From emilbergg at gmail.com  Sun Mar  3 14:01:41 2013
From: emilbergg at gmail.com (Emil Berg)
Date: Sun, 3 Mar 2013 13:01:41 +0200
Subject: [Freeswitch-dev] Development Info
Message-ID: <CAAbVz80A4+cj6pZV=hxS3_XiAejgj6TeENynhAYnHch=3BHXYg@mail.gmail.com>

Hello,

I'm new to freewitch and I would like to develop an endpoint.
Is there any book / tutorial / other info about development in Freeswitch?
I couldn't find any info on the internet, but there are relevant courses,
which take place once a year, so there must be some presntations / other
documents.

I'll be glad if someone can tell me where to download / buy the books or
documents.

Thanks,
Emil.
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From gvvsubhashkumar at gmail.com  Mon Mar  4 10:43:07 2013
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Mon, 4 Mar 2013 13:13:07 +0530
Subject: [Freeswitch-dev] Unable to recreate Log Files
Message-ID: <CAFvodxULBGuC=fSGfy187TFi9FidkTtc+P4YLv=iC4Q6Qeo56g@mail.gmail.com>

Hi,

I used the latest binaries(Windows Installer),version info is given below
and observered that when it is not able to rename the log file it is keep
on writing the logs in freeswitch.log file which is crossing the limt set
in logconf.xml file.

The error seen in the freeswitch log file is

[CRIT] mod_logfile.c:164 Error renaming log from C:/Program
Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program
Files/FreeSWITCH/log/freeswitch.log.5 [No error]

And the version info is

Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z

*NOTE* : This issue is critical as we are facing it at production systems.

Thanks,
Subhash.
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From steveayre at gmail.com  Mon Mar  4 16:45:26 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 4 Mar 2013 13:45:26 +0000
Subject: [Freeswitch-dev] Unable to recreate Log Files
In-Reply-To: <CAFvodxULBGuC=fSGfy187TFi9FidkTtc+P4YLv=iC4Q6Qeo56g@mail.gmail.com>
References: <CAFvodxULBGuC=fSGfy187TFi9FidkTtc+P4YLv=iC4Q6Qeo56g@mail.gmail.com>
Message-ID: <CAFiqYunn7RJP-bTnU3+m9sRrPDrnMRxM113BEc0n4ZGAC1yymQ@mail.gmail.com>

Does the FS user have write permissions to the C:/Program
Files/FreeSWITCH/log folder?



On 4 March 2013 07:43, Subhash <gvvsubhashkumar at gmail.com> wrote:

> Hi,
>
> I used the latest binaries(Windows Installer),version info is given below
> and observered that when it is not able to rename the log file it is keep
> on writing the logs in freeswitch.log file which is crossing the limt set
> in logconf.xml file.
>
> The error seen in the freeswitch log file is
>
> [CRIT] mod_logfile.c:164 Error renaming log from C:/Program
> Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program
> Files/FreeSWITCH/log/freeswitch.log.5 [No error]
>
> And the version info is
>
> Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z
>
> *NOTE* : This issue is critical as we are facing it at production systems.
>
> 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-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From gvvsubhashkumar at gmail.com  Mon Mar  4 17:03:43 2013
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Mon, 4 Mar 2013 19:33:43 +0530
Subject: [Freeswitch-dev] Unable to recreate Log Files
In-Reply-To: <CAFiqYunn7RJP-bTnU3+m9sRrPDrnMRxM113BEc0n4ZGAC1yymQ@mail.gmail.com>
References: <CAFvodxULBGuC=fSGfy187TFi9FidkTtc+P4YLv=iC4Q6Qeo56g@mail.gmail.com>
	<CAFiqYunn7RJP-bTnU3+m9sRrPDrnMRxM113BEc0n4ZGAC1yymQ@mail.gmail.com>
Message-ID: <CAFvodxWh2oOgMOPU=xz-912-PyQSSD51XOL3irn+vWE1P6o2ZA@mail.gmail.com>

Yes it has permissions.
On Mar 4, 2013 7:22 PM, "Steven Ayre" <steveayre at gmail.com> wrote:

> Does the FS user have write permissions to the C:/Program
> Files/FreeSWITCH/log folder?
>
>
>
> On 4 March 2013 07:43, Subhash <gvvsubhashkumar at gmail.com> wrote:
>
>> Hi,
>>
>> I used the latest binaries(Windows Installer),version info is given below
>> and observered that when it is not able to rename the log file it is keep
>> on writing the logs in freeswitch.log file which is crossing the limt set
>> in logconf.xml file.
>>
>> The error seen in the freeswitch log file is
>>
>> [CRIT] mod_logfile.c:164 Error renaming log from C:/Program
>> Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program
>> Files/FreeSWITCH/log/freeswitch.log.5 [No error]
>>
>> And the version info is
>>
>> Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z
>>
>> *NOTE* : This issue is critical as we are facing it at production
>> systems.
>>
>> 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-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Mon Mar  4 17:20:59 2013
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Mar 2013 09:20:59 -0500
Subject: [Freeswitch-dev] Development Info
In-Reply-To: <CAAbVz80A4+cj6pZV=hxS3_XiAejgj6TeENynhAYnHch=3BHXYg@mail.gmail.com>
References: <CAAbVz80A4+cj6pZV=hxS3_XiAejgj6TeENynhAYnHch=3BHXYg@mail.gmail.com>
Message-ID: <2EEC01B4-EBD5-4CD6-B3A4-A205C5444655@jerris.com>

What kind of endpoint are you thinking of developing? we can probably push you in the right direction.

Mike

On Mar 3, 2013, at 6:01 AM, Emil Berg <emilbergg at gmail.com> wrote:

> Hello,
> 
> I'm new to freewitch and I would like to develop an endpoint.
> Is there any book / tutorial / other info about development in Freeswitch?
> I couldn't find any info on the internet, but there are relevant courses, which take place once a year, so there must be some presntations / other documents.
> 
> I'll be glad if someone can tell me where to download / buy the books or documents.
> 



From gvvsubhashkumar at gmail.com  Mon Mar  4 18:50:39 2013
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Mon, 4 Mar 2013 21:20:39 +0530
Subject: [Freeswitch-dev] Unable to recreate Log Files
In-Reply-To: <CAFvodxWh2oOgMOPU=xz-912-PyQSSD51XOL3irn+vWE1P6o2ZA@mail.gmail.com>
References: <CAFvodxULBGuC=fSGfy187TFi9FidkTtc+P4YLv=iC4Q6Qeo56g@mail.gmail.com>
	<CAFiqYunn7RJP-bTnU3+m9sRrPDrnMRxM113BEc0n4ZGAC1yymQ@mail.gmail.com>
	<CAFvodxWh2oOgMOPU=xz-912-PyQSSD51XOL3irn+vWE1P6o2ZA@mail.gmail.com>
Message-ID: <CAFvodxXM5JEbbDsz+H4rARXoE_c+Wa50SFkLpZvL__MAWs8gFQ@mail.gmail.com>

Are there any specific reasons for this?


Thanks,
Subhash.


On Mon, Mar 4, 2013 at 7:33 PM, Subhash <gvvsubhashkumar at gmail.com> wrote:

> Yes it has permissions.
> On Mar 4, 2013 7:22 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>
>> Does the FS user have write permissions to the C:/Program
>> Files/FreeSWITCH/log folder?
>>
>>
>>
>> On 4 March 2013 07:43, Subhash <gvvsubhashkumar at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I used the latest binaries(Windows Installer),version info is given
>>> below and observered that when it is not able to rename the log file it is
>>> keep on writing the logs in freeswitch.log file which is crossing the limt
>>> set in logconf.xml file.
>>>
>>> The error seen in the freeswitch log file is
>>>
>>> [CRIT] mod_logfile.c:164 Error renaming log from C:/Program
>>> Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program
>>> Files/FreeSWITCH/log/freeswitch.log.5 [No error]
>>>
>>> And the version info is
>>>
>>> Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z
>>>
>>> *NOTE* : This issue is critical as we are facing it at production
>>> systems.
>>>
>>> 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-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
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From anthony.minessale at gmail.com  Mon Mar  4 23:57:27 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Mar 2013 14:57:27 -0600
Subject: [Freeswitch-dev] Unable to recreate Log Files
In-Reply-To: <CAFvodxXM5JEbbDsz+H4rARXoE_c+Wa50SFkLpZvL__MAWs8gFQ@mail.gmail.com>
References: <CAFvodxULBGuC=fSGfy187TFi9FidkTtc+P4YLv=iC4Q6Qeo56g@mail.gmail.com>
	<CAFiqYunn7RJP-bTnU3+m9sRrPDrnMRxM113BEc0n4ZGAC1yymQ@mail.gmail.com>
	<CAFvodxWh2oOgMOPU=xz-912-PyQSSD51XOL3irn+vWE1P6o2ZA@mail.gmail.com>
	<CAFvodxXM5JEbbDsz+H4rARXoE_c+Wa50SFkLpZvL__MAWs8gFQ@mail.gmail.com>
Message-ID: <CAKbxfG_L9_LyTjETNh2VZ21HPw5=0+mVMd=qgrPCKOWiTXCSnA@mail.gmail.com>

Do you have some idea why it's unable to rename it?  The log snippet
suggests no real reason but there must be one.

For instance if its unable to rename the file because the disk is full, its
really the same problem so you need to have a process to delete old logs.



On Mon, Mar 4, 2013 at 9:50 AM, Subhash <gvvsubhashkumar at gmail.com> wrote:

> Are there any specific reasons for this?
>
>
> Thanks,
> Subhash.
>
>
> On Mon, Mar 4, 2013 at 7:33 PM, Subhash <gvvsubhashkumar at gmail.com> wrote:
>
>> Yes it has permissions.
>> On Mar 4, 2013 7:22 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>>
>>> Does the FS user have write permissions to the C:/Program
>>> Files/FreeSWITCH/log folder?
>>>
>>>
>>>
>>> On 4 March 2013 07:43, Subhash <gvvsubhashkumar at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> I used the latest binaries(Windows Installer),version info is given
>>>> below and observered that when it is not able to rename the log file it is
>>>> keep on writing the logs in freeswitch.log file which is crossing the limt
>>>> set in logconf.xml file.
>>>>
>>>> The error seen in the freeswitch log file is
>>>>
>>>> [CRIT] mod_logfile.c:164 Error renaming log from C:/Program
>>>> Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program
>>>> Files/FreeSWITCH/log/freeswitch.log.5 [No error]
>>>>
>>>> And the version info is
>>>>
>>>> Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z
>>>>
>>>> *NOTE* : This issue is critical as we are facing it at production
>>>> systems.
>>>>
>>>> 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-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at 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 dxj19831029 at gmail.com  Tue Mar  5 10:36:36 2013
From: dxj19831029 at gmail.com (Xijing Dai)
Date: Tue, 5 Mar 2013 15:36:36 +0800
Subject: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?
Message-ID: <CAMsCbX9uZPcEvVzoec_1JZBBN-f4FnTNgtbKpSg2dVGD=L7o2w@mail.gmail.com>

hey


Is there any plan to upgrade freeswitch to use Sofia-SIP
1.12.11<http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.11.txt>
 ?

Cheers
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From POlsson at enghouse.com  Tue Mar  5 10:44:08 2013
From: POlsson at enghouse.com (Peter Olsson)
Date: Tue, 5 Mar 2013 07:44:08 +0000
Subject: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?
Message-ID: <1FFF97C269757C458224B7C895F35F15235CC8@cantor.std.visionutv.se>

I believe that the Sofia library within the FS source tree has more updates than this version - most of the updates in the last official Sofia release came from the FS developers anyway. So going to 1.12.11 would probably be a downgrade.

I'm not 100% sure about this though...

/Peter

Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Xijing Dai
Skickat: den 5 mars 2013 08:37
Till: freeswitch-dev at lists.freeswitch.org
?mne: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?

hey


Is there any plan to upgrade freeswitch to use Sofia-SIP 1.12.11<http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.11.txt> ?

Cheers
!DSPAM:51359cdf32761263612615!
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From gvvsubhashkumar at gmail.com  Tue Mar  5 11:06:14 2013
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Tue, 5 Mar 2013 13:36:14 +0530
Subject: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?
In-Reply-To: <1FFF97C269757C458224B7C895F35F15235CC8@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F15235CC8@cantor.std.visionutv.se>
Message-ID: <CAFvodxUhopQCU5kdUNG89LjQPKTp6ZJGCrZPYLEj-ijJF3C8cw@mail.gmail.com>

Does sofia-sip supports rtp rfc also?
On Mar 5, 2013 1:22 PM, "Peter Olsson" <POlsson at enghouse.com> wrote:

>  I believe that the Sofia library within the FS source tree has more
> updates than this version ? most of the updates in the last official Sofia
> release came from the FS developers anyway. So going to 1.12.11 would
> probably be a downgrade.****
>
> ** **
>
> I?m not 100% sure about this though...****
>
> ** **
>
> /Peter****
>
> ** **
>
> *Fr?n:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] *F?r *Xijing Dai
> *Skickat:* den 5 mars 2013 08:37
> *Till:* freeswitch-dev at lists.freeswitch.org
> *?mne:* [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?****
>
> ** **
>
> hey****
>
> ** **
>
> ** **
>
> Is there any plan to upgrade freeswitch to use Sofia-SIP 1.12.11<http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.11.txt>
>  ?****
>
> ** **
>
> Cheers****
>
> !DSPAM:51359cdf32761263612615! ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Mar  5 11:56:36 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 5 Mar 2013 08:56:36 +0000
Subject: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?
In-Reply-To: <CAFvodxUhopQCU5kdUNG89LjQPKTp6ZJGCrZPYLEj-ijJF3C8cw@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F15235CC8@cantor.std.visionutv.se>
	<CAFvodxUhopQCU5kdUNG89LjQPKTp6ZJGCrZPYLEj-ijJF3C8cw@mail.gmail.com>
Message-ID: <CAFiqYu=XJtrUW7QR=hCHc-gBQTfkOiAhXAcHFSBJyTJL5wxHoA@mail.gmail.com>

Sofia-SIP only provides the SIP signalling layer. FreeSWITCH uses its own
RTP stack for media, and in the progress of moving to crtp.

The current combination of libraries is well tested and stable, and as
Peter points out contain many patches that are improvements over upstream.

Because of this stability the developers aren't making any major updates to
the library versions at the moment. Master (1.3) is the development branch
for v1.2.stable. They're currently working towards the next stable 1.2
release, after which they plan to create a new 1.4 branch. 1.2 will
continue to be 'stable' while in 1.4 they will start doing major updates to
the libraries. That's likely to introduce a period of instability in 1.4
that you won't have if you remain on 1.2, the price for running on the
bleeding edge.

Is there a particular rea

-Steve



On 5 March 2013 08:06, Subhash <gvvsubhashkumar at gmail.com> wrote:

> Does sofia-sip supports rtp rfc also?
> On Mar 5, 2013 1:22 PM, "Peter Olsson" <POlsson at enghouse.com> wrote:
>
>>  I believe that the Sofia library within the FS source tree has more
>> updates than this version ? most of the updates in the last official Sofia
>> release came from the FS developers anyway. So going to 1.12.11 would
>> probably be a downgrade.****
>>
>> ** **
>>
>> I?m not 100% sure about this though...****
>>
>> ** **
>>
>> /Peter****
>>
>> ** **
>>
>> *Fr?n:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
>> freeswitch-dev-bounces at lists.freeswitch.org] *F?r *Xijing Dai
>> *Skickat:* den 5 mars 2013 08:37
>> *Till:* freeswitch-dev at lists.freeswitch.org
>> *?mne:* [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?****
>>
>> ** **
>>
>> hey****
>>
>> ** **
>>
>> ** **
>>
>> Is there any plan to upgrade freeswitch to use Sofia-SIP 1.12.11<http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.11.txt>
>>  ?****
>>
>> ** **
>>
>> Cheers****
>>
>> !DSPAM:51359cdf32761263612615! ****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Mar  5 11:57:25 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 5 Mar 2013 08:57:25 +0000
Subject: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?
In-Reply-To: <CAFiqYu=XJtrUW7QR=hCHc-gBQTfkOiAhXAcHFSBJyTJL5wxHoA@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F15235CC8@cantor.std.visionutv.se>
	<CAFvodxUhopQCU5kdUNG89LjQPKTp6ZJGCrZPYLEj-ijJF3C8cw@mail.gmail.com>
	<CAFiqYu=XJtrUW7QR=hCHc-gBQTfkOiAhXAcHFSBJyTJL5wxHoA@mail.gmail.com>
Message-ID: <CAFiqYumGDDXm9aocYbNuMLvDiCe7k2-4iRC59CVagkD6Xts1nQ@mail.gmail.com>

Is there a particular reason you're asking about Sofia-SIP 1.12.11? Are
there any specific new bugfixes/features in that version you're after?

