, but these are *
informational* variables ("where did the call come from?) and therefore
unchangeable.
They should be presented differently than the others, I'd imagine...

-Avi


On Sat, Dec 29, 2012 at 8:21 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> I've made some further improvements.
>
> The channel variables page now contains placeholders for all the variables
> that were missing (approx 119 added).
>
> http://wiki.freeswitch.org/wiki/Channel_Variables
>
> I have also created/updated the following variable documentation pages
> with the info collected from mod_xml_curl docs
>
> http://wiki.freeswitch.org/wiki/Variable_sip_auth_realm
> http://wiki.freeswitch.org/wiki/Variable_sip_auth_username
> http://wiki.freeswitch.org/wiki/Variable_sip_auth_method
> http://wiki.freeswitch.org/wiki/Variable_sip_user_agent
> http://wiki.freeswitch.org/wiki/Variable_sip_request_host
> http://wiki.freeswitch.org/wiki/Variable_sip_request_port
> http://wiki.freeswitch.org/wiki/Variable_sip_profile
> http://wiki.freeswitch.org/wiki/Variable_sip_from_port
> http://wiki.freeswitch.org/wiki/Variable_sip_from_user
> http://wiki.freeswitch.org/wiki/Variable_sip_from_host
> http://wiki.freeswitch.org/wiki/Variable_sip_to_host
> http://wiki.freeswitch.org/wiki/Variable_sip_to_port
> http://wiki.freeswitch.org/wiki/Variable_sip_to_user
> http://wiki.freeswitch.org/wiki/Variable_sip_contact_host
> http://wiki.freeswitch.org/wiki/Variable_sip_contact_port
> http://wiki.freeswitch.org/wiki/Variable_sip_contact_user
>
> As such, I have also removed the field descriptions from mod_xml_curl and
> placed the necessary references back to the correct documentation.
>
> Could someone please review this and make sure it's in line with
> quality/expectations etc.
>
> Cal
>
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From vbvbrj at gmail.com  Sat Dec 29 21:51:23 2012
From: vbvbrj at gmail.com (Mimiko)
Date: Sat, 29 Dec 2012 20:51:23 +0200
Subject: [Freeswitch-users] Specifying file permission for audio records.
Message-ID: <50DF3BAB.4040209@gmail.com>

Hello.

I want audio files recorded from callcenter to be available to a web 
server. Both webserver and FS runs on same server. The storage directory 
for holding recorded audio files is on the same server. FS run in 
unprivileged mode, and the directory for recorded audio files have 
permission 775. Webserver runs unprivileged under another user.

The problem is that created recorded audio files by FS get the 770 
permission. How to change this behavior, so permission of created files 
will inherit directorys permision?

-- 
Mimiko desu.


From cal.leeming at simplicitymedialtd.co.uk  Sat Dec 29 22:08:00 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sat, 29 Dec 2012 19:08:00 +0000
Subject: [Freeswitch-users] mod_xml_curl + variables documentation
In-Reply-To: <CALNZuwmK19xpW0txsb7ToqDtQF_9gZKjguFf7ehH4VkRkRTVHg@mail.gmail.com>
References: <CAHKQagGWH7vt1AXtQMvpwcWU_ZkTiWxn00xVZfjbbs8m=RtagA@mail.gmail.com>
	<CDA1C659-889E-428C-B713-0BDA11AA7193@gmail.com>
	<CAHKQagF4jK5pTBReedetuetnoZEk0Oqfb7mO8WyNgpWPhF=dRw@mail.gmail.com>
	<CAHKQagG_-cN6wPza6_n5G1H14MUCeBR2-xGzTydH0yENreQOgg@mail.gmail.com>
	<CALNZuwmK19xpW0txsb7ToqDtQF_9gZKjguFf7ehH4VkRkRTVHg@mail.gmail.com>
Message-ID: <CAHKQagGBAFcoLzprDOFsdEA-HvbqgUk9enq1oqZrbVZ6Y_uSCg@mail.gmail.com>

Ah bollocks :/

Should I revert my changes?? I'm happy to re-edit stuff, I'd just need a
nudge in the right direction to say where stuff goes etc.

Cal

On Sat, Dec 29, 2012 at 6:37 PM, Avi Marcus <avi at avimarcus.net> wrote:

> The only thing is.. it looks like this list is read-only variable.
>
> Most channel variable can be changed to get a result, but these are *
> informational* variables ("where did the call come from?) and therefore
> unchangeable.
> They should be presented differently than the others, I'd imagine...
>
> -Avi
>
>
> On Sat, Dec 29, 2012 at 8:21 PM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> I've made some further improvements.
>>
>> The channel variables page now contains placeholders for all the
>> variables that were missing (approx 119 added).
>>
>> http://wiki.freeswitch.org/wiki/Channel_Variables
>>
>> I have also created/updated the following variable documentation pages
>> with the info collected from mod_xml_curl docs
>>
>> http://wiki.freeswitch.org/wiki/Variable_sip_auth_realm
>> http://wiki.freeswitch.org/wiki/Variable_sip_auth_username
>> http://wiki.freeswitch.org/wiki/Variable_sip_auth_method
>> http://wiki.freeswitch.org/wiki/Variable_sip_user_agent
>> http://wiki.freeswitch.org/wiki/Variable_sip_request_host
>> http://wiki.freeswitch.org/wiki/Variable_sip_request_port
>> http://wiki.freeswitch.org/wiki/Variable_sip_profile
>> http://wiki.freeswitch.org/wiki/Variable_sip_from_port
>> http://wiki.freeswitch.org/wiki/Variable_sip_from_user
>> http://wiki.freeswitch.org/wiki/Variable_sip_from_host
>> http://wiki.freeswitch.org/wiki/Variable_sip_to_host
>> http://wiki.freeswitch.org/wiki/Variable_sip_to_port
>> http://wiki.freeswitch.org/wiki/Variable_sip_to_user
>> http://wiki.freeswitch.org/wiki/Variable_sip_contact_host
>> http://wiki.freeswitch.org/wiki/Variable_sip_contact_port
>> http://wiki.freeswitch.org/wiki/Variable_sip_contact_user
>>
>> As such, I have also removed the field descriptions from mod_xml_curl and
>> placed the necessary references back to the correct documentation.
>>
>> Could someone please review this and make sure it's in line with
>> quality/expectations etc.
>>
>> Cal
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 areski at gmail.com  Sat Dec 29 23:48:20 2012
From: areski at gmail.com (Areski)
Date: Sat, 29 Dec 2012 21:48:20 +0100
Subject: [Freeswitch-users] stop a running lua script from luarun
In-Reply-To: <A391089F-7CCF-4766-A151-73A9A7C2C4A7@gmail.com>
References: <4F13F527-1F4A-41D5-81B6-4D7578D2426D@gmail.com>
	<CAKzWOxVsPdvHtF1Hf238qH=OjxjL=1LvRF_0bb=_D0Pzp=5k=A@mail.gmail.com>
	<A391089F-7CCF-4766-A151-73A9A7C2C4A7@gmail.com>
Message-ID: <CAMf7LBh=Ycb52y-YKUusEmnxQ412JwCHvuzFxkApU3hTWV53pA@mail.gmail.com>

I use to send a custom even to stop a lua script which run an infinite loop
:
luarun myscript.lua stop

if argv[1] then
    i=1
    while argv[i] do
        if argv[i] == "stop" then
            --Send Stop message
            local event = freeswitch.Event("custom", "lua::stop_event")
            event:addHeader("Action", "stop")
            event:fire()
            logger("Sent stop message to lua script")
            return
        end
        i=i+1
    end
    return
end

and then in the loop I listen for those events!

you can find a full example here :
https://github.com/Star2Billing/newfies-dialer/blob/develop/lua/listener.lua



On Wed, Dec 26, 2012 at 8:12 PM, Muhammad Naseer Bhatti
<nbhatti at gmail.com>wrote:

>
> No, I am running this loop myself. But I found other way to terminate the
> script, like look for a condition  in each iteration of the loop and if
> condition is true terminate the loop. But the question remains the same, if
> there is a way to figure out which scripts are running  to stop them if
> possible. Since all process are spawn by freeswitch top -H is not a handy
> solution.
>
>
> Thanks,
> --
> Muhammad Naseer Bhatti
>
>
>
> On Dec 26, 2012, at 8:14 PM, Michael Collins <msc at freeswitch.org> wrote:
>
> Have you been able to determine why it's stuck in an infinite loop? That
> would be a better long-term solution. In the meantime I think you can do
> top -H to see individual threads but I don't know how easy/difficult it is
> to pick out which one is running Lua.
> -MC
>
> On Sun, Dec 23, 2012 at 2:21 AM, Muhammad Naseer Bhatti <nbhatti at gmail.com
> > wrote:
>
>>
>> Looks like simple one, but I can't figure it out if there a way to stop a
>> lua script invoked via luarun which is stuck in an infinite loop and won't
>> terminate itself? Of if we can come up with a way to monitor the thread and
>> stop it?
>>
>> Thanks,
>> --
>> Muhammad Naseer Bhatti
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
> http://www.ClueCon.com <http://www.cluecon.com/>
> http://www.OSTAG.org <http://www.ostag.org/>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Kinds regards,
/Areski

----
*Arezqui Belaid*,
areski at gmail.com / +34650784355

Founder at Star2Billing (www.star2billing.com)
Author @ A2Billing (www.a2billing.net), @ Newfies-Dialer (
www.newfies-dialer.org), @ CDR-Stats (www.cdr-stats.org)
----
Twitter : http://twitter.com/areskib / LinkedIN :
http://www.linkedin.com/in/areski
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From lloyd.aloysius at gmail.com  Sun Dec 30 00:17:44 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Sat, 29 Dec 2012 16:17:44 -0500
Subject: [Freeswitch-users] xml_curl - switch.conf
In-Reply-To: <CAHKQagFe7qUJ7r3mcKJpFcai1cw=tzeiAVOAeCe1RE-4sEj7QQ@mail.gmail.com>
References: <CAGvuWYYyBv=ywC39WQ4MkG8C6rknsoXQP9XcrV7+srk+PQ2Uuw@mail.gmail.com>
	<CAHKQagFe7qUJ7r3mcKJpFcai1cw=tzeiAVOAeCe1RE-4sEj7QQ@mail.gmail.com>
Message-ID: <CAGvuWYbnDYCyo1KT1Yx9FH7JCbcn+vkgrvEJzJ9QzzaFFR6MMA@mail.gmail.com>

- freeswitch start without a switch.conf
- by default without switch.conf,use the sqlite database for core

-we can provide the switch.conf parameters and values through xml_curl.
using the post_load_switch.conf
-there is no way to provide the core-db-dsn through the
post_load_switch.conf
-so i have now only one line in my switch.conf that is <param
name="core-db-dsn" value="dsn:user:password" />




On Sat, Dec 29, 2012 at 1:03 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Yes you can send back switch.conf, but you need to do it via post_load.
>
> Take a look at "post_load configuration" under;
> http://wiki.freeswitch.org/wiki/Mod_xml_curl
>
> Although it does seem to take note of the variables sent back, I'm not
> 100% if all of them are applied in post_load (i.e. there may be some that
> don't get applied via this method).
>
> You'll have to use some trial and error to see (update the docs if there
> is anything specific that doesn't get applied using post_load).
>
> Let me know if this doesn't make any sense
>
> Cal
>
>
> On Sat, Dec 29, 2012 at 4:58 PM, Lloyd Aloysius <lloyd.aloysius at gmail.com>wrote:
>
>> Hi,
>>
>> Can we provide the switch.conf through xml_curl. I see freeswitch not
>> requesting the switch.conf?
>>
>> I would like to eliminate all the .conf files except the xml_curl.conf.
>> Does this possible? Which .conf files cannot use through the xml_curl?
>>
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 gabe at gundy.org  Sun Dec 30 03:30:32 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 29 Dec 2012 17:30:32 -0700
Subject: [Freeswitch-users] Specifying file permission for audio records.
In-Reply-To: <50DF3BAB.4040209@gmail.com>
References: <50DF3BAB.4040209@gmail.com>
Message-ID: <CAMwZ-tLTfJcG39tM6YRO7RmnHPy2P_=KD=S_zf86W11ZHO0VLw@mail.gmail.com>

On Sat, Dec 29, 2012 at 11:51 AM, Mimiko <vbvbrj at gmail.com> wrote:
> I want audio files recorded from callcenter to be available to a web
> server. Both webserver and FS runs on same server. The storage directory
> for holding recorded audio files is on the same server. FS run in
> unprivileged mode, and the directory for recorded audio files have
> permission 775. Webserver runs unprivileged under another user.

Something like this?

http://en.wikipedia.org/wiki/Setuid#setuid_and_setgid_on_directories


Gabe


From lloyd.aloysius at gmail.com  Sun Dec 30 05:50:08 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Sat, 29 Dec 2012 21:50:08 -0500
Subject: [Freeswitch-users] xml_curl - switch.conf
In-Reply-To: <CAGvuWYbnDYCyo1KT1Yx9FH7JCbcn+vkgrvEJzJ9QzzaFFR6MMA@mail.gmail.com>
References: <CAGvuWYYyBv=ywC39WQ4MkG8C6rknsoXQP9XcrV7+srk+PQ2Uuw@mail.gmail.com>
	<CAHKQagFe7qUJ7r3mcKJpFcai1cw=tzeiAVOAeCe1RE-4sEj7QQ@mail.gmail.com>
	<CAGvuWYbnDYCyo1KT1Yx9FH7JCbcn+vkgrvEJzJ9QzzaFFR6MMA@mail.gmail.com>
Message-ID: <CAGvuWYaMYN60TQukMSPnk2L5z9KiPRi80_m-UGrtbCmR6v7yXw@mail.gmail.com>

This may help someone in future. I completely switch everything using
xml_curl

But we need to the following 3 files in
the /usr/local/freeswitch/conf/autoload_configs

1. *modules.conf.xml* - only one module need to be there. Other modules can
be load using post_load_modules_conf request

<load module="mod_xml_curl"/>

2.*switch.conf* - only one line need to be there. Other configuration load
thorugh the post_load_switch_conf request

<param name="core-db-dsn" value="dsn:user:password" />

3. *xml_curl.con.**xm*l


*
*


On Sat, Dec 29, 2012 at 4:17 PM, Lloyd Aloysius <lloyd.aloysius at gmail.com>wrote:

> - freeswitch start without a switch.conf
> - by default without switch.conf,use the sqlite database for core
>
> -we can provide the switch.conf parameters and values through xml_curl.
> using the post_load_switch.conf
> -there is no way to provide the core-db-dsn through the
> post_load_switch.conf
> -so i have now only one line in my switch.conf that is <param
> name="core-db-dsn" value="dsn:user:password" />
>
>
>
>
> On Sat, Dec 29, 2012 at 1:03 PM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> Yes you can send back switch.conf, but you need to do it via post_load.
>>
>> Take a look at "post_load configuration" under;
>> http://wiki.freeswitch.org/wiki/Mod_xml_curl
>>
>> Although it does seem to take note of the variables sent back, I'm not
>> 100% if all of them are applied in post_load (i.e. there may be some that
>> don't get applied via this method).
>>
>> You'll have to use some trial and error to see (update the docs if there
>> is anything specific that doesn't get applied using post_load).
>>
>> Let me know if this doesn't make any sense
>>
>> Cal
>>
>>
>> On Sat, Dec 29, 2012 at 4:58 PM, Lloyd Aloysius <lloyd.aloysius at gmail.com
>> > wrote:
>>
>>> Hi,
>>>
>>> Can we provide the switch.conf through xml_curl. I see freeswitch not
>>> requesting the switch.conf?
>>>
>>> I would like to eliminate all the .conf files except the xml_curl.conf.
>>> Does this possible? Which .conf files cannot use through the xml_curl?
>>>
>>>
>>> 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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 a.afzali2003 at gmail.com  Sun Dec 30 13:06:31 2012
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sun, 30 Dec 2012 13:36:31 +0330
Subject: [Freeswitch-users] Mod event zmq Status
Message-ID: <CABK0fC5aH7N+Gi1j4m2SLnPzTrr_oWmW5pTvgkEbP+VRcFCm5g@mail.gmail.com>

Hi Guys,

As stated in the wiki there are chances for crash in case of fork() system
calls. Does it mean that is not suitable to use in production ?

BEST,
-- Afshin
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From juanito1982 at gmail.com  Sun Dec 30 19:18:37 2012
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Sun, 30 Dec 2012 17:18:37 +0100
Subject: [Freeswitch-users] Mod event zmq Status
In-Reply-To: <CABK0fC5aH7N+Gi1j4m2SLnPzTrr_oWmW5pTvgkEbP+VRcFCm5g@mail.gmail.com>
References: <CABK0fC5aH7N+Gi1j4m2SLnPzTrr_oWmW5pTvgkEbP+VRcFCm5g@mail.gmail.com>
Message-ID: <CAHLYDCtUteQKRzHVb-L=387wA-m-oAgA5z6k=jqfgT3Jt0quyQ@mail.gmail.com>

No, it has lacks of stability so I would not use it in a production
enviroment.

