From daniel.neubert at solomo.de  Sun May  1 02:13:43 2011
From: daniel.neubert at solomo.de (Daniel Neubert)
Date: Sun, 1 May 2011 00:13:43 +0200
Subject: [Freeswitch-dev] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTinKvt9MZ22HO4sENyQ52BK-ccu+2Q@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>	<4DBC0065.1060206@solomo.de>	<BANLkTikwmyo3HpSZPanV6Gic=UQy95x28w@mail.gmail.com>	<BANLkTimwV6u3_gB_kDmTVbrwZR-61V9naQ@mail.gmail.com>
	<BANLkTinKvt9MZ22HO4sENyQ52BK-ccu+2Q@mail.gmail.com>
Message-ID: <4DBC8997.70801@solomo.de>

Thanks for your Feedback! I failed on reading the wiki page (mixed up 
with the old version...) - shame on me :(

Using Ubuntu 10.04 LTS solved all issues - the system is working fine!

Thanks for your really good work with the install.pl script!

Best regards / Mit freundlichen Gr??en,
Daniel Neubert

On 30.04.2011 15:02, Giovanni Maruzzelli wrote:
> Also, don't forget to install all the required packages listed in the
> wikipage (eg: kernel headers, it seems not finding include files).
> Btw, I see you're on 32 bit. Never tested on 32 bit. Maybe there is a
> problem with 32 bit?
> Please open a Jira if you can't find a solution, or file a jira with
> the solution you found.
> Thanks again,
> -giovanni
>
> On 4/30/11, Giovanni Maruzzelli<gmaruzz at celliax.org>  wrote:
>> The supported linux distro are listed in the wiki page: centos 5.x and
>> 6.x, Ubuntu 10.04.
>> Maybe for newer or custom kernels you have to slightly modify the OSS
>> driver code.
>> If you do modify it successfully, please send a patch via
>> http://jira.freeswitch.org
>>
>> Thanks in advance,
>> -giovanni
>>
>> On 4/30/11, Giovanni Maruzzelli<gmaruzz at celliax.org>  wrote:
>>> 1) is very very bad to answer a mailing list post with something
>>> unrelated
>>> 2) use the new installer and the latest git to avoid any such problem
>>> 3) if you want or need to use the old way, refer to the old wikipage
>>> for the perfect install (first row in wiki page tell you where the old
>>> page is)
>>> 4) anyway, for bug, issues, etc, open a jira issue on
>>> http://jira.freeswitch.org
>>>
>>> Have a nice weekend and don't hiijack the threads ;)
>>>
>>> -giovanni
>>>
>>>
>>>
>>> On 4/30/11, Daniel Neubert<daniel.neubert at solomo.de>  wrote:
>>>> Thanks for your great work! I've been playing around with skypopen in my
>>>> spare time for a few days now.
>>>>
>>>> Every time I send a call to the skypopen endpoint, it fails with
>>>>
>>>> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:364
>>>> (skypopen/skype101/MySkypeUser) State ROUTING going to sleep
>>>> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:325
>>>> (skypopen/skype101/MySkypeUser) Running State Change CS_CONSUME_MEDIA
>>>> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:383
>>>> (skypopen/skype101/MySkypeUser) State CONSUME_MEDIA
>>>> 2011-04-30 14:15:53.532923 [DEBUG] mod_skypopen.c:747
>>>> [32b8f10|9350fb9] [DEBUG_SKYPE  747  ][skype101       ][IDLE,IDLE]
>>>> skype101 CHANNEL CONSUME_MEDIA
>>>> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:383
>>>> (skypopen/skype101/MySkypeUser) State CONSUME_MEDIA going to sleep
>>>> 2011-04-30 14:15:53.532923 [DEBUG] skypopen_protocol.c:173
>>>> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][IDLE,IDLE]
>>>> READING: |||CALL 39 STATUS UNPLACED|||
>>>> 2011-04-30 14:15:53.532923 [DEBUG] skypopen_protocol.c:714
>>>> [32b8f10|9350fb9] [DEBUG_SKYPE  714  ][skype101       ][DIALING,UNPLACD]
>>>> skype_call: 39 is now UNPLACED
>>>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173
>>>> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][DIALING,UNPLACD]
>>>> READING: |||CALL 39 STATUS ROUTING|||
>>>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:709
>>>> [32b8f10|9350fb9] [DEBUG_SKYPE  709  ][skype101       ][DIALING,ROUTING]
>>>> skype_call: 39 is now ROUTING
>>>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173
>>>> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][DIALING,ROUTING]
>>>> READING: |||CALL 39 FAILUREREASON 7|||
>>>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:540
>>>> [32b8f10|9350fb9] [DEBUG_SKYPE  540  ][skype101       ][DIALING,ROUTING]
>>>> Skype FAILED on skype_call 39. Let's wait for the FAILED message.
>>>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173
>>>> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][DIALING,ROUTING]
>>>> READING: |||CALL 39 VAA_INPUT_STATUS FALSE|||
>>>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173
>>>> [32b8f10|9350fb9] [DEBUG_SKYPE  173  ][skype101       ][DIALING,ROUTING]
>>>> READING: |||CALL 39 STATUS FAILED|||
>>>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:683
>>>> [32b8f10|9350fb9] [DEBUG_SKYPE  683  ][skype101       ][DIALING,FAILED]
>>>> we tried to call Skype on skype_call 39 and Skype has now FAILED
>>>> 2011-04-30 14:15:53.612936 [DEBUG] mod_skypopen.c:1413
>>>> [32b8f10|9350fb9] [DEBUG_SKYPE  1413 ][skype101       ][DOWN,FAILED]
>>>> skype call ended
>>>> 2011-04-30 14:15:53.612936 [DEBUG] switch_channel.c:2572
>>>> (skypopen/skype101/MySkypeUser) Callstate Change RINGING ->  HANGUP
>>>>
>>>> I assume that FAILUREREASON 7 indicates an issue regarding the audio
>>>> interface, correct?
>>>>
>>>> I've tried to modify these values (tried 0,1 and 2 (which was default) )
>>>> - but did not change anything.
>>>>
>>>> <CaptureDevice>2</CaptureDevice>
>>>> <RingDevice>2</RingDevice>
>>>> <SoundDevice>2</SoundDevice>
>>>>
>>>> Could you give me a hint?
>>>>
>>>> Best regards / Mit freundlichen Gr??en,
>>>> Daniel Neubert
>>>>
>>>> On 30.04.2011 10:31, Giovanni Maruzzelli wrote:
>>>>> Dear FreeSWITCHers,
>>>>>
>>>>> after a fair amount of effort, I ended up with a new way to install
>>>>> and use mod_skypopen on Linux.
>>>>>
>>>>> No more looking around the internet for the lost 2.0.0.72 Skype client
>>>>> for
>>>>> ALSA.
>>>>>
>>>>> First, we can use the readily available Skype client for OSS.
>>>>>
>>>>> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS
>>>>> driver, that's very easy to compile and install, and do not need to
>>>>> mess with the operating system installation.)
>>>>>
>>>>> Second, I wrote an installer that automatically do all the tedious
>>>>> work for you: download and install the skype client, create the config
>>>>> directory for Skype clients, create the config file for mod_skypopen,
>>>>> create the script that launches the Skype clients.
>>>>>
>>>>> I hope those improvements will lower the barriers for Skype calls on
>>>>> FreeSWITCH.
>>>>>
>>>>> Actually is ludicrously simple now, and after you compile FreeSWITCH,
>>>>> mod_skypopen and the skypopen.ko OSS driver it will take like less
>>>>> than one minute to have a complete installation of mod_skypopen ready
>>>>> to make and receive calls.
>>>>>
>>>>> All automatic, no more need to fiddle around with the Skype client
>>>>> download, configurations, authorization, etc.
>>>>>
>>>>> Is all well tested, but maybe there are still some bugs, and maybe the
>>>>> docs are not clear/easy enough.
>>>>>
>>>>> Please have a look at the new and improved wiki page and let me know
>>>>> what do you think about (and maybe test the procedures).
>>>>>
>>>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk
>>>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux
>>>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR
>>>>>
>>>>> You must update to the latest git to have all the goodies.
>>>>>
>>>>> Thank you all for your support,
>>>>>
>>>>> -giovanni
>>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>> --
>>> Sent from my mobile device
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>> --
>> Sent from my mobile device
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110501/2564dcf0/attachment.html 