-Steve




On 5 March 2013 08:56, Steven Ayre <steveayre at gmail.com> wrote:

> Sofia-SIP only provides the SIP signalling layer. FreeSWITCH uses its own
> RTP stack for media, and in the progress of moving to crtp.
>
> The current combination of libraries is well tested and stable, and as
> Peter points out contain many patches that are improvements over upstream.
>
> Because of this stability the developers aren't making any major updates
> to the library versions at the moment. Master (1.3) is the development
> branch for v1.2.stable. They're currently working towards the next stable
> 1.2 release, after which they plan to create a new 1.4 branch. 1.2 will
> continue to be 'stable' while in 1.4 they will start doing major updates to
> the libraries. That's likely to introduce a period of instability in 1.4
> that you won't have if you remain on 1.2, the price for running on the
> bleeding edge.
>
> Is there a particular rea
>
> -Steve
>
>
>
> On 5 March 2013 08:06, Subhash <gvvsubhashkumar at gmail.com> wrote:
>
>> Does sofia-sip supports rtp rfc also?
>> On Mar 5, 2013 1:22 PM, "Peter Olsson" <POlsson at enghouse.com> wrote:
>>
>>>  I believe that the Sofia library within the FS source tree has more
>>> updates than this version ? most of the updates in the last official Sofia
>>> release came from the FS developers anyway. So going to 1.12.11 would
>>> probably be a downgrade.****
>>>
>>> ** **
>>>
>>> I?m not 100% sure about this though...****
>>>
>>> ** **
>>>
>>> /Peter****
>>>
>>> ** **
>>>
>>> *Fr?n:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-dev-bounces at lists.freeswitch.org] *F?r *Xijing Dai
>>> *Skickat:* den 5 mars 2013 08:37
>>> *Till:* freeswitch-dev at lists.freeswitch.org
>>> *?mne:* [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?****
>>>
>>> ** **
>>>
>>> hey****
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> Is there any plan to upgrade freeswitch to use Sofia-SIP 1.12.11<http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.11.txt>
>>>  ?****
>>>
>>> ** **
>>>
>>> Cheers****
>>>
>>> !DSPAM:51359cdf32761263612615! ****
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
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From dxj19831029 at gmail.com  Tue Mar  5 13:18:36 2013
From: dxj19831029 at gmail.com (Xijing Dai)
Date: Tue, 5 Mar 2013 18:18:36 +0800
Subject: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?
In-Reply-To: <CAFiqYumGDDXm9aocYbNuMLvDiCe7k2-4iRC59CVagkD6Xts1nQ@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F15235CC8@cantor.std.visionutv.se>
	<CAFvodxUhopQCU5kdUNG89LjQPKTp6ZJGCrZPYLEj-ijJF3C8cw@mail.gmail.com>
	<CAFiqYu=XJtrUW7QR=hCHc-gBQTfkOiAhXAcHFSBJyTJL5wxHoA@mail.gmail.com>
	<CAFiqYumGDDXm9aocYbNuMLvDiCe7k2-4iRC59CVagkD6Xts1nQ@mail.gmail.com>
Message-ID: <CAMsCbX8u-ym0y5ZvZpZnQF7C3F+bSFP3Qw7Yam+Tz_TkO4L7JA@mail.gmail.com>

I had a very random sip error in the freeswitch, and I want to try
sofia-sip 1.12.11 to see if it's fixed there, but there are a few build
errors?

Is there any race condition chance inside mod_sofia?

We make a few calls from the same softphone simultaneously, it includes
pause/resume/terminate on any call at any time.

The outbound endpoint is going though loopback inside freeswitch.

Sometimes, the paused inbound call won't be able to resume, and after some
deep tracing inside freeswitch, it seems to me that the sip message for
resuming call is handled on outbound non-loopback channel.
And I used wireshark to catch the packets, it shows that the call is in
trying stage.

This happens very randomly.

Do you guys have any idea?

Cheers









On Tue, Mar 5, 2013 at 4:57 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Is there a particular reason you're asking about Sofia-SIP 1.12.11? Are
> there any specific new bugfixes/features in that version you're after?
>
> -Steve
>
>
>
>
> On 5 March 2013 08:56, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Sofia-SIP only provides the SIP signalling layer. FreeSWITCH uses its own
>> RTP stack for media, and in the progress of moving to crtp.
>>
>> The current combination of libraries is well tested and stable, and as
>> Peter points out contain many patches that are improvements over upstream.
>>
>> Because of this stability the developers aren't making any major updates
>> to the library versions at the moment. Master (1.3) is the development
>> branch for v1.2.stable. They're currently working towards the next stable
>> 1.2 release, after which they plan to create a new 1.4 branch. 1.2 will
>> continue to be 'stable' while in 1.4 they will start doing major updates to
>> the libraries. That's likely to introduce a period of instability in 1.4
>> that you won't have if you remain on 1.2, the price for running on the
>> bleeding edge.
>>
>> Is there a particular rea
>>
>> -Steve
>>
>>
>>
>> On 5 March 2013 08:06, Subhash <gvvsubhashkumar at gmail.com> wrote:
>>
>>> Does sofia-sip supports rtp rfc also?
>>> On Mar 5, 2013 1:22 PM, "Peter Olsson" <POlsson at enghouse.com> wrote:
>>>
>>>>  I believe that the Sofia library within the FS source tree has more
>>>> updates than this version ? most of the updates in the last official Sofia
>>>> release came from the FS developers anyway. So going to 1.12.11 would
>>>> probably be a downgrade.****
>>>>
>>>> ** **
>>>>
>>>> I?m not 100% sure about this though...****
>>>>
>>>> ** **
>>>>
>>>> /Peter****
>>>>
>>>> ** **
>>>>
>>>> *Fr?n:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
>>>> freeswitch-dev-bounces at lists.freeswitch.org] *F?r *Xijing Dai
>>>> *Skickat:* den 5 mars 2013 08:37
>>>> *Till:* freeswitch-dev at lists.freeswitch.org
>>>> *?mne:* [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?****
>>>>
>>>> ** **
>>>>
>>>> hey****
>>>>
>>>> ** **
>>>>
>>>> ** **
>>>>
>>>> Is there any plan to upgrade freeswitch to use Sofia-SIP 1.12.11<http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.11.txt>
>>>>  ?****
>>>>
>>>> ** **
>>>>
>>>> Cheers****
>>>>
>>>> !DSPAM:51359cdf32761263612615! ****
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From POlsson at enghouse.com  Tue Mar  5 14:17:28 2013
From: POlsson at enghouse.com (Peter Olsson)
Date: Tue, 5 Mar 2013 11:17:28 +0000
Subject: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?
Message-ID: <1FFF97C269757C458224B7C895F35F15235E1A@cantor.std.visionutv.se>

If this is an actual bug, it?s probably within FreeSWITCH itself. Please read instructions on the wiki for how to report problems, and file a Jira.

/Peter

Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Xijing Dai
Skickat: den 5 mars 2013 11:19
Till: freeswitch-dev at lists.freeswitch.org
?mne: Re: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?

I had a very random sip error in the freeswitch, and I want to try sofia-sip 1.12.11 to see if it's fixed there, but there are a few build errors?

Is there any race condition chance inside mod_sofia?

We make a few calls from the same softphone simultaneously, it includes pause/resume/terminate on any call at any time.

The outbound endpoint is going though loopback inside freeswitch.

Sometimes, the paused inbound call won't be able to resume, and after some deep tracing inside freeswitch, it seems to me that the sip message for resuming call is handled on outbound non-loopback channel.
And I used wireshark to catch the packets, it shows that the call is in trying stage.

This happens very randomly.

Do you guys have any idea?

Cheers









On Tue, Mar 5, 2013 at 4:57 PM, Steven Ayre <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
Is there a particular reason you're asking about Sofia-SIP 1.12.11? Are there any specific new bugfixes/features in that version you're after?

-Steve



On 5 March 2013 08:56, Steven Ayre <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
Sofia-SIP only provides the SIP signalling layer. FreeSWITCH uses its own RTP stack for media, and in the progress of moving to crtp.

The current combination of libraries is well tested and stable, and as Peter points out contain many patches that are improvements over upstream.

Because of this stability the developers aren't making any major updates to the library versions at the moment. Master (1.3) is the development branch for v1.2.stable. They're currently working towards the next stable 1.2 release, after which they plan to create a new 1.4 branch. 1.2 will continue to be 'stable' while in 1.4 they will start doing major updates to the libraries. That's likely to introduce a period of instability in 1.4 that you won't have if you remain on 1.2, the price for running on the bleeding edge.

Is there a particular rea

-Steve



On 5 March 2013 08:06, Subhash <gvvsubhashkumar at gmail.com<mailto:gvvsubhashkumar at gmail.com>> wrote:

Does sofia-sip supports rtp rfc also?
On Mar 5, 2013 1:22 PM, "Peter Olsson" <POlsson at enghouse.com<mailto:POlsson at enghouse.com>> wrote:
I believe that the Sofia library within the FS source tree has more updates than this version ? most of the updates in the last official Sofia release came from the FS developers anyway. So going to 1.12.11 would probably be a downgrade.

I?m not 100% sure about this though...

/Peter

Fr?n: freeswitch-dev-bounces at lists.freeswitch.org<mailto:freeswitch-dev-bounces at lists.freeswitch.org> [mailto:freeswitch-dev-bounces at lists.freeswitch.org<mailto:freeswitch-dev-bounces at lists.freeswitch.org>] F?r Xijing Dai
Skickat: den 5 mars 2013 08:37
Till: freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>
?mne: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?

hey


Is there any plan to upgrade freeswitch to use Sofia-SIP 1.12.11<http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.11.txt> ?

Cheers

_________________________________________________________________________
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-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org<mailto:FreeSWITCH-dev at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
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-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org<mailto:FreeSWITCH-dev at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
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-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org<mailto:FreeSWITCH-dev at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org

!DSPAM:5135c29932761784685714!
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From steveayre at gmail.com  Tue Mar  5 15:10:14 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 5 Mar 2013 12:10:14 +0000
Subject: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?
In-Reply-To: <CAMsCbX8u-ym0y5ZvZpZnQF7C3F+bSFP3Qw7Yam+Tz_TkO4L7JA@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F15235CC8@cantor.std.visionutv.se>
	<CAFvodxUhopQCU5kdUNG89LjQPKTp6ZJGCrZPYLEj-ijJF3C8cw@mail.gmail.com>
	<CAFiqYu=XJtrUW7QR=hCHc-gBQTfkOiAhXAcHFSBJyTJL5wxHoA@mail.gmail.com>
	<CAFiqYumGDDXm9aocYbNuMLvDiCe7k2-4iRC59CVagkD6Xts1nQ@mail.gmail.com>
	<CAMsCbX8u-ym0y5ZvZpZnQF7C3F+bSFP3Qw7Yam+Tz_TkO4L7JA@mail.gmail.com>
Message-ID: <CAFiqYuk5e2XSP-HhpHaub6hce0Uuv2Ss_OUDY8+8dGUtF+Jnmw@mail.gmail.com>

Trying 1.2.11 would be far from trivial and would actually remove a large
number of bug fixes. Also simply installing sofia-sip on your system won't
work, as FS uses the version in its libs directory.

If you find a bug always:

1) Try to reproduce it on the **current** head of the master branch
2) If still present, report it to http://jira.freeswitch.org/

The developers are very good at finding and fixing bugs.

-Steve



On 5 March 2013 10:18, Xijing Dai <dxj19831029 at gmail.com> wrote:

> I had a very random sip error in the freeswitch, and I want to try
> sofia-sip 1.12.11 to see if it's fixed there, but there are a few build
> errors?
>
> Is there any race condition chance inside mod_sofia?
>
> We make a few calls from the same softphone simultaneously, it includes
> pause/resume/terminate on any call at any time.
>
> The outbound endpoint is going though loopback inside freeswitch.
>
> Sometimes, the paused inbound call won't be able to resume, and after some
> deep tracing inside freeswitch, it seems to me that the sip message for
> resuming call is handled on outbound non-loopback channel.
> And I used wireshark to catch the packets, it shows that the call is in
> trying stage.
>
> This happens very randomly.
>
> Do you guys have any idea?
>
> Cheers
>
>
>
>
>
>
>
>
>
> On Tue, Mar 5, 2013 at 4:57 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Is there a particular reason you're asking about Sofia-SIP 1.12.11? Are
>> there any specific new bugfixes/features in that version you're after?
>>
>> -Steve
>>
>>
>>
>>
>> On 5 March 2013 08:56, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Sofia-SIP only provides the SIP signalling layer. FreeSWITCH uses its
>>> own RTP stack for media, and in the progress of moving to crtp.
>>>
>>> The current combination of libraries is well tested and stable, and as
>>> Peter points out contain many patches that are improvements over upstream.
>>>
>>> Because of this stability the developers aren't making any major updates
>>> to the library versions at the moment. Master (1.3) is the development
>>> branch for v1.2.stable. They're currently working towards the next stable
>>> 1.2 release, after which they plan to create a new 1.4 branch. 1.2 will
>>> continue to be 'stable' while in 1.4 they will start doing major updates to
>>> the libraries. That's likely to introduce a period of instability in 1.4
>>> that you won't have if you remain on 1.2, the price for running on the
>>> bleeding edge.
>>>
>>> Is there a particular rea
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 5 March 2013 08:06, Subhash <gvvsubhashkumar at gmail.com> wrote:
>>>
>>>> Does sofia-sip supports rtp rfc also?
>>>> On Mar 5, 2013 1:22 PM, "Peter Olsson" <POlsson at enghouse.com> wrote:
>>>>
>>>>>  I believe that the Sofia library within the FS source tree has more
>>>>> updates than this version ? most of the updates in the last official Sofia
>>>>> release came from the FS developers anyway. So going to 1.12.11 would
>>>>> probably be a downgrade.****
>>>>>
>>>>> ** **
>>>>>
>>>>> I?m not 100% sure about this though...****
>>>>>
>>>>> ** **
>>>>>
>>>>> /Peter****
>>>>>
>>>>> ** **
>>>>>
>>>>> *Fr?n:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
>>>>> freeswitch-dev-bounces at lists.freeswitch.org] *F?r *Xijing Dai
>>>>> *Skickat:* den 5 mars 2013 08:37
>>>>> *Till:* freeswitch-dev at lists.freeswitch.org
>>>>> *?mne:* [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?***
>>>>> *
>>>>>
>>>>> ** **
>>>>>
>>>>> hey****
>>>>>
>>>>> ** **
>>>>>
>>>>> ** **
>>>>>
>>>>> Is there any plan to upgrade freeswitch to use Sofia-SIP 1.12.11<http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.11.txt>
>>>>>  ?****
>>>>>
>>>>> ** **
>>>>>
>>>>> Cheers****
>>>>>
>>>>> !DSPAM:51359cdf32761263612615! ****
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-dev mailing list
>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Tue Mar  5 18:27:10 2013
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 05 Mar 2013 09:27:10 -0600
Subject: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?
In-Reply-To: <CAMsCbX8u-ym0y5ZvZpZnQF7C3F+bSFP3Qw7Yam+Tz_TkO4L7JA@mail.gmail.com>
Message-ID: <CD5B6AEE.6D513%krice@freeswitch.org>

Open a Ticket in Jira... Include the debug information and make sure you are
testing against current git master head...

Replacing the sofia-sip we have in tree with 1.12.11 will do you no good, a)
many of the patches in the last release actually came from FreeSWITCH, 2) we
have many more patches since then to go into sofia-sip which you will be
missing as 1.12.11 was released nearly 2 years ago.