Regars

2012/12/30 afshin afzali <a.afzali2003 at gmail.com>

> Hi Guys,
>
> As stated in the wiki there are chances for crash in case of fork() system
> calls. Does it mean that is not suitable to use in production ?
>
> BEST,
> -- Afshin
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 david.villasmil.work at gmail.com  Sun Dec 30 19:38:14 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 30 Dec 2012 17:38:14 +0100
Subject: [Freeswitch-users] Kudos to the FreSWITCH team!
Message-ID: <CAFGRPVqkGgxkvCxQzJBzMDb4Ed3Rpm7Fk6rCFJNtO+s-jyQz6Q@mail.gmail.com>

Hello FS team,


I'm just writing thie short email to congratulate you guys for the amazing
work you're doing with FS since the very beginning.

As some of you may know, some time ago I developed the FreeSWITC-Billing
platform on top of FreeSWITCH (although it's been some time since I updated
it).

Some of my customers have been using it for 2 years now and have good
traffic running on it.

Long story short, I was called by 2 of my customers who were in turn called
by a couple of their providers asking to please, please, pretty please slow
doen the rate with which my customers were sending calls to them or to
limit the channels, because my customers were bringing their soft-switches
DOWN... I don't know the  other switch's brand , but I know one is a
Quintum.

So, like I said: KUDOS!


Cheers to all and Happy New Year!!


David Villasmil
https://github.com/davidcsi/FreeSWITCH-Billing
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From cal.leeming at simplicitymedialtd.co.uk  Sun Dec 30 20:57:20 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 30 Dec 2012 17:57:20 +0000
Subject: [Freeswitch-users] mod_xml_curl + variables documentation
In-Reply-To: <CAHKQagGBAFcoLzprDOFsdEA-HvbqgUk9enq1oqZrbVZ6Y_uSCg@mail.gmail.com>
References: <CAHKQagGWH7vt1AXtQMvpwcWU_ZkTiWxn00xVZfjbbs8m=RtagA@mail.gmail.com>
	<CDA1C659-889E-428C-B713-0BDA11AA7193@gmail.com>
	<CAHKQagF4jK5pTBReedetuetnoZEk0Oqfb7mO8WyNgpWPhF=dRw@mail.gmail.com>
	<CAHKQagG_-cN6wPza6_n5G1H14MUCeBR2-xGzTydH0yENreQOgg@mail.gmail.com>
	<CALNZuwmK19xpW0txsb7ToqDtQF_9gZKjguFf7ehH4VkRkRTVHg@mail.gmail.com>
	<CAHKQagGBAFcoLzprDOFsdEA-HvbqgUk9enq1oqZrbVZ6Y_uSCg@mail.gmail.com>
Message-ID: <CAHKQagGyiE9sm1Kn7hqWuq-C-wXaMGUtENTEjJ=jdTUCbom97g@mail.gmail.com>

Just reviewed the Channel Variables wiki again, it seems that the
"Information Variables" section already had some overlapping variables
(ones which may be considered read only etc).

Therefore, I'm going to leave my changes in there for now, and no doubt
they will get separated out in the future.

I had accidently included some mod_xml_curl specific variables in there,
and those have now been removed.

Cal

On Sat, Dec 29, 2012 at 7:08 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Ah bollocks :/
>
> Should I revert my changes?? I'm happy to re-edit stuff, I'd just need a
> nudge in the right direction to say where stuff goes etc.
>
> Cal
>
> On Sat, Dec 29, 2012 at 6:37 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> The only thing is.. it looks like this list is read-only variable.
>>
>> Most channel variable can be changed to get a result, but these are *
>> informational* variables ("where did the call come from?) and therefore
>> unchangeable.
>> They should be presented differently than the others, I'd imagine...
>>
>> -Avi
>>
>>
>> On Sat, Dec 29, 2012 at 8:21 PM, Cal Leeming [Simplicity Media Ltd] <
>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>
>>> I've made some further improvements.
>>>
>>> The channel variables page now contains placeholders for all the
>>> variables that were missing (approx 119 added).
>>>
>>> http://wiki.freeswitch.org/wiki/Channel_Variables
>>>
>>> I have also created/updated the following variable documentation pages
>>> with the info collected from mod_xml_curl docs
>>>
>>> http://wiki.freeswitch.org/wiki/Variable_sip_auth_realm
>>> http://wiki.freeswitch.org/wiki/Variable_sip_auth_username
>>> http://wiki.freeswitch.org/wiki/Variable_sip_auth_method
>>> http://wiki.freeswitch.org/wiki/Variable_sip_user_agent
>>> http://wiki.freeswitch.org/wiki/Variable_sip_request_host
>>> http://wiki.freeswitch.org/wiki/Variable_sip_request_port
>>> http://wiki.freeswitch.org/wiki/Variable_sip_profile
>>> http://wiki.freeswitch.org/wiki/Variable_sip_from_port
>>> http://wiki.freeswitch.org/wiki/Variable_sip_from_user
>>> http://wiki.freeswitch.org/wiki/Variable_sip_from_host
>>> http://wiki.freeswitch.org/wiki/Variable_sip_to_host
>>> http://wiki.freeswitch.org/wiki/Variable_sip_to_port
>>> http://wiki.freeswitch.org/wiki/Variable_sip_to_user
>>> http://wiki.freeswitch.org/wiki/Variable_sip_contact_host
>>> http://wiki.freeswitch.org/wiki/Variable_sip_contact_port
>>> http://wiki.freeswitch.org/wiki/Variable_sip_contact_user
>>>
>>> As such, I have also removed the field descriptions from mod_xml_curl
>>> and placed the necessary references back to the correct documentation.
>>>
>>> Could someone please review this and make sure it's in line with
>>> quality/expectations etc.
>>>
>>> Cal
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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 cal.leeming at simplicitymedialtd.co.uk  Sun Dec 30 21:04:57 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 30 Dec 2012 18:04:57 +0000
Subject: [Freeswitch-users] mod_xml_curl - missing dialplan variable
 (domain of originating user for blind transfer)
In-Reply-To: <CAHKQagG36-_01XRP_75VPqmN_2QxpSbvtCKm7NXEC=2+EczEWQ@mail.gmail.com>
References: <CAHKQagGSpjOzhxFo-5ga-XqrPz77+gdrjYU4T7d7vUyJLv-PNw@mail.gmail.com>
	<CAHKQagFk4=zjboAJJuYx2gqX6vcZvYR-3MBKCJi-j7bfOSWtoA@mail.gmail.com>
	<CAHKQagFrWbRft0wLiPsML22K61mfM+Z3Qh1ZJC+aKiebnNB+GA@mail.gmail.com>
	<CAHKQagHvQwWCOvdM0abqrM1xz65kFz9hY6J+fMKbJjH=Ot=kgQ@mail.gmail.com>
	<CAHKQagHowbH_Zr8YPWGDsbug4m1ggcaTY=LdgzYf9yvJq15qFA@mail.gmail.com>
	<CAKbxfG8vrDExvtbt-W--u0AizwTR+S0x7XqoPrA1FXms=Yq-gw@mail.gmail.com>
	<CAHKQagG36-_01XRP_75VPqmN_2QxpSbvtCKm7NXEC=2+EczEWQ@mail.gmail.com>
Message-ID: <CAHKQagGSrZZ3YysjMh51V7zPbw-dPjdTE=5XqKMQgwRMp-B4Rg@mail.gmail.com>

I have started to move some of this information into the wiki.

However, my previous comments about
forcing sip_auth_realm/sip_auth_username are generally considered unsafe
from a security point of view.

If you force those variables, under some circumstances a blind transferred
call from an external gateway could later on be considered an authorized
internal user and lead to undesired behaviour.

Forcing sip_invite_domain however is acceptable, as some request context
will lose this information during a blind transfer or bridge.

I am currently building up a spreadsheet of different call scenarios, along
with the request variables for each one, so it is easy to see how the
variables change depending on what event is happening.

Cal

On Tue, Nov 20, 2012 at 1:21 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Hi Anthony,
>
> Thanks for the reply, yeah I spent some time looking at alternative ways
> and came up with one that seems to get the job done
>
> I am convinced now that there is no bug as such - but one thing that is
> absolutely clear, is that mod_xml_curl is in desperate need of
> normalization.. even if it was fully documented, some of the variants don't
> make logical sense. However, the information is there, and as long as you
> do the correct conditional checks, then it will work flawlessly.
>
> To make the authenticated domain stick when passing through to a gateway,
> we use the following;
>
> <action application="bridge"
> data="{sip_auth_realm=${sip_auth_realm}}{sip_auth_username=${sip_auth_username}}{sip_invite_domain=${sip_from_host}}sofia/gateway/{{
> dst.gateway.name
> }}/{{dst.gateway.fs_dial_prefix}}{{dst.dst_number_e164}}"/>
>
> The most important part of the above being;
>
> {sip_auth_realm=${sip_auth_realm}}{sip_auth_username=${sip_auth_username}}{sip_invite_domain=${sip_from_host}}
>
> However, this alone doesn't deal with the fact that you have to look in
> different places for the correct variables depending on what the current
> call context is.
>
> To a certain extent, your own business logic will also determine which
> variables should be used, and a combination of conditional checks may need
> to be used in order to figure out which variable you should be using in the
> first place lol.
>
> Here are the patterns we have found so far - this is just an information
> dump for now, and later down the road I will update the mod_xml_curl
> documentation.
>
> # Ensure that variable_sip_auth_username / variable_sip_auth_realm
> #
> # If Call-Direction is inbound, then;
> # src_user = variable_sip_auth_username
> # src_domain = variable_sip_auth_realm
> # dst_user = variable_sip_to_user
> #
> # If Call-Direction is outbound, then;
> # originate_user = variable_sip_auth_username
> # originate_domain = variable_sip_auth_realm
> # src_user = variable_sip_to_user
> # src_domain = variable_sip_to_host
> # dst_user = Caller-Destination-Number
>
> * gateway to gateway (442476100401 > 442476100402)
> * domain to gateway - blind xfer to gateway (2000 > 442476100401
> > 442476100402)
> * domain to gateway - blind xfer to domain (2000 > 442476100401 > 2002)
> * domain to gateway (2000 > 442476100401)
> * domain to domain (2000 > 2001)
>
> ----
>
> # Check if variable_sip_to_host is present and known gateway
> #
> # If variable_dialed_user and variable_dialed_domain are present;
> # originate_user = variable_dialed_user
> # originate_domain = variable_dialed_domain
> # src_user = variable_sip_from_user
> # src_domain = variable_sip_to_host
> # dst_user = Caller-Destination-Number
> #
> # If not present;
> # src_user = variable_sip_from_user
> # src_domain = variable_sip_to_host
> # dst_user = variable_sip_to_user
>
> * gateway to domain (442476100401 > 2002)
> * gateway to domain - blind xfer to gateway (442476100401 > 2000
> > 442476100402)
> * gateway to domain - blind xfer to domain (442476100401 > 2000 > 2002)
>
> ----
>
> # check for variable_dialed_domain
> # If match;
> # src_user = variable_dialed_user
> # src_domain  = variable_dialed_domain
> # dst_user = Caller-Destination-Number
>
> * domain to domain to gateway (2000 > 2001 > 442476100402)
>
>
> On Tue, Nov 20, 2012 at 2:55 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> One thing you can do is set the variable when you do know it, from the
>> dialplan.  So it will be there in all the subsequent events.
>>
>>
>>
>> On Mon, Nov 19, 2012 at 1:34 AM, Cal Leeming [Simplicity Media Ltd] <
>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>
>>> Last update, then I'm really going to sleep!
>>>
>>> Apologies for the noise btw, in hindsight I should have collected all
>>> this info and posted in one go.
>>>
>>> I've tried enabling auth-all-packets (along with auth_calls), as I
>>> thought maybe having authentication on REFER packets might make a
>>> difference, but sadly it had no effect (the SIP headers show
>>> proxy-authorization in the REFER, but nothing extra showed in mod_xml_curl)
>>>
>>> I've managed to narrow down the circumstances in which this happens;
>>>
>>> CORRECT:
>>>
>>> * User receives call from gateway, blind transfer to another user (shows
>>> correctly as variable_dialed_user/variable_dialed_domain)
>>> * User receives call from another user, blind transfer to gateway (shows
>>> correctly as variable_dialed_user/variable_dialed_domain)
>>> * User receives call from another user, blind transfer to another
>>> user (shows correctly as variable_dialed_user/variable_dialed_domain)
>>> * User makes call to another user, blind transfer to another user (shows
>>> correctly as variable_dialed_user/variable_dialed_domain)
>>> * User makes call to another user, blind transfer to a gateway (shows
>>> correctly as variable_dialed_user/variable_dialed_domain)
>>>
>>> MISSING:
>>>
>>> * User makes call to a gateway, blind transfer to another gateway (no
>>> clean variables for domain)
>>> * User makes call to a gateway, blind transfer to another user (no clean
>>> variables for domain)
>>>
>>> So, the problem seems to happen specifically when you blind transfer a
>>> call that is already in progress on a gateway, where the call was
>>> originated by a user and not a gateway.
>>>
>>> I did a bit more looking through code, added a few switch_log_printf()
>>> calls, and found that the following method is NOT called in those two
>>> scenarios where these variables are missing;
>>> mod_dptools.c: "switch_call_cause_t user_outgoing_channel"
>>>
>>> This is about as far as I can go on this, as I just don't know enough
>>> about C to give any more in-depth info :/
>>>
>>> Cal
>>>
>>> On Mon, Nov 19, 2012 at 5:52 AM, Cal Leeming [Simplicity Media Ltd] <
>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>
>>>> Sorry, another update.. after tinkering with the SIP headers, we found
>>>> that we're able to pass any user/host along in an INVITE, and this is
>>>> passed onto mod_xml_curl.
>>>>
>>>> To fix this particular part of the problem, we enabled auth_calls and
>>>> this gives us correct/clean variables we can trust, specifically;
>>>>
>>>>  u'variable_sip_auth_username': u'2000',
>>>>  u'variable_user_name': u'2000',
>>>>
>>>> However, when attempting to do the blind transfer again, this variables
>>>> are all missing.
>>>>
>>>> At this point I'm convinced that attempting to extract the user/domain
>>>> from the Refer headers is probably not the right thing to do... But I'm
>>>> still no closer to figuring out what the correct approach should be to
>>>> expose the authenticated user/domain to mod_xml_curl.
>>>>
>>>> Cal
>>>>
>>>>
>>>> On Mon, Nov 19, 2012 at 4:58 AM, Cal Leeming [Simplicity Media Ltd] <
>>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>
>>>>> Another quick update on this before I pass out from lack of sleep..!
>>>>>
>>>>> Just had a look through the mod_sofia.c/h source and found the
>>>>> following;
>>>>>
>>>>> mod_sofia.c/mod_sofia.h
>>>>> #define SOFIA_REFER_TO_VARIABLE "sip_refer_to"
>>>>> if (!zstr(full_ref_by)) {
>>>>> switch_channel_set_variable(t_channel, SOFIA_SIP_HEADER_PREFIX
>>>>> "Referred-By", full_ref_by);
>>>>> }
>>>>> if (!zstr(full_ref_to)) {
>>>>> switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
>>>>> full_ref_to);
>>>>> }
>>>>> if (!zstr(full_ref_to)) {
>>>>>  switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
>>>>> full_ref_to);
>>>>> }
>>>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>>>>> SWITCH_LOG_DEBUG, "Process REFER to [%s@%s]\n", exten, (char *)
>>>>> refer_to->r_url->url_host);
>>>>>
>>>>> If the correct approach is deemed to be patching code, then I figured
>>>>> it could be as simple as this;
>>>>>
>>>>> switch_channel_set_variable(t_channel, "Referred-By-User", exten);
>>>>> switch_channel_set_variable(t_channel, "Referred-By-Domain", (char *)
>>>>> refer_to->r_url->url_host);
>>>>>
>>>>> This is pretty much where my knowledge of C ends, I can (just about)
>>>>> read and copy chunks of C code, but that's about it :)
>>>>>
>>>>> Cal
>>>>>
>>>>> On Mon, Nov 19, 2012 at 4:38 AM, Cal Leeming [Simplicity Media Ltd] <
>>>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>>
>>>>>> Not sure if this is relevant but thought I'd point it out.
>>>>>>
>>>>>> The following field seems to contain the IP of the domain we were
>>>>>> expecting ('c1881.voiceflare.co.uk')
>>>>>>
>>>>>>  u'variable_sip_from_host': u'89.238.182.137',
>>>>>>
>>>>>> Normally, this field would contain the hostname and not the IP.
>>>>>>
>>>>>> Cal
>>>>>>
>>>>>> On Mon, Nov 19, 2012 at 4:34 AM, Cal Leeming [Simplicity Media Ltd] <
>>>>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>>>
>>>>>>> Hi guys,
>>>>>>>
>>>>>>> In a nut shell, it appears that when attempting to perform a blind
>>>>>>> transfer under certain conditions (explained below), mod_xml_curl does not
>>>>>>> expose the originating domain in a clean format.
>>>>>>>
>>>>>>> My initial plan was to find the point where these variable were
>>>>>>> being generated, look at what was available, then add an extra variable for
>>>>>>> the domain and submit a patch.
>>>>>>>
>>>>>>> Sadly my C isn't great and I hit a brick wall, so if anyone can help
>>>>>>> out, I will ensure the mod_xml_curl documentation is updated and/or assist
>>>>>>> with any patching/testing required.
>>>>>>>
>>>>>>>  Please take the following scenario;
>>>>>>>
>>>>>>> * Extension 2000 calls an external number via a gateway (i.e. bridge
>>>>>>> sofia/gateway/name/e164_number_here).
>>>>>>> * Call connects fine, audio stays good, no disconnection problems
>>>>>>> etc.
>>>>>>> * Call is blind transferred to another extension
>>>>>>>
>>>>>>> As a result, the following is determined;
>>>>>>>
>>>>>>> * User initiating the blind transfer is 2000
>>>>>>> * Domain initiating the blind transfer is c1881.voiceflare.co.uk
>>>>>>> * Destination number of the call is 447866123456
>>>>>>> * Number to blind transfer to is 2001
>>>>>>> * Call to mod_xml_curl is made
>>>>>>>
>>>>>>> It makes reference to the User in the following 'clean' variables
>>>>>>> (by clean, I mean variables that just contain 2000 and don't require
>>>>>>> mangling to extract the info);
>>>>>>>
>>>>>>>  u'Caller-ANI': u'2000',
>>>>>>>  u'Caller-Username': u'2000',
>>>>>>>  u'Caller-Caller-ID-Number': u'2000',
>>>>>>>  u'Hunt-ANI': u'2000',
>>>>>>>  u'Hunt-Caller-ID-Number': u'2000',
>>>>>>>  u'Hunt-Username': u'2000',
>>>>>>>  u'variable_last_sent_callee_id_number': u'2000',
>>>>>>>  u'variable_sip_from_user': u'2000',
>>>>>>>
>>>>>>> It also has the User in the following unclean variables;
>>>>>>>
>>>>>>>  u'variable_bridge_channel': u'sofia/external/
>>>>>>> 2000 at c1881.voiceflare.co.uk:5060',
>>>>>>>  u'variable_sip_from_uri': u'2000 at 89.238.182.137',
>>>>>>>  u'variable_sip_full_from': u'"foxx" <sip:2000 at 89.238.182.137
>>>>>>> >;tag=XryjFQp3rB2NF',
>>>>>>>  u'variable_sip_h_Referred-By': u'"foxx" <
>>>>>>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>>>>>>
>>>>>>> However, it only references the domain in the following unclean
>>>>>>> variables;
>>>>>>>
>>>>>>>  u'variable_bridge_channel': u'sofia/external/
>>>>>>> 2000 at c1881.voiceflare.co.uk:5060',
>>>>>>>  u'variable_sip_h_Referred-By': u'"foxx" <
>>>>>>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>>>>>>  u'variable_sip_refer_to': u'<sip:2001 at c1881.voiceflare.co.uk:5060
>>>>>>> >',
>>>>>>>
>>>>>>> Lets say that we want to determine the user/domain that has
>>>>>>> initiated this transfer, doing so would mean mangling with one of those
>>>>>>> above variables, which seems a bit dirty (plus it is not clear which is the
>>>>>>> correct one to use).
>>>>>>>
>>>>>>> I have attached the SIP trace of the entire blind transfer event,
>>>>>>> and the full mod_xml_curl request being sent.
>>>>>>>
>>>>>>> If anyone could answer the following, it'd be much appreciated;
>>>>>>>
>>>>>>> * Should there be an improvement patch that exposes the domain of
>>>>>>> the user that originated the blind transfer?
>>>>>>> * Are there better/alternative ways to extracting the domain of the
>>>>>>> user that originated the blind transfer?
>>>>>>>
>>>>>>> Many thanks
>>>>>>>
>>>>>>> Cal
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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  Sun Dec 30 21:27:40 2012
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sun, 30 Dec 2012 21:57:40 +0330
Subject: [Freeswitch-users] Mod event zmq Status
In-Reply-To: <CAHLYDCtUteQKRzHVb-L=387wA-m-oAgA5z6k=jqfgT3Jt0quyQ@mail.gmail.com>
References: <CABK0fC5aH7N+Gi1j4m2SLnPzTrr_oWmW5pTvgkEbP+VRcFCm5g@mail.gmail.com>
	<CAHLYDCtUteQKRzHVb-L=387wA-m-oAgA5z6k=jqfgT3Jt0quyQ@mail.gmail.com>
Message-ID: <CABK0fC5r4dynsnat_twTgwjRM+oTNc+6Qc48W_5aj__Np3LfjA@mail.gmail.com>