From msc at freeswitch.org  Mon May  2 23:04:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 May 2011 12:04:11 -0700
Subject: [Freeswitch-dev] FreeSWITCH Cookbook: resources
Message-ID: <BANLkTi=sJdtc-00KbxAzomtiyq_8+_K6hw@mail.gmail.com>

Hello all!

The response to my request for reviewers is overwhelming - thank you so
much! We have more than enough reviewers. However, any of you who wish to
read the draft and offer feedback are still welcome to do so. You just won't
be "official" Packt reviewers.

I'd like to focus attention on testing resources. For those of you who have
data center resources and such I would like to ask for your assistance. It
would be helpful to have a spare server (low power == okay, cuz it's only
for a few simultaneous calls max) with a DID & public IP address. This will
help us test various scenarios and make sure that we can do box2box and NAT
scenarios.

Please contact me off list if you have any servers, DIDs, etc. that could be
used for this purpose.

Thanks again to everyone who has volunteered to help! I would much rather
have to sift through dozens of emails from people who want to help than have
to keep begging and have no one step up. ;)  You guys are awesome - keep up
the good work.

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110502/76fbd212/attachment-0001.html 

From dujinfang at gmail.com  Tue May  3 04:27:44 2011
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 3 May 2011 08:27:44 +0800
Subject: [Freeswitch-dev] FreeSWITCH Cookbook: resources
In-Reply-To: <BANLkTi=sJdtc-00KbxAzomtiyq_8+_K6hw@mail.gmail.com>
References: <BANLkTi=sJdtc-00KbxAzomtiyq_8+_K6hw@mail.gmail.com>
Message-ID: <3B214F233ACA44E7816385080AA75FE1@gmail.com>

We have a server that currently no load, it's in Softlayer and let me know your pubkey if you want to use that. Also let me know how long you want to use that, a few weeks is fine.

It's Ubuntu Karmic 64bit.


Perhaps I don't have time to review all the chapters, but still would like to review and contribute ideas for the recipe book.

Thanks.

-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow
On Tuesday, May 3, 2011 at 3:04 AM, Michael Collins wrote: 
> Hello all!
> 
> The response to my request for reviewers is overwhelming - thank you so much! We have more than enough reviewers. However, any of you who wish to read the draft and offer feedback are still welcome to do so. You just won't be "official" Packt reviewers. 
> 
> I'd like to focus attention on testing resources. For those of you who have data center resources and such I would like to ask for your assistance. It would be helpful to have a spare server (low power == okay, cuz it's only for a few simultaneous calls max) with a DID & public IP address. This will help us test various scenarios and make sure that we can do box2box and NAT scenarios. 
> 
> Please contact me off list if you have any servers, DIDs, etc. that could be used for this purpose.
> 
> Thanks again to everyone who has volunteered to help! I would much rather have to sift through dozens of emails from people who want to help than have to keep begging and have no one step up. ;) You guys are awesome - keep up the good work. 
> 
> -MC 
> _______________________________________________
> 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
> 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110503/74bfe7c9/attachment.html 

From msc at freeswitch.org  Wed May  4 19:27:31 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 4 May 2011 08:27:31 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <BANLkTi=H5hdhzP8nh8JgnX3trEjBjsUtvQ@mail.gmail.com>

Hey folks,

We're having an easy conference call today. The agenda is light:
http://wiki.freeswitch.org/wiki/FS_weekly_2011_05_04

Feel free to join and hang out with us!

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110504/c6b96ab8/attachment.html 

From anton.vazir at gmail.com  Fri May  6 23:35:14 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 7 May 2011 00:35:14 +0500
Subject: [Freeswitch-dev] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <4DBC03EA.6050902@solomo.de>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<4DBC0065.1060206@solomo.de> <4DBC03EA.6050902@solomo.de>
Message-ID: <BANLkTikbOiy-NV7v1+nRQe8oiLz0i+DZhA@mail.gmail.com>

Hi Daniel!

Have you been able to compile the module for 2.6.38?