K

On 3/5/13 4:18 AM, "Xijing Dai" <dxj19831029 at gmail.com> wrote:

> I had a very random sip error in the freeswitch, and I want to try sofia-sip
> 1.12.11 to see if it's fixed there, but there are a few build errors?
> 
> Is there any race condition chance inside mod_sofia?
> 
> We make a few calls from the same softphone?simultaneously, it includes
> pause/resume/terminate on any call at any time.
> 
> The outbound endpoint is going though loopback inside freeswitch.
> 
> Sometimes, the paused inbound call won't be able to resume, and after some
> deep tracing inside freeswitch, it seems to me that the sip message for
> resuming call is handled on outbound non-loopback channel.
> And I used wireshark to catch the packets, it shows that the call is in trying
> stage.
> 
> This happens very randomly.
> 
> Do you guys have any idea?
> 
> Cheers
> 
> 
> 
> 
> 
> 
> 
> 
> 
> On Tue, Mar 5, 2013 at 4:57 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> Is there a particular reason you're asking about Sofia-SIP 1.12.11? Are there
>> any specific new bugfixes/features in?that version you're after?
>> 
>> -Steve
>> 
>> 
>> 
>> 
>> On 5 March 2013 08:56, Steven Ayre <steveayre at gmail.com> wrote:
>>> Sofia-SIP only provides the SIP signalling layer. FreeSWITCH uses its own
>>> RTP stack for media, and in the progress of moving to crtp.
>>> 
>>> The current combination of libraries is well tested and stable, and as Peter
>>> points out contain many patches that are improvements over upstream.
>>> 
>>> Because of this stability the developers aren't making any major updates to
>>> the library versions at the moment. Master (1.3) is the development branch
>>> for v1.2.stable. They're currently working towards the next stable 1.2
>>> release, after which they plan to create a new 1.4 branch. 1.2 will continue
>>> to be 'stable' while in 1.4 they will start doing major updates to the
>>> libraries. That's likely to introduce a period of instability in 1.4 that
>>> you won't have if you remain on 1.2, the price for running on the bleeding
>>> edge.
>>> 
>>> Is there a particular rea
>>> 
>>> -Steve
>>> 
>>> 
>>> 
>>> On 5 March 2013 08:06, Subhash <gvvsubhashkumar at gmail.com> wrote:
>>>> 
>>>> Does sofia-sip supports rtp rfc also?
>>>> 
>>>> On Mar 5, 2013 1:22 PM, "Peter Olsson" <POlsson at enghouse.com> wrote:
>>>>> I believe that the Sofia library within the FS source tree has more
>>>>> updates than this version ? most of the updates in the last official Sofia
>>>>> release came from the FS developers anyway. So going to 1.12.11 would
>>>>> probably be a downgrade.
>>>>> ?
>>>>> I?m not 100% sure about this though...
>>>>> ?
>>>>> /Peter
>>>>> ?
>>>>> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org
>>>>> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Xijing Dai
>>>>> Skickat: den 5 mars 2013 08:37
>>>>> Till: freeswitch-dev at lists.freeswitch.org
>>>>> ?mne: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?
>>>>> ?
>>>>> hey
>>>>> 
>>>>> ?
>>>>> 
>>>>> ?
>>>>> 
>>>>> Is there any plan to upgrade freeswitch to use?Sofia-SIP 1.12.11
>>>>> <http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.11.txt>
>>>>> ??
>>>>> 
>>>>> ?
>>>>> 
>>>>> Cheers
>>>>> !DSPAM:51359cdf32761263612615!
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-dev mailing list
>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>> http://www.freeswitch.org
>>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>> 
>>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> 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 mike at jerris.com  Tue Mar  5 18:52:51 2013
From: mike at jerris.com (Michael Jerris)
Date: Tue, 5 Mar 2013 10:52:51 -0500
Subject: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?
In-Reply-To: <CD5B6AEE.6D513%krice@freeswitch.org>
References: <CD5B6AEE.6D513%krice@freeswitch.org>
Message-ID: <B364F725-99A8-4971-9B18-7CC574EB6EC1@jerris.com>

We have all the updates in sofia tree plus many, minus one I just committed to the sofia tree last weekend (that I'll pull in this week).  That commit was the first commit to the sofia tree in 18 months.


On Mar 5, 2013, at 10:27 AM, Ken Rice <krice at freeswitch.org> wrote:

> Open a Ticket in Jira... Include the debug information and make sure you are testing against current git master head...
> 
> Replacing the sofia-sip we have in tree with 1.12.11 will do you no good, a) many of the patches in the last release actually came from FreeSWITCH, 2) we have many more patches since then to go into sofia-sip which you will be missing as 1.12.11 was released nearly 2 years ago.
> 
> K
> 
> On 3/5/13 4:18 AM, "Xijing Dai" <dxj19831029 at gmail.com> wrote:
> 
>> I had a very random sip error in the freeswitch, and I want to try sofia-sip 1.12.11 to see if it's fixed there, but there are a few build errors?
>> 
>> Is there any race condition chance inside mod_sofia?
>> 
>> We make a few calls from the same softphone simultaneously, it includes pause/resume/terminate on any call at any time.
>> 
>> The outbound endpoint is going though loopback inside freeswitch.
>> 
>> Sometimes, the paused inbound call won't be able to resume, and after some deep tracing inside freeswitch, it seems to me that the sip message for resuming call is handled on outbound non-loopback channel.
>> And I used wireshark to catch the packets, it shows that the call is in trying stage.
>> 
>> This happens very randomly.
>> 
>> Do you guys have any idea?
>> 
>> Cheers
>> 
>> 
>> On Tue, Mar 5, 2013 at 4:57 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>> Is there a particular reason you're asking about Sofia-SIP 1.12.11? Are there any specific new bugfixes/features in that version you're after?
>>> 
>>> -Steve
>>> 
>>> 
>>> 
>>> 
>>> On 5 March 2013 08:56, Steven Ayre <steveayre at gmail.com> wrote:
>>>> Sofia-SIP only provides the SIP signalling layer. FreeSWITCH uses its own RTP stack for media, and in the progress of moving to crtp.
>>>> 
>>>> The current combination of libraries is well tested and stable, and as Peter points out contain many patches that are improvements over upstream.
>>>> 
>>>> Because of this stability the developers aren't making any major updates to the library versions at the moment. Master (1.3) is the development branch for v1.2.stable. They're currently working towards the next stable 1.2 release, after which they plan to create a new 1.4 branch. 1.2 will continue to be 'stable' while in 1.4 they will start doing major updates to the libraries. That's likely to introduce a period of instability in 1.4 that you won't have if you remain on 1.2, the price for running on the bleeding edge.
>>>> 
>>>> Is there a particular rea
>>>> 
>>>> -Steve
>>>> 
>>>> 
>>>> 
>>>> On 5 March 2013 08:06, Subhash <gvvsubhashkumar at gmail.com> wrote:
>>>>> 
>>>>> Does sofia-sip supports rtp rfc also?
>>>>> 
>>>>> On Mar 5, 2013 1:22 PM, "Peter Olsson" <POlsson at enghouse.com> wrote:
>>>>>> I believe that the Sofia library within the FS source tree has more updates than this version ? most of the updates in the last official Sofia release came from the FS developers anyway. So going to 1.12.11 would probably be a downgrade.
>>>>>>  
>>>>>> I?m not 100% sure about this though...
>>>>>>  
>>>>>> /Peter
>>>>>>  
>>>>>> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Xijing Dai
>>>>>> Skickat: den 5 mars 2013 08:37
>>>>>> Till: freeswitch-dev at lists.freeswitch.org
>>>>>> ?mne: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?
>>>>>> 
>>>>>> 
>>>>>> Is there any plan to upgrade freeswitch to use Sofia-SIP 1.12.11 <http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.11.txt>  ?
>>>>>> 
>>>>>>   
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From dxj19831029 at gmail.com  Tue Mar  5 19:19:16 2013
From: dxj19831029 at gmail.com (Xijing Dai)
Date: Wed, 6 Mar 2013 00:19:16 +0800
Subject: [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?
In-Reply-To: <B364F725-99A8-4971-9B18-7CC574EB6EC1@jerris.com>
References: <CD5B6AEE.6D513%krice@freeswitch.org>
	<B364F725-99A8-4971-9B18-7CC574EB6EC1@jerris.com>
Message-ID: <CAMsCbX8OLhUVQcTn4HWN60xGt9-OpH9o-9ubVHvTj7i20xb-xw@mail.gmail.com>

Thanks guys, I will try to reproduce it and report in jira.

Cheers

On Tue, Mar 5, 2013 at 11:52 PM, Michael Jerris <mike at jerris.com> wrote:

> We have all the updates in sofia tree plus many, minus one I just
> committed to the sofia tree last weekend (that I'll pull in this week).
>  That commit was the first commit to the sofia tree in 18 months.
>
>
> On Mar 5, 2013, at 10:27 AM, Ken Rice <krice at freeswitch.org> wrote:
>
>  Open a Ticket in Jira... Include the debug information and make sure you
> are testing against current git master head...
>
> Replacing the sofia-sip we have in tree with 1.12.11 will do you no good,
> a) many of the patches in the last release actually came from FreeSWITCH,
> 2) we have many more patches since then to go into sofia-sip which you will
> be missing as 1.12.11 was released nearly 2 years ago.
>
> K
>
> On 3/5/13 4:18 AM, "Xijing Dai" <dxj19831029 at gmail.com> wrote:
>
> I had a very random sip error in the freeswitch, and I want to try
> sofia-sip 1.12.11 to see if it's fixed there, but there are a few build
> errors?
>
> Is there any race condition chance inside mod_sofia?
>
> We make a few calls from the same softphone simultaneously, it includes
> pause/resume/terminate on any call at any time.
>
> The outbound endpoint is going though loopback inside freeswitch.
>
> Sometimes, the paused inbound call won't be able to resume, and after some
> deep tracing inside freeswitch, it seems to me that the sip message for
> resuming call is handled on outbound non-loopback channel.
> And I used wireshark to catch the packets, it shows that the call is in
> trying stage.
>
> This happens very randomly.
>
> Do you guys have any idea?
>
> Cheers
>
>
> On Tue, Mar 5, 2013 at 4:57 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
> Is there a particular reason you're asking about Sofia-SIP 1.12.11? Are
> there any specific new bugfixes/features in that version you're after?
>
> -Steve
>
>
>
>
> On 5 March 2013 08:56, Steven Ayre <steveayre at gmail.com> wrote:
>
> Sofia-SIP only provides the SIP signalling layer. FreeSWITCH uses its own
> RTP stack for media, and in the progress of moving to crtp.
>
> The current combination of libraries is well tested and stable, and as
> Peter points out contain many patches that are improvements over upstream.
>
> Because of this stability the developers aren't making any major updates
> to the library versions at the moment. Master (1.3) is the development
> branch for v1.2.stable. They're currently working towards the next stable
> 1.2 release, after which they plan to create a new 1.4 branch. 1.2 will
> continue to be 'stable' while in 1.4 they will start doing major updates to
> the libraries. That's likely to introduce a period of instability in 1.4
> that you won't have if you remain on 1.2, the price for running on the
> bleeding edge.
>
> Is there a particular rea
>
> -Steve
>
>
>
> On 5 March 2013 08:06, Subhash <gvvsubhashkumar at gmail.com> wrote:
>
>
> Does sofia-sip supports rtp rfc also?
>
> On Mar 5, 2013 1:22 PM, "Peter Olsson" <POlsson at enghouse.com> wrote:
>
> I believe that the Sofia library within the FS source tree has more
> updates than this version ? most of the updates in the last official Sofia
> release came from the FS developers anyway. So going to 1.12.11 would
> probably be a downgrade.
>
> I?m not 100% sure about this though...
>
> /Peter
>
> *Fr?n:* freeswitch-dev-bounces at lists.freeswitch.org [
> mailto:freeswitch-dev-bounces at lists.freeswitch.org<freeswitch-dev-bounces at lists.freeswitch.org>]
> *F?r *Xijing Dai
> *Skickat:* den 5 mars 2013 08:37
> *Till:* freeswitch-dev at lists.freeswitch.org
> *?mne:* [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?
>
>
> Is there any plan to upgrade freeswitch to use Sofia-SIP 1.12.11 <
> http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.11.txt>
>  ?
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From fdelawarde at wirelessmundi.com  Tue Mar  5 19:23:31 2013
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 05 Mar 2013 17:23:31 +0100
Subject: [Freeswitch-dev] [Freeswitch-users] Maximum message size
In-Reply-To: <CAOmzhXk=fnM3=pQ3fbcLJ2g6ocGuZBhgnKubvE2o76yYHiw1eA@mail.gmail.com>
References: <CAOmzhXk=fnM3=pQ3fbcLJ2g6ocGuZBhgnKubvE2o76yYHiw1eA@mail.gmail.com>
Message-ID: <1362500611.17612.83.camel@luna.madrid.commsmundi.com>

Hi Kevin,

According to RFC-3261 (18.1.1), the stack should be able to support
packets of up to 65535 bytes.

I'm not sure sofia follows French telcos interop rules.

Cordialement,
Fran?ois.

On Tue, 2013-03-05 at 16:53 +0100, Kevin Mathy wrote:
> Hi List, 
> 
> 
> Another question ;-) , as described in FFT Doc 10.001, the maximum
> message size is 2048 bytes for SIP messages, and 1024 bytes for SDP
> bodies; could you, please, confirm that FS is compliant with that ? 
> If not, what the maximum message size supported / sent by FS ? 
> 
> 
> Thanks a lot, 
> 
> 
> Bien cordialement, 
> Best Regards, 
> 
> 
> Kevin MATHY
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From k.mathy at hexanet.fr  Tue Mar  5 18:53:42 2013
From: k.mathy at hexanet.fr (Kevin Mathy)
Date: Tue, 5 Mar 2013 16:53:42 +0100
Subject: [Freeswitch-dev] Maximum message size
Message-ID: <CAOmzhXk=fnM3=pQ3fbcLJ2g6ocGuZBhgnKubvE2o76yYHiw1eA@mail.gmail.com>

Hi List,

Another question ;-) , as described in FFT Doc 10.001, the maximum message
size is 2048 bytes for SIP messages, and 1024 bytes for SDP bodies; could
you, please, confirm that FS is compliant with that ?
If not, what the maximum message size supported / sent by FS ?

Thanks a lot,
*

*
*
Bien cordialement,
*
*
Best Regards,

**Kevin MATHY*
*
*

-- 

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From msc at freeswitch.org  Wed Mar  6 20:03:37 2013
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Mar 2013 09:03:37 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly Conference Call
Message-ID: <CAKzWOxVgmsEZT6nrRqgBUe63GF=rzYUHxpwCaLm8zM=E8TPH8g@mail.gmail.com>

Hi All!

Today's agenda page is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_06

We have a few news items to discuss and then Ken and I are going to be
doing a refresher on VoIP sec with ZRTP and also touching upon SRTP and
TLS. Oh, and we'll have new ClueCon 2012 videos released as well!

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 Mar  6 22:38:02 2013
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Mar 2013 11:38:02 -0800
Subject: [Freeswitch-dev] Announcement: FreeSWITCH 1.2.6 Is Now Available
Message-ID: <CAKzWOxWRHCFJAHhVz14m28n08nupdWerC07kLgz=rS3ZpGAYfw@mail.gmail.com>

Hello all,

We just wanted to let everyone know that FreeSWITCH 1.2.6 has been
released<http://freeswitch.org/node/439>.
Many have been waiting for this version so that they can put it into
production systems. This new version has numerous bug fixes and the team
has spent a lot of time and energy tracking down and eliminating
hard-to-find memory leaks. Please update to this version as soon as you
reasonably can.

Thanks for being a great community!

-- 
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  Thu Mar  7 00:53:38 2013
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 06 Mar 2013 15:53:38 -0600
Subject: [Freeswitch-dev] Announcement: FreeSWITCH 1.2.7 Is Now
 Available (Of course we find 2 things right after we release lol)
In-Reply-To: <CAKzWOxWRHCFJAHhVz14m28n08nupdWerC07kLgz=rS3ZpGAYfw@mail.gmail.com>
Message-ID: <CD5D1702.6D5CE%krice@freeswitch.org>

And as Murphy?s Law would have it we found 2 nasty bugs today and fixed
them...

Bug 1) Pretty bad memory leak when using TLS. This is now Fixed
Bug 2) Dead Lock in mod_sofia around the database handlers. This is now
Fixed

So FreeSWITCH 1.2.7 is up and ready for you to grab. Sorry for the rapid rev
bump but these are bad enough that we wanted to get them in your hands ASAP.

Of course you can grab the source tarball at
http://files.freeswitch.org/freeswitch-1.2.7.tar.bz2

Have Fun
Ken


On 3/6/13 1:38 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> Hello all,
> 
> We just wanted to let everyone know that FreeSWITCH 1.2.6 has been released
> <http://freeswitch.org/node/439> . Many have been waiting for this version so
> that they can put it into production systems. This new version has numerous
> bug fixes and the team has spent a lot of time and energy tracking down and
> eliminating hard-to-find memory leaks. Please update to this version as soon
> as you reasonably can.
> 
> Thanks for being a great community!