Thanks :)


On Sun, Dec 30, 2012 at 7:48 PM, Juan Antonio Iba?ez Santorum <
juanito1982 at gmail.com> wrote:

> No, it has lacks of stability so I would not use it in a production
> enviroment.
>
> Regars
>
> 2012/12/30 afshin afzali <a.afzali2003 at gmail.com>
>
>> Hi Guys,
>>
>> As stated in the wiki there are chances for crash in case of fork()
>> system calls. Does it mean that is not suitable to use in production ?
>>
>> BEST,
>> -- Afshin
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cal.leeming at simplicitymedialtd.co.uk  Sun Dec 30 22:12:37 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 30 Dec 2012 19:12:37 +0000
Subject: [Freeswitch-users] Naming convention - sip_profile_name /
	sofia_profile_name / sip_gateway_name
Message-ID: <CAHKQagEGJHphMxXExq6DiJo0E5oSw_4rMopVBgK2p_iZRPzP=A@mail.gmail.com>

Hello,

The following ticket has been raised in regards to the somewhat confusion
naming convention of sip_profile_name / sofia_profile_name /
sip_gateway_name.

http://jira.freeswitch.org/browse/FS-4980

To avoid confusion, I have also updated the documentation to reflect the
current behavior;

http://wiki.freeswitch.org/wiki/Variable_sofia_profile_name
http://wiki.freeswitch.org/wiki/Variable_sip_profile_name
http://wiki.freeswitch.org/wiki/Variable_sip_gateway_name

If anyone spots any mistakes in the above, please let me know.

Thanks

Cal
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From drk at drkngs.net  Sun Dec 30 22:38:07 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Sun, 30 Dec 2012 11:38:07 -0800
Subject: [Freeswitch-users] mod_xml_curl - missing dialplan variable
 (domain of originating user for blind transfer)
In-Reply-To: <CAHKQagGSrZZ3YysjMh51V7zPbw-dPjdTE=5XqKMQgwRMp-B4Rg@mail.gmail.com>
Message-ID: <20121230193807.7bfe4ec1@mail.tritonwest.net>

 
Since this is for use in external generated XML, and for the most part it should be code that is generating it, the easiest way to handle calls from gateways, is to back them with users in the directory. When A calll is from a gateway, simply do a set_user in your dialplan, and then a transfer back to the number dialed in the context ${user_context}, which will be set after you do the set_user.   
   
This way you don't have to do anything special for calls that come from a gateway, transfers and other things just work right.  
   
--Dave

      _____  

  From: Cal Leeming [Simplicity Media Ltd] [mailto:cal.leeming at simplicitymedialtd.co.uk]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Sun, 30 Dec 2012 10:04:57 -0800
Subject: Re: [Freeswitch-users] mod_xml_curl - missing dialplan variable (domain of originating user for blind transfer)

I have started to move some of this information into the wiki.  

  
However, my previous comments about forcing sip_auth_realm/sip_auth_username are generally considered unsafe from a security point of view.  

  
If you force those variables, under some circumstances a blind transferred call from an external gateway could later on be considered an authorized internal user and lead to undesired behaviour.  

  
Forcing sip_invite_domain however is acceptable, as some request context will lose this information during a blind transfer or bridge.  

  
I am currently building up a spreadsheet of different call scenarios, along with the request variables for each one, so it is easy to see how the variables change depending on what event is happening.  

  
Cal  

  
On Tue, Nov 20, 2012 at 1:21 PM, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:
  Hi Anthony,  

  
Thanks for the reply, yeah I spent some time looking at alternative ways and came up with one that seems to get the job done  

  
I am convinced now that there is no bug as such - but one thing that is absolutely clear, is that mod_xml_curl is in desperate need of normalization.. even if it was fully documented, some of the variants don't make logical sense. However, the information is there, and as long as you do the correct conditional checks, then it will work flawlessly.  

  
To make the authenticated domain stick when passing through to a gateway, we use the following;  

  
<action application="bridge" data="{sip_auth_realm=${sip_auth_realm}}{sip_auth_username=${sip_auth_username}}{sip_invite_domain=${sip_from_host}}sofia/gateway/{{dst.gateway.name}}/{{dst.gateway.fs_dial_prefix}}{{dst.dst_number_e164}}"/>

The most important part of the above being;  
{sip_auth_realm=${sip_auth_realm}}{sip_auth_username=${sip_auth_username}}{sip_invite_domain=${sip_from_host}}  

  
However, this alone doesn't deal with the fact that you have to look in different places for the correct variables depending on what the current call context is.  

  
To a certain extent, your own business logic will also determine which variables should be used, and a combination of conditional checks may need to be used in order to figure out which variable you should be using in the first place lol.  

  
Here are the patterns we have found so far - this is just an information dump for now, and later down the road I will update the mod_xml_curl documentation.  

  
# Ensure that variable_sip_auth_username / variable_sip_auth_realm  
#  
# If Call-Direction is inbound, then;  
# src_user = variable_sip_auth_username  
# src_domain = variable_sip_auth_realm  
# dst_user = variable_sip_to_user  
#  
# If Call-Direction is outbound, then;  
# originate_user = variable_sip_auth_username  
# originate_domain = variable_sip_auth_realm  
# src_user = variable_sip_to_user  
# src_domain = variable_sip_to_host  
# dst_user = Caller-Destination-Number  

  
* gateway to gateway (442476100401 > 442476100402)  
* domain to gateway - blind xfer to gateway (2000 > 442476100401 > 442476100402)  
* domain to gateway - blind xfer to domain (2000 > 442476100401 > 2002)  
* domain to gateway (2000 > 442476100401)  
* domain to domain (2000 > 2001)  

  
----  

  
# Check if variable_sip_to_host is present and known gateway  
#  
# If variable_dialed_user and variable_dialed_domain are present;  
# originate_user = variable_dialed_user  
# originate_domain = variable_dialed_domain  
# src_user = variable_sip_from_user  
# src_domain = variable_sip_to_host  
# dst_user = Caller-Destination-Number  
#  
# If not present;  
# src_user = variable_sip_from_user  
# src_domain = variable_sip_to_host  
# dst_user = variable_sip_to_user  

  
  
* gateway to domain (442476100401 > 2002)  
* gateway to domain - blind xfer to gateway (442476100401 > 2000 > 442476100402)  
* gateway to domain - blind xfer to domain (442476100401 > 2000 > 2002)  

  
----  

  
# check for variable_dialed_domain  
# If match;  
# src_user = variable_dialed_user  
# src_domain  = variable_dialed_domain  
# dst_user = Caller-Destination-Number  

  
  
* domain to domain to gateway (2000 > 2001 > 442476100402)  
  
  

  

  
On Tue, Nov 20, 2012 at 2:55 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
  One thing you can do is set the variable when you do know it, from the dialplan.  So it will be there in all the subsequent events.  

  


  
  
  
On Mon, Nov 19, 2012 at 1:34 AM, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:
    
  
Last update, then I'm really going to sleep!  

  
  
Apologies for the noise btw, in hindsight I should have collected all this info and posted in one go.  

  
I've tried enabling auth-all-packets (along with auth_calls), as I thought maybe having authentication on REFER packets might make a difference, but sadly it had no effect (the SIP headers show proxy-authorization in the REFER, but nothing extra showed in mod_xml_curl)  

  
I've managed to narrow down the circumstances in which this happens;  

  
CORRECT:  

* User receives call from gateway, blind transfer to another user (shows correctly as variable_dialed_user/variable_dialed_domain)  
* User receives call from another user, blind transfer to gateway (shows correctly as variable_dialed_user/variable_dialed_domain)  
* User receives call from another user, blind transfer to another user (shows correctly as variable_dialed_user/variable_dialed_domain)  
* User makes call to another user, blind transfer to another user (shows correctly as variable_dialed_user/variable_dialed_domain)  
* User makes call to another user, blind transfer to a gateway (shows correctly as variable_dialed_user/variable_dialed_domain)  

  
MISSING:  

  
* User makes call to a gateway, blind transfer to another gateway (no clean variables for domain)  
* User makes call to a gateway, blind transfer to another user (no clean variables for domain)  

  
So, the problem seems to happen specifically when you blind transfer a call that is already in progress on a gateway, where the call was originated by a user and not a gateway.  

  
I did a bit more looking through code, added a few switch_log_printf() calls, and found that the following method is NOT called in those two scenarios where these variables are missing;  
mod_dptools.c: "switch_call_cause_t user_outgoing_channel"  

  
This is about as far as I can go on this, as I just don't know enough about C to give any more in-depth info :/  

  
Cal  
  
  

  
On Mon, Nov 19, 2012 at 5:52 AM, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:
  Sorry, another update.. after tinkering with the SIP headers, we found that we're able to pass any user/host along in an INVITE, and this is passed onto mod_xml_curl.  

  
To fix this particular part of the problem, we enabled auth_calls and this gives us correct/clean variables we can trust, specifically;  

  
  
 u'variable_sip_auth_username': u'2000',  
 u'variable_user_name': u'2000',  

  
However, when attempting to do the blind transfer again, this variables are all missing.  

  
At this point I'm convinced that attempting to extract the user/domain from the Refer headers is probably not the right thing to do... But I'm still no closer to figuring out what the correct approach should be to expose the authenticated user/domain to mod_xml_curl.  

  
Cal  
  
  

  

  
On Mon, Nov 19, 2012 at 4:58 AM, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:
  Another quick update on this before I pass out from lack of sleep..!  

  
Just had a look through the mod_sofia.c/h source and found the following;  

  
  
mod_sofia.c/mod_sofia.h  
#define SOFIA_REFER_TO_VARIABLE "sip_refer_to"  
if (!zstr(full_ref_by)) {  
switch_channel_set_variable(t_channel, SOFIA_SIP_HEADER_PREFIX "Referred-By", full_ref_by);  
}  
if (!zstr(full_ref_to)) {  
switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE, full_ref_to);  
}  
if (!zstr(full_ref_to)) {  
switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE, full_ref_to);  
}  
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Process REFER to [%s@%s]\n", exten, (char *) refer_to->r_url->url_host);  

  
If the correct approach is deemed to be patching code, then I figured it could be as simple as this;  

  
  
switch_channel_set_variable(t_channel, "Referred-By-User", exten);  
switch_channel_set_variable(t_channel, "Referred-By-Domain", (char *) refer_to->r_url->url_host);  

  
This is pretty much where my knowledge of C ends, I can (just about) read and copy chunks of C code, but that's about it :)  

  
Cal  
  

  
On Mon, Nov 19, 2012 at 4:38 AM, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:
  Not sure if this is relevant but thought I'd point it out.  

  
The following field seems to contain the IP of the domain we were expecting ('c1881.voiceflare.co.uk')  

  
  
 u'variable_sip_from_host': u'89.238.182.137',  

  
Normally, this field would contain the hostname and not the IP.  

  
Cal  
  

  
On Mon, Nov 19, 2012 at 4:34 AM, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:
  Hi guys,  

  
In a nut shell, it appears that when attempting to perform a blind transfer under certain conditions (explained below), mod_xml_curl does not expose the originating domain in a clean format.  
  

  
  
My initial plan was to find the point where these variable were being generated, look at what was available, then add an extra variable for the domain and submit a patch.  

  
Sadly my C isn't great and I hit a brick wall, so if anyone can help out, I will ensure the mod_xml_curl documentation is updated and/or assist with any patching/testing required.  
  