2011/4/30 Daniel Neubert <daniel.neubert at solomo.de>:
> Looks like the issue is located in building the skypopen.ko module:
>
> make -C /lib/modules/2.6.38-8-generic/build
> M=/usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss
> LDDINC=/usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/../include
> modules
> make[1]: Betrete Verzeichnis '/usr/src/linux-headers-2.6.38-8-generic'
> CC [M] /usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/main.o
> /usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/main.c:315:2:
> error: unknown field ?ioctl? specified in initializer
> /usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/main.c:315:2:
> warning: initialization from incompatible pointer type
> make[2]: ***
> [/usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/main.o]
> Fehler 1
> make[1]: ***
> [_module_/usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss]
> Fehler 2
> make[1]: Verlasse Verzeichnis '/usr/src/linux-headers-2.6.38-8-generic'
> make: *** [modules] Fehler 2
>
> I guess using kernel 2.6.38-8 is not supported?
>
> 2.6.38-8-generic #42-Ubuntu SMP Mon Apr 11 03:31:50 UTC 2011 i686 i686
> i386 GNU/Linux
>
> Best regards / Mit freundlichen Gr??en,
> Daniel Neubert
>
>
>
> On 30.04.2011 14:28, Daniel Neubert wrote:
>> Thanks for your great work! I've been playing around with skypopen in my
>> spare time for a few days now.
>>
>> Every time I send a call to the skypopen endpoint, it fails with
>>
>> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:364
>> (skypopen/skype101/MySkypeUser) State ROUTING going to sleep
>> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:325
>> (skypopen/skype101/MySkypeUser) Running State Change CS_CONSUME_MEDIA
>> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:383
>> (skypopen/skype101/MySkypeUser) State CONSUME_MEDIA
>> 2011-04-30 14:15:53.532923 [DEBUG] mod_skypopen.c:747
>> [32b8f10|9350fb9] [DEBUG_SKYPE ?747 ?][skype101 ? ? ? ][IDLE,IDLE]
>> skype101 CHANNEL CONSUME_MEDIA
>> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:383
>> (skypopen/skype101/MySkypeUser) State CONSUME_MEDIA going to sleep
>> 2011-04-30 14:15:53.532923 [DEBUG] skypopen_protocol.c:173
>> [32b8f10|9350fb9] [DEBUG_SKYPE ?173 ?][skype101 ? ? ? ][IDLE,IDLE]
>> READING: |||CALL 39 STATUS UNPLACED|||
>> 2011-04-30 14:15:53.532923 [DEBUG] skypopen_protocol.c:714
>> [32b8f10|9350fb9] [DEBUG_SKYPE ?714 ?][skype101 ? ? ? ][DIALING,UNPLACD]
>> skype_call: 39 is now UNPLACED
>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173
>> [32b8f10|9350fb9] [DEBUG_SKYPE ?173 ?][skype101 ? ? ? ][DIALING,UNPLACD]
>> READING: |||CALL 39 STATUS ROUTING|||
>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:709
>> [32b8f10|9350fb9] [DEBUG_SKYPE ?709 ?][skype101 ? ? ? ][DIALING,ROUTING]
>> skype_call: 39 is now ROUTING
>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173
>> [32b8f10|9350fb9] [DEBUG_SKYPE ?173 ?][skype101 ? ? ? ][DIALING,ROUTING]
>> READING: |||CALL 39 FAILUREREASON 7|||
>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:540
>> [32b8f10|9350fb9] [DEBUG_SKYPE ?540 ?][skype101 ? ? ? ][DIALING,ROUTING]
>> Skype FAILED on skype_call 39. Let's wait for the FAILED message.
>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173
>> [32b8f10|9350fb9] [DEBUG_SKYPE ?173 ?][skype101 ? ? ? ][DIALING,ROUTING]
>> READING: |||CALL 39 VAA_INPUT_STATUS FALSE|||
>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173
>> [32b8f10|9350fb9] [DEBUG_SKYPE ?173 ?][skype101 ? ? ? ][DIALING,ROUTING]
>> READING: |||CALL 39 STATUS FAILED|||
>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:683
>> [32b8f10|9350fb9] [DEBUG_SKYPE ?683 ?][skype101 ? ? ? ][DIALING,FAILED]
>> we tried to call Skype on skype_call 39 and Skype has now FAILED
>> 2011-04-30 14:15:53.612936 [DEBUG] mod_skypopen.c:1413
>> [32b8f10|9350fb9] [DEBUG_SKYPE ?1413 ][skype101 ? ? ? ][DOWN,FAILED]
>> skype call ended
>> 2011-04-30 14:15:53.612936 [DEBUG] switch_channel.c:2572
>> (skypopen/skype101/MySkypeUser) Callstate Change RINGING -> ?HANGUP
>>
>> I assume that FAILUREREASON 7 indicates an issue regarding the audio
>> interface, correct?
>>
>> I've tried to modify these values (tried 0,1 and 2 (which was default) )
>> - but did not change anything.
>>
>> <CaptureDevice>2</CaptureDevice>
>> <RingDevice>2</RingDevice>
>> <SoundDevice>2</SoundDevice>
>>
>> Could you give me a hint?
>>
>> Best regards / Mit freundlichen Gr??en,
>> Daniel Neubert
>>
>> On 30.04.2011 10:31, Giovanni Maruzzelli wrote:
>>> Dear FreeSWITCHers,
>>>
>>> after a fair amount of effort, I ended up with a new way to install
>>> and use mod_skypopen on Linux.
>>>
>>> No more looking around the internet for the lost 2.0.0.72 Skype client for ALSA.
>>>
>>> First, we can use the readily available Skype client for OSS.
>>>
>>> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS
>>> driver, that's very easy to compile and install, and do not need to
>>> mess with the operating system installation.)
>>>
>>> Second, I wrote an installer that automatically do all the tedious
>>> work for you: download and install the skype client, create the config
>>> directory for Skype clients, create the config file for mod_skypopen,
>>> create the script that launches the Skype clients.
>>>
>>> I hope those improvements will lower the barriers for Skype calls on FreeSWITCH.
>>>
>>> Actually is ludicrously simple now, and after you compile FreeSWITCH,
>>> mod_skypopen and the skypopen.ko OSS driver it will take like less
>>> than one minute to have a complete installation of mod_skypopen ready
>>> to make and receive calls.
>>>
>>> All automatic, no more need to fiddle around with the Skype client
>>> download, configurations, authorization, etc.
>>>
>>> Is all well tested, but maybe there are still some bugs, and maybe the
>>> docs are not clear/easy enough.
>>>
>>> Please have a look at the new and improved wiki page and let me know
>>> what do you think about (and maybe test the procedures).
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk
>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux
>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR
>>>
>>> You must update to the latest git to have all the goodies.
>>>
>>> Thank you all for your support,
>>>
>>> -giovanni
>>>
>
> _______________________________________________
> 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 anton.vazir at gmail.com  Fri May  6 23:37:58 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 7 May 2011 00:37:58 +0500
Subject: [Freeswitch-dev] skypopen.ko compile failed for 2.6.36.4
Message-ID: <BANLkTikh2+DmFMiOYAwbZNF+9JnGQbpeBw@mail.gmail.com>

Hi Giovanni,

Seems you do not support yet 2.6.36 kernel? Could you please have a
look, maybe it needs a trivial change to compile under 2.6.36 ?

root at lab3:/usr/src/freeswitch/src/mod/endpoints/mod_skypopen/oss# make
make -C /lib/modules/2.6.36.4/build
M=/usr/src/freeswitch/src/mod/endpoints/mod_skypopen/oss
LDDINC=/usr/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/../include
modules
make[1]: Entering directory `/usr/src/linux-headers-2.6.36.4'
  CC [M]  /usr/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/main.o
/usr/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/main.c:315:
error: unknown field ?ioctl? specified in initializer
/usr/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/main.c:315:
warning: initialization from incompatible pointer type
make[2]: *** [/usr/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/main.o]
Error 1
make[1]: *** [_module_/usr/src/freeswitch/src/mod/endpoints/mod_skypopen/oss]
Error 2
make[1]: Leaving directory `/usr/src/linux-headers-2.6.36.4'
make: *** [modules] Error 2

Sincerely,
Anton.