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

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From ec.amazumdar at tatapowersed.com  Thu Mar  7 07:32:51 2013
From: ec.amazumdar at tatapowersed.com (Anindo Mazumdar)
Date: Thu, 7 Mar 2013 10:02:51 +0530
Subject: [Freeswitch-dev] Configuring FREESwitch as ENUM server
Message-ID: <201303070432.r274Wpfo002972@blr.tatapowersed.com>

Hello All,

We are trying to configure FREESwitch as ENUM server. We were only able to find the following wiki doc explaining how to configure FREESwitch(http://wiki.freeswitch.org/wiki/Mod_enum), but were unable to comprehend it because of unfamiliarity with FREESwitch. Is anyone knowing of a better approach for solving the same issue?
-- 
With Regards,

Anindo Mazumdar,

From krice at freeswitch.org  Thu Mar  7 07:42:03 2013
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 06 Mar 2013 22:42:03 -0600
Subject: [Freeswitch-dev] Configuring FREESwitch as ENUM server
In-Reply-To: <201303070432.r274Wpfo002972@blr.tatapowersed.com>
Message-ID: <CD5D76BB.6D5F0%krice@freeswitch.org>

FreeSWITCH is not an ENUM server... It is a requester of ENUM
information.... An ENUM Server by definition is a DNS server...

You should be looking at how to configure your prefered DNS service software
to provide ENUM information, then via mod_enum FreeSWITCH will query that
DNS server for the information

Configuring an ENUM enabled DNS server is beyond the scope of FreeSWITCH,
but there may be some information on the FreeSWITCH wiki on how to do that


On 3/6/13 10:32 PM, "Anindo Mazumdar" <ec.amazumdar at tatapowersed.com> wrote:

> Hello All,
> 
> We are trying to configure FREESwitch as ENUM server. We were only able to
> find the following wiki doc explaining how to configure
> FREESwitch(http://wiki.freeswitch.org/wiki/Mod_enum), but were unable to
> comprehend it because of unfamiliarity with FREESwitch. Is anyone knowing of a
> better approach for solving the same issue?

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




From sparklezou at 163.com  Thu Mar  7 08:58:28 2013
From: sparklezou at 163.com (sparklezou)
Date: Thu, 7 Mar 2013 13:58:28 +0800
Subject: [Freeswitch-dev] About the "hardware fax detection" on Sangoma
	A101, support on Freeswitch or NOT
Message-ID: <7953eb01.2d60.13d436dbb30.Coremail.sparklezou@163.com>

Hi Develop Team,

Is there any pre-defined extension name in the dialplan to handle the HW detect fax?

Seems such function implemented in asterisk. review following email.

Thanks!

2013-03-07



sparklezou



????sparklezou
?????2013-02-17 16:34
???About the "hardware fax detection" on Sangoma A101, support on Freeswitch or NOT
????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>
???

Hi All,

The Sangoma A101 board support "hardware fax detection". It configured in /etc/wanpipe/wanpipe1.conf configureation file.

TDMV_HW_FAX_DETECT = YES # YES: receive fax 1100hz events from hardware

Is there any configuration on the dial plan, if detect the fax, goes to the predefined dial plan context.

something the same in asterisk http://wiki.sangoma.com/wanpipe-linux-asterisk-fax-detect

Thanks!

2013-02-17



sparklezou
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From lconroy at insensate.co.uk  Thu Mar  7 14:00:36 2013
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Thu, 7 Mar 2013 11:00:36 +0000
Subject: [Freeswitch-dev] Configuring FREESwitch as ENUM server
In-Reply-To: <CD5D76BB.6D5F0%krice@freeswitch.org>
References: <CD5D76BB.6D5F0%krice@freeswitch.org>
Message-ID: <8B9D9529-7C20-40C9-9BC2-2EC3542A6412@insensate.co.uk>

Hi There,
 Strictly speaking, there is no such thing as an ENUM server --
as Ken says, it's just a DNS server that is authoritative for domains within (I'd guess in your case) 1.9.e164.arpa.

=> If you *really* mean ENUM, you need to look for a scheme to provision NAPTR data into those domains, and need to get the "owner" of the parent zone to point to your DNS servers for those domains.

Freeswitch (or any other sensible softswitch) will check for ENUM records if you configure it, and calls "will just work".

all the best,
  Lawrence

On 7 Mar 2013, at 04:42, Ken Rice wrote:

> FreeSWITCH is not an ENUM server... It is a requester of ENUM
> information.... An ENUM Server by definition is a DNS server...
> 
> You should be looking at how to configure your prefered DNS service software
> to provide ENUM information, then via mod_enum FreeSWITCH will query that
> DNS server for the information
> 
> Configuring an ENUM enabled DNS server is beyond the scope of FreeSWITCH,
> but there may be some information on the FreeSWITCH wiki on how to do that
> 
> 
> On 3/6/13 10:32 PM, "Anindo Mazumdar" <ec.amazumdar at tatapowersed.com> wrote:
> 
>> Hello All,
>> 
>> We are trying to configure FREESwitch as ENUM server. We were only able to
>> find the following wiki doc explaining how to configure
>> FREESwitch(http://wiki.freeswitch.org/wiki/Mod_enum), but were unable to
>> comprehend it because of unfamiliarity with FREESwitch. Is anyone knowing of a
>> better approach for solving the same issue?
> 
> -- 
> 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-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From mike at jerris.com  Thu Mar  7 17:11:39 2013
From: mike at jerris.com (Michael Jerris)
Date: Thu, 7 Mar 2013 09:11:39 -0500
Subject: [Freeswitch-dev] About the "hardware fax detection" on Sangoma
	A101, support on Freeswitch or NOT
In-Reply-To: <7953eb01.2d60.13d436dbb30.Coremail.sparklezou@163.com>
References: <7953eb01.2d60.13d436dbb30.Coremail.sparklezou@163.com>
Message-ID: <E6A3A4C6-4C8F-45F7-B47C-43616DC5AF25@jerris.com>

A function to do what? None of the fax dialplan applications are specific to hardware detect that I can think of

On Mar 7, 2013, at 12:58 AM, sparklezou <sparklezou at 163.com> wrote:

> Hi Develop Team,
>  
> Is there any pre-defined extension name in the dialplan to handle the HW detect fax?
>  
> Seems such function implemented in asterisk. review following email.
>  
> Thanks!
>  
> 2013-03-07
> sparklezou
> ????sparklezou
> ?????2013-02-17 16:34
> ???About the "hardware fax detection" on Sangoma A101, support on Freeswitch or NOT
> ????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>
> ???
>  
> Hi All,
>  
> The Sangoma A101 board support "hardware fax detection". It configured in /etc/wanpipe/wanpipe1.conf configureation file.
> 
> TDMV_HW_FAX_DETECT = YES # YES: receive fax 1100hz events from hardware
> 
> Is there any configuration on the dial plan, if detect the fax, goes to the predefined dial plan context.
> 
> something the same in asterisk http://wiki.sangoma.com/wanpipe-linux-asterisk-fax-detect
> 
> Thanks!
>  
> 2013-02-17
> sparklezou

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From v.kovalyshyn at gmail.com  Mon Mar 11 10:24:00 2013
From: v.kovalyshyn at gmail.com (=?UTF-8?B?0JLRltGC0LDQu9GW0Lkg0JrQvtCy0LDQu9C40YjQuNC9?=)
Date: Mon, 11 Mar 2013 09:24:00 +0200
Subject: [Freeswitch-dev] Memory Leak FreeSWITCH on Windows
Message-ID: <CANb-=miDKCo=UMLsj=ReLwr_xs9Hhm=9PzGHMTRHp27Phu6jMA@mail.gmail.com>

Hi Develop Team!

We've using Freesiwtch on Windows 7 64bit. FreeSWITCH compiled from source
(lates git and I've try 1.2.7) using Visual Studio C++ 2010 Express.

FreeSWITCHConsole.exe start with about 21Mb in memory.
But after 10-15 hourse (I tested without calls during this sunday)
FreeSWITCHConsole.exe is over 800Mb and not responding....

How to debbug reason of the memory leak on Windows?

--
Vitaly Kovalyshyn
http://vk.it-sfera.com.ua/
http://wiki.webitel.com/
Twitter: @kovalyshyn <https://twitter.com/kovalyshyn>
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From POlsson at enghouse.com  Mon Mar 11 10:45:23 2013
From: POlsson at enghouse.com (Peter Olsson)
Date: Mon, 11 Mar 2013 07:45:23 +0000
Subject: [Freeswitch-dev] Memory Leak FreeSWITCH on Windows
Message-ID: <1FFF97C269757C458224B7C895F35F1523B431@cantor.std.visionutv.se>

There is no easy way to find memory leaks on Windows, at least not with the standard tools. I personally usually end up running the same things on Linux, under valgrind.

First I would make sure to rule out everything outside FS, make a simple dialplan ? remove any javascript, Lua scripts etc, and use the sqlite core database (if you are using ODBC). I?ve been running FS for some years on Windows, and I update to current head versions at least once a week ? and right now I don?t see any memory leak problems at all ? so it seems to me that it?s related to your specific configuration.

/Peter


Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r ??????? ?????????
Skickat: den 11 mars 2013 08:24
Till: freeswitch-dev at lists.freeswitch.org
?mne: [Freeswitch-dev] Memory Leak FreeSWITCH on Windows

Hi Develop Team!

We've using Freesiwtch on Windows 7 64bit. FreeSWITCH compiled from source (lates git and I've try 1.2.7) using Visual Studio C++ 2010 Express.

FreeSWITCHConsole.exe start with about 21Mb in memory.
But after 10-15 hourse (I tested without calls during this sunday) FreeSWITCHConsole.exe is over 800Mb and not responding....

How to debbug reason of the memory leak on Windows?

--
Vitaly Kovalyshyn
http://vk.it-sfera.com.ua/
http://wiki.webitel.com/
Twitter: @kovalyshyn<https://twitter.com/kovalyshyn>
!DSPAM:513d82da32761296515016!
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From v.kovalyshyn at gmail.com  Mon Mar 11 10:59:24 2013
From: v.kovalyshyn at gmail.com (=?UTF-8?B?0JLRltGC0LDQu9GW0Lkg0JrQvtCy0LDQu9C40YjQuNC9?=)
Date: Mon, 11 Mar 2013 09:59:24 +0200
Subject: [Freeswitch-dev] Memory Leak FreeSWITCH on Windows
In-Reply-To: <1FFF97C269757C458224B7C895F35F1523B431@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1523B431@cantor.std.visionutv.se>
Message-ID: <CANb-=mid4yxOLXnxo+O+UV-M+mg4mZV8c=8GWLjw5KexC9XC-Q@mail.gmail.com>

Thank You! I'll try Your recommenfations...

I used:
mod_xml_curl - for dialplan from my MS SQL
mod_xml_cdr - for posting CDR to my MS SQL
ODBC pgSQL for mod_callcenter
and some Lua scripts inside dialplan.....


--
Vitaly Kovalyshyn
http://vk.it-sfera.com.ua/
http://wiki.webitel.com/
Twitter: @kovalyshyn <https://twitter.com/kovalyshyn>


2013/3/11 Peter Olsson <POlsson at enghouse.com>

>  There is no easy way to find memory leaks on Windows, at least not with
> the standard tools. I personally usually end up running the same things on
> Linux, under valgrind.****
>
> ** **
>
> First I would make sure to rule out everything outside FS, make a simple
> dialplan ? remove any javascript, Lua scripts etc, and use the sqlite core
> database (if you are using ODBC). I?ve been running FS for some years on
> Windows, and I update to current head versions at least once a week ? and
> right now I don?t see any memory leak problems at all ? so it seems to me
> that it?s related to your specific configuration.****
>
> ** **
>
> /Peter****
>
> ** **
>
> ** **
>
> *Fr?n:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] *F?r *??????? ?????????
> *Skickat:* den 11 mars 2013 08:24
> *Till:* freeswitch-dev at lists.freeswitch.org
> *?mne:* [Freeswitch-dev] Memory Leak FreeSWITCH on Windows****
>
> ** **
>
> Hi Develop Team!****
>
> ** **
>
> We've using Freesiwtch on Windows 7 64bit. FreeSWITCH compiled from source
> (lates git and I've try 1.2.7) using Visual Studio C++ 2010 Express. ****
>
> ** **
>
> FreeSWITCHConsole.exe start with about 21Mb in memory.****
>
> But after 10-15 hourse (I tested without calls during this sunday)
> FreeSWITCHConsole.exe is over 800Mb and not responding....****
>
> ** **
>
> How to debbug reason of the memory leak on Windows?****
>
>
> ****
>
> --
> Vitaly Kovalyshyn
> http://vk.it-sfera.com.ua/****
>
> http://wiki.webitel.com/****
>
> Twitter: @kovalyshyn <https://twitter.com/kovalyshyn>****
>
> !DSPAM:513d82da32761296515016! ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From Alexander.Haugg at c4b.de  Mon Mar 11 10:05:33 2013
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Mon, 11 Mar 2013 07:05:33 +0000
Subject: [Freeswitch-dev] Feature Request: Execute an executable to get
 configuration / dialplan
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC814D21070@vc4bex2.corp.c4b.de>

Hi all,

last week i have written and tested a new feature to get the xml config / dialplan from an executable.
At the first i use it only for the dialplan but i think it could be a general feature.

Is the Jira system the right place to post my patches and description for this as "New Feature"?

Why this feature request?
I had a look on the mod_xml_curl but this was not exat this what i need, i need a tool that make it possible to have the config only virtual and don't need a http deamon.

What do you think for this?

Thanks Alex
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From mike at jerris.com  Mon Mar 11 15:16:47 2013
From: mike at jerris.com (Michael Jerris)
Date: Mon, 11 Mar 2013 08:16:47 -0400
Subject: [Freeswitch-dev] Feature Request: Execute an executable to get
	configuration / dialplan
In-Reply-To: <B60DA91F5EA0E24A96E73CDE43CC9EC814D21070@vc4bex2.corp.c4b.de>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC814D21070@vc4bex2.corp.c4b.de>
Message-ID: <CDE1D993-D8E7-4C78-8557-AF8ED9E4A41A@jerris.com>

is this different than:
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Preprocessor_Commands

exec - Execute a shell command
<X-PRE-PROCESS cmd="exec" data="echo XML config loaded at `date` >> /var/log/fs_config.log"/>

On Mar 11, 2013, at 3:05 AM, Alexander Haugg <Alexander.Haugg at c4b.de> wrote:

> Hi all,
>  
> last week i have written and tested a new feature to get the xml config / dialplan from an executable.
> At the first i use it only for the dialplan but i think it could be a general feature.
>  
> Is the Jira system the right place to post my patches and description for this as ?New Feature??
>  
> Why this feature request?
> I had a look on the mod_xml_curl but this was not exat this what i need, i need a tool that make it possible to have the config only virtual and don?t need a http deamon.
>  
> What do you think for this?
>  
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From Alexander.Haugg at c4b.de  Mon Mar 11 16:08:03 2013
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Mon, 11 Mar 2013 13:08:03 +0000
Subject: [Freeswitch-dev] Feature Request: Execute an executable to
	get	configuration / dialplan
In-Reply-To: <CDE1D993-D8E7-4C78-8557-AF8ED9E4A41A@jerris.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC814D21070@vc4bex2.corp.c4b.de>
	<CDE1D993-D8E7-4C78-8557-AF8ED9E4A41A@jerris.com>
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC814D24157@vc4bex2.corp.c4b.de>

It is not the same. The cmd=exec store the return value in a file, my solution example have the dialplan only temporarily and it works on Windows (not tested on linux but the adaptings for linux are tiny)  (at the moment not forked).

You can do a look in my patches.

Von: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris
Gesendet: Montag, 11. M?rz 2013 13:17
An: freeswitch-dev at lists.freeswitch.org
Betreff: Re: [Freeswitch-dev] Feature Request: Execute an executable to get configuration / dialplan

is this different than:
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Preprocessor_Commands

?  exec - Execute a shell command

<X-PRE-PROCESS cmd="exec" data="echo XML config loaded at `date` >> /var/log/fs_config.log"/>

On Mar 11, 2013, at 3:05 AM, Alexander Haugg <Alexander.Haugg at c4b.de<mailto:Alexander.Haugg at c4b.de>> wrote:


Hi all,

last week i have written and tested a new feature to get the xml config / dialplan from an executable.
At the first i use it only for the dialplan but i think it could be a general feature.

Is the Jira system the right place to post my patches and description for this as "New Feature"?

Why this feature request?
I had a look on the mod_xml_curl but this was not exat this what i need, i need a tool that make it possible to have the config only virtual and don't need a http deamon.

What do you think for this?

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From Alexander.Haugg at c4b.de  Mon Mar 11 16:21:04 2013
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Mon, 11 Mar 2013 13:21:04 +0000
Subject: [Freeswitch-dev] Feature Request: Execute an executable to
	get	configuration / dialplan
In-Reply-To: <CDE1D993-D8E7-4C78-8557-AF8ED9E4A41A@jerris.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC814D21070@vc4bex2.corp.c4b.de>
	<CDE1D993-D8E7-4C78-8557-AF8ED9E4A41A@jerris.com>
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC814D24188@vc4bex2.corp.c4b.de>

But you a right, i can complete the windows part for the cmd=exec?