  
Please take the following scenario;  

  
* Extension 2000 calls an external number via a gateway (i.e. bridge sofia/gateway/name/e164_number_here).  
* Call connects fine, audio stays good, no disconnection problems etc.  
* Call is blind transferred to another extension  

  
As a result, the following is determined;  

  
* User initiating the blind transfer is 2000  
* Domain initiating the blind transfer is c1881.voiceflare.co.uk  
* Destination number of the call is 447866123456  
* Number to blind transfer to is 2001  
* Call to mod_xml_curl is made  

  
It makes reference to the User in the following 'clean' variables (by clean, I mean variables that just contain 2000 and don't require mangling to extract the info);  

  
  
 u'Caller-ANI': u'2000',  
  
 u'Caller-Username': u'2000',  
  
 u'Caller-Caller-ID-Number': u'2000',  
  
 u'Hunt-ANI': u'2000',  
  
 u'Hunt-Caller-ID-Number': u'2000',  
  
 u'Hunt-Username': u'2000',  
  
 u'variable_last_sent_callee_id_number': u'2000',  
  
  
 u'variable_sip_from_user': u'2000',  
  

  
It also has the User in the following unclean variables;  

  
 u'variable_bridge_channel': u'sofia/external/2000 at c1881.voiceflare.co.uk:5060',  
  
  
 u'variable_sip_from_uri': u'2000 at 89.238.182.137',  
 u'variable_sip_full_from': u'"foxx" <sip:2000 at 89.238.182.137>;tag=XryjFQp3rB2NF',  
  
 u'variable_sip_h_Referred-By': u'"foxx" <sip:2000 at c1881.voiceflare.co.uk:5060>',  

  
However, it only references the domain in the following unclean variables;  

  
 u'variable_bridge_channel': u'sofia/external/2000 at c1881.voiceflare.co.uk:5060',  
 u'variable_sip_h_Referred-By': u'"foxx" <sip:2000 at c1881.voiceflare.co.uk:5060>',  
 u'variable_sip_refer_to': u'<sip:2001 at c1881.voiceflare.co.uk:5060>',  

  
Lets say that we want to determine the user/domain that has initiated this transfer, doing so would mean mangling with one of those above variables, which seems a bit dirty (plus it is not clear which is the correct one to use).  

  
I have attached the SIP trace of the entire blind transfer event, and the full mod_xml_curl request being sent.  

  
  
If anyone could answer the following, it'd be much appreciated;  

  
* Should there be an improvement patch that exposes the domain of the user that originated the blind transfer?  
* Are there better/alternative ways to extracting the domain of the user that originated the blind transfer?  

  
Many thanks  

Cal  






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-- 
Anthony Minessale II

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From cal.leeming at simplicitymedialtd.co.uk  Sun Dec 30 23:12:55 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sun, 30 Dec 2012 20:12:55 +0000
Subject: [Freeswitch-users] mod_xml_curl - missing dialplan variable
 (domain of originating user for blind transfer)
In-Reply-To: <20121230193807.7bfe4ec1@mail.tritonwest.net>
References: <CAHKQagGSrZZ3YysjMh51V7zPbw-dPjdTE=5XqKMQgwRMp-B4Rg@mail.gmail.com>
	<20121230193807.7bfe4ec1@mail.tritonwest.net>
Message-ID: <CAHKQagHXLO9HrbDcevVmV-3_jTQOiYHMAXZDQ_j7ePL7fU5=qw@mail.gmail.com>

>From what I can tell - that's not entirely correct from a security stand
point.

Let's say a call comes in from an external gateway (DDI 442476100401), it's
routed to user 2001.

If you do 'set_user' then you'd be effectively saying that Leg A of the
call is an authorized user - is this correct?

If so - this could have all sorts of implications when performing
sanity/security checks on later events.

Please let me know if I'm wrong, as I haven't used 'set_user' much.

Cal

On Sun, Dec 30, 2012 at 7:38 PM, Dave R. Kompel <drk at drkngs.net> wrote:

> **
>  Since this is for use in external generated XML, and for the most part
> it should be code that is generating it, the easiest way to handle calls
> from gateways, is to back them with users in the directory. When A calll is
> from a gateway, simply do a set_user in your dialplan, and then a transfer
> back to the number dialed in the context ${user_context}, which will be set
> after you do the set_user.
>
> This way you don't have to do anything special for calls that come from a
> gateway, transfers and other things just work right.
>
> --Dave
>
>
>  ------------------------------
> *From:* Cal Leeming [Simplicity Media Ltd] [mailto:
> cal.leeming at simplicitymedialtd.co.uk]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> *Sent:* Sun, 30 Dec 2012 10:04:57 -0800
> *Subject:* Re: [Freeswitch-users] mod_xml_curl - missing dialplan
> variable (domain of originating user for blind transfer)
>
>
> I have started to move some of this information into the wiki.
>
> However, my previous comments about
> forcing sip_auth_realm/sip_auth_username are generally considered unsafe
> from a security point of view.
>
> If you force those variables, under some circumstances a blind transferred
> call from an external gateway could later on be considered an authorized
> internal user and lead to undesired behaviour.
>
> Forcing sip_invite_domain however is acceptable, as some request context
> will lose this information during a blind transfer or bridge.
>
> I am currently building up a spreadsheet of different call scenarios,
> along with the request variables for each one, so it is easy to see how the
> variables change depending on what event is happening.
>
> Cal
>
> On Tue, Nov 20, 2012 at 1:21 PM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> Hi Anthony,
>>
>> Thanks for the reply, yeah I spent some time looking at alternative ways
>> and came up with one that seems to get the job done
>>
>> I am convinced now that there is no bug as such - but one thing that is
>> absolutely clear, is that mod_xml_curl is in desperate need of
>> normalization.. even if it was fully documented, some of the variants don't
>> make logical sense. However, the information is there, and as long as you
>> do the correct conditional checks, then it will work flawlessly.
>>
>> To make the authenticated domain stick when passing through to a gateway,
>> we use the following;
>>
>> <action application="bridge"
>> data="{sip_auth_realm=${sip_auth_realm}}{sip_auth_username=${sip_auth_username}}{sip_invite_domain=${sip_from_host}}sofia/gateway/{{
>> dst.gateway.name
>> }}/{{dst.gateway.fs_dial_prefix}}{{dst.dst_number_e164}}"/>
>>
>> The most important part of the above being;
>>
>> {sip_auth_realm=${sip_auth_realm}}{sip_auth_username=${sip_auth_username}}{sip_invite_domain=${sip_from_host}}
>>
>> However, this alone doesn't deal with the fact that you have to look in
>> different places for the correct variables depending on what the current
>> call context is.
>>
>> To a certain extent, your own business logic will also determine which
>> variables should be used, and a combination of conditional checks may need
>> to be used in order to figure out which variable you should be using in the
>> first place lol.
>>
>> Here are the patterns we have found so far - this is just an information
>> dump for now, and later down the road I will update the mod_xml_curl
>> documentation.
>>
>> # Ensure that variable_sip_auth_username / variable_sip_auth_realm
>> #
>> # If Call-Direction is inbound, then;
>> # src_user = variable_sip_auth_username
>> # src_domain = variable_sip_auth_realm
>> # dst_user = variable_sip_to_user
>> #
>> # If Call-Direction is outbound, then;
>> # originate_user = variable_sip_auth_username
>> # originate_domain = variable_sip_auth_realm
>> # src_user = variable_sip_to_user
>> # src_domain = variable_sip_to_host
>> # dst_user = Caller-Destination-Number
>>
>> * gateway to gateway (442476100401 > 442476100402)
>> * domain to gateway - blind xfer to gateway (2000 > 442476100401
>> > 442476100402)
>> * domain to gateway - blind xfer to domain (2000 > 442476100401 > 2002)
>> * domain to gateway (2000 > 442476100401)
>> * domain to domain (2000 > 2001)
>>
>> ----
>>
>> # Check if variable_sip_to_host is present and known gateway
>> #
>> # If variable_dialed_user and variable_dialed_domain are present;
>> # originate_user = variable_dialed_user
>> # originate_domain = variable_dialed_domain
>> # src_user = variable_sip_from_user
>> # src_domain = variable_sip_to_host
>> # dst_user = Caller-Destination-Number
>> #
>> # If not present;
>> # src_user = variable_sip_from_user
>> # src_domain = variable_sip_to_host
>> # dst_user = variable_sip_to_user
>>
>>  * gateway to domain (442476100401 > 2002)
>> * gateway to domain - blind xfer to gateway (442476100401 > 2000
>> > 442476100402)
>> * gateway to domain - blind xfer to domain (442476100401 > 2000 > 2002)
>>
>> ----
>>
>> # check for variable_dialed_domain
>> # If match;
>> # src_user = variable_dialed_user
>> # src_domain  = variable_dialed_domain
>> # dst_user = Caller-Destination-Number
>>
>>  * domain to domain to gateway (2000 > 2001 > 442476100402)
>>
>>
>> On Tue, Nov 20, 2012 at 2:55 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> One thing you can do is set the variable when you do know it, from the
>>> dialplan.  So it will be there in all the subsequent events.
>>>
>>>
>>>
>>>  On Mon, Nov 19, 2012 at 1:34 AM, Cal Leeming [Simplicity Media Ltd] <
>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>
>>>>  Last update, then I'm really going to sleep!
>>>>
>>>>  Apologies for the noise btw, in hindsight I should have collected all
>>>> this info and posted in one go.
>>>>
>>>> I've tried enabling auth-all-packets (along with auth_calls), as I
>>>> thought maybe having authentication on REFER packets might make a
>>>> difference, but sadly it had no effect (the SIP headers show
>>>> proxy-authorization in the REFER, but nothing extra showed in mod_xml_curl)
>>>>
>>>> I've managed to narrow down the circumstances in which this happens;
>>>>
>>>> CORRECT:
>>>>
>>>> * User receives call from gateway, blind transfer to another
>>>> user (shows correctly as variable_dialed_user/variable_dialed_domain)
>>>> * User receives call from another user, blind transfer to
>>>> gateway (shows correctly as variable_dialed_user/variable_dialed_domain)
>>>> * User receives call from another user, blind transfer to another
>>>> user (shows correctly as variable_dialed_user/variable_dialed_domain)
>>>> * User makes call to another user, blind transfer to another
>>>> user (shows correctly as variable_dialed_user/variable_dialed_domain)
>>>> * User makes call to another user, blind transfer to a gateway (shows
>>>> correctly as variable_dialed_user/variable_dialed_domain)
>>>>
>>>> MISSING:
>>>>
>>>> * User makes call to a gateway, blind transfer to another gateway (no
>>>> clean variables for domain)
>>>> * User makes call to a gateway, blind transfer to another user (no
>>>> clean variables for domain)
>>>>
>>>> So, the problem seems to happen specifically when you blind transfer a
>>>> call that is already in progress on a gateway, where the call was
>>>> originated by a user and not a gateway.
>>>>
>>>> I did a bit more looking through code, added a few switch_log_printf()
>>>> calls, and found that the following method is NOT called in those two
>>>> scenarios where these variables are missing;
>>>> mod_dptools.c: "switch_call_cause_t user_outgoing_channel"
>>>>
>>>> This is about as far as I can go on this, as I just don't know enough
>>>> about C to give any more in-depth info :/
>>>>
>>>> Cal
>>>>
>>>> On Mon, Nov 19, 2012 at 5:52 AM, Cal Leeming [Simplicity Media Ltd] <
>>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>
>>>>> Sorry, another update.. after tinkering with the SIP headers, we found
>>>>> that we're able to pass any user/host along in an INVITE, and this is
>>>>> passed onto mod_xml_curl.
>>>>>
>>>>> To fix this particular part of the problem, we enabled auth_calls and
>>>>> this gives us correct/clean variables we can trust, specifically;
>>>>>
>>>>>   u'variable_sip_auth_username': u'2000',
>>>>>  u'variable_user_name': u'2000',
>>>>>
>>>>> However, when attempting to do the blind transfer again, this
>>>>> variables are all missing.
>>>>>
>>>>> At this point I'm convinced that attempting to extract the user/domain
>>>>> from the Refer headers is probably not the right thing to do... But I'm
>>>>> still no closer to figuring out what the correct approach should be to
>>>>> expose the authenticated user/domain to mod_xml_curl.
>>>>>
>>>>> Cal
>>>>>
>>>>>
>>>>> On Mon, Nov 19, 2012 at 4:58 AM, Cal Leeming [Simplicity Media Ltd] <
>>>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>>
>>>>>> Another quick update on this before I pass out from lack of sleep..!
>>>>>>
>>>>>> Just had a look through the mod_sofia.c/h source and found the
>>>>>> following;
>>>>>>
>>>>>>  mod_sofia.c/mod_sofia.h
>>>>>> #define SOFIA_REFER_TO_VARIABLE "sip_refer_to"
>>>>>> if (!zstr(full_ref_by)) {
>>>>>> switch_channel_set_variable(t_channel, SOFIA_SIP_HEADER_PREFIX
>>>>>> "Referred-By", full_ref_by);
>>>>>> }
>>>>>> if (!zstr(full_ref_to)) {
>>>>>> switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
>>>>>> full_ref_to);
>>>>>> }
>>>>>> if (!zstr(full_ref_to)) {
>>>>>> switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
>>>>>> full_ref_to);
>>>>>> }
>>>>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>>>>>> SWITCH_LOG_DEBUG, "Process REFER to [%s@%s]\n", exten, (char *)
>>>>>> refer_to->r_url->url_host);
>>>>>>
>>>>>> If the correct approach is deemed to be patching code, then I figured
>>>>>> it could be as simple as this;
>>>>>>
>>>>>>  switch_channel_set_variable(t_channel, "Referred-By-User", exten);
>>>>>> switch_channel_set_variable(t_channel, "Referred-By-Domain", (char *)
>>>>>> refer_to->r_url->url_host);
>>>>>>
>>>>>> This is pretty much where my knowledge of C ends, I can (just about)
>>>>>> read and copy chunks of C code, but that's about it :)
>>>>>>
>>>>>> Cal
>>>>>>
>>>>>> On Mon, Nov 19, 2012 at 4:38 AM, Cal Leeming [Simplicity Media Ltd] <
>>>>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>>>
>>>>>>> Not sure if this is relevant but thought I'd point it out.
>>>>>>>
>>>>>>> The following field seems to contain the IP of the domain we were
>>>>>>> expecting ('c1881.voiceflare.co.uk')
>>>>>>>
>>>>>>>   u'variable_sip_from_host': u'89.238.182.137',
>>>>>>>
>>>>>>> Normally, this field would contain the hostname and not the IP.
>>>>>>>
>>>>>>> Cal
>>>>>>>
>>>>>>> On Mon, Nov 19, 2012 at 4:34 AM, Cal Leeming [Simplicity Media Ltd]
>>>>>>> <cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>>>>
>>>>>>>> Hi guys,
>>>>>>>>
>>>>>>>> In a nut shell, it appears that when attempting to perform a blind
>>>>>>>> transfer under certain conditions (explained below), mod_xml_curl does not
>>>>>>>> expose the originating domain in a clean format.
>>>>>>>>
>>>>>>>>  My initial plan was to find the point where these variable were
>>>>>>>> being generated, look at what was available, then add an extra variable for
>>>>>>>> the domain and submit a patch.
>>>>>>>>
>>>>>>>> Sadly my C isn't great and I hit a brick wall, so if anyone can
>>>>>>>> help out, I will ensure the mod_xml_curl documentation is updated and/or
>>>>>>>> assist with any patching/testing required.
>>>>>>>>
>>>>>>>> Please take the following scenario;
>>>>>>>>
>>>>>>>> * Extension 2000 calls an external number via a gateway (i.e.
>>>>>>>> bridge sofia/gateway/name/e164_number_here).
>>>>>>>> * Call connects fine, audio stays good, no disconnection problems
>>>>>>>> etc.
>>>>>>>> * Call is blind transferred to another extension
>>>>>>>>
>>>>>>>> As a result, the following is determined;
>>>>>>>>
>>>>>>>> * User initiating the blind transfer is 2000
>>>>>>>> * Domain initiating the blind transfer is c1881.voiceflare.co.uk
>>>>>>>> * Destination number of the call is 447866123456
>>>>>>>> * Number to blind transfer to is 2001
>>>>>>>> * Call to mod_xml_curl is made
>>>>>>>>
>>>>>>>> It makes reference to the User in the following 'clean' variables
>>>>>>>> (by clean, I mean variables that just contain 2000 and don't require
>>>>>>>> mangling to extract the info);
>>>>>>>>
>>>>>>>>   u'Caller-ANI': u'2000',
>>>>>>>>   u'Caller-Username': u'2000',
>>>>>>>>   u'Caller-Caller-ID-Number': u'2000',
>>>>>>>>   u'Hunt-ANI': u'2000',
>>>>>>>>   u'Hunt-Caller-ID-Number': u'2000',
>>>>>>>>   u'Hunt-Username': u'2000',
>>>>>>>>   u'variable_last_sent_callee_id_number': u'2000',
>>>>>>>>   u'variable_sip_from_user': u'2000',
>>>>>>>>
>>>>>>>> It also has the User in the following unclean variables;
>>>>>>>>
>>>>>>>>  u'variable_bridge_channel': u'sofia/external/
>>>>>>>> 2000 at c1881.voiceflare.co.uk:5060',
>>>>>>>>   u'variable_sip_from_uri': u'2000 at 89.238.182.137',
>>>>>>>>  u'variable_sip_full_from': u'"foxx" <sip:2000 at 89.238.182.137
>>>>>>>> >;tag=XryjFQp3rB2NF',
>>>>>>>>   u'variable_sip_h_Referred-By': u'"foxx" <
>>>>>>>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>>>>>>>
>>>>>>>> However, it only references the domain in the following unclean
>>>>>>>> variables;
>>>>>>>>
>>>>>>>>  u'variable_bridge_channel': u'sofia/external/
>>>>>>>> 2000 at c1881.voiceflare.co.uk:5060',
>>>>>>>>  u'variable_sip_h_Referred-By': u'"foxx" <
>>>>>>>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>>>>>>>  u'variable_sip_refer_to': u'<sip:2001 at c1881.voiceflare.co.uk:5060
>>>>>>>> >',
>>>>>>>>
>>>>>>>> Lets say that we want to determine the user/domain that has
>>>>>>>> initiated this transfer, doing so would mean mangling with one of those
>>>>>>>> above variables, which seems a bit dirty (plus it is not clear which is the
>>>>>>>> correct one to use).
>>>>>>>>
>>>>>>>> I have attached the SIP trace of the entire blind transfer event,
>>>>>>>> and the full mod_xml_curl request being sent.
>>>>>>>>
>>>>>>>>  If anyone could answer the following, it'd be much appreciated;
>>>>>>>>
>>>>>>>> * Should there be an improvement patch that exposes the domain of
>>>>>>>> the user that originated the blind transfer?
>>>>>>>> * Are there better/alternative ways to extracting the domain of the
>>>>>>>> user that originated the blind transfer?
>>>>>>>>
>>>>>>>> Many thanks
>>>>>>>>
>>>>>>>> Cal
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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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>
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From ben at langfeld.co.uk  Sun Dec 30 23:47:34 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Sun, 30 Dec 2012 20:47:34 +0000
Subject: [Freeswitch-users] uuid_record and RECORD_STEREO
In-Reply-To: <CAKzWOxX+167M=k7acjY45H=WA2_eprwbs3S=L-9VZyTDCbQ4FQ@mail.gmail.com>
References: <89F15D50-E189-4D86-9320-102F086DF18F@gmail.com>
	<CAKzWOxX+167M=k7acjY45H=WA2_eprwbs3S=L-9VZyTDCbQ4FQ@mail.gmail.com>
Message-ID: <CAAyX+KHVjRmvY_6sYAAi+5SxXZdOc72cn4_DdV466rpb7JYX+w@mail.gmail.com>

If uuid_setvar is used, it affects all future invocations of uuid_record.
This is global channel state. What we need is to invoke uuid_record with
that var set locally for that invocation, such that it doesn't affect
future invocations (a local variable).