From anton.vazir at gmail.com  Sat May  7 00:10:51 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 7 May 2011 01:10:51 +0500
Subject: [Freeswitch-dev] skypopen.ko compile failed for 2.6.36.4
In-Reply-To: <BANLkTikh2+DmFMiOYAwbZNF+9JnGQbpeBw@mail.gmail.com>
References: <BANLkTikh2+DmFMiOYAwbZNF+9JnGQbpeBw@mail.gmail.com>
Message-ID: <BANLkTi=fQ1DdfNB6cnw6HJ+Xm=GK55Bv+A@mail.gmail.com>

IOCTL system in new kernels changed, it's described here

http://lwn.net/Articles/119652/

so we change line 315 of main.c of skypopen:

Old code
.ioctl =    skypopen_ioctl

new code
.compat_ioctl =    skypopen_ioctl

Than it compiles and loads into the kernel.
I'm just going to test if it actually works.

sorry for not providing patch, lasy...


From anton.vazir at gmail.com  Sat May  7 00:12:56 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 7 May 2011 01:12:56 +0500
Subject: [Freeswitch-dev] skypopen.ko compile failed for 2.6.36.4
In-Reply-To: <BANLkTi=fQ1DdfNB6cnw6HJ+Xm=GK55Bv+A@mail.gmail.com>
References: <BANLkTikh2+DmFMiOYAwbZNF+9JnGQbpeBw@mail.gmail.com>
	<BANLkTi=fQ1DdfNB6cnw6HJ+Xm=GK55Bv+A@mail.gmail.com>
Message-ID: <BANLkTi=PO8zYTKbhDgNq6kJ8wVeiTShexg@mail.gmail.com>

Actually new IOCTL should avoid BKL, but it's written that
conpat_ioctl should nto use BKL too, if it is not so - the code should
be changed to use
unlocked_ioct instead to avoid BKL

2011/5/7 Anton VG <anton.vazir at gmail.com>:
> IOCTL system in new kernels changed, it's described here
>
> http://lwn.net/Articles/119652/
>
> so we change line 315 of main.c of skypopen:
>
> Old code
> .ioctl = ? ?skypopen_ioctl
>
> new code
> .compat_ioctl = ? ?skypopen_ioctl
>
> Than it compiles and loads into the kernel.
> I'm just going to test if it actually works.
>
> sorry for not providing patch, lasy...
>


From anton.vazir at gmail.com  Sat May  7 00:39:23 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 7 May 2011 01:39:23 +0500
Subject: [Freeswitch-dev] skypopen.ko compile failed for 2.6.36.4
In-Reply-To: <BANLkTi=PO8zYTKbhDgNq6kJ8wVeiTShexg@mail.gmail.com>
References: <BANLkTikh2+DmFMiOYAwbZNF+9JnGQbpeBw@mail.gmail.com>
	<BANLkTi=fQ1DdfNB6cnw6HJ+Xm=GK55Bv+A@mail.gmail.com>
	<BANLkTi=PO8zYTKbhDgNq6kJ8wVeiTShexg@mail.gmail.com>
Message-ID: <BANLkTikLiUqhhCyU8JX0u2OabEkcXD5wMQ@mail.gmail.com>

It works, but somehow strange. FS demo talks sssslllloooowwww, like
timing issue..., (do not happen if I call from SIP phone in parallel),
and also was OK with previous skypopen version for ALSA. Would be nice
if someone could test it too.

2011/5/7 Anton VG <anton.vazir at gmail.com>:
> Actually new IOCTL should avoid BKL, but it's written that
> conpat_ioctl should nto use BKL too, if it is not so - the code should
> be changed to use
> unlocked_ioct instead to avoid BKL
>
> 2011/5/7 Anton VG <anton.vazir at gmail.com>:
>> IOCTL system in new kernels changed, it's described here
>>
>> http://lwn.net/Articles/119652/
>>
>> so we change line 315 of main.c of skypopen:
>>
>> Old code
>> .ioctl = ? ?skypopen_ioctl
>>
>> new code
>> .compat_ioctl = ? ?skypopen_ioctl
>>
>> Than it compiles and loads into the kernel.
>> I'm just going to test if it actually works.
>>
>> sorry for not providing patch, lasy...
>>
>


From anton.vazir at gmail.com  Sat May  7 01:01:08 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 7 May 2011 02:01:08 +0500
Subject: [Freeswitch-dev] skypopen.ko compile failed for 2.6.36.4
In-Reply-To: <BANLkTikLiUqhhCyU8JX0u2OabEkcXD5wMQ@mail.gmail.com>
References: <BANLkTikh2+DmFMiOYAwbZNF+9JnGQbpeBw@mail.gmail.com>
	<BANLkTi=fQ1DdfNB6cnw6HJ+Xm=GK55Bv+A@mail.gmail.com>
	<BANLkTi=PO8zYTKbhDgNq6kJ8wVeiTShexg@mail.gmail.com>
	<BANLkTikLiUqhhCyU8JX0u2OabEkcXD5wMQ@mail.gmail.com>
Message-ID: <BANLkTi=tDH7yHQQ1M4Xx_Ocy10KrxphEFA@mail.gmail.com>

Giovanni,
Ok, I divided the hrtimer by 2 (main.c, line 95), as shown below, and
now it's sounds almost normal, maybe just a slightly faster. in whole
fs-demo, skype gone forward for a 1 second.

hrtimer_forward(&dev->timer_inq, now, ktime_set(0, SKYPOPEN_SLEEP * 1000000/2));

btw, unlocked_ioctl works too.