Von: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris
Gesendet: Montag, 11. M?rz 2013 13:17
An: freeswitch-dev at lists.freeswitch.org
Betreff: Re: [Freeswitch-dev] Feature Request: Execute an executable to get configuration / dialplan

is this different than:
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Preprocessor_Commands

?  exec - Execute a shell command

<X-PRE-PROCESS cmd="exec" data="echo XML config loaded at `date` >> /var/log/fs_config.log"/>

On Mar 11, 2013, at 3:05 AM, Alexander Haugg <Alexander.Haugg at c4b.de<mailto:Alexander.Haugg at c4b.de>> wrote:


Hi all,

last week i have written and tested a new feature to get the xml config / dialplan from an executable.
At the first i use it only for the dialplan but i think it could be a general feature.

Is the Jira system the right place to post my patches and description for this as "New Feature"?

Why this feature request?
I had a look on the mod_xml_curl but this was not exat this what i need, i need a tool that make it possible to have the config only virtual and don't need a http deamon.

What do you think for this?

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From sparklezou at 163.com  Mon Mar 11 17:03:38 2013
From: sparklezou at 163.com (sparklezou)
Date: Mon, 11 Mar 2013 22:03:38 +0800
Subject: [Freeswitch-dev] About the "hardware fax detection" on
	Sangoma	A101, support on Freeswitch or NOT
In-Reply-To: <E6A3A4C6-4C8F-45F7-B47C-43616DC5AF25@jerris.com>
References: <E6A3A4C6-4C8F-45F7-B47C-43616DC5AF25@jerris.com>
Message-ID: <7bd471d.975f.13d59c35c86.Coremail.sparklezou@163.com>

Hi Michael,

Seem there should be on notification from the FreeTDM lib, HW fax detect.

Then Freeswitch could handle the incoming fax via the pre-defined dialplan context. refer to

http://wiki.sangoma.com/wanpipe-linux-asterisk-fax-detect#asterisk-configuration


2013-03-11



sparklezou



????Michael Jerris
?????2013-03-07 22:11
???Re: [Freeswitch-dev] About the "hardware fax detection" on Sangoma A101, support on Freeswitch or NOT
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

A function to do what? None of the fax dialplan applications are specific to hardware detect that I can think of


On Mar 7, 2013, at 12:58 AM, sparklezou <sparklezou at 163.com> wrote:


Hi Develop Team,

Is there any pre-defined extension name in the dialplan to handle the HW detect fax?

Seems such function implemented in asterisk. review following email.

Thanks!

2013-03-07



sparklezou



????sparklezou
?????2013-02-17 16:34
???About the "hardware fax detection" on Sangoma A101, support on Freeswitch or NOT
????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>
???

Hi All,

The Sangoma A101 board support "hardware fax detection". It configured in /etc/wanpipe/wanpipe1.conf configureation file.

TDMV_HW_FAX_DETECT = YES # YES: receive fax 1100hz events from hardware

Is there any configuration on the dial plan, if detect the fax, goes to the predefined dial plan context.

something the same in asterisk http://wiki.sangoma.com/wanpipe-linux-asterisk-fax-detect

Thanks!

2013-02-17



sparklezou
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From anthony.minessale at gmail.com  Mon Mar 11 18:20:21 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Mar 2013 10:20:21 -0500
Subject: [Freeswitch-dev] Feature Request: Execute an executable to get
 configuration / dialplan
In-Reply-To: <B60DA91F5EA0E24A96E73CDE43CC9EC814D24188@vc4bex2.corp.c4b.de>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC814D21070@vc4bex2.corp.c4b.de>
	<CDE1D993-D8E7-4C78-8557-AF8ED9E4A41A@jerris.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC814D24188@vc4bex2.corp.c4b.de>
Message-ID: <CAKbxfG9LL7bCFfKWHNdFmn3Nmop_2aC8w8E70g9=vxTwEnBUqw@mail.gmail.com>

Somehow we continue to fail at spreading the message that this should go
into Jira.
Please file it at http://jira.freeswitch.org for further discussion.



On Mon, Mar 11, 2013 at 8:21 AM, Alexander Haugg <Alexander.Haugg at c4b.de>wrote:

>  But you a right, i can complete the windows part for the cmd=exec?****
>
> ** **
>
> *Von:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] *Im Auftrag von *Michael
> Jerris
> *Gesendet:* Montag, 11. M?rz 2013 13:17
> *An:* freeswitch-dev at lists.freeswitch.org
> *Betreff:* Re: [Freeswitch-dev] Feature Request: Execute an executable to
> get configuration / dialplan****
>
> ** **
>
> is this different than:****
>
> http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Preprocessor_Commands
> ****
>
> ** **
>
> **?  **exec - Execute a shell command****
>
> <X-PRE-PROCESS cmd="exec" data="echo XML config loaded at `date` >> /var/log/fs_config.log"/>****
>
>  ** **
>
> On Mar 11, 2013, at 3:05 AM, Alexander Haugg <Alexander.Haugg at c4b.de>
> wrote:****
>
>
>
> ****
>
> Hi all,****
>
>  ****
>
> last week i have written and tested a new feature to get the xml config /
> dialplan from an executable.****
>
> At the first i use it only for the dialplan but i think it could be a
> general feature.****
>
>  ****
>
> Is the Jira system the right place to post my patches and description for
> this as ?New Feature??****
>
>  ****
>
> Why this feature request?****
>
> I had a look on the mod_xml_curl but this was not exat this what i need, i
> need a tool that make it possible to have the config only virtual and don?t
> need a http deamon.****
>
>  ****
>
> What do you think for this?****
>
>  ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at 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 marketing at cluecon.com  Mon Mar 11 22:06:35 2013
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 11 Mar 2013 12:06:35 -0700
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxV-jtsKVB=ghk6q-uoxuDYnnocvpW4uJAp7YatkXQjiDQ@mail.gmail.com>

Hello folks! News and Notes is back after a hiatus last week.

The important news from the past week is that of the recent releases of
both FreeSWITCH version 1.2.6 <http://freeswitch.org/node/439> and version
1.2.7. As Ken Rice mentioned in his
announcement<http://lists.freeswitch.org/pipermail/freeswitch-users/2013-March/093229.html>,
Murphy's Law <http://en.wikipedia.org/wiki/Murphy%27s_law> struck again.
Quite literally within minutes of tagging and releasing version 1.2.6 the
team discovered some serious issues and very quickly released version
1.2.7. Needless-to-say, if you somehow ended up on 1.2.6 then please
immediately move to 1.2.7.

Last week on the community conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_06>we reviewed
some of the important aspects of
ZRTP <http://wiki.freeswitch.org/wiki/ZRTP> vs.
SRTP<http://wiki.freeswitch.org/wiki/SRTP>.
It is easy to setup ZRTP, so we encourage you to do so. The more people we
have using it the better our implementation will be. Also, it will be
easier to encourage hardware vendors to include ZRTP when they realize just
how pervasive it is. On this week's
call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_13>we are
having long-time telephony OSS programmer Areski join us to
highlight the improvements in Newfies Dialer<http://www.newfies-dialer.org/>.
A lot has changed in the year since we last heard from him. We look forward
to hearing more.

In ClueCon news we have uploaded more videos from 2012:
* Seven Du - Building a Command and Dispatch System<http://youtu.be/FsSenee8Fn8>
* Chad Phillips - Fun With VoIP <http://youtu.be/cZSRkkbf1D0>

Stay tuned for more information on ClueCon 2013 - it's coming up fast. In
the meantime, have a great week!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From msc at freeswitch.org  Wed Mar 13 19:08:31 2013
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Mar 2013 09:08:31 -0700
Subject: [Freeswitch-dev] FreeSWITCH Weekly Conference Call
Message-ID: <CAKzWOxVR9xT8rPemQvRUyiYxf8wjtQ9_BcE=0kwdDNhi4D394g@mail.gmail.com>

Hello folks,

Today's weekly conference call agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_13

Today we have Areski coming to discuss Newfies dialer. We hope you can make
it!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From thierry.devel at gmail.com  Tue Mar 19 01:43:47 2013
From: thierry.devel at gmail.com (Thierry Panthier)
Date: Tue, 19 Mar 2013 09:43:47 +1100
Subject: [Freeswitch-dev] mod_fifo
Message-ID: <CA+6yP1EHq7OPA0OxOo-3qzY-ZrDAFp-JyLcS1Aa3_VJTsiRujw@mail.gmail.com>

Hi,

I'm currently using a FreeSWITCH version from the stable branch (checked
out on 29/01/2013), which works perfectly fine.

Yesterday I tried to use version 1.2.7 (with the exact same configuration)
but my fifo stopped working. Basically every call never leaves the queue,
they hang in there *waiting* forever.

I compared these versions and I noticed the following modification:

- fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_TRUE*);
+ fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_FALSE*);

What was the motivation for this change?

If I revert the changes (use the old mod_fifo) everything works fine!

Is anyone out there having a similar problem?

Regards,

Thierry Panthier
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From dujinfang at gmail.com  Tue Mar 19 02:30:39 2013
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 19 Mar 2013 07:30:39 +0800
Subject: [Freeswitch-dev] mod_fifo
In-Reply-To: <CA+6yP1EHq7OPA0OxOo-3qzY-ZrDAFp-JyLcS1Aa3_VJTsiRujw@mail.gmail.com>
References: <CA+6yP1EHq7OPA0OxOo-3qzY-ZrDAFp-JyLcS1Aa3_VJTsiRujw@mail.gmail.com>
Message-ID: <B41D4CA0C3BC461EB7767C7DA10E1ECC@gmail.com>

check http://jira.freeswitch.org/browse/FS-4903 , I think it works for me. I'm not using this in production though.

you can comment or reopen, or create a new jira and link to this. 

-- 
Seven Du
http://www.freeswitch.org.cn
http://about.me/dujinfang
http://www.dujinfang.com


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


On Tuesday, March 19, 2013 at 6:43 AM, Thierry Panthier wrote:

> Hi,
> 
> I'm currently using a FreeSWITCH version from the stable branch (checked out on 29/01/2013), which works perfectly fine.
> 
> Yesterday I tried to use version 1.2.7 (with the exact same configuration) but my fifo stopped working. Basically every call never leaves the queue, they hang in there waiting forever.
> 
> I compared these versions and I noticed the following modification:
> 
> - fifo_execute_sql_queued(&sql, SWITCH_TRUE, SWITCH_TRUE);
> + fifo_execute_sql_queued(&sql, SWITCH_TRUE, SWITCH_FALSE);
> 
> What was the motivation for this change?
> 
> If I revert the changes (use the old mod_fifo) everything works fine!
> 
> Is anyone out there having a similar problem?
> 
> Regards,
> 
> Thierry Panthier 
> _________________________________________________________________________
> 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-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org (mailto:FreeSWITCH-dev at lists.freeswitch.org)
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
> 
> 


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From thierry.devel at gmail.com  Tue Mar 19 03:11:13 2013
From: thierry.devel at gmail.com (Thierry Panthier)
Date: Tue, 19 Mar 2013 11:11:13 +1100
Subject: [Freeswitch-dev] mod_fifo
In-Reply-To: <B41D4CA0C3BC461EB7767C7DA10E1ECC@gmail.com>
References: <CA+6yP1EHq7OPA0OxOo-3qzY-ZrDAFp-JyLcS1Aa3_VJTsiRujw@mail.gmail.com>
	<B41D4CA0C3BC461EB7767C7DA10E1ECC@gmail.com>
Message-ID: <CA+6yP1FSjweGT9+Xq6Ox8d5wDoOg0+Wmj9pKsGh62UREsBe-Vw@mail.gmail.com>

Thanks! I'll get the logs of both working and non-working versions and I'll
re-open this issue.

Regards,

Thierry


On Tue, Mar 19, 2013 at 10:30 AM, Seven Du <dujinfang at gmail.com> wrote:

>  check http://jira.freeswitch.org/browse/FS-4903 , I think it works for
> me. I'm not using this in production though.
>
> you can comment or reopen, or create a new jira and link to this.
>
> --
> Seven Du
> http://www.freeswitch.org.cn
> http://about.me/dujinfang
> http://www.dujinfang.com
>
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Tuesday, March 19, 2013 at 6:43 AM, Thierry Panthier wrote:
>
> Hi,
>
> I'm currently using a FreeSWITCH version from the stable branch (checked
> out on 29/01/2013), which works perfectly fine.
>
> Yesterday I tried to use version 1.2.7 (with the exact same configuration)
> but my fifo stopped working. Basically every call never leaves the queue,
> they hang in there *waiting* forever.
>
> I compared these versions and I noticed the following modification:
>
> - fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_TRUE*);
> + fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_FALSE*);
>
> What was the motivation for this change?
>
> If I revert the changes (use the old mod_fifo) everything works fine!
>
> Is anyone out there having a similar problem?
>
> Regards,
>
> Thierry Panthier
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Mar 19 17:11:03 2013
From: mike at jerris.com (Michael Jerris)
Date: Tue, 19 Mar 2013 10:11:03 -0400
Subject: [Freeswitch-dev] mod_fifo
In-Reply-To: <CA+6yP1EHq7OPA0OxOo-3qzY-ZrDAFp-JyLcS1Aa3_VJTsiRujw@mail.gmail.com>
References: <CA+6yP1EHq7OPA0OxOo-3qzY-ZrDAFp-JyLcS1Aa3_VJTsiRujw@mail.gmail.com>
Message-ID: <DAF72726-50B4-4AA9-BFA6-7E28B8F6C7C1@jerris.com>

The changes were related to fixing some performance bottlenecks in mod_fifo related to db operations.  Is there any particular one that is causing the problem or are you reverting the entire change?

Please update details on http://jira.freeswitch.org/browse/FS-4903

Thanks
Mike

On Mar 18, 2013, at 6:43 PM, Thierry Panthier <thierry.devel at gmail.com> wrote:

> Hi,
> 
> I'm currently using a FreeSWITCH version from the stable branch (checked out on 29/01/2013), which works perfectly fine.
> 
> Yesterday I tried to use version 1.2.7 (with the exact same configuration) but my fifo stopped working. Basically every call never leaves the queue, they hang in there waiting forever.
> 
> I compared these versions and I noticed the following modification:
> 
> - fifo_execute_sql_queued(&sql, SWITCH_TRUE, SWITCH_TRUE);
> + fifo_execute_sql_queued(&sql, SWITCH_TRUE, SWITCH_FALSE);
> 
> What was the motivation for this change?
> 
> If I revert the changes (use the old mod_fifo) everything works fine!
> 
> Is anyone out there having a similar problem?
> 

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From nneul at mst.edu  Tue Mar 19 17:07:20 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Tue, 19 Mar 2013 09:07:20 -0500
Subject: [Freeswitch-dev] 'make current' doesn't clean libs dirs
Message-ID: <51487118.1010102@mst.edu>

Noticed this while trying to test the fax support and found that mod_spandsp wouldn't load due to functions not being 
defined.

It looks like the 'make clean' and 'make modwipe' leave quite a few .o files around in the libs dirs. I'd imagine that 
most of the time, this has no impact, but when something in there changes......

-- Nathan

------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From anthony.minessale at gmail.com  Wed Mar 20 00:20:07 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Mar 2013 16:20:07 -0500
Subject: [Freeswitch-dev] 'make current' doesn't clean libs dirs
In-Reply-To: <51487118.1010102@mst.edu>
References: <51487118.1010102@mst.edu>
Message-ID: <CAKbxfG-pO6CAn=qMGg50bGtwLrPiftLT7sBv2MZKdjiNykYgqQ@mail.gmail.com>

Use git clean -fdx
On Mar 19, 2013 5:13 PM, "Nathan Neulinger" <nneul at mst.edu> wrote:

> Noticed this while trying to test the fax support and found that
> mod_spandsp wouldn't load due to functions not being
> defined.
>
> It looks like the 'make clean' and 'make modwipe' leave quite a few .o
> files around in the libs dirs. I'd imagine that
> most of the time, this has no impact, but when something in there
> changes......
>
> -- Nathan
>
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From thierry.devel at gmail.com  Wed Mar 20 06:17:32 2013
From: thierry.devel at gmail.com (Thierry Panthier)
Date: Wed, 20 Mar 2013 14:17:32 +1100
Subject: [Freeswitch-dev] mod_fifo
In-Reply-To: <DAF72726-50B4-4AA9-BFA6-7E28B8F6C7C1@jerris.com>
References: <CA+6yP1EHq7OPA0OxOo-3qzY-ZrDAFp-JyLcS1Aa3_VJTsiRujw@mail.gmail.com>
	<DAF72726-50B4-4AA9-BFA6-7E28B8F6C7C1@jerris.com>
Message-ID: <CA+6yP1GaOoOOjKcx66xp6QGBG0O6Kpz3ubm+um10Oo59-9+eCQ@mail.gmail.com>

Hi Michael,

I reverted all occurrences of the SQL execute commands:

- fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_TRUE*);
+ fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_FALSE*);

I haven't yet had the time to investigate the issue. All that I know is
that, after the update from my current version to 1.2.7, my fifo stopped
working. And that reverting those changes in mod_fifo fixed the issue.