Regards,
Ben Langfeld


On 27 December 2012 17:45, Michael Collins <msc at freeswitch.org> wrote:

> I'm not sure I understand the question. Could you elaborate? What do you
> mean by "set a global state on the channel"?
>
> -MC
>
>
> On Thu, Dec 27, 2012 at 3:55 AM, Luca Pradovera <luca.pradovera at gmail.com>wrote:
>
>> Hello,
>> is there a way to invoke uuid_record through bgapi, setting RECORD_STEREO
>> to true as a local parameter instead of using uuid_setvar, over inbound
>> Event Socket?
>> I would prefer to not set a global state on the channel if possible.
>> Thanks!
>>
>> --
>> Luca Pradovera
>> luca.pradovera at gmail.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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 dujinfang at gmail.com  Mon Dec 31 02:21:51 2012
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 31 Dec 2012 07:21:51 +0800
Subject: [Freeswitch-users] uuid_record and RECORD_STEREO
In-Reply-To: <CAAyX+KHVjRmvY_6sYAAi+5SxXZdOc72cn4_DdV466rpb7JYX+w@mail.gmail.com>
References: <89F15D50-E189-4D86-9320-102F086DF18F@gmail.com>
	<CAKzWOxX+167M=k7acjY45H=WA2_eprwbs3S=L-9VZyTDCbQ4FQ@mail.gmail.com>
	<CAAyX+KHVjRmvY_6sYAAi+5SxXZdOc72cn4_DdV466rpb7JYX+w@mail.gmail.com>
Message-ID: <4C61A84750854AF4822B728CEE0B3A56@gmail.com>

uuid_setvar with no val will unset the var 

-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Monday, December 31, 2012 at 4:47 AM, Ben Langfeld wrote:

> If uuid_setvar is used, it affects all future invocations of uuid_record. This is global channel state. What we need is to invoke uuid_record with that var set locally for that invocation, such that it doesn't affect future invocations (a local variable).
> 
> Regards,
> Ben Langfeld 
> 
> On 27 December 2012 17:45, Michael Collins <msc at freeswitch.org (mailto:msc at freeswitch.org)> wrote:
> > I'm not sure I understand the question. Could you elaborate? What do you mean by "set a global state on the channel"?
> > 
> > -MC
> > 
> > 
> > On Thu, Dec 27, 2012 at 3:55 AM, Luca Pradovera <luca.pradovera at gmail.com (mailto:luca.pradovera at gmail.com)> wrote:
> > > Hello,
> > > is there a way to invoke uuid_record through bgapi, setting RECORD_STEREO to true as a local parameter instead of using uuid_setvar, over inbound Event Socket?
> > > I would prefer to not set a global state on the channel if possible.
> > > Thanks!
> > > 
> > > --
> > > Luca Pradovera
> > > luca.pradovera at gmail.com (mailto:luca.pradovera at gmail.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
> > > 
> > > 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
> > 
> > 
> > 
> > -- 
> > Michael S Collins
> > Twitter: @mercutioviz
> > http://www.FreeSWITCH.org
> > http://www.ClueCon.com
> > http://www.OSTAG.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
> > 
> > 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
> 
> 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 drk at drkngs.net  Mon Dec 31 02:27:13 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Sun, 30 Dec 2012 15:27:13 -0800
Subject: [Freeswitch-users] mod_xml_curl - missing dialplan variable
 (domain of originating user for blind transfer)
In-Reply-To: <CAHKQagHXLO9HrbDcevVmV-3_jTQOiYHMAXZDQ_j7ePL7fU5=qw@mail.gmail.com>
Message-ID: <20121230232713.0fac881b@mail.tritonwest.net>

No, you want to have a gateway user that you use that has the right context, so they can't dial out on their own, but the user CAN transfer them to something external if you want.  
   
--Dave
      _____  

  From: Cal Leeming [Simplicity Media Ltd] [mailto:cal.leeming at simplicitymedialtd.co.uk]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Sun, 30 Dec 2012 12:12:55 -0800
Subject: Re: [Freeswitch-users] mod_xml_curl - missing dialplan variable (domain of originating user for blind transfer)

>From what I can tell - that's not entirely correct from a security stand point.  

  
Let's say a call comes in from an external gateway (DDI 442476100401), it's routed to user 2001.  

  
If you do 'set_user' then you'd be effectively saying that Leg A of the call is an authorized user - is this correct?  

  
If so - this could have all sorts of implications when performing sanity/security checks on later events.  

  
Please let me know if I'm wrong, as I haven't used 'set_user' much.  

  
Cal  

  
On Sun, Dec 30, 2012 at 7:38 PM, Dave R. Kompel <drk at drkngs.net> wrote:
    
  
  
Since this is for use in external generated XML, and for the most part it should be code that is generating it, the easiest way to handle calls from gateways, is to back them with users in the directory. When A calll is from a gateway, simply do a set_user in your dialplan, and then a transfer back to the number dialed in the context ${user_context}, which will be set after you do the set_user.   
   
This way you don't have to do anything special for calls that come from a gateway, transfers and other things just work right.  
   
--Dave

      _____  

  From: Cal Leeming [Simplicity Media Ltd] [mailto:cal.leeming at simplicitymedialtd.co.uk]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Sun, 30 Dec 2012 10:04:57 -0800
Subject: Re: [Freeswitch-users] mod_xml_curl - missing dialplan variable (domain of originating user for blind transfer)  
  


I have started to move some of this information into the wiki.   

  
However, my previous comments about forcing sip_auth_realm/sip_auth_username are generally considered unsafe from a security point of view.  

  
If you force those variables, under some circumstances a blind transferred call from an external gateway could later on be considered an authorized internal user and lead to undesired behaviour.  

  
Forcing sip_invite_domain however is acceptable, as some request context will lose this information during a blind transfer or bridge.  

  
I am currently building up a spreadsheet of different call scenarios, along with the request variables for each one, so it is easy to see how the variables change depending on what event is happening.  

  
Cal  

  
On Tue, Nov 20, 2012 at 1:21 PM, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:
  Hi Anthony,   

  
Thanks for the reply, yeah I spent some time looking at alternative ways and came up with one that seems to get the job done  

  
I am convinced now that there is no bug as such - but one thing that is absolutely clear, is that mod_xml_curl is in desperate need of normalization.. even if it was fully documented, some of the variants don't make logical sense. However, the information is there, and as long as you do the correct conditional checks, then it will work flawlessly.  

  
To make the authenticated domain stick when passing through to a gateway, we use the following;  

  
<action application="bridge" data="{sip_auth_realm=${sip_auth_realm}}{sip_auth_username=${sip_auth_username}}{sip_invite_domain=${sip_from_host}}sofia/gateway/{{dst.gateway.name}}/{{dst.gateway.fs_dial_prefix}}{{dst.dst_number_e164}}"/>

The most important part of the above being;  
{sip_auth_realm=${sip_auth_realm}}{sip_auth_username=${sip_auth_username}}{sip_invite_domain=${sip_from_host}}  

  
However, this alone doesn't deal with the fact that you have to look in different places for the correct variables depending on what the current call context is.  

  
To a certain extent, your own business logic will also determine which variables should be used, and a combination of conditional checks may need to be used in order to figure out which variable you should be using in the first place lol.  

  
Here are the patterns we have found so far - this is just an information dump for now, and later down the road I will update the mod_xml_curl documentation.  

  
# Ensure that variable_sip_auth_username / variable_sip_auth_realm  
#  
# If Call-Direction is inbound, then;  
# src_user = variable_sip_auth_username  
# src_domain = variable_sip_auth_realm  
# dst_user = variable_sip_to_user  
#  
# If Call-Direction is outbound, then;  
# originate_user = variable_sip_auth_username  
# originate_domain = variable_sip_auth_realm  
# src_user = variable_sip_to_user  
# src_domain = variable_sip_to_host  
# dst_user = Caller-Destination-Number  

  
* gateway to gateway (442476100401 > 442476100402)  
* domain to gateway - blind xfer to gateway (2000 > 442476100401 > 442476100402)  
* domain to gateway - blind xfer to domain (2000 > 442476100401 > 2002)  
* domain to gateway (2000 > 442476100401)  
* domain to domain (2000 > 2001)  

  
----  

  
# Check if variable_sip_to_host is present and known gateway  
#  
# If variable_dialed_user and variable_dialed_domain are present;  
# originate_user = variable_dialed_user  
# originate_domain = variable_dialed_domain  
# src_user = variable_sip_from_user  
# src_domain = variable_sip_to_host  
# dst_user = Caller-Destination-Number  
#  
# If not present;  
# src_user = variable_sip_from_user  
# src_domain = variable_sip_to_host  
# dst_user = variable_sip_to_user  

  
  
* gateway to domain (442476100401 > 2002)  
* gateway to domain - blind xfer to gateway (442476100401 > 2000 > 442476100402)  
* gateway to domain - blind xfer to domain (442476100401 > 2000 > 2002)  

  
----  

  
# check for variable_dialed_domain  
# If match;  
# src_user = variable_dialed_user  
# src_domain  = variable_dialed_domain  
# dst_user = Caller-Destination-Number  

  
  
* domain to domain to gateway (2000 > 2001 > 442476100402)  
  
  

  

  
On Tue, Nov 20, 2012 at 2:55 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
  One thing you can do is set the variable when you do know it, from the dialplan.  So it will be there in all the subsequent events.   

  


  
  
  
On Mon, Nov 19, 2012 at 1:34 AM, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:
    
  
Last update, then I'm really going to sleep!   

  
  
Apologies for the noise btw, in hindsight I should have collected all this info and posted in one go.  

  
I've tried enabling auth-all-packets (along with auth_calls), as I thought maybe having authentication on REFER packets might make a difference, but sadly it had no effect (the SIP headers show proxy-authorization in the REFER, but nothing extra showed in mod_xml_curl)  

  
I've managed to narrow down the circumstances in which this happens;  

  
CORRECT:  

* User receives call from gateway, blind transfer to another user (shows correctly as variable_dialed_user/variable_dialed_domain)  
* User receives call from another user, blind transfer to gateway (shows correctly as variable_dialed_user/variable_dialed_domain)  
* User receives call from another user, blind transfer to another user (shows correctly as variable_dialed_user/variable_dialed_domain)  
* User makes call to another user, blind transfer to another user (shows correctly as variable_dialed_user/variable_dialed_domain)  
* User makes call to another user, blind transfer to a gateway (shows correctly as variable_dialed_user/variable_dialed_domain)  

  
MISSING:  

  
* User makes call to a gateway, blind transfer to another gateway (no clean variables for domain)  
* User makes call to a gateway, blind transfer to another user (no clean variables for domain)  

  
So, the problem seems to happen specifically when you blind transfer a call that is already in progress on a gateway, where the call was originated by a user and not a gateway.  

  
I did a bit more looking through code, added a few switch_log_printf() calls, and found that the following method is NOT called in those two scenarios where these variables are missing;  
mod_dptools.c: "switch_call_cause_t user_outgoing_channel"  

  
This is about as far as I can go on this, as I just don't know enough about C to give any more in-depth info :/  

  
Cal  
  
  

  
On Mon, Nov 19, 2012 at 5:52 AM, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:
  Sorry, another update.. after tinkering with the SIP headers, we found that we're able to pass any user/host along in an INVITE, and this is passed onto mod_xml_curl.   

  
To fix this particular part of the problem, we enabled auth_calls and this gives us correct/clean variables we can trust, specifically;  

  
  
 u'variable_sip_auth_username': u'2000',  
 u'variable_user_name': u'2000',  

  
However, when attempting to do the blind transfer again, this variables are all missing.  

  
At this point I'm convinced that attempting to extract the user/domain from the Refer headers is probably not the right thing to do... But I'm still no closer to figuring out what the correct approach should be to expose the authenticated user/domain to mod_xml_curl.  

  
Cal  
  
  

  

  
On Mon, Nov 19, 2012 at 4:58 AM, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:
  Another quick update on this before I pass out from lack of sleep..!   

  
Just had a look through the mod_sofia.c/h source and found the following;  

  
  
mod_sofia.c/mod_sofia.h  
#define SOFIA_REFER_TO_VARIABLE "sip_refer_to"  
if (!zstr(full_ref_by)) {  
switch_channel_set_variable(t_channel, SOFIA_SIP_HEADER_PREFIX "Referred-By", full_ref_by);  
}  
if (!zstr(full_ref_to)) {  
switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE, full_ref_to);  
}  
if (!zstr(full_ref_to)) {  
switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE, full_ref_to);  
}  
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Process REFER to [%s@%s]\n", exten, (char *) refer_to->r_url->url_host);  

  
If the correct approach is deemed to be patching code, then I figured it could be as simple as this;  

  
  
switch_channel_set_variable(t_channel, "Referred-By-User", exten);  
switch_channel_set_variable(t_channel, "Referred-By-Domain", (char *) refer_to->r_url->url_host);  

  
This is pretty much where my knowledge of C ends, I can (just about) read and copy chunks of C code, but that's about it :)  

  
Cal  
  

  
On Mon, Nov 19, 2012 at 4:38 AM, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:
  Not sure if this is relevant but thought I'd point it out.   

  
The following field seems to contain the IP of the domain we were expecting ('c1881.voiceflare.co.uk')  

  
  
 u'variable_sip_from_host': u'89.238.182.137',  

  
Normally, this field would contain the hostname and not the IP.  

  
Cal  
  

  
On Mon, Nov 19, 2012 at 4:34 AM, Cal Leeming [Simplicity Media Ltd] <cal.leeming at simplicitymedialtd.co.uk> wrote:
  Hi guys,   

  
In a nut shell, it appears that when attempting to perform a blind transfer under certain conditions (explained below), mod_xml_curl does not expose the originating domain in a clean format.  
  

  
  
My initial plan was to find the point where these variable were being generated, look at what was available, then add an extra variable for the domain and submit a patch.  

  
Sadly my C isn't great and I hit a brick wall, so if anyone can help out, I will ensure the mod_xml_curl documentation is updated and/or assist with any patching/testing required.  
  