From anton.vazir at gmail.com  Sat May  7 11:12:37 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 7 May 2011 12:12:37 +0500
Subject: [Freeswitch-dev] is there any info abount which git tag/ Freeswitch
	release is stable?
Message-ID: <BANLkTinNo1JvE+Lf=nH8FyCRPD1DSEHoMA@mail.gmail.com>

Guys, here is a little bit of critics,

There is no info - what version of freeswitch should be used in production.
the wiki pages

http://wiki.freeswitch.org/wiki/Installation_Guide
http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix

says that latest GIT is "very" stable, but in 5 pulls i did, 2 times I
immediately hit regressions:
first time - non working DNS resolver in mod_sofia
second - broken ESL outbound (as of this night)

at the same time, it's mentioned that 1.0.6 is not the "latest stable"
anymore, but in files, there is only 1.0.6

so it would be very helpful if there is some way to find out, what is
the latest GIT tag or release could be downloaded
to be usable, it's just not possible while in development, to have GIT
always stable. None can do it, so there really should be atleast
"testing" tag/brunch,
which is known working over serveral weeks (debian testing can be an example)

Hope this piece of critics not insults anyone,
FS is a great stuff,
Regards,
Anton.


From anton.vazir at gmail.com  Sat May  7 13:03:30 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 7 May 2011 14:03:30 +0500
Subject: [Freeswitch-dev] is there any info abount which git tag/
	Freeswitch release is stable?
In-Reply-To: <BANLkTinNo1JvE+Lf=nH8FyCRPD1DSEHoMA@mail.gmail.com>
References: <BANLkTinNo1JvE+Lf=nH8FyCRPD1DSEHoMA@mail.gmail.com>
Message-ID: <BANLkTinNrWdaT93a6Jg4mwwmMkqhh1_mww@mail.gmail.com>

regarding the broken ESL - it was my bug, my apologies.
As for stable, or pre-stable version/tag/branch... question persists...


From daniel.bryars at aeriandi.com  Sun May  8 01:47:58 2011
From: daniel.bryars at aeriandi.com (Daniel Bryars)
Date: Sat, 7 May 2011 22:47:58 +0100
Subject: [Freeswitch-dev] is there any info abount which git tag/
 Freeswitch	release is stable?
In-Reply-To: <BANLkTinNo1JvE+Lf=nH8FyCRPD1DSEHoMA@mail.gmail.com>
References: <BANLkTinNo1JvE+Lf=nH8FyCRPD1DSEHoMA@mail.gmail.com>
Message-ID: <2CC6B62FEE3A634CAD6B5D44FC9315E24FA6693F55@aeramsdc1.AERIANDI.LAN>

I don't think it's impossible - just difficult. My view is based on my experience where we "almost" continuously deploy our web sites to our live environment; we continuously deploy after every check in (waiting 15 minutes for a quiescent period) to stage and then manually push to live weekly (it used to be ad hoc, now it's every week, I hope soon it will be every day). To do this we've needed key unit tests (not full coverage by any means) and automated regression testing.  And, most importantly, a change in developer mind set (we now know that as soon as we check in some code it's going to wind itself up on a live machine with live customers without any QA team to rely on).

We're building a new solution with FS (on Windows) and ideally would like to adopt the same strategy. Ideally I don't want a "stable release" I want to always pull the latest version and run that in production.

Right now for our FS project we've got the build process deploying to our stage environment (no live customers yet) using some Powershell scripts and Microsoft Web Deploy. We intend to build on this over the next 4 to 6 weeks. The plan is to pull from GIT, apply our own regression tests and if they pass push out to live (automatically).

Currently we have plans to test only the functionality which we actually use - but depending on how things go I'm up for increasing the coverage and providing the results of our tests publicly. Perhaps this will provide a useful guide to the current state of the build and help provide what you're asking for (but I can only do this on Windows.)

I'll keep you posted with our progress, (and I'll also be at the next ClueCon conference - if you're going then perhaps we can have a little brainstorm then.)

Daniel


-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anton VG
Sent: 07 May 2011 08:13
To: freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-dev] is there any info abount which git tag/ Freeswitch release is stable?

Guys, here is a little bit of critics,

There is no info - what version of freeswitch should be used in production.
the wiki pages

http://wiki.freeswitch.org/wiki/Installation_Guide
http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix

says that latest GIT is "very" stable, but in 5 pulls i did, 2 times I immediately hit regressions:
first time - non working DNS resolver in mod_sofia second - broken ESL outbound (as of this night)

at the same time, it's mentioned that 1.0.6 is not the "latest stable"
anymore, but in files, there is only 1.0.6

so it would be very helpful if there is some way to find out, what is the latest GIT tag or release could be downloaded to be usable, it's just not possible while in development, to have GIT always stable. None can do it, so there really should be atleast "testing" tag/brunch, which is known working over serveral weeks (debian testing can be an example)

Hope this piece of critics not insults anyone, FS is a great stuff, Regards, Anton.

_______________________________________________
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


The information transmitted is intended only for the person or entity
to which it is addressed and may contain confidential and/or privileged
material. Any review, retransmission, dissemination or other use of, or
taking of any action in reliance upon, this information by persons or
entities other than the intended recipient is prohibited. If you received
this in error, please contact the sender and delete the material from
your computer.


From anthony.minessale at gmail.com  Sun May  8 01:57:06 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 7 May 2011 16:57:06 -0500
Subject: [Freeswitch-dev] is there any info abount which git tag/
 Freeswitch release is stable?
In-Reply-To: <2CC6B62FEE3A634CAD6B5D44FC9315E24FA6693F55@aeramsdc1.AERIANDI.LAN>
References: <BANLkTinNo1JvE+Lf=nH8FyCRPD1DSEHoMA@mail.gmail.com>
	<2CC6B62FEE3A634CAD6B5D44FC9315E24FA6693F55@aeramsdc1.AERIANDI.LAN>
Message-ID: <BANLkTikNbA1hgon8OCnK0cD+55p4MXY8Og@mail.gmail.com>

current HEAD is a candidate for a stable branch.