I'm sorry for the delay but I'll post my config and the logs in the bug
description as soon as possible.

Regards,

Thierry




On Wed, Mar 20, 2013 at 1:11 AM, Michael Jerris <mike at jerris.com> wrote:

> The changes were related to fixing some performance bottlenecks in
> mod_fifo related to db operations.  Is there any particular one that is
> causing the problem or are you reverting the entire change?
>
> Please update details on http://jira.freeswitch.org/browse/FS-4903
>
> Thanks
> Mike
>
> On Mar 18, 2013, at 6:43 PM, Thierry Panthier <thierry.devel at gmail.com>
> wrote:
>
> Hi,
>
> I'm currently using a FreeSWITCH version from the stable branch (checked
> out on 29/01/2013), which works perfectly fine.
>
> Yesterday I tried to use version 1.2.7 (with the exact same configuration)
> but my fifo stopped working. Basically every call never leaves the queue,
> they hang in there *waiting* forever.
>
> I compared these versions and I noticed the following modification:
>
> - fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_TRUE*);
> + fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_FALSE*);
>
> What was the motivation for this change?
>
> If I revert the changes (use the old mod_fifo) everything works fine!
>
> Is anyone out there having a similar problem?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From nneul at mst.edu  Wed Mar 20 00:36:40 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Tue, 19 Mar 2013 16:36:40 -0500
Subject: [Freeswitch-dev] 'make current' doesn't clean libs dirs
In-Reply-To: <CAKbxfG-pO6CAn=qMGg50bGtwLrPiftLT7sBv2MZKdjiNykYgqQ@mail.gmail.com>
References: <51487118.1010102@mst.edu>
	<CAKbxfG-pO6CAn=qMGg50bGtwLrPiftLT7sBv2MZKdjiNykYgqQ@mail.gmail.com>
Message-ID: <5148DA68.3020408@mst.edu>

That will also purge modules config correct?

I was mainly going on the advice to use 'make current' to regenerate from updated git head, which doesn't appear to be 
sufficient if any changes have been made in the libraries. In my case, it wasn't even rebuilding the spandsp libs, even 
though the *.c files showed a newer timestamp.

Given that, there may be other dependency issues as well.

-- Nathan

On 03/19/2013 04:20 PM, Anthony Minessale wrote:
> git clean -fdx

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From anthony.minessale at gmail.com  Wed Mar 20 20:57:05 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 20 Mar 2013 12:57:05 -0500
Subject: [Freeswitch-dev] 'make current' doesn't clean libs dirs
In-Reply-To: <5148DA68.3020408@mst.edu>
References: <51487118.1010102@mst.edu>
	<CAKbxfG-pO6CAn=qMGg50bGtwLrPiftLT7sBv2MZKdjiNykYgqQ@mail.gmail.com>
	<5148DA68.3020408@mst.edu>
Message-ID: <CAKbxfG9__tsSJGxdBsK4XguFD3WkWagfsdot2cf53TbG3t4HZA@mail.gmail.com>

Yes this is how it is, the build system on builds libs once unless .update
is modified.  Make sure will do like make current with full clean.
On Mar 20, 2013 3:50 AM, "Nathan Neulinger" <nneul at mst.edu> wrote:

> That will also purge modules config correct?
>
> I was mainly going on the advice to use 'make current' to regenerate from
> updated git head, which doesn't appear to be
> sufficient if any changes have been made in the libraries. In my case, it
> wasn't even rebuilding the spandsp libs, even
> though the *.c files showed a newer timestamp.
>
> Given that, there may be other dependency issues as well.
>
> -- Nathan
>
> On 03/19/2013 04:20 PM, Anthony Minessale wrote:
> > git clean -fdx
>
> --
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From ashish at nms.co.in  Thu Mar 21 08:12:03 2013
From: ashish at nms.co.in (Ashish gautam)
Date: Thu, 21 Mar 2013 10:42:03 +0530
Subject: [Freeswitch-dev] freetdm channel restarting error
Message-ID: <CADBDwyHabfkhW=XnNXHmsVmtAU5Lsx6PL0DVqFzVja3R6LoFuQ@mail.gmail.com>

Hi,

I have set up and installed a PRI line on my freeswitch machine. I am
getting these warnings and errors when I start FS:

2013-03-21 10:38:11.294272 [WARNING] ftdm_io.c:3018 [s1c3][1:3] Channel not
opened, proceeding anyway
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c4][1:4] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c5][1:5] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c6][1:6] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c7][1:7] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c8][1:8] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c9][1:9] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c10][1:10] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c11][1:11] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c12][1:12] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c13][1:13] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c14][1:14] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c15][1:15] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c17][1:17] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c18][1:18] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c19][1:19] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.274272 [WARNING] ftmod_libpri.c:1954 [s1c20][1:20] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c21][1:21] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c22][1:22] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c23][1:23] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c24][1:24] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c25][1:25] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c26][1:26] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c27][1:27] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c28][1:28] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c29][1:29] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c30][1:30] --
T316 timed out, resending RESTART request
2013-03-21 10:38:36.294276 [WARNING] ftmod_libpri.c:1954 [s1c31][1:31] --
T316 timed out, resending RESTART request
2013-03-21 10:38:41.314272 [NOTICE] ftmod_libpri.c:2159 [s1c4][1:4] --
Restart of channel completed
2013-03-21 10:38:41.314272 [WARNING] ftdm_io.c:3018 [s1c4][1:4] Channel not
opened, proceeding anyway
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c5][1:5] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c6][1:6] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c7][1:7] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c8][1:8] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c9][1:9] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c10][1:10] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c11][1:11] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c12][1:12] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c13][1:13] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c14][1:14] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c15][1:15] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c17][1:17] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c18][1:18] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c19][1:19] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c20][1:20] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c21][1:21] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c22][1:22] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c23][1:23] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c24][1:24] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c25][1:25] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c26][1:26] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c27][1:27] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c28][1:28] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c29][1:29] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c30][1:30] -- T316
timed out, channel reached restart attempt limit '3' and is suspended
2013-03-21 10:39:06.294272 [ERR] ftmod_libpri.c:1950 [s1c31][1:31] -- T316
timed out, channel reached restart attempt limit '3' and is suspended

Please help me to get out of this.
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From jcherukuri_necc at yahoo.com  Thu Mar 21 22:40:54 2013
From: jcherukuri_necc at yahoo.com (Jyotshna Cherukuri)
Date: Thu, 21 Mar 2013 12:40:54 -0700 (PDT)
Subject: [Freeswitch-dev] Freeswitch's capability of switching RTP
	transmission to new destination's address without any SIP
	signaling update
Message-ID: <1363894854.61597.YahooMailNeo@web161302.mail.bf1.yahoo.com>

Hi,

I am wondering if?current version of FS has the capability to dynamically switch RTP trasnmission to the source address of RTP packets if the source address of RTP packets gets changed automatically while a call is already in progress . This might happen when a mobile handset gets a different IP on a access point reconnect during an ongoing call. ?

Any help is greatly appreciated

Thanks in advance
Regards
Jyotshna
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From anthony.minessale at gmail.com  Thu Mar 21 23:44:20 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Mar 2013 15:44:20 -0500
Subject: [Freeswitch-dev] Freeswitch's capability of switching RTP
 transmission to new destination's address without any SIP signaling update
In-Reply-To: <1363894854.61597.YahooMailNeo@web161302.mail.bf1.yahoo.com>
References: <1363894854.61597.YahooMailNeo@web161302.mail.bf1.yahoo.com>
Message-ID: <CAKbxfG8CD-F3Ph6WApFTJDs0Gvqt6vW5Vi53iKQVnX77DtoNAA@mail.gmail.com>

yes if there is a re-invite it will.


On Thu, Mar 21, 2013 at 2:40 PM, Jyotshna Cherukuri <
jcherukuri_necc at yahoo.com> wrote:

> Hi,
>
> I am wondering if current version of FS has the capability to dynamically
> switch RTP trasnmission to the source address of RTP packets if the source
> address of RTP packets gets changed automatically while a call is already
> in progress . This might happen when a mobile handset gets a different IP
> on a access point reconnect during an ongoing call.
>
> Any help is greatly appreciated
>
> Thanks in advance
> Regards
> Jyotshna
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at 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  Fri Mar 22 00:07:37 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 21 Mar 2013 21:07:37 +0000
Subject: [Freeswitch-dev] Freeswitch's capability of switching RTP
 transmission to new destination's address without any SIP signaling update
In-Reply-To: <CAKbxfG8CD-F3Ph6WApFTJDs0Gvqt6vW5Vi53iKQVnX77DtoNAA@mail.gmail.com>
References: <1363894854.61597.YahooMailNeo@web161302.mail.bf1.yahoo.com>
	<CAKbxfG8CD-F3Ph6WApFTJDs0Gvqt6vW5Vi53iKQVnX77DtoNAA@mail.gmail.com>
Message-ID: <CAFiqYum9pO7akK45nfdh=nsto0zxz3iB_gvwZzV8SCAVhSeW6w@mail.gmail.com>

Since your subject says without SIP signalling update, no. The phone needs
to detect its IP change and tell FS.

FS needs to know where to send signalling messages to as well as media -
ringing, answer, hangup etc. That means it needs to know the new IP & port
for the SIP part too.

The closest to what you're asking is RTP auto-adjust, which is a workaround
for devices unable to handle NAT correctly. But I don't think it should
allow it mid-call, or that could allow someone to hijack your phone call.

-Steve



On 21 March 2013 20:44, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> yes if there is a re-invite it will.
>
>
> On Thu, Mar 21, 2013 at 2:40 PM, Jyotshna Cherukuri <
> jcherukuri_necc at yahoo.com> wrote:
>
>> Hi,
>>
>> I am wondering if current version of FS has the capability to
>> dynamically switch RTP trasnmission to the source address of RTP packets if
>> the source address of RTP packets gets changed automatically while a call
>> is already in progress . This might happen when a mobile handset gets a
>> different IP on a access point reconnect during an ongoing call.
>>
>> Any help is greatly appreciated
>>
>> Thanks in advance
>> Regards
>> Jyotshna
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at 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-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From wpichler at yosd.at  Thu Mar 21 18:03:12 2013
From: wpichler at yosd.at (Wolfgang Pichler)
Date: Thu, 21 Mar 2013 16:03:12 +0100
Subject: [Freeswitch-dev] sofia_glue - sdp with m=image 0
Message-ID: <CACgr81NsnFi7ZzqECG1e-foBae1-Ntjt09zwBdnFa_95jRckLQ@mail.gmail.com>

Hi all,

i have encountered a strange behaviour - and i think i have found a
possible problem within sofia_glue

Following scenario:

Carrier sends INVITE with T38 Offer -> Freeswitch does forward offer ->
asterisk does forward offer -> ATA

In this scenario it happend to me - that the T38 Offer (with corrent
m=image and port in sdp) got forwarded by freeswitch to asterisk, asterisk
did forwarded without the T38 Offer (don't know why) - so also without the
m=image sdp part - the ATA did answered correctly (without m=image part) ->
asterisk did created a new sdp with m=image 0, without T38 parts - and
returned this to freeswitch.

Freeswitch did hangup the call - because sofia_glue did extracted the port
0 as audio port - and this is not a legal port...

Sending port 0 is according to rfc legal - so freeswitch should ignore it
in this case.

the question is - if both values are given - which one is the value we need
? I think the audio port is the port which is of higher priority.

So the workflow should be:
 - Try to extract audio port from m=audio part.
 - If you got it - and it is a valid port then go on
 - else try to extract audio port from m=image part...

The function in question is sofia_glue_tech_proxy_remote_addr - in
sofia_glue.c

Is it generaly a good idea to use the m=image part as audio port ? Why is
there no direct image port member in the pvt structure as with the video
port ?

Hope someone can help here...

br,
Wolfgang
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From anthony.minessale at gmail.com  Fri Mar 22 04:32:22 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Mar 2013 20:32:22 -0500
Subject: [Freeswitch-dev] sofia_glue - sdp with m=image 0
In-Reply-To: <CACgr81NsnFi7ZzqECG1e-foBae1-Ntjt09zwBdnFa_95jRckLQ@mail.gmail.com>
References: <CACgr81NsnFi7ZzqECG1e-foBae1-Ntjt09zwBdnFa_95jRckLQ@mail.gmail.com>
Message-ID: <CAKbxfG_7-7rkJvtaM2kQJkQBvBUMSNjQZuYPah9iLaPEjMkMJw@mail.gmail.com>

post a trace with "sofia global siptrace on"
its too hard to diagnose without the exact trace.


On Thu, Mar 21, 2013 at 10:03 AM, Wolfgang Pichler <wpichler at yosd.at> wrote:

> Hi all,
>
> i have encountered a strange behaviour - and i think i have found a
> possible problem within sofia_glue
>
> Following scenario:
>
> Carrier sends INVITE with T38 Offer -> Freeswitch does forward offer ->
> asterisk does forward offer -> ATA
>
> In this scenario it happend to me - that the T38 Offer (with corrent
> m=image and port in sdp) got forwarded by freeswitch to asterisk, asterisk
> did forwarded without the T38 Offer (don't know why) - so also without the
> m=image sdp part - the ATA did answered correctly (without m=image part) ->
> asterisk did created a new sdp with m=image 0, without T38 parts - and
> returned this to freeswitch.
>
> Freeswitch did hangup the call - because sofia_glue did extracted the port
> 0 as audio port - and this is not a legal port...
>
> Sending port 0 is according to rfc legal - so freeswitch should ignore it
> in this case.
>
> the question is - if both values are given - which one is the value we
> need ? I think the audio port is the port which is of higher priority.
>
> So the workflow should be:
>  - Try to extract audio port from m=audio part.
>  - If you got it - and it is a valid port then go on
>  - else try to extract audio port from m=image part...
>
> The function in question is sofia_glue_tech_proxy_remote_addr - in
> sofia_glue.c
>
> Is it generaly a good idea to use the m=image part as audio port ? Why is
> there no direct image port member in the pvt structure as with the video
> port ?
>
> Hope someone can help here...
>
> br,
> Wolfgang
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at 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 wpichler at yosd.at  Fri Mar 22 10:01:37 2013
From: wpichler at yosd.at (Wolfgang Pichler)
Date: Fri, 22 Mar 2013 08:01:37 +0100
Subject: [Freeswitch-dev] sofia_glue - sdp with m=image 0
In-Reply-To: <CAKbxfG_7-7rkJvtaM2kQJkQBvBUMSNjQZuYPah9iLaPEjMkMJw@mail.gmail.com>
References: <CACgr81NsnFi7ZzqECG1e-foBae1-Ntjt09zwBdnFa_95jRckLQ@mail.gmail.com>
	<CAKbxfG_7-7rkJvtaM2kQJkQBvBUMSNjQZuYPah9iLaPEjMkMJw@mail.gmail.com>
Message-ID: <CACgr81OfONMvLYPVKgrsciYH9iW2=HObgkEhJkUp_k4abDPLjA@mail.gmail.com>

sorry - i do have already reconfigured the audiocodes gateway which was
sending this - and it is already in production...

But i think the whole thing is also visible without trace, it is more a
basic implementation question.

The question is - is it really a good idea to overwrite the audio port -
when there is also a image port in the sdp ?

Here is the function i mean:



switch_status_t sofia_glue_tech_proxy_remote_addr(private_object_t
*tech_pvt, const char *sdp_str)
{
        const char *err;
        char rip[RA_PTR_LEN] = "";
        char rp[RA_PTR_LEN] = "";
        char rvp[RA_PTR_LEN] = "";
        char *p, *ip_ptr = NULL, *port_ptr = NULL, *vid_port_ptr = NULL,
*pe;
        int x;
        const char *val;
        switch_status_t status = SWITCH_STATUS_FALSE;

        if (zstr(sdp_str)) {
                sdp_str = tech_pvt->remote_sdp_str;
        }

        if (zstr(sdp_str)) {
                goto end;
        }

        if ((p = (char *) switch_stristr("c=IN IP4 ", sdp_str)) || (p =
(char *) switch_stristr("c=IN IP6 ", sdp_str))) {
                ip_ptr = p + 9;
        }

        if ((p = (char *) switch_stristr("m=audio ", sdp_str))) {
                port_ptr = p + 8;
        }

        if ((p = (char *) switch_stristr("m=image ", sdp_str))) {
                port_ptr = p + 8;
        }

        if ((p = (char *) switch_stristr("m=video ", sdp_str))) {
                vid_port_ptr = p + 8;
        }


As you can see - it first checks for audio port - writes it to port_ptr,
then checks for the image port - and if it does exists (also if it exists -
but is zero - which is legal) - it does overwrite the audio port.