  
Please take the following scenario;  

  
* Extension 2000 calls an external number via a gateway (i.e. bridge sofia/gateway/name/e164_number_here).  
* Call connects fine, audio stays good, no disconnection problems etc.  
* Call is blind transferred to another extension  

  
As a result, the following is determined;  

  
* User initiating the blind transfer is 2000  
* Domain initiating the blind transfer is c1881.voiceflare.co.uk  
* Destination number of the call is 447866123456  
* Number to blind transfer to is 2001  
* Call to mod_xml_curl is made  

  
It makes reference to the User in the following 'clean' variables (by clean, I mean variables that just contain 2000 and don't require mangling to extract the info);  

  
  
 u'Caller-ANI': u'2000',  
  
 u'Caller-Username': u'2000',  
  
 u'Caller-Caller-ID-Number': u'2000',  
  
 u'Hunt-ANI': u'2000',  
  
 u'Hunt-Caller-ID-Number': u'2000',  
  
 u'Hunt-Username': u'2000',  
  
 u'variable_last_sent_callee_id_number': u'2000',  
  
  
 u'variable_sip_from_user': u'2000',  
  

  
It also has the User in the following unclean variables;  

  
 u'variable_bridge_channel': u'sofia/external/2000 at c1881.voiceflare.co.uk:5060',  
  
  
 u'variable_sip_from_uri': u'2000 at 89.238.182.137',  
 u'variable_sip_full_from': u'"foxx" <sip:2000 at 89.238.182.137>;tag=XryjFQp3rB2NF',  
  
 u'variable_sip_h_Referred-By': u'"foxx" <sip:2000 at c1881.voiceflare.co.uk:5060>',  

  
However, it only references the domain in the following unclean variables;  

  
 u'variable_bridge_channel': u'sofia/external/2000 at c1881.voiceflare.co.uk:5060',  
 u'variable_sip_h_Referred-By': u'"foxx" <sip:2000 at c1881.voiceflare.co.uk:5060>',  
 u'variable_sip_refer_to': u'<sip:2001 at c1881.voiceflare.co.uk:5060>',  

  
Lets say that we want to determine the user/domain that has initiated this transfer, doing so would mean mangling with one of those above variables, which seems a bit dirty (plus it is not clear which is the correct one to use).  

  
I have attached the SIP trace of the entire blind transfer event, and the full mod_xml_curl request being sent.  

  
  
If anyone could answer the following, it'd be much appreciated;  

  
* Should there be an improvement patch that exposes the domain of the user that originated the blind transfer?  
* Are there better/alternative ways to extracting the domain of the user that originated the blind transfer?  

  
Many thanks  

Cal  






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_________________________________________________________________________
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From magnus.kelly at gmail.com  Mon Dec 31 02:29:50 2012
From: magnus.kelly at gmail.com (Magnus)
Date: Sun, 30 Dec 2012 23:29:50 +0000
Subject: [Freeswitch-users] Issues with updating freeswitch with yum?
Message-ID: <7A5A217E-6D3B-4ED9-9856-E75382AF2223@gmail.com>

Hello,

When trying to yum update freeswitch I get below error code ?

Package freeswitch-application-distributor-1.2.5.1-1.el6.x86_64.rpm is not signed

Any thoughts on how to proceed?

Thanks
Magnus




From krice at freeswitch.org  Mon Dec 31 02:57:26 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 30 Dec 2012 17:57:26 -0600
Subject: [Freeswitch-users] Issues with updating freeswitch with yum?
In-Reply-To: <7A5A217E-6D3B-4ED9-9856-E75382AF2223@gmail.com>
Message-ID: <CD063106.6855C%krice@freeswitch.org>

Don't use the RPMs... They are dated and I havent had a chance to update
them... That will be happening shortly into the new years...

If you must --nogpgcheck will bypass the error


On 12/30/12 5:29 PM, "Magnus" <magnus.kelly at gmail.com> wrote:

> Hello,
> 
> When trying to yum update freeswitch I get below error code ?
> 
> Package freeswitch-application-distributor-1.2.5.1-1.el6.x86_64.rpm is not
> signed
> 
> Any thoughts on how to proceed?
> 
> Thanks
> Magnus
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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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http://www.FreeSWITCH.org
http://www.ClueCon.com
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irc.freenode.net #freeswitch




From cal.leeming at simplicitymedialtd.co.uk  Mon Dec 31 03:41:16 2012
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Mon, 31 Dec 2012 00:41:16 +0000
Subject: [Freeswitch-users] mod_xml_curl - missing dialplan variable
 (domain of originating user for blind transfer)
In-Reply-To: <20121230232713.0fac881b@mail.tritonwest.net>
References: <CAHKQagHXLO9HrbDcevVmV-3_jTQOiYHMAXZDQ_j7ePL7fU5=qw@mail.gmail.com>
	<20121230232713.0fac881b@mail.tritonwest.net>
Message-ID: <CAHKQagHUcuAAiB20OAVmcq6vN9gE0tpq4t1KatUY7U57GQGO6Q@mail.gmail.com>

Sorry, I might have misunderstood but that doesn't clarify the potential
security flaw of using set_user.

I agree that the call needs correct context, but forcing a call from a
gateway to use the context of the user they are calling doesn't sound like
the right thing to do.

Am I missing something here??

Cal

On Sun, Dec 30, 2012 at 11:27 PM, Dave R. Kompel <drk at drkngs.net> wrote:

> **
> No, you want to have a gateway user that you use that has the right
> context, so they can't dial out on their own, but the user CAN transfer
> them to something external if you want.
>
> --Dave
>
>  ------------------------------
> *From:* Cal Leeming [Simplicity Media Ltd] [mailto:
> cal.leeming at simplicitymedialtd.co.uk]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> *Sent:* Sun, 30 Dec 2012 12:12:55 -0800
>
> *Subject:* Re: [Freeswitch-users] mod_xml_curl - missing dialplan
> variable (domain of originating user for blind transfer)
>
> >From what I can tell - that's not entirely correct from a security stand
> point.
>
> Let's say a call comes in from an external gateway (DDI 442476100401),
> it's routed to user 2001.
>
> If you do 'set_user' then you'd be effectively saying that Leg A of the
> call is an authorized user - is this correct?
>
> If so - this could have all sorts of implications when performing
> sanity/security checks on later events.
>
> Please let me know if I'm wrong, as I haven't used 'set_user' much.
>
> Cal
>
> On Sun, Dec 30, 2012 at 7:38 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>
>> **
>>  Since this is for use in external generated XML, and for the most part
>> it should be code that is generating it, the easiest way to handle calls
>> from gateways, is to back them with users in the directory. When A calll is
>> from a gateway, simply do a set_user in your dialplan, and then a transfer
>> back to the number dialed in the context ${user_context}, which will be set
>> after you do the set_user.
>>
>> This way you don't have to do anything special for calls that come from a
>> gateway, transfers and other things just work right.
>>
>> --Dave
>>
>>
>>  ------------------------------
>> *From:* Cal Leeming [Simplicity Media Ltd] [mailto:
>> cal.leeming at simplicitymedialtd.co.uk]
>> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org
>> ]
>> *Sent:* Sun, 30 Dec 2012 10:04:57 -0800
>> *Subject:* Re: [Freeswitch-users] mod_xml_curl - missing dialplan
>> variable (domain of originating user for blind transfer)
>>
>>
>> I have started to move some of this information into the wiki.
>>
>> However, my previous comments about
>> forcing sip_auth_realm/sip_auth_username are generally considered unsafe
>> from a security point of view.
>>
>> If you force those variables, under some circumstances a blind
>> transferred call from an external gateway could later on be considered an
>> authorized internal user and lead to undesired behaviour.
>>
>> Forcing sip_invite_domain however is acceptable, as some request context
>> will lose this information during a blind transfer or bridge.
>>
>> I am currently building up a spreadsheet of different call scenarios,
>> along with the request variables for each one, so it is easy to see how the
>> variables change depending on what event is happening.
>>
>> Cal
>>
>> On Tue, Nov 20, 2012 at 1:21 PM, Cal Leeming [Simplicity Media Ltd] <
>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>
>>> Hi Anthony,
>>>
>>> Thanks for the reply, yeah I spent some time looking at alternative ways
>>> and came up with one that seems to get the job done
>>>
>>> I am convinced now that there is no bug as such - but one thing that is
>>> absolutely clear, is that mod_xml_curl is in desperate need of
>>> normalization.. even if it was fully documented, some of the variants don't
>>> make logical sense. However, the information is there, and as long as you
>>> do the correct conditional checks, then it will work flawlessly.
>>>
>>> To make the authenticated domain stick when passing through to a
>>> gateway, we use the following;
>>>
>>> <action application="bridge"
>>> data="{sip_auth_realm=${sip_auth_realm}}{sip_auth_username=${sip_auth_username}}{sip_invite_domain=${sip_from_host}}sofia/gateway/{{
>>> dst.gateway.name
>>> }}/{{dst.gateway.fs_dial_prefix}}{{dst.dst_number_e164}}"/>
>>>
>>> The most important part of the above being;
>>>
>>> {sip_auth_realm=${sip_auth_realm}}{sip_auth_username=${sip_auth_username}}{sip_invite_domain=${sip_from_host}}
>>>
>>> However, this alone doesn't deal with the fact that you have to look in
>>> different places for the correct variables depending on what the current
>>> call context is.
>>>
>>> To a certain extent, your own business logic will also determine which
>>> variables should be used, and a combination of conditional checks may need
>>> to be used in order to figure out which variable you should be using in the
>>> first place lol.
>>>
>>> Here are the patterns we have found so far - this is just an information
>>> dump for now, and later down the road I will update the mod_xml_curl
>>> documentation.
>>>
>>> # Ensure that variable_sip_auth_username / variable_sip_auth_realm
>>> #
>>> # If Call-Direction is inbound, then;
>>> # src_user = variable_sip_auth_username
>>> # src_domain = variable_sip_auth_realm
>>> # dst_user = variable_sip_to_user
>>> #
>>> # If Call-Direction is outbound, then;
>>> # originate_user = variable_sip_auth_username
>>> # originate_domain = variable_sip_auth_realm
>>> # src_user = variable_sip_to_user
>>> # src_domain = variable_sip_to_host
>>> # dst_user = Caller-Destination-Number
>>>
>>> * gateway to gateway (442476100401 > 442476100402)
>>> * domain to gateway - blind xfer to gateway (2000 > 442476100401
>>> > 442476100402)
>>> * domain to gateway - blind xfer to domain (2000 > 442476100401 > 2002)
>>> * domain to gateway (2000 > 442476100401)
>>> * domain to domain (2000 > 2001)
>>>
>>> ----
>>>
>>> # Check if variable_sip_to_host is present and known gateway
>>> #
>>> # If variable_dialed_user and variable_dialed_domain are present;
>>> # originate_user = variable_dialed_user
>>> # originate_domain = variable_dialed_domain
>>> # src_user = variable_sip_from_user
>>> # src_domain = variable_sip_to_host
>>> # dst_user = Caller-Destination-Number
>>> #
>>> # If not present;
>>> # src_user = variable_sip_from_user
>>> # src_domain = variable_sip_to_host
>>> # dst_user = variable_sip_to_user
>>>
>>>  * gateway to domain (442476100401 > 2002)
>>> * gateway to domain - blind xfer to gateway (442476100401 > 2000
>>> > 442476100402)
>>> * gateway to domain - blind xfer to domain (442476100401 > 2000 > 2002)
>>>
>>> ----
>>>
>>> # check for variable_dialed_domain
>>> # If match;
>>> # src_user = variable_dialed_user
>>> # src_domain  = variable_dialed_domain
>>> # dst_user = Caller-Destination-Number
>>>
>>>  * domain to domain to gateway (2000 > 2001 > 442476100402)
>>>
>>>
>>> On Tue, Nov 20, 2012 at 2:55 AM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> One thing you can do is set the variable when you do know it, from the
>>>> dialplan.  So it will be there in all the subsequent events.
>>>>
>>>>
>>>>
>>>>  On Mon, Nov 19, 2012 at 1:34 AM, Cal Leeming [Simplicity Media Ltd] <
>>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>
>>>>>  Last update, then I'm really going to sleep!
>>>>>
>>>>>  Apologies for the noise btw, in hindsight I should have collected
>>>>> all this info and posted in one go.
>>>>>
>>>>> I've tried enabling auth-all-packets (along with auth_calls), as I
>>>>> thought maybe having authentication on REFER packets might make a
>>>>> difference, but sadly it had no effect (the SIP headers show
>>>>> proxy-authorization in the REFER, but nothing extra showed in mod_xml_curl)
>>>>>
>>>>> I've managed to narrow down the circumstances in which this happens;
>>>>>
>>>>> CORRECT:
>>>>>
>>>>> * User receives call from gateway, blind transfer to another
>>>>> user (shows correctly as variable_dialed_user/variable_dialed_domain)
>>>>> * User receives call from another user, blind transfer to
>>>>> gateway (shows correctly as variable_dialed_user/variable_dialed_domain)
>>>>> * User receives call from another user, blind transfer to another
>>>>> user (shows correctly as variable_dialed_user/variable_dialed_domain)
>>>>> * User makes call to another user, blind transfer to another
>>>>> user (shows correctly as variable_dialed_user/variable_dialed_domain)
>>>>> * User makes call to another user, blind transfer to a gateway (shows
>>>>> correctly as variable_dialed_user/variable_dialed_domain)
>>>>>
>>>>> MISSING:
>>>>>
>>>>> * User makes call to a gateway, blind transfer to another gateway (no
>>>>> clean variables for domain)
>>>>> * User makes call to a gateway, blind transfer to another user (no
>>>>> clean variables for domain)
>>>>>
>>>>> So, the problem seems to happen specifically when you blind transfer a
>>>>> call that is already in progress on a gateway, where the call was
>>>>> originated by a user and not a gateway.
>>>>>
>>>>> I did a bit more looking through code, added a few switch_log_printf()
>>>>> calls, and found that the following method is NOT called in those two
>>>>> scenarios where these variables are missing;
>>>>> mod_dptools.c: "switch_call_cause_t user_outgoing_channel"
>>>>>
>>>>> This is about as far as I can go on this, as I just don't know enough
>>>>> about C to give any more in-depth info :/
>>>>>
>>>>> Cal
>>>>>
>>>>> On Mon, Nov 19, 2012 at 5:52 AM, Cal Leeming [Simplicity Media Ltd] <
>>>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>>
>>>>>> Sorry, another update.. after tinkering with the SIP headers, we
>>>>>> found that we're able to pass any user/host along in an INVITE, and this is
>>>>>> passed onto mod_xml_curl.
>>>>>>
>>>>>> To fix this particular part of the problem, we enabled auth_calls and
>>>>>> this gives us correct/clean variables we can trust, specifically;
>>>>>>
>>>>>>   u'variable_sip_auth_username': u'2000',
>>>>>>  u'variable_user_name': u'2000',
>>>>>>
>>>>>> However, when attempting to do the blind transfer again, this
>>>>>> variables are all missing.
>>>>>>
>>>>>> At this point I'm convinced that attempting to extract the
>>>>>> user/domain from the Refer headers is probably not the right thing to do...
>>>>>> But I'm still no closer to figuring out what the correct approach should be
>>>>>> to expose the authenticated user/domain to mod_xml_curl.
>>>>>>
>>>>>> Cal
>>>>>>
>>>>>>
>>>>>> On Mon, Nov 19, 2012 at 4:58 AM, Cal Leeming [Simplicity Media Ltd] <
>>>>>> cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>>>
>>>>>>> Another quick update on this before I pass out from lack of sleep..!
>>>>>>>
>>>>>>> Just had a look through the mod_sofia.c/h source and found the
>>>>>>> following;
>>>>>>>
>>>>>>>  mod_sofia.c/mod_sofia.h
>>>>>>> #define SOFIA_REFER_TO_VARIABLE "sip_refer_to"
>>>>>>> if (!zstr(full_ref_by)) {
>>>>>>> switch_channel_set_variable(t_channel, SOFIA_SIP_HEADER_PREFIX
>>>>>>> "Referred-By", full_ref_by);
>>>>>>> }
>>>>>>> if (!zstr(full_ref_to)) {
>>>>>>> switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
>>>>>>> full_ref_to);
>>>>>>> }
>>>>>>> if (!zstr(full_ref_to)) {
>>>>>>> switch_channel_set_variable(t_channel, SOFIA_REFER_TO_VARIABLE,
>>>>>>> full_ref_to);
>>>>>>> }
>>>>>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>>>>>>> SWITCH_LOG_DEBUG, "Process REFER to [%s@%s]\n", exten, (char *)
>>>>>>> refer_to->r_url->url_host);
>>>>>>>
>>>>>>> If the correct approach is deemed to be patching code, then I
>>>>>>> figured it could be as simple as this;
>>>>>>>
>>>>>>>  switch_channel_set_variable(t_channel, "Referred-By-User", exten);
>>>>>>> switch_channel_set_variable(t_channel, "Referred-By-Domain", (char
>>>>>>> *) refer_to->r_url->url_host);
>>>>>>>
>>>>>>> This is pretty much where my knowledge of C ends, I can (just about)
>>>>>>> read and copy chunks of C code, but that's about it :)
>>>>>>>
>>>>>>> Cal
>>>>>>>
>>>>>>> On Mon, Nov 19, 2012 at 4:38 AM, Cal Leeming [Simplicity Media Ltd]
>>>>>>> <cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>>>>
>>>>>>>> Not sure if this is relevant but thought I'd point it out.
>>>>>>>>
>>>>>>>> The following field seems to contain the IP of the domain we were
>>>>>>>> expecting ('c1881.voiceflare.co.uk')
>>>>>>>>
>>>>>>>>   u'variable_sip_from_host': u'89.238.182.137',
>>>>>>>>
>>>>>>>> Normally, this field would contain the hostname and not the IP.
>>>>>>>>
>>>>>>>> Cal
>>>>>>>>
>>>>>>>> On Mon, Nov 19, 2012 at 4:34 AM, Cal Leeming [Simplicity Media Ltd]
>>>>>>>> <cal.leeming at simplicitymedialtd.co.uk> wrote:
>>>>>>>>
>>>>>>>>> Hi guys,
>>>>>>>>>
>>>>>>>>> In a nut shell, it appears that when attempting to perform a blind
>>>>>>>>> transfer under certain conditions (explained below), mod_xml_curl does not
>>>>>>>>> expose the originating domain in a clean format.
>>>>>>>>>
>>>>>>>>>  My initial plan was to find the point where these variable were
>>>>>>>>> being generated, look at what was available, then add an extra variable for
>>>>>>>>> the domain and submit a patch.
>>>>>>>>>
>>>>>>>>> Sadly my C isn't great and I hit a brick wall, so if anyone can
>>>>>>>>> help out, I will ensure the mod_xml_curl documentation is updated and/or
>>>>>>>>> assist with any patching/testing required.
>>>>>>>>>
>>>>>>>>> Please take the following scenario;
>>>>>>>>>
>>>>>>>>> * Extension 2000 calls an external number via a gateway (i.e.
>>>>>>>>> bridge sofia/gateway/name/e164_number_here).
>>>>>>>>> * Call connects fine, audio stays good, no disconnection problems
>>>>>>>>> etc.
>>>>>>>>> * Call is blind transferred to another extension
>>>>>>>>>
>>>>>>>>> As a result, the following is determined;
>>>>>>>>>
>>>>>>>>> * User initiating the blind transfer is 2000
>>>>>>>>> * Domain initiating the blind transfer is c1881.voiceflare.co.uk
>>>>>>>>> * Destination number of the call is 447866123456
>>>>>>>>> * Number to blind transfer to is 2001
>>>>>>>>> * Call to mod_xml_curl is made
>>>>>>>>>
>>>>>>>>> It makes reference to the User in the following 'clean' variables
>>>>>>>>> (by clean, I mean variables that just contain 2000 and don't require
>>>>>>>>> mangling to extract the info);
>>>>>>>>>
>>>>>>>>>   u'Caller-ANI': u'2000',
>>>>>>>>>   u'Caller-Username': u'2000',
>>>>>>>>>   u'Caller-Caller-ID-Number': u'2000',
>>>>>>>>>   u'Hunt-ANI': u'2000',
>>>>>>>>>   u'Hunt-Caller-ID-Number': u'2000',
>>>>>>>>>   u'Hunt-Username': u'2000',
>>>>>>>>>   u'variable_last_sent_callee_id_number': u'2000',
>>>>>>>>>   u'variable_sip_from_user': u'2000',
>>>>>>>>>
>>>>>>>>> It also has the User in the following unclean variables;
>>>>>>>>>
>>>>>>>>>  u'variable_bridge_channel': u'sofia/external/
>>>>>>>>> 2000 at c1881.voiceflare.co.uk:5060',
>>>>>>>>>   u'variable_sip_from_uri': u'2000 at 89.238.182.137',
>>>>>>>>>  u'variable_sip_full_from': u'"foxx" <sip:2000 at 89.238.182.137
>>>>>>>>> >;tag=XryjFQp3rB2NF',
>>>>>>>>>   u'variable_sip_h_Referred-By': u'"foxx" <
>>>>>>>>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>>>>>>>>
>>>>>>>>> However, it only references the domain in the following unclean
>>>>>>>>> variables;
>>>>>>>>>
>>>>>>>>>  u'variable_bridge_channel': u'sofia/external/
>>>>>>>>> 2000 at c1881.voiceflare.co.uk:5060',
>>>>>>>>>  u'variable_sip_h_Referred-By': u'"foxx" <
>>>>>>>>> sip:2000 at c1881.voiceflare.co.uk:5060>',
>>>>>>>>>  u'variable_sip_refer_to': u'<sip:2001 at c1881.voiceflare.co.uk:5060
>>>>>>>>> >',
>>>>>>>>>
>>>>>>>>> Lets say that we want to determine the user/domain that has
>>>>>>>>> initiated this transfer, doing so would mean mangling with one of those
>>>>>>>>> above variables, which seems a bit dirty (plus it is not clear which is the
>>>>>>>>> correct one to use).
>>>>>>>>>
>>>>>>>>> I have attached the SIP trace of the entire blind transfer event,
>>>>>>>>> and the full mod_xml_curl request being sent.
>>>>>>>>>
>>>>>>>>>  If anyone could answer the following, it'd be much appreciated;
>>>>>>>>>
>>>>>>>>> * Should there be an improvement patch that exposes the domain of
>>>>>>>>> the user that originated the blind transfer?
>>>>>>>>> * Are there better/alternative ways to extracting the domain of
>>>>>>>>> the user that originated the blind transfer?
>>>>>>>>>
>>>>>>>>> Many thanks
>>>>>>>>>
>>>>>>>>> Cal
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting 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
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> 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
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gerald.weber at besharp.at  Mon Dec 31 15:30:24 2012
From: gerald.weber at besharp.at (Gerald Weber)
Date: Mon, 31 Dec 2012 12:30:24 +0000
Subject: [Freeswitch-users] mod_event_erlang and invalid process type
Message-ID: <D1925DECD316894DAADB84076C1B7D827E2F4852@exchange.besharp.at>