On Sat, May 7, 2011 at 4:47 PM, Daniel Bryars
<daniel.bryars at aeriandi.com> wrote:
> I don't think it's impossible - just difficult. My view is based on my experience where we "almost" continuously deploy our web sites to our live environment; we continuously deploy after every check in (waiting 15 minutes for a quiescent period) to stage and then manually push to live weekly (it used to be ad hoc, now it's every week, I hope soon it will be every day). To do this we've needed key unit tests (not full coverage by any means) and automated regression testing. ?And, most importantly, a change in developer mind set (we now know that as soon as we check in some code it's going to wind itself up on a live machine with live customers without any QA team to rely on).
>
> We're building a new solution with FS (on Windows) and ideally would like to adopt the same strategy. Ideally I don't want a "stable release" I want to always pull the latest version and run that in production.
>
> Right now for our FS project we've got the build process deploying to our stage environment (no live customers yet) using some Powershell scripts and Microsoft Web Deploy. We intend to build on this over the next 4 to 6 weeks. The plan is to pull from GIT, apply our own regression tests and if they pass push out to live (automatically).
>
> Currently we have plans to test only the functionality which we actually use - but depending on how things go I'm up for increasing the coverage and providing the results of our tests publicly. Perhaps this will provide a useful guide to the current state of the build and help provide what you're asking for (but I can only do this on Windows.)
>
> I'll keep you posted with our progress, (and I'll also be at the next ClueCon conference - if you're going then perhaps we can have a little brainstorm then.)
>
> Daniel
>
>
> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anton VG
> Sent: 07 May 2011 08:13
> To: freeswitch-dev at lists.freeswitch.org
> Subject: [Freeswitch-dev] is there any info abount which git tag/ Freeswitch release is stable?
>
> Guys, here is a little bit of critics,
>
> There is no info - what version of freeswitch should be used in production.
> the wiki pages
>
> http://wiki.freeswitch.org/wiki/Installation_Guide
> http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix
>
> says that latest GIT is "very" stable, but in 5 pulls i did, 2 times I immediately hit regressions:
> first time - non working DNS resolver in mod_sofia second - broken ESL outbound (as of this night)
>
> at the same time, it's mentioned that 1.0.6 is not the "latest stable"
> anymore, but in files, there is only 1.0.6
>
> so it would be very helpful if there is some way to find out, what is the latest GIT tag or release could be downloaded to be usable, it's just not possible while in development, to have GIT always stable. None can do it, so there really should be atleast "testing" tag/brunch, which is known working over serveral weeks (debian testing can be an example)
>
> Hope this piece of critics not insults anyone, FS is a great stuff, Regards, Anton.
>
> _______________________________________________
> 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
>
>
> The information transmitted is intended only for the person or entity
> to which it is addressed and may contain confidential and/or privileged
> material. Any review, retransmission, dissemination or other use of, or
> taking of any action in reliance upon, this information by persons or
> entities other than the intended recipient is prohibited. If you received
> this in error, please contact the sender and delete the material from
> your computer.
>
> _______________________________________________
> 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


From jmesquita at freeswitch.org  Sun May  8 04:23:02 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 7 May 2011 21:23:02 -0300
Subject: [Freeswitch-dev] is there any info abount which git tag/
 Freeswitch release is stable?
In-Reply-To: <BANLkTikNbA1hgon8OCnK0cD+55p4MXY8Og@mail.gmail.com>
References: <BANLkTinNo1JvE+Lf=nH8FyCRPD1DSEHoMA@mail.gmail.com>
	<2CC6B62FEE3A634CAD6B5D44FC9315E24FA6693F55@aeramsdc1.AERIANDI.LAN>
	<BANLkTikNbA1hgon8OCnK0cD+55p4MXY8Og@mail.gmail.com>
Message-ID: <BANLkTintPOGStozBV7NhCV5v9Op4SZUqSw@mail.gmail.com>

Anton, I consider myself a part of this community as well as an active
contributor. Whoever knows me personally or has spoken to me for more then 5
minutes is able to tell the passion that I hold for OSS and, of course,
FreeSWITCH.

Maybe this passion is too blindly but I believe that there are some
importants points being overseen here by most of those who write these types
of email I see pop up on the mailing list every now and again.

1. FreeSWITCH's core is developed mainly by 3 or 4 guys. We are talking
about less then *5 *people where one of them is the person who has made over
60% of the code in there.
2. FreeSWITCH does not generate direct profit for anyone who has been
actively contributing that I know of. We all add a twist to the product  and
that is where our value added lies on.
3. No one is working for no one, this is one of the things that makes it a
community instead of a company.

My point is that it can be a little offending when I see those comments that
are almost demanding and that doesn't have a proposition to go with.

I don't want to make any enemies and don't intend to offend either, but
instead of making a short response to this critic, I would think that we can
all benefit from more ppl helping on the project. Maybe the right question
could have been, who is the tester on the project and how can I help test to
get a stable branch?

I am really sorry if I seem aggressive, it is not the intention. I simply
took the opportunity of this email to write about my feelings towards emails
like those and not yours exclusively.

I hope my critics do not offend anyone. :-)

Regards,
Jo?o Mesquita



On Sat, May 7, 2011 at 6:57 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> current HEAD is a candidate for a stable branch.
>
> On Sat, May 7, 2011 at 4:47 PM, Daniel Bryars
> <daniel.bryars at aeriandi.com> wrote:
> > I don't think it's impossible - just difficult. My view is based on my
> experience where we "almost" continuously deploy our web sites to our live
> environment; we continuously deploy after every check in (waiting 15 minutes
> for a quiescent period) to stage and then manually push to live weekly (it
> used to be ad hoc, now it's every week, I hope soon it will be every day).
> To do this we've needed key unit tests (not full coverage by any means) and
> automated regression testing.  And, most importantly, a change in developer
> mind set (we now know that as soon as we check in some code it's going to
> wind itself up on a live machine with live customers without any QA team to
> rely on).
> >
> > We're building a new solution with FS (on Windows) and ideally would like
> to adopt the same strategy. Ideally I don't want a "stable release" I want
> to always pull the latest version and run that in production.
> >
> > Right now for our FS project we've got the build process deploying to our
> stage environment (no live customers yet) using some Powershell scripts and
> Microsoft Web Deploy. We intend to build on this over the next 4 to 6 weeks.
> The plan is to pull from GIT, apply our own regression tests and if they
> pass push out to live (automatically).
> >
> > Currently we have plans to test only the functionality which we actually
> use - but depending on how things go I'm up for increasing the coverage and
> providing the results of our tests publicly. Perhaps this will provide a
> useful guide to the current state of the build and help provide what you're
> asking for (but I can only do this on Windows.)
> >
> > I'll keep you posted with our progress, (and I'll also be at the next
> ClueCon conference - if you're going then perhaps we can have a little
> brainstorm then.)
> >
> > Daniel
> >
> >
> > -----Original Message-----
> > From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anton VG
> > Sent: 07 May 2011 08:13
> > To: freeswitch-dev at lists.freeswitch.org
> > Subject: [Freeswitch-dev] is there any info abount which git tag/
> Freeswitch release is stable?
> >
> > Guys, here is a little bit of critics,
> >
> > There is no info - what version of freeswitch should be used in
> production.
> > the wiki pages
> >
> > http://wiki.freeswitch.org/wiki/Installation_Guide
> > http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix
> >
> > says that latest GIT is "very" stable, but in 5 pulls i did, 2 times I
> immediately hit regressions:
> > first time - non working DNS resolver in mod_sofia second - broken ESL
> outbound (as of this night)
> >
> > at the same time, it's mentioned that 1.0.6 is not the "latest stable"
> > anymore, but in files, there is only 1.0.6
> >
> > so it would be very helpful if there is some way to find out, what is the
> latest GIT tag or release could be downloaded to be usable, it's just not
> possible while in development, to have GIT always stable. None can do it, so
> there really should be atleast "testing" tag/brunch, which is known working
> over serveral weeks (debian testing can be an example)
> >
> > Hope this piece of critics not insults anyone, FS is a great stuff,
> Regards, Anton.
> >
> > _______________________________________________
> > 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
> >
> >
> > The information transmitted is intended only for the person or entity
> > to which it is addressed and may contain confidential and/or privileged
> > material. Any review, retransmission, dissemination or other use of, or
> > taking of any action in reliance upon, this information by persons or
> > entities other than the intended recipient is prohibited. If you received
> > this in error, please contact the sender and delete the material from
> > your computer.
> >
> > _______________________________________________
> > 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
>
> _______________________________________________
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110507/b966abe8/attachment-0001.html 