So calls with sdp

m=audio 12123
and
m=image 0

will get dropped by freeswitch !

Thanks for you attention...

br,
Wolfgang

2013/3/22 Anthony Minessale <anthony.minessale at gmail.com>

> post a trace with "sofia global siptrace on"
> its too hard to diagnose without the exact trace.
>
>
> On Thu, Mar 21, 2013 at 10:03 AM, Wolfgang Pichler <wpichler at yosd.at>wrote:
>
>> Hi all,
>>
>> i have encountered a strange behaviour - and i think i have found a
>> possible problem within sofia_glue
>>
>> Following scenario:
>>
>> Carrier sends INVITE with T38 Offer -> Freeswitch does forward offer ->
>> asterisk does forward offer -> ATA
>>
>> In this scenario it happend to me - that the T38 Offer (with corrent
>> m=image and port in sdp) got forwarded by freeswitch to asterisk, asterisk
>> did forwarded without the T38 Offer (don't know why) - so also without the
>> m=image sdp part - the ATA did answered correctly (without m=image part) ->
>> asterisk did created a new sdp with m=image 0, without T38 parts - and
>> returned this to freeswitch.
>>
>> Freeswitch did hangup the call - because sofia_glue did extracted the
>> port 0 as audio port - and this is not a legal port...
>>
>> Sending port 0 is according to rfc legal - so freeswitch should ignore it
>> in this case.
>>
>> the question is - if both values are given - which one is the value we
>> need ? I think the audio port is the port which is of higher priority.
>>
>> So the workflow should be:
>>  - Try to extract audio port from m=audio part.
>>  - If you got it - and it is a valid port then go on
>>  - else try to extract audio port from m=image part...
>>
>> The function in question is sofia_glue_tech_proxy_remote_addr - in
>> sofia_glue.c
>>
>> Is it generaly a good idea to use the m=image part as audio port ? Why is
>> there no direct image port member in the pvt structure as with the video
>> port ?
>>
>> Hope someone can help here...
>>
>> br,
>> Wolfgang
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at 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-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Mar 22 19:54:48 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 22 Mar 2013 11:54:48 -0500
Subject: [Freeswitch-dev] sofia_glue - sdp with m=image 0
In-Reply-To: <CACgr81OfONMvLYPVKgrsciYH9iW2=HObgkEhJkUp_k4abDPLjA@mail.gmail.com>
References: <CACgr81NsnFi7ZzqECG1e-foBae1-Ntjt09zwBdnFa_95jRckLQ@mail.gmail.com>
	<CAKbxfG_7-7rkJvtaM2kQJkQBvBUMSNjQZuYPah9iLaPEjMkMJw@mail.gmail.com>
	<CACgr81OfONMvLYPVKgrsciYH9iW2=HObgkEhJkUp_k4abDPLjA@mail.gmail.com>
Message-ID: <CAKbxfG-T7qb5Vdky6zmKXhQ6L14Ldk8qJno0Lr1=9e0ihnZReQ@mail.gmail.com>

The log would have helped explain that you are using proxy_media mode which
is an important fact.

Please try latest HEAD.  And from now on report issues like this to
http://jira.freeswitch.org

Bugs should not be fixed via the mailing list.




On Fri, Mar 22, 2013 at 2:01 AM, Wolfgang Pichler <wpichler at yosd.at> wrote:

> sorry - i do have already reconfigured the audiocodes gateway which was
> sending this - and it is already in production...
>
> But i think the whole thing is also visible without trace, it is more a
> basic implementation question.
>
> The question is - is it really a good idea to overwrite the audio port -
> when there is also a image port in the sdp ?
>
> Here is the function i mean:
>
>
>
> switch_status_t sofia_glue_tech_proxy_remote_addr(private_object_t
> *tech_pvt, const char *sdp_str)
> {
>         const char *err;
>         char rip[RA_PTR_LEN] = "";
>         char rp[RA_PTR_LEN] = "";
>         char rvp[RA_PTR_LEN] = "";
>         char *p, *ip_ptr = NULL, *port_ptr = NULL, *vid_port_ptr = NULL,
> *pe;
>         int x;
>         const char *val;
>         switch_status_t status = SWITCH_STATUS_FALSE;
>
>         if (zstr(sdp_str)) {
>                 sdp_str = tech_pvt->remote_sdp_str;
>         }
>
>         if (zstr(sdp_str)) {
>                 goto end;
>         }
>
>         if ((p = (char *) switch_stristr("c=IN IP4 ", sdp_str)) || (p =
> (char *) switch_stristr("c=IN IP6 ", sdp_str))) {
>                 ip_ptr = p + 9;
>         }
>
>         if ((p = (char *) switch_stristr("m=audio ", sdp_str))) {
>                 port_ptr = p + 8;
>         }
>
>         if ((p = (char *) switch_stristr("m=image ", sdp_str))) {
>                 port_ptr = p + 8;
>         }
>
>         if ((p = (char *) switch_stristr("m=video ", sdp_str))) {
>                 vid_port_ptr = p + 8;
>         }
>
>
> As you can see - it first checks for audio port - writes it to port_ptr,
> then checks for the image port - and if it does exists (also if it exists -
> but is zero - which is legal) - it does overwrite the audio port.
>
> So calls with sdp
>
> m=audio 12123
> and
> m=image 0
>
> will get dropped by freeswitch !
>
> Thanks for you attention...
>
> br,
> Wolfgang
>
> 2013/3/22 Anthony Minessale <anthony.minessale at gmail.com>
>
>> post a trace with "sofia global siptrace on"
>> its too hard to diagnose without the exact trace.
>>
>>
>> On Thu, Mar 21, 2013 at 10:03 AM, Wolfgang Pichler <wpichler at yosd.at>wrote:
>>
>>> Hi all,
>>>
>>> i have encountered a strange behaviour - and i think i have found a
>>> possible problem within sofia_glue
>>>
>>> Following scenario:
>>>
>>> Carrier sends INVITE with T38 Offer -> Freeswitch does forward offer ->
>>> asterisk does forward offer -> ATA
>>>
>>> In this scenario it happend to me - that the T38 Offer (with corrent
>>> m=image and port in sdp) got forwarded by freeswitch to asterisk, asterisk
>>> did forwarded without the T38 Offer (don't know why) - so also without the
>>> m=image sdp part - the ATA did answered correctly (without m=image part) ->
>>> asterisk did created a new sdp with m=image 0, without T38 parts - and
>>> returned this to freeswitch.
>>>
>>> Freeswitch did hangup the call - because sofia_glue did extracted the
>>> port 0 as audio port - and this is not a legal port...
>>>
>>> Sending port 0 is according to rfc legal - so freeswitch should ignore
>>> it in this case.
>>>
>>> the question is - if both values are given - which one is the value we
>>> need ? I think the audio port is the port which is of higher priority.
>>>
>>> So the workflow should be:
>>>  - Try to extract audio port from m=audio part.
>>>  - If you got it - and it is a valid port then go on
>>>  - else try to extract audio port from m=image part...
>>>
>>> The function in question is sofia_glue_tech_proxy_remote_addr - in
>>> sofia_glue.c
>>>
>>> Is it generaly a good idea to use the m=image part as audio port ? Why
>>> is there no direct image port member in the pvt structure as with the video
>>> port ?
>>>
>>> Hope someone can help here...
>>>
>>> br,
>>> Wolfgang
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at 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-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at 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 ashish at nms.co.in  Fri Mar 22 13:53:43 2013
From: ashish at nms.co.in (Ashish gautam)
Date: Fri, 22 Mar 2013 16:23:43 +0530
Subject: [Freeswitch-dev] Unexpected call drop on PSTN network
Message-ID: <CADBDwyGXrMeuC=pXbuzDC5SBa7wdaSxwztZGoFsEQebUtXmT9A@mail.gmail.com>

Hi,

I am facing a strange issue with making outgoing calls to PSTN netwrok
through freetdm. The calls are successfully made to some Telecom networks
while they just drop abruptly and hangup on other networks. The calls are
being generated through inbound Even Socket originate action and put in a
public context which executes a perl script in the dialplan.

Please help me out. Thanks in advance.

--
Ashish
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From marketing at cluecon.com  Tue Mar 26 02:34:03 2013
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 25 Mar 2013 16:34:03 -0700
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxUj+xyp65DKgSjpiHNoPWKLB9jHDknYSrLTGpF+HGiS_Q@mail.gmail.com>

Greetings!

We are back after a brief hiatus. Our friend and colleague Brian West took
some much-deserved time off last week so the rest of us were a bit busier
than usual. Brian does an amazing amount of work on FreeSWITCH, CudaTel,
and ClueCon, so we're definitely glad he's back to work!

On last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_20>we talked
about Debian packaging with a little VoIP security and ZRTP
thrown in for good measure. Interestingly, a number of people stayed on the
call until 6pm EST! You are welcome to stay in the public conference room
for as long as you like. This
week<http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_27>we have Mark
Crane scheduled to give us an update on FusionPBX. It's been
about 12 months since we last heard from him. We are looking forward to
seeing the improvements that have been added in the past year.

We are also gearing up for ClueCon 2013 <http://www.cluecon.com> so save
the date: August 6-8, 2013. If you have any questions about being a
speaker, sponsor, or attendee then by all means contact us at this email
address. Remember that WebRTC is a big topic in the news right now and we
are looking for folks to talk about WebRTC in their open source telephony
projects and solutions.

Have a great week and we'll talk to you again in April!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From thierry.devel at gmail.com  Tue Mar 26 09:05:57 2013
From: thierry.devel at gmail.com (Thierry Panthier)
Date: Tue, 26 Mar 2013 17:05:57 +1100
Subject: [Freeswitch-dev] mod_fifo
In-Reply-To: <CA+6yP1GaOoOOjKcx66xp6QGBG0O6Kpz3ubm+um10Oo59-9+eCQ@mail.gmail.com>
References: <CA+6yP1EHq7OPA0OxOo-3qzY-ZrDAFp-JyLcS1Aa3_VJTsiRujw@mail.gmail.com>
	<DAF72726-50B4-4AA9-BFA6-7E28B8F6C7C1@jerris.com>
	<CA+6yP1GaOoOOjKcx66xp6QGBG0O6Kpz3ubm+um10Oo59-9+eCQ@mail.gmail.com>
Message-ID: <CA+6yP1HO=+v2qZ93oVZZf09ZLNh+aOZ3jX4QvMr3u-LRB=_7aw@mail.gmail.com>

Hi,

I have just uploaded all the information but I don't think I have
permission to re-open the bug.

Regards,

Thierry Panthier




On Wed, Mar 20, 2013 at 2:17 PM, Thierry Panthier
<thierry.devel at gmail.com>wrote:

> Hi Michael,
>
> I reverted all occurrences of the SQL execute commands:
>
> - fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_TRUE*);
> + fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_FALSE*);
>
> I haven't yet had the time to investigate the issue. All that I know is
> that, after the update from my current version to 1.2.7, my fifo stopped
> working. And that reverting those changes in mod_fifo fixed the issue.
>
> I'm sorry for the delay but I'll post my config and the logs in the bug
> description as soon as possible.
>
> Regards,
>
> Thierry
>
>
>
>
> On Wed, Mar 20, 2013 at 1:11 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> The changes were related to fixing some performance bottlenecks in
>> mod_fifo related to db operations.  Is there any particular one that is
>> causing the problem or are you reverting the entire change?
>>
>> Please update details on http://jira.freeswitch.org/browse/FS-4903
>>
>> Thanks
>> Mike
>>
>> On Mar 18, 2013, at 6:43 PM, Thierry Panthier <thierry.devel at gmail.com>
>> wrote:
>>
>> Hi,
>>
>> I'm currently using a FreeSWITCH version from the stable branch (checked
>> out on 29/01/2013), which works perfectly fine.
>>
>> Yesterday I tried to use version 1.2.7 (with the exact same
>> configuration) but my fifo stopped working. Basically every call never
>> leaves the queue, they hang in there *waiting* forever.
>>
>> I compared these versions and I noticed the following modification:
>>
>> - fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_TRUE*);
>> + fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_FALSE*);
>>
>> What was the motivation for this change?
>>
>> If I revert the changes (use the old mod_fifo) everything works fine!
>>
>> Is anyone out there having a similar problem?
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
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From wpichler at yosd.at  Tue Mar 26 08:31:57 2013
From: wpichler at yosd.at (Wolfgang Pichler)
Date: Tue, 26 Mar 2013 06:31:57 +0100
Subject: [Freeswitch-dev] sofia_glue - sdp with m=image 0
In-Reply-To: <CAKbxfG-T7qb5Vdky6zmKXhQ6L14Ldk8qJno0Lr1=9e0ihnZReQ@mail.gmail.com>
References: <CACgr81NsnFi7ZzqECG1e-foBae1-Ntjt09zwBdnFa_95jRckLQ@mail.gmail.com>
	<CAKbxfG_7-7rkJvtaM2kQJkQBvBUMSNjQZuYPah9iLaPEjMkMJw@mail.gmail.com>
	<CACgr81OfONMvLYPVKgrsciYH9iW2=HObgkEhJkUp_k4abDPLjA@mail.gmail.com>
	<CAKbxfG-T7qb5Vdky6zmKXhQ6L14Ldk8qJno0Lr1=9e0ihnZReQ@mail.gmail.com>
Message-ID: <CACgr81P=msw3srijtb+gqgm95YwC091sAQ4orekqWGqZ0RjRtg@mail.gmail.com>

Hi,

i will use the bug tracker next time - sorry.

I have posted to the mailing list - because i was not sure if it really is
a bug - or if it was intended behaviour...

The changes in latest HEAD do solve the problem for me.