Hi all,

trying to learn erlang i tried to connect to freeswitch using mod_event_erlang and receive some events:

erlang shell:

[root at fstest ~]# erl -name test33 at freeswitch.besharp.at -setcookie ClueCon
Erlang R15B03 (erts-5.9.3.1) [source] [64-bit] [smp:2:2] [async-threads:0] [hipe] [kernel-poll:false]

Eshell V5.9.3.1  (abort with ^G)
(test33 at freeswitch.besharp.at)1> Pid = fsevents:start().
<0.46.0>
(test33 at freeswitch.besharp.at)2> fsevents:init('freeswitch at freeswitch.besharp.at').
{event,'ALL'}
(test33 at freeswitch.besharp.at)3>


Console output:

2012-12-31 13:18:36.900506 [DEBUG] handle_msg.c:301 ALL events enabled
2012-12-31 13:18:36.900506 [DEBUG] handle_msg.c:314 enable event ALL
2012-12-31 13:18:51.480577 [ERR] ei_helpers.c:274 Invalid process type 0! ((null),, <0.0.0>)
2012-12-31 13:18:51.480577 [ERR] ei_helpers.c:274 Invalid process type 0! ((null),, <0.0.0>)
2012-12-31 13:18:51.500492 [ERR] ei_helpers.c:274 Invalid process type 0! ((null),, <0.0.0>)

Can anyone tell me what i'm doing wrong ? Console log keeps saying invalid process type (until i kill the erlang process)
and i dont receive any events.

Erlang Version: 5.9.3.1
Freeswitch: latest HEAD from 2012-12-31

The fsevents module:

-module(fsevents).
-export([start/0, init/1, loop/0]).

start() ->
                Pid = spawn(?MODULE, loop, []),
                register(?MODULE,Pid),
                Pid.

init(Node) ->
                {foo, Node} ! {event, 'ALL'}.

loop() ->
                receive
                               stop ->
                                               ok;
                               _Anything ->
                                               io:format("received ~p~n", [_Anything]),
                                               loop()
                end.


The communication seems to work basically:


(test33 at freeswitch.besharp.at)4> {foo, freeswitch at freeswitch.besharp.at} ! {api, status, ""}.
{api,status,[]}
(test33 at freeswitch.besharp.at)5> receive Y -> Y after 1000 -> timeout end.
{ok,"UP 0 years, 0 days, 1 hour, 20 minutes, 52 seconds, 534 milliseconds, 225 microseconds\nFreeSWITCH (Version 1.3.13b git 8859eb0 2012-12-30 19:08:55Z) is ready\n0 session(s) since startup\n0 session(s) - 0 out of max 30 per sec \n1000 session(s) max\nmin idle cpu 0.00/99.00\nCurrent Stack Size/Max 240K/8192K\n"}
(test33 at freeswitch.besharp.at)3<mailto:test33 at freeswitch.besharp.at)3>>

Console output:

freeswitch at fstest> load mod_erlang_event
2012-12-31 13:17:21.420611 [INFO] mod_enum.c:872 ENUM Reloaded
2012-12-31 13:17:21.420611 [INFO] switch_time.c:1165 Timezone reloaded 530 definitions
2012-12-31 13:17:21.420611 [DEBUG] mod_erlang_event.c:1816 sections 16
2012-12-31 13:17:21.420611 [CONSOLE] switch_loadable_module.c:1348 Successfully Loaded [mod_erlang_event]
2012-12-31 13:17:21.420611 [NOTICE] switch_loadable_module.c:254 Adding Application 'erlang'
2012-12-31 13:17:21.420611 [NOTICE] switch_loadable_module.c:254 Adding Application 'erlang_sendmsg'
2012-12-31 13:17:21.420611 [NOTICE] switch_loadable_module.c:298 Adding API Function 'erlang'

+OK Reloading XML
+OK

2012-12-31 13:17:21.420611 [DEBUG] mod_erlang_event.c:1915 Socket 48 up listening on 0.0.0.0:8031
2012-12-31 13:17:21.420611 [DEBUG] mod_erlang_event.c:1946 Connected to epmd and published erlang cnode at freeswitch at freeswitch.besharp.at
freeswitch at fstest> 2012-12-31 13:17:43.380527 [DEBUG] mod_erlang_event.c:1986 Launching listener, connection from node test33 at freeswitch.besharp.at, ip 192.168.20.73
2012-12-31 13:17:43.380527 [DEBUG] mod_erlang_event.c:1009 Connection Open from 192.168.20.73

Thanks / regards and Happy New Year :)
Gw

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From ben at langfeld.co.uk  Mon Dec 31 16:03:18 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Mon, 31 Dec 2012 13:03:18 +0000
Subject: [Freeswitch-users] uuid_record and RECORD_STEREO
In-Reply-To: <4C61A84750854AF4822B728CEE0B3A56@gmail.com>
References: <89F15D50-E189-4D86-9320-102F086DF18F@gmail.com>
	<CAKzWOxX+167M=k7acjY45H=WA2_eprwbs3S=L-9VZyTDCbQ4FQ@mail.gmail.com>
	<CAAyX+KHVjRmvY_6sYAAi+5SxXZdOc72cn4_DdV466rpb7JYX+w@mail.gmail.com>
	<4C61A84750854AF4822B728CEE0B3A56@gmail.com>
Message-ID: <CAAyX+KG4YMRgz9SocH-jm2Uzi5k-FKTFTP9cC_GmJdqig3o8Kw@mail.gmail.com>

Right, but that's not the same thing... I don't want to unset a global
value. I want to avoid setting a global in the first place. The reasons for
this should be obvious (global state + concurrency = bad).

Regards,
Ben Langfeld


On 30 December 2012 23:21, Seven Du <dujinfang at gmail.com> wrote:

>  uuid_setvar with no val will unset the var
>
> --
> Seven Du
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Monday, December 31, 2012 at 4:47 AM, Ben Langfeld wrote:
>
> If uuid_setvar is used, it affects all future invocations of uuid_record.
> This is global channel state. What we need is to invoke uuid_record with
> that var set locally for that invocation, such that it doesn't affect
> future invocations (a local variable).
>
> Regards,
> Ben Langfeld
>
>
> On 27 December 2012 17:45, Michael Collins <msc at freeswitch.org> wrote:
>
> I'm not sure I understand the question. Could you elaborate? What do you
> mean by "set a global state on the channel"?
>
> -MC
>
>
> On Thu, Dec 27, 2012 at 3:55 AM, Luca Pradovera <luca.pradovera at gmail.com>wrote:
>
> Hello,
> is there a way to invoke uuid_record through bgapi, setting RECORD_STEREO
> to true as a local parameter instead of using uuid_setvar, over inbound
> Event Socket?
> I would prefer to not set a global state on the channel if possible.
> Thanks!
>
> --
> Luca Pradovera
> luca.pradovera at gmail.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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 gerald.weber at besharp.at  Mon Dec 31 16:12:16 2012
From: gerald.weber at besharp.at (Gerald Weber)
Date: Mon, 31 Dec 2012 13:12:16 +0000
Subject: [Freeswitch-users] mod_event_erlang and invalid process type
In-Reply-To: <D1925DECD316894DAADB84076C1B7D827E2F4852@exchange.besharp.at>
References: <D1925DECD316894DAADB84076C1B7D827E2F4852@exchange.besharp.at>
Message-ID: <D1925DECD316894DAADB84076C1B7D827E2F489A@exchange.besharp.at>

Some more infos:

Changed the init/1 function to:

init(Node) ->
                {foo, Node} ! register_event_handler,
                {foo, Node} ! register_log_handler,
                {foo, Node} ! {event, 'ALL'},
                {foo, Node} ! {set_log_level, debug}.

Now the "invalid process type" error message disappeared, but i still dont get any events.

Some erlang console command outputs:

freeswitch at fstest> erlang listeners

Listener to test33 at freeswitch.besharp.at with 0 outbound sessions

freeswitch at fstest> erlang bindings

No bindings

freeswitch at fstest> erlang handlers

Listener test33 at freeswitch.besharp.at:
--------------------------------
CLONE
CHANNEL_CREATE
CHANNEL_DESTROY
CHANNEL_STATE
CHANNEL_CALLSTATE
CHANNEL_ANSWER
CHANNEL_HANGUP
CHANNEL_HANGUP_COMPLETE
CHANNEL_EXECUTE
CHANNEL_EXECUTE_COMPLETE
CHANNEL_HOLD
CHANNEL_UNHOLD
CHANNEL_BRIDGE
CHANNEL_UNBRIDGE
CHANNEL_PROGRESS
CHANNEL_PROGRESS_MEDIA
CHANNEL_OUTGOING
CHANNEL_PARK
CHANNEL_UNPARK
CHANNEL_APPLICATION
CHANNEL_ORIGINATE
CHANNEL_UUID
API
LOG
INBOUND_CHAN
OUTBOUND_CHAN
STARTUP
SHUTDOWN
PUBLISH
UNPUBLISH
TALK
NOTALK
SESSION_CRASH
MODULE_LOAD
MODULE_UNLOAD
DTMF
MESSAGE
PRESENCE_IN
NOTIFY_IN
PRESENCE_OUT
PRESENCE_PROBE
MESSAGE_WAITING
MESSAGE_QUERY
ROSTER
CODEC
BACKGROUND_JOB
DETECTED_SPEECH
DETECTED_TONE
PRIVATE_COMMAND
HEARTBEAT
TRAP
ADD_SCHEDULE
DEL_SCHEDULE
EXE_SCHEDULE
RE_SCHEDULE
RELOADXML
NOTIFY
SEND_MESSAGE
RECV_MESSAGE
REQUEST_PARAMS
CHANNEL_DATA
GENERAL
COMMAND
SESSION_HEARTBEAT
CLIENT_DISCONNECTED
SERVER_DISCONNECTED
SEND_INFO
RECV_INFO
RECV_RTCP_MESSAGE
CALL_SECURE
NAT
RECORD_START
RECORD_STOP
PLAYBACK_START
PLAYBACK_STOP
CALL_UPDATE
FAILURE
SOCKET_DATA
MEDIA_BUG_START
MEDIA_BUG_STOP
CONFERENCE_DATA_QUERY
CONFERENCE_DATA
CALL_SETUP_REQ
CALL_SETUP_RESULT
CUSTOM:

Thanks
gw

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Gerald Weber
Gesendet: Montag, 31. Dezember 2012 13:30
An: freeswitch-users at lists.freeswitch.org
Betreff: [Freeswitch-users] mod_event_erlang and invalid process type

Hi all,

trying to learn erlang i tried to connect to freeswitch using mod_event_erlang and receive some events:

erlang shell:

[root at fstest ~]# erl -name test33 at freeswitch.besharp.at<mailto:test33 at freeswitch.besharp.at> -setcookie ClueCon
Erlang R15B03 (erts-5.9.3.1) [source] [64-bit] [smp:2:2] [async-threads:0] [hipe] [kernel-poll:false]

Eshell V5.9.3.1  (abort with ^G)
(test33 at freeswitch.besharp.at)1<mailto:test33 at freeswitch.besharp.at)1>> Pid = fsevents:start().
<0.46.0>
(test33 at freeswitch.besharp.at)2<mailto:test33 at freeswitch.besharp.at)2>> fsevents:init('freeswitch at freeswitch.besharp.at').
{event,'ALL'}
(test33 at freeswitch.besharp.at)3<mailto:test33 at freeswitch.besharp.at)3>>


Console output:

2012-12-31 13:18:36.900506 [DEBUG] handle_msg.c:301 ALL events enabled
2012-12-31 13:18:36.900506 [DEBUG] handle_msg.c:314 enable event ALL
2012-12-31 13:18:51.480577 [ERR] ei_helpers.c:274 Invalid process type 0! ((null),, <0.0.0>)
2012-12-31 13:18:51.480577 [ERR] ei_helpers.c:274 Invalid process type 0! ((null),, <0.0.0>)
2012-12-31 13:18:51.500492 [ERR] ei_helpers.c:274 Invalid process type 0! ((null),, <0.0.0>)

Can anyone tell me what i'm doing wrong ? Console log keeps saying invalid process type (until i kill the erlang process)
and i dont receive any events.