From anton.vazir at gmail.com  Sun May  8 12:38:43 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sun, 8 May 2011 13:38:43 +0500
Subject: [Freeswitch-dev] is there any info abount which git tag/
 Freeswitch release is stable?
In-Reply-To: <BANLkTintPOGStozBV7NhCV5v9Op4SZUqSw@mail.gmail.com>
References: <BANLkTinNo1JvE+Lf=nH8FyCRPD1DSEHoMA@mail.gmail.com>
	<2CC6B62FEE3A634CAD6B5D44FC9315E24FA6693F55@aeramsdc1.AERIANDI.LAN>
	<BANLkTikNbA1hgon8OCnK0cD+55p4MXY8Og@mail.gmail.com>
	<BANLkTintPOGStozBV7NhCV5v9Op4SZUqSw@mail.gmail.com>
Message-ID: <BANLkTin5O-DtV0Be6ttTtVZK+2AYVpV8JQ@mail.gmail.com>

Jo?o,
your point is clear, and for sure it is very impressive, what work can
be done by that modest
number of developers. Don't get me wrong, there was no intention to
sound "demanding", just thoughts.
I myself watching FS progress from the moment Anthony have announced it.
FS do not have backing company as some other, this makes some things a
little more difficult
But I trust it would be in a great benefit for FS to gain bigger
community, and little more organization would not bother...
Though it requires men power, which seems is just not enough for the moment.

BTW, somehow strange, FS is not included to Debian distro, not even in
UNSTABLE, although, FS support a debian build out of GIT with no extra
work.
Just Debian is one of the most recognized distro for servers, and
having FS included would add to it's popularity quite sufficiently.
Is there no willing maintainer or some other reason?

BR,
Anton.

2011/5/8 Jo?o Mesquita <jmesquita at freeswitch.org>:
> Anton, I consider myself a part of this community as well as an active
> contributor. Whoever knows me personally or has spoken to me for more then 5
> minutes is able to tell the passion that I hold for OSS and, of course,
> FreeSWITCH.
> Maybe this passion is too blindly but I believe that there are some
> importants points being overseen here by most of those who write these types
> of email I see pop up on the mailing list every now and again.
> 1. FreeSWITCH's core is developed mainly by 3 or 4 guys. We are talking
> about less then 5 people where one of them is the person who has made over
> 60% of the code in there.
> 2. FreeSWITCH does not generate direct profit for anyone who has been
> actively contributing that I know of. We all add a twist to the product ?and
> that is where our value added lies on.
> 3. No one is working for no one, this is one of the things that makes it a
> community instead of a company.
> My point is that it can be a little offending when I see those comments that
> are almost demanding and that doesn't have a proposition to go with.
> I don't want to make any enemies and don't intend to offend either, but
> instead of making a short response to this critic, I would think that we can
> all benefit from more ppl helping on the project. Maybe the right question
> could have been, who is the tester on the project and how can I help test to
> get a stable branch?
>
> I am really sorry if I seem aggressive, it is not the intention. I simply
> took the opportunity of this email to write about my feelings towards emails
> like those and not yours exclusively.
>
> I hope my critics do not offend anyone. :-)
> Regards,
> Jo?o Mesquita
>
>
>
> On Sat, May 7, 2011 at 6:57 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> current HEAD is a candidate for a stable branch.
>>
>> On Sat, May 7, 2011 at 4:47 PM, Daniel Bryars
>> <daniel.bryars at aeriandi.com> wrote:
>> > I don't think it's impossible - just difficult. My view is based on my
>> > experience where we "almost" continuously deploy our web sites to our live
>> > environment; we continuously deploy after every check in (waiting 15 minutes
>> > for a quiescent period) to stage and then manually push to live weekly (it
>> > used to be ad hoc, now it's every week, I hope soon it will be every day).
>> > To do this we've needed key unit tests (not full coverage by any means) and
>> > automated regression testing. ?And, most importantly, a change in developer
>> > mind set (we now know that as soon as we check in some code it's going to
>> > wind itself up on a live machine with live customers without any QA team to
>> > rely on).
>> >
>> > We're building a new solution with FS (on Windows) and ideally would
>> > like to adopt the same strategy. Ideally I don't want a "stable release" I
>> > want to always pull the latest version and run that in production.
>> >
>> > Right now for our FS project we've got the build process deploying to
>> > our stage environment (no live customers yet) using some Powershell scripts
>> > and Microsoft Web Deploy. We intend to build on this over the next 4 to 6
>> > weeks. The plan is to pull from GIT, apply our own regression tests and if
>> > they pass push out to live (automatically).
>> >
>> > Currently we have plans to test only the functionality which we actually
>> > use - but depending on how things go I'm up for increasing the coverage and
>> > providing the results of our tests publicly. Perhaps this will provide a
>> > useful guide to the current state of the build and help provide what you're
>> > asking for (but I can only do this on Windows.)
>> >
>> > I'll keep you posted with our progress, (and I'll also be at the next
>> > ClueCon conference - if you're going then perhaps we can have a little
>> > brainstorm then.)
>> >
>> > Daniel
>> >
>> >
>> > -----Original Message-----
>> > From: freeswitch-dev-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anton VG
>> > Sent: 07 May 2011 08:13
>> > To: freeswitch-dev at lists.freeswitch.org
>> > Subject: [Freeswitch-dev] is there any info abount which git tag/
>> > Freeswitch release is stable?
>> >
>> > Guys, here is a little bit of critics,
>> >
>> > There is no info - what version of freeswitch should be used in
>> > production.
>> > the wiki pages
>> >
>> > http://wiki.freeswitch.org/wiki/Installation_Guide
>> > http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix
>> >
>> > says that latest GIT is "very" stable, but in 5 pulls i did, 2 times I
>> > immediately hit regressions:
>> > first time - non working DNS resolver in mod_sofia second - broken ESL
>> > outbound (as of this night)
>> >
>> > at the same time, it's mentioned that 1.0.6 is not the "latest stable"
>> > anymore, but in files, there is only 1.0.6
>> >
>> > so it would be very helpful if there is some way to find out, what is
>> > the latest GIT tag or release could be downloaded to be usable, it's just
>> > not possible while in development, to have GIT always stable. None can do
>> > it, so there really should be atleast "testing" tag/brunch, which is known
>> > working over serveral weeks (debian testing can be an example)
>> >
>> > Hope this piece of critics not insults anyone, FS is a great stuff,
>> > Regards, Anton.
>> >
>> > _______________________________________________
>> > 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
>> >
>> >
>> > The information transmitted is intended only for the person or entity
>> > to which it is addressed and may contain confidential and/or privileged
>> > material. Any review, retransmission, dissemination or other use of, or
>> > taking of any action in reliance upon, this information by persons or
>> > entities other than the intended recipient is prohibited. If you
>> > received
>> > this in error, please contact the sender and delete the material from
>> > your computer.
>> >
>> > _______________________________________________
>> > 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
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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 michal.bielicki at seventhsignal.de  Sun May  8 14:36:39 2011
From: michal.bielicki at seventhsignal.de (=?utf-8?B?TWljaGFsIEJpZWxpY2tp?=)
Date: Sun, 08 May 2011 12:36:39 +0200
Subject: [Freeswitch-dev] =?utf-8?q?Antw=2E=3A__is_there_any_info_abount_w?=
	=?utf-8?q?hich_git_tag/_Freeswitch_release_is_stable=3F?=
Message-ID: <mailman.4.1304851004.24552.freeswitch-dev@lists.freeswitch.org>