br,
Wolfgang

2013/3/22 Anthony Minessale <anthony.minessale at gmail.com>

> The log would have helped explain that you are using proxy_media mode
> which is an important fact.
>
> Please try latest HEAD.  And from now on report issues like this to
> http://jira.freeswitch.org
>
> Bugs should not be fixed via the mailing list.
>
>
>
>
> On Fri, Mar 22, 2013 at 2:01 AM, Wolfgang Pichler <wpichler at yosd.at>wrote:
>
>> sorry - i do have already reconfigured the audiocodes gateway which was
>> sending this - and it is already in production...
>>
>> But i think the whole thing is also visible without trace, it is more a
>> basic implementation question.
>>
>> The question is - is it really a good idea to overwrite the audio port -
>> when there is also a image port in the sdp ?
>>
>> Here is the function i mean:
>>
>>
>>
>> switch_status_t sofia_glue_tech_proxy_remote_addr(private_object_t
>> *tech_pvt, const char *sdp_str)
>> {
>>         const char *err;
>>         char rip[RA_PTR_LEN] = "";
>>         char rp[RA_PTR_LEN] = "";
>>         char rvp[RA_PTR_LEN] = "";
>>         char *p, *ip_ptr = NULL, *port_ptr = NULL, *vid_port_ptr = NULL,
>> *pe;
>>         int x;
>>         const char *val;
>>         switch_status_t status = SWITCH_STATUS_FALSE;
>>
>>         if (zstr(sdp_str)) {
>>                 sdp_str = tech_pvt->remote_sdp_str;
>>         }
>>
>>         if (zstr(sdp_str)) {
>>                 goto end;
>>         }
>>
>>         if ((p = (char *) switch_stristr("c=IN IP4 ", sdp_str)) || (p =
>> (char *) switch_stristr("c=IN IP6 ", sdp_str))) {
>>                 ip_ptr = p + 9;
>>         }
>>
>>         if ((p = (char *) switch_stristr("m=audio ", sdp_str))) {
>>                 port_ptr = p + 8;
>>         }
>>
>>         if ((p = (char *) switch_stristr("m=image ", sdp_str))) {
>>                 port_ptr = p + 8;
>>         }
>>
>>         if ((p = (char *) switch_stristr("m=video ", sdp_str))) {
>>                 vid_port_ptr = p + 8;
>>         }
>>
>>
>> As you can see - it first checks for audio port - writes it to port_ptr,
>> then checks for the image port - and if it does exists (also if it exists -
>> but is zero - which is legal) - it does overwrite the audio port.
>>
>> So calls with sdp
>>
>> m=audio 12123
>> and
>> m=image 0
>>
>> will get dropped by freeswitch !
>>
>> Thanks for you attention...
>>
>> br,
>> Wolfgang
>>
>> 2013/3/22 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>> post a trace with "sofia global siptrace on"
>>> its too hard to diagnose without the exact trace.
>>>
>>>
>>> On Thu, Mar 21, 2013 at 10:03 AM, Wolfgang Pichler <wpichler at yosd.at>wrote:
>>>
>>>> Hi all,
>>>>
>>>> i have encountered a strange behaviour - and i think i have found a
>>>> possible problem within sofia_glue
>>>>
>>>> Following scenario:
>>>>
>>>> Carrier sends INVITE with T38 Offer -> Freeswitch does forward offer ->
>>>> asterisk does forward offer -> ATA
>>>>
>>>> In this scenario it happend to me - that the T38 Offer (with corrent
>>>> m=image and port in sdp) got forwarded by freeswitch to asterisk, asterisk
>>>> did forwarded without the T38 Offer (don't know why) - so also without the
>>>> m=image sdp part - the ATA did answered correctly (without m=image part) ->
>>>> asterisk did created a new sdp with m=image 0, without T38 parts - and
>>>> returned this to freeswitch.
>>>>
>>>> Freeswitch did hangup the call - because sofia_glue did extracted the
>>>> port 0 as audio port - and this is not a legal port...
>>>>
>>>> Sending port 0 is according to rfc legal - so freeswitch should ignore
>>>> it in this case.
>>>>
>>>> the question is - if both values are given - which one is the value we
>>>> need ? I think the audio port is the port which is of higher priority.
>>>>
>>>> So the workflow should be:
>>>>  - Try to extract audio port from m=audio part.
>>>>  - If you got it - and it is a valid port then go on
>>>>  - else try to extract audio port from m=image part...
>>>>
>>>> The function in question is sofia_glue_tech_proxy_remote_addr - in
>>>> sofia_glue.c
>>>>
>>>> Is it generaly a good idea to use the m=image part as audio port ? Why
>>>> is there no direct image port member in the pvt structure as with the video
>>>> port ?
>>>>
>>>> Hope someone can help here...
>>>>
>>>> br,
>>>> Wolfgang
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at 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-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at 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-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From ashish at nms.co.in  Tue Mar 26 12:12:04 2013
From: ashish at nms.co.in (Ashish gautam)
Date: Tue, 26 Mar 2013 14:42:04 +0530
Subject: [Freeswitch-dev] Unexpected call drop on PSTN network
Message-ID: <CADBDwyF8=+=2VaOGqGg4GwSNkqeffwtLpLOoTEsARfAhA1WQLg@mail.gmail.com>

Hi,

The issue got resolved. I did not set "ignore_early_media=true" in the dial
string. Everything is fine after that.

Thanks!

On Tue, Mar 26, 2013 at 5:04 AM, <
freeswitch-dev-request at lists.freeswitch.org> wrote:

> Send FreeSWITCH-dev mailing list submissions to
>         freeswitch-dev at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> or, via email, send a message with subject or body 'help' to
>         freeswitch-dev-request at lists.freeswitch.org
>
> You can reach the person managing the list at
>         freeswitch-dev-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-dev digest..."
>
> Today's Topics:
>
>    1. Re: sofia_glue - sdp with m=image 0 (Anthony Minessale)
>    2. Unexpected call drop on PSTN network (Ashish gautam)
>    3. FreeSWITCH Weekly News and Notes (Michael Collins)
>
>
> ---------- Forwarded message ----------
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-dev at lists.freeswitch.org
> Cc:
> Date: Fri, 22 Mar 2013 11:54:48 -0500
> Subject: Re: [Freeswitch-dev] sofia_glue - sdp with m=image 0
> The log would have helped explain that you are using proxy_media mode
> which is an important fact.
>
> Please try latest HEAD.  And from now on report issues like this to
> http://jira.freeswitch.org
>
> Bugs should not be fixed via the mailing list.
>
>
>
>
> On Fri, Mar 22, 2013 at 2:01 AM, Wolfgang Pichler <wpichler at yosd.at>wrote:
>
>> sorry - i do have already reconfigured the audiocodes gateway which was
>> sending this - and it is already in production...
>>
>> But i think the whole thing is also visible without trace, it is more a
>> basic implementation question.
>>
>> The question is - is it really a good idea to overwrite the audio port -
>> when there is also a image port in the sdp ?
>>
>> Here is the function i mean:
>>
>>
>>
>> switch_status_t sofia_glue_tech_proxy_remote_addr(private_object_t
>> *tech_pvt, const char *sdp_str)
>> {
>>         const char *err;
>>         char rip[RA_PTR_LEN] = "";
>>         char rp[RA_PTR_LEN] = "";
>>         char rvp[RA_PTR_LEN] = "";
>>         char *p, *ip_ptr = NULL, *port_ptr = NULL, *vid_port_ptr = NULL,
>> *pe;
>>         int x;
>>         const char *val;
>>         switch_status_t status = SWITCH_STATUS_FALSE;
>>
>>         if (zstr(sdp_str)) {
>>                 sdp_str = tech_pvt->remote_sdp_str;
>>         }
>>
>>         if (zstr(sdp_str)) {
>>                 goto end;
>>         }
>>
>>         if ((p = (char *) switch_stristr("c=IN IP4 ", sdp_str)) || (p =
>> (char *) switch_stristr("c=IN IP6 ", sdp_str))) {
>>                 ip_ptr = p + 9;
>>         }
>>
>>         if ((p = (char *) switch_stristr("m=audio ", sdp_str))) {
>>                 port_ptr = p + 8;
>>         }
>>
>>         if ((p = (char *) switch_stristr("m=image ", sdp_str))) {
>>                 port_ptr = p + 8;
>>         }
>>
>>         if ((p = (char *) switch_stristr("m=video ", sdp_str))) {
>>                 vid_port_ptr = p + 8;
>>         }
>>
>>
>> As you can see - it first checks for audio port - writes it to port_ptr,
>> then checks for the image port - and if it does exists (also if it exists -
>> but is zero - which is legal) - it does overwrite the audio port.
>>
>> So calls with sdp
>>
>> m=audio 12123
>> and
>> m=image 0
>>
>> will get dropped by freeswitch !
>>
>> Thanks for you attention...
>>
>> br,
>> Wolfgang
>>
>> 2013/3/22 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>> post a trace with "sofia global siptrace on"
>>> its too hard to diagnose without the exact trace.
>>>
>>>
>>> On Thu, Mar 21, 2013 at 10:03 AM, Wolfgang Pichler <wpichler at yosd.at>wrote:
>>>
>>>> Hi all,
>>>>
>>>> i have encountered a strange behaviour - and i think i have found a
>>>> possible problem within sofia_glue
>>>>
>>>> Following scenario:
>>>>
>>>> Carrier sends INVITE with T38 Offer -> Freeswitch does forward offer ->
>>>> asterisk does forward offer -> ATA
>>>>
>>>> In this scenario it happend to me - that the T38 Offer (with corrent
>>>> m=image and port in sdp) got forwarded by freeswitch to asterisk, asterisk
>>>> did forwarded without the T38 Offer (don't know why) - so also without the
>>>> m=image sdp part - the ATA did answered correctly (without m=image part) ->
>>>> asterisk did created a new sdp with m=image 0, without T38 parts - and
>>>> returned this to freeswitch.
>>>>
>>>> Freeswitch did hangup the call - because sofia_glue did extracted the
>>>> port 0 as audio port - and this is not a legal port...
>>>>
>>>> Sending port 0 is according to rfc legal - so freeswitch should ignore
>>>> it in this case.
>>>>
>>>> the question is - if both values are given - which one is the value we
>>>> need ? I think the audio port is the port which is of higher priority.
>>>>
>>>> So the workflow should be:
>>>>  - Try to extract audio port from m=audio part.
>>>>  - If you got it - and it is a valid port then go on
>>>>  - else try to extract audio port from m=image part...
>>>>
>>>> The function in question is sofia_glue_tech_proxy_remote_addr - in
>>>> sofia_glue.c
>>>>
>>>> Is it generaly a good idea to use the m=image part as audio port ? Why
>>>> is there no direct image port member in the pvt structure as with the video
>>>> port ?
>>>>
>>>> Hope someone can help here...
>>>>
>>>> br,
>>>> Wolfgang
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at 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-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at 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
>
>
> ---------- Forwarded message ----------
> From: Ashish gautam <ashish at nms.co.in>
> To: freeswitch-dev at lists.freeswitch.org
> Cc:
> Date: Fri, 22 Mar 2013 16:23:43 +0530
> Subject: [Freeswitch-dev] Unexpected call drop on PSTN network
> Hi,
>
> I am facing a strange issue with making outgoing calls to PSTN netwrok
> through freetdm. The calls are successfully made to some Telecom networks
> while they just drop abruptly and hangup on other networks. The calls are
> being generated through inbound Even Socket originate action and put in a
> public context which executes a perl script in the dialplan.
>
> Please help me out. Thanks in advance.
>
> --
> Ashish
>
>
> ---------- Forwarded message ----------
> From: Michael Collins <marketing at cluecon.com>
> To: freeswitch-users at lists.freeswitch.org,
> freeswitch-dev at lists.freeswitch.org,
> freeswitch-cluecon at lists.freeswitch.org
> Cc:
> Date: Mon, 25 Mar 2013 16:34:03 -0700
> Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
> Greetings!
>
> We are back after a brief hiatus. Our friend and colleague Brian West took
> some much-deserved time off last week so the rest of us were a bit busier
> than usual. Brian does an amazing amount of work on FreeSWITCH, CudaTel,
> and ClueCon, so we're definitely glad he's back to work!
>
> On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_20>we talked about Debian packaging with a little VoIP security and ZRTP
> thrown in for good measure. Interestingly, a number of people stayed on the
> call until 6pm EST! You are welcome to stay in the public conference room
> for as long as you like. This week<http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_27>we have Mark Crane scheduled to give us an update on FusionPBX. It's been
> about 12 months since we last heard from him. We are looking forward to
> seeing the improvements that have been added in the past year.
>
> We are also gearing up for ClueCon 2013 <http://www.cluecon.com> so save
> the date: August 6-8, 2013. If you have any questions about being a
> speaker, sponsor, or attendee then by all means contact us at this email
> address. Remember that WebRTC is a big topic in the news right now and we
> are looking for folks to talk about WebRTC in their open source telephony
> projects and solutions.
>
> Have a great week and we'll talk to you again in April!
>
> --
> Michael S Collins
> ClueCon Team
> http://www.cluecon.com
> 877-7-4ACLUE
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Mar 26 19:26:22 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Mar 2013 11:26:22 -0500
Subject: [Freeswitch-dev] mod_fifo
In-Reply-To: <CA+6yP1HO=+v2qZ93oVZZf09ZLNh+aOZ3jX4QvMr3u-LRB=_7aw@mail.gmail.com>
References: <CA+6yP1EHq7OPA0OxOo-3qzY-ZrDAFp-JyLcS1Aa3_VJTsiRujw@mail.gmail.com>
	<DAF72726-50B4-4AA9-BFA6-7E28B8F6C7C1@jerris.com>
	<CA+6yP1GaOoOOjKcx66xp6QGBG0O6Kpz3ubm+um10Oo59-9+eCQ@mail.gmail.com>
	<CA+6yP1HO=+v2qZ93oVZZf09ZLNh+aOZ3jX4QvMr3u-LRB=_7aw@mail.gmail.com>
Message-ID: <CAKbxfG_dGk9D6PkoT2BaK9Rj32nb5xcDBJF7ShGhR3AbzOkQwA@mail.gmail.com>

Please try to keep issues/topics either in Jira or this list.  Its hard to
jump back and forth.  Once something is in jira, typically its not
necessary to use the mailing list thread.


On Tue, Mar 26, 2013 at 1:05 AM, Thierry Panthier
<thierry.devel at gmail.com>wrote:

> Hi,
>
> I have just uploaded all the information but I don't think I have
> permission to re-open the bug.
>
> Regards,
>
> Thierry Panthier
>
>
>
>
> On Wed, Mar 20, 2013 at 2:17 PM, Thierry Panthier <thierry.devel at gmail.com
> > wrote:
>
>> Hi Michael,
>>
>> I reverted all occurrences of the SQL execute commands:
>>
>> - fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_TRUE*);
>> + fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_FALSE*);
>>
>> I haven't yet had the time to investigate the issue. All that I know is
>> that, after the update from my current version to 1.2.7, my fifo stopped
>> working. And that reverting those changes in mod_fifo fixed the issue.
>>
>> I'm sorry for the delay but I'll post my config and the logs in the bug
>> description as soon as possible.
>>
>> Regards,
>>
>> Thierry
>>
>>
>>
>>
>> On Wed, Mar 20, 2013 at 1:11 AM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> The changes were related to fixing some performance bottlenecks in
>>> mod_fifo related to db operations.  Is there any particular one that is
>>> causing the problem or are you reverting the entire change?
>>>
>>> Please update details on http://jira.freeswitch.org/browse/FS-4903
>>>
>>> Thanks
>>> Mike
>>>
>>> On Mar 18, 2013, at 6:43 PM, Thierry Panthier <thierry.devel at gmail.com>
>>> wrote:
>>>
>>> Hi,
>>>
>>> I'm currently using a FreeSWITCH version from the stable branch (checked
>>> out on 29/01/2013), which works perfectly fine.
>>>
>>> Yesterday I tried to use version 1.2.7 (with the exact same
>>> configuration) but my fifo stopped working. Basically every call never
>>> leaves the queue, they hang in there *waiting* forever.
>>>
>>> I compared these versions and I noticed the following modification:
>>>
>>> - fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_TRUE*);
>>> + fifo_execute_sql_queued(&sql, SWITCH_TRUE, *SWITCH_FALSE*);
>>>
>>> What was the motivation for this change?
>>>
>>> If I revert the changes (use the old mod_fifo) everything works fine!
>>>
>>> Is anyone out there having a similar problem?
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at 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 Mar 27 18:58:20 2013
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Mar 2013 08:58:20 -0700
Subject: [Freeswitch-dev] FreeSWITCH Community Conference Call Today
Message-ID: <CAKzWOxWmbFAddP+XHgqhgugH4Pr6gpOeUEmXkNE8ibfcaCm-hA@mail.gmail.com>

Hello all,

Today's conference call agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_27

We have a few quick items to discuss and then we will have Mark Crane give
us an update on FusionPBX.

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 georgi_mei at abv.bg  Thu Mar 28 11:12:06 2013
From: georgi_mei at abv.bg (Georgi Stefanov)
Date: Thu, 28 Mar 2013 10:12:06 +0200 (EET)
Subject: [Freeswitch-dev] Sending RTP data without SIP
Message-ID: <944388170.80102.1364458326768.JavaMail.apache@mail23.abv.bg>

 Hello All,

Recently, I am having fun with freeswitch.
I need to implement a module, which sends RTP data to previously specified IP and PORT. I do not want to have SIP/SDP session.

Which is the best or just good enough way to do that?

I understand my module should act as an endpoint then I need probably would need to create the session object, set the needed codecs, load (for example) the desired file for playing and start sending RTP.

I had some debugging actions on freeswitch and found that functions sofia_glue_negotiate_sdp, sofia_glue_tech_choose_port are dealing with local and remote site of RTP connections.

Could you advice me ?


From mike at jerris.com  Thu Mar 28 18:47:35 2013
From: mike at jerris.com (Michael Jerris)
Date: Thu, 28 Mar 2013 11:47:35 -0400
Subject: [Freeswitch-dev] Sending RTP data without SIP
In-Reply-To: <944388170.80102.1364458326768.JavaMail.apache@mail23.abv.bg>
References: <944388170.80102.1364458326768.JavaMail.apache@mail23.abv.bg>
Message-ID: <201E664D-82AE-4AC7-B1ED-331D2F7D41BC@jerris.com>

this mostly exists already.  check out rtp.c in the mod_sofia directory.

On Mar 28, 2013, at 4:12 AM, Georgi Stefanov <georgi_mei at abv.bg> wrote:

> Hello All,
> 
> Recently, I am having fun with freeswitch.
> I need to implement a module, which sends RTP data to previously specified IP and PORT. I do not want to have SIP/SDP session.
> 
> Which is the best or just good enough way to do that?
> 
> I understand my module should act as an endpoint then I need probably would need to create the session object, set the needed codecs, load (for example) the desired file for playing and start sending RTP.
> 
> I had some debugging actions on freeswitch and found that functions sofia_glue_negotiate_sdp, sofia_glue_tech_choose_port are dealing with local and remote site of RTP connections.
> 
> Could you advice me ?