Erlang Version: 5.9.3.1
Freeswitch: latest HEAD from 2012-12-31

The fsevents module:

-module(fsevents).
-export([start/0, init/1, loop/0]).

start() ->
                Pid = spawn(?MODULE, loop, []),
                register(?MODULE,Pid),
                Pid.

init(Node) ->
                {foo, Node} ! {event, 'ALL'}.

loop() ->
                receive
                               stop ->
                                               ok;
                               _Anything ->
                                               io:format("received ~p~n", [_Anything]),
                                               loop()
                end.


The communication seems to work basically:


(test33 at freeswitch.besharp.at)4<mailto:test33 at freeswitch.besharp.at)4>> {foo, freeswitch at freeswitch.besharp.at<mailto:freeswitch at freeswitch.besharp.at>} ! {api, status, ""}.
{api,status,[]}
(test33 at freeswitch.besharp.at)5<mailto:test33 at freeswitch.besharp.at)5>> receive Y -> Y after 1000 -> timeout end.
{ok,"UP 0 years, 0 days, 1 hour, 20 minutes, 52 seconds, 534 milliseconds, 225 microseconds\nFreeSWITCH (Version 1.3.13b git 8859eb0 2012-12-30 19:08:55Z) is ready\n0 session(s) since startup\n0 session(s) - 0 out of max 30 per sec \n1000 session(s) max\nmin idle cpu 0.00/99.00\nCurrent Stack Size/Max 240K/8192K\n"}
(test33 at freeswitch.besharp.at)3<mailto:test33 at freeswitch.besharp.at)3>>

Console output:

freeswitch at fstest> load mod_erlang_event
2012-12-31 13:17:21.420611 [INFO] mod_enum.c:872 ENUM Reloaded
2012-12-31 13:17:21.420611 [INFO] switch_time.c:1165 Timezone reloaded 530 definitions
2012-12-31 13:17:21.420611 [DEBUG] mod_erlang_event.c:1816 sections 16
2012-12-31 13:17:21.420611 [CONSOLE] switch_loadable_module.c:1348 Successfully Loaded [mod_erlang_event]
2012-12-31 13:17:21.420611 [NOTICE] switch_loadable_module.c:254 Adding Application 'erlang'
2012-12-31 13:17:21.420611 [NOTICE] switch_loadable_module.c:254 Adding Application 'erlang_sendmsg'
2012-12-31 13:17:21.420611 [NOTICE] switch_loadable_module.c:298 Adding API Function 'erlang'

+OK Reloading XML
+OK

2012-12-31 13:17:21.420611 [DEBUG] mod_erlang_event.c:1915 Socket 48 up listening on 0.0.0.0:8031
2012-12-31 13:17:21.420611 [DEBUG] mod_erlang_event.c:1946 Connected to epmd and published erlang cnode at freeswitch at freeswitch.besharp.at<mailto:freeswitch at freeswitch.besharp.at>
freeswitch at fstest> 2012-12-31 13:17:43.380527 [DEBUG] mod_erlang_event.c:1986 Launching listener, connection from node test33 at freeswitch.besharp.at<mailto:test33 at freeswitch.besharp.at>, ip 192.168.20.73
2012-12-31 13:17:43.380527 [DEBUG] mod_erlang_event.c:1009 Connection Open from 192.168.20.73

Thanks / regards and Happy New Year :)
Gw

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From steveayre at gmail.com  Mon Dec 31 16:58:13 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 31 Dec 2012 13:58:13 +0000
Subject: [Freeswitch-users] uuid_record and RECORD_STEREO
In-Reply-To: <CAAyX+KG4YMRgz9SocH-jm2Uzi5k-FKTFTP9cC_GmJdqig3o8Kw@mail.gmail.com>
References: <89F15D50-E189-4D86-9320-102F086DF18F@gmail.com>
	<CAKzWOxX+167M=k7acjY45H=WA2_eprwbs3S=L-9VZyTDCbQ4FQ@mail.gmail.com>
	<CAAyX+KHVjRmvY_6sYAAi+5SxXZdOc72cn4_DdV466rpb7JYX+w@mail.gmail.com>
	<4C61A84750854AF4822B728CEE0B3A56@gmail.com>
	<CAAyX+KG4YMRgz9SocH-jm2Uzi5k-FKTFTP9cC_GmJdqig3o8Kw@mail.gmail.com>
Message-ID: <CAFiqYun8Ori1L6tX48XCv0zrKK7Tu3RTgqZzYL+fyg8BiB6ecA@mail.gmail.com>

uuid_setvar sets a channel variable. But this *isn't* a *global* variable -
it doesn't affect any other calls.

You're correct though, it'll affect any future uuid_record invocations on
the *same* channel. But you probably shouldn't be executing stuff on the
same channel from multiple scripts so that shouldn't cause a problem.

The recording only reads the setting from the channel variable, so setting
the channel variable is the only way.




On 31 December 2012 13:03, Ben Langfeld <ben at langfeld.co.uk> wrote:

> Right, but that's not the same thing... I don't want to unset a global
> value. I want to avoid setting a global in the first place. The reasons for
> this should be obvious (global state + concurrency = bad).
>
> Regards,
> Ben Langfeld
>
>
> On 30 December 2012 23:21, Seven Du <dujinfang at gmail.com> wrote:
>
>>  uuid_setvar with no val will unset the var
>>
>> --
>> Seven Du
>> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>>
>> On Monday, December 31, 2012 at 4:47 AM, Ben Langfeld wrote:
>>
>> If uuid_setvar is used, it affects all future invocations of uuid_record.
>> This is global channel state. What we need is to invoke uuid_record with
>> that var set locally for that invocation, such that it doesn't affect
>> future invocations (a local variable).
>>
>> Regards,
>> Ben Langfeld
>>
>>
>> On 27 December 2012 17:45, Michael Collins <msc at freeswitch.org> wrote:
>>
>> I'm not sure I understand the question. Could you elaborate? What do you
>> mean by "set a global state on the channel"?
>>
>> -MC
>>
>>
>> On Thu, Dec 27, 2012 at 3:55 AM, Luca Pradovera <luca.pradovera at gmail.com
>> > wrote:
>>
>> Hello,
>> is there a way to invoke uuid_record through bgapi, setting RECORD_STEREO
>> to true as a local parameter instead of using uuid_setvar, over inbound
>> Event Socket?
>> I would prefer to not set a global state on the channel if possible.
>> Thanks!
>>
>> --
>> Luca Pradovera
>> luca.pradovera at gmail.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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ben at langfeld.co.uk  Mon Dec 31 17:38:36 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Mon, 31 Dec 2012 14:38:36 +0000
Subject: [Freeswitch-users] uuid_record and RECORD_STEREO
In-Reply-To: <CAFiqYun8Ori1L6tX48XCv0zrKK7Tu3RTgqZzYL+fyg8BiB6ecA@mail.gmail.com>
References: <89F15D50-E189-4D86-9320-102F086DF18F@gmail.com>
	<CAKzWOxX+167M=k7acjY45H=WA2_eprwbs3S=L-9VZyTDCbQ4FQ@mail.gmail.com>
	<CAAyX+KHVjRmvY_6sYAAi+5SxXZdOc72cn4_DdV466rpb7JYX+w@mail.gmail.com>
	<4C61A84750854AF4822B728CEE0B3A56@gmail.com>
	<CAAyX+KG4YMRgz9SocH-jm2Uzi5k-FKTFTP9cC_GmJdqig3o8Kw@mail.gmail.com>
	<CAFiqYun8Ori1L6tX48XCv0zrKK7Tu3RTgqZzYL+fyg8BiB6ecA@mail.gmail.com>
Message-ID: <CAAyX+KFRrq82xhPJEmA3ax3t7FUssvzC-4FcvJ0Q+udS2hF_vg@mail.gmail.com>

It's global in the scope of the channel (from the perspective of individual
bugs). I don't think "you shouldn't be invoking this async interface from
multiple places" is a fantastic argument. The fact is it *can* be, and so
we should tighten up this race condition if we can. Can anyone provide
guidance on how to add this as an argument to uuid_record, while falling
back to the channel variable?

Regards,
Ben Langfeld


On 31 December 2012 13:58, Steven Ayre <steveayre at gmail.com> wrote:

> uuid_setvar sets a channel variable. But this *isn't* a *global* variable
> - it doesn't affect any other calls.
>
> You're correct though, it'll affect any future uuid_record invocations on
> the *same* channel. But you probably shouldn't be executing stuff on the
> same channel from multiple scripts so that shouldn't cause a problem.
>
> The recording only reads the setting from the channel variable, so setting
> the channel variable is the only way.
>
>
>
>
> On 31 December 2012 13:03, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
>> Right, but that's not the same thing... I don't want to unset a global
>> value. I want to avoid setting a global in the first place. The reasons for
>> this should be obvious (global state + concurrency = bad).
>>
>> Regards,
>> Ben Langfeld
>>
>>
>> On 30 December 2012 23:21, Seven Du <dujinfang at gmail.com> wrote:
>>
>>>  uuid_setvar with no val will unset the var
>>>
>>> --
>>> Seven Du
>>> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>>>
>>> On Monday, December 31, 2012 at 4:47 AM, Ben Langfeld wrote:
>>>
>>> If uuid_setvar is used, it affects all future invocations of
>>> uuid_record. This is global channel state. What we need is to invoke
>>> uuid_record with that var set locally for that invocation, such that it
>>> doesn't affect future invocations (a local variable).
>>>
>>> Regards,
>>> Ben Langfeld
>>>
>>>
>>> On 27 December 2012 17:45, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>> I'm not sure I understand the question. Could you elaborate? What do you
>>> mean by "set a global state on the channel"?
>>>
>>> -MC
>>>
>>>
>>> On Thu, Dec 27, 2012 at 3:55 AM, Luca Pradovera <
>>> luca.pradovera at gmail.com> wrote:
>>>
>>> Hello,
>>> is there a way to invoke uuid_record through bgapi, setting
>>> RECORD_STEREO to true as a local parameter instead of using uuid_setvar,
>>> over inbound Event Socket?
>>> I would prefer to not set a global state on the channel if possible.
>>> Thanks!
>>>
>>> --
>>> Luca Pradovera
>>> luca.pradovera at gmail.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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> Michael S Collins
>>> Twitter: @mercutioviz
>>> http://www.FreeSWITCH.org
>>> http://www.ClueCon.com
>>> http://www.OSTAG.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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 abaci64 at gmail.com  Mon Dec 31 19:39:36 2012
From: abaci64 at gmail.com (Abaci)
Date: Mon, 31 Dec 2012 11:39:36 -0500
Subject: [Freeswitch-users] Problem with Polycom
Message-ID: <50E1BFC8.9060503@gmail.com>

We have OpenSIPS in front of FreeSWITCH, phones register to OpenSIPS, 
OpenSIPS sends all calls to FreeSWITCH. the way we have it set up is 
that both OpenSIPS and FreeSWITCH auth all the invites (we have it like 
this for a reason). this setup works fine (with Yealink  at least).

Problem we have is with Polycom phones, the phone registers just fine 
but can't make calls. When OpenSIPS gets the INVITE it responds with a 
407, to which the Polycom responds, OpenSIPS is then sending the call to 
FreeSWITCH which sends a 407 again, OpenSIPS proxies it back to the 
polycom, however the Polycom does not respond to the challenge.

I don't think this is a FreeSWITCH problem, I would still like to hear 
if anyone has an idea how I can get this working (config option?)

Thanks


From cmrienzo at gmail.com  Mon Dec 31 19:42:19 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Mon, 31 Dec 2012 11:42:19 -0500
Subject: [Freeswitch-users] uuid_record and RECORD_STEREO
In-Reply-To: <CAAyX+KFRrq82xhPJEmA3ax3t7FUssvzC-4FcvJ0Q+udS2hF_vg@mail.gmail.com>
References: <89F15D50-E189-4D86-9320-102F086DF18F@gmail.com>
	<CAKzWOxX+167M=k7acjY45H=WA2_eprwbs3S=L-9VZyTDCbQ4FQ@mail.gmail.com>
	<CAAyX+KHVjRmvY_6sYAAi+5SxXZdOc72cn4_DdV466rpb7JYX+w@mail.gmail.com>
	<4C61A84750854AF4822B728CEE0B3A56@gmail.com>
	<CAAyX+KG4YMRgz9SocH-jm2Uzi5k-FKTFTP9cC_GmJdqig3o8Kw@mail.gmail.com>
	<CAFiqYun8Ori1L6tX48XCv0zrKK7Tu3RTgqZzYL+fyg8BiB6ecA@mail.gmail.com>
	<CAAyX+KFRrq82xhPJEmA3ax3t7FUssvzC-4FcvJ0Q+udS2hF_vg@mail.gmail.com>
Message-ID: <CAD-qRkiVkm3GXff13wwxzoM91uomXLr0KCFOXBqz5rquZZyfcQ@mail.gmail.com>

Most (if not all) uuid_ API commands are implemented in mod_commands.c.
 Session recording is implemented in switch_ivr_async.c.  Changing
uuid_record shouldn't be too difficult.

Chris


On Mon, Dec 31, 2012 at 9:38 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> It's global in the scope of the channel (from the perspective of
> individual bugs). I don't think "you shouldn't be invoking this async
> interface from multiple places" is a fantastic argument. The fact is it
> *can* be, and so we should tighten up this race condition if we can. Can
> anyone provide guidance on how to add this as an argument to uuid_record,
> while falling back to the channel variable?
>
> Regards,
> Ben Langfeld
>
>
> On 31 December 2012 13:58, Steven Ayre <steveayre at gmail.com> wrote:
>
>> uuid_setvar sets a channel variable. But this *isn't* a *global*variable - it doesn't affect any other calls.
>>
>> You're correct though, it'll affect any future uuid_record invocations on
>> the *same* channel. But you probably shouldn't be executing stuff on the
>> same channel from multiple scripts so that shouldn't cause a problem.
>>
>> The recording only reads the setting from the channel variable, so
>> setting the channel variable is the only way.
>>
>>
>>
>>
>> On 31 December 2012 13:03, Ben Langfeld <ben at langfeld.co.uk> wrote:
>>
>>> Right, but that's not the same thing... I don't want to unset a global
>>> value. I want to avoid setting a global in the first place. The reasons for
>>> this should be obvious (global state + concurrency = bad).
>>>
>>> Regards,
>>> Ben Langfeld
>>>
>>>
>>> On 30 December 2012 23:21, Seven Du <dujinfang at gmail.com> wrote:
>>>
>>>>  uuid_setvar with no val will unset the var
>>>>
>>>> --
>>>> Seven Du
>>>> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>>>>
>>>> On Monday, December 31, 2012 at 4:47 AM, Ben Langfeld wrote:
>>>>
>>>> If uuid_setvar is used, it affects all future invocations of
>>>> uuid_record. This is global channel state. What we need is to invoke
>>>> uuid_record with that var set locally for that invocation, such that it
>>>> doesn't affect future invocations (a local variable).
>>>>
>>>> Regards,
>>>> Ben Langfeld
>>>>
>>>>
>>>> On 27 December 2012 17:45, Michael Collins <msc at freeswitch.org> wrote:
>>>>
>>>> I'm not sure I understand the question. Could you elaborate? What do
>>>> you mean by "set a global state on the channel"?
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Thu, Dec 27, 2012 at 3:55 AM, Luca Pradovera <
>>>> luca.pradovera at gmail.com> wrote:
>>>>
>>>> Hello,
>>>> is there a way to invoke uuid_record through bgapi, setting
>>>> RECORD_STEREO to true as a local parameter instead of using uuid_setvar,
>>>> over inbound Event Socket?
>>>> I would prefer to not set a global state on the channel if possible.
>>>> Thanks!
>>>>
>>>> --
>>>> Luca Pradovera
>>>> luca.pradovera at gmail.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
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Michael S Collins
>>>> Twitter: @mercutioviz
>>>> http://www.FreeSWITCH.org
>>>> http://www.ClueCon.com
>>>> http://www.OSTAG.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting 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
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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