Debian dies Not accept packages without versions. Git releases are not accepted

Gesendet mit meinem HTC

----- Reply message -----
Von: "Anton VG" <anton.vazir at gmail.com>
An: <freeswitch-dev at lists.freeswitch.org>
Betreff: [Freeswitch-dev] is there any info abount which git tag/ Freeswitch release is stable?
Datum: So., Mai. 8, 2011 10:38


Jo?o,
your point is clear, and for sure it is very impressive, what work can
be done by that modest
number of developers. Don't get me wrong, there was no intention to
sound "demanding", just thoughts.
I myself watching FS progress from the moment Anthony have announced it.
FS do not have backing company as some other, this makes some things a
little more difficult
But I trust it would be in a great benefit for FS to gain bigger
community, and little more organization would not bother...
Though it requires men power, which seems is just not enough for the moment.

BTW, somehow strange, FS is not included to Debian distro, not even in
UNSTABLE, although, FS support a debian build out of GIT with no extra
work.
Just Debian is one of the most recognized distro for servers, and
having FS included would add to it's popularity quite sufficiently.
Is there no willing maintainer or some other reason?

BR,
Anton.

2011/5/8 Jo?o Mesquita <jmesquita at freeswitch.org>:
> Anton, I consider myself a part of this community as well as an active
> contributor. Whoever knows me personally or has spoken to me for more then 5
> minutes is able to tell the passion that I hold for OSS and, of course,
> FreeSWITCH.
> Maybe this passion is too blindly but I believe that there are some
> importants points being overseen here by most of those who write these types
> of email I see pop up on the mailing list every now and again.
> 1. FreeSWITCH's core is developed mainly by 3 or 4 guys. We are talking
> about less then 5 people where one of them is the person who has made over
> 60% of the code in there.
> 2. FreeSWITCH does not generate direct profit for anyone who has been
> actively contributing that I know of. We all add a twist to the product ?and
> that is where our value added lies on.
> 3. No one is working for no one, this is one of the things that makes it a
> community instead of a company.
> My point is that it can be a little offending when I see those comments that
> are almost demanding and that doesn't have a proposition to go with.
> I don't want to make any enemies and don't intend to offend either, but
> instead of making a short response to this critic, I would think that we can
> all benefit from more ppl helping on the project. Maybe the right question
> could have been, who is the tester on the project and how can I help test to
> get a stable branch?
>
> I am really sorry if I seem aggressive, it is not the intention. I simply
> took the opportunity of this email to write about my feelings towards emails
> like those and not yours exclusively.
>
> I hope my critics do not offend anyone. :-)
> Regards,
> Jo?o Mesquita
>
>
>
> On Sat, May 7, 2011 at 6:57 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> current HEAD is a candidate for a stable branch.
>>
>> On Sat, May 7, 2011 at 4:47 PM, Daniel Bryars
>> <daniel.bryars at aeriandi.com> wrote:
>> > I don't think it's impossible - just difficult. My view is based on my
>> > experience where we "almost" continuously deploy our web sites to our live
>> > environment; we continuously deploy after every check in (waiting 15 minutes
>> > for a quiescent period) to stage and then manually push to live weekly (it
>> > used to be ad hoc, now it's every week, I hope soon it will be every day).
>> > To do this we've needed key unit tests (not full coverage by any means) and
>> > automated regression testing. ?And, most importantly, a change in developer
>> > mind set (we now know that as soon as we check in some code it's going to
>> > wind itself up on a live machine with live customers without any QA team to
>> > rely on).
>> >
>> > We're building a new solution with FS (on Windows) and ideally would
>> > like to adopt the same strategy. Ideally I don't want a "stable release" I
>> > want to always pull the latest version and run that in production.
>> >
>> > Right now for our FS project we've got the build process deploying to
>> > our stage environment (no live customers yet) using some Powershell scripts
>> > and Microsoft Web Deploy. We intend to build on this over the next 4 to 6
>> > weeks. The plan is to pull from GIT, apply our own regression tests and if
>> > they pass push out to live (automatically).
>> >
>> > Currently we have plans to test only the functionality which we actually
>> > use - but depending on how things go I'm up for increasing the coverage and
>> > providing the results of our tests publicly. Perhaps this will provide a
>> > useful guide to the current state of the build and help provide what you're
>> > asking for (but I can only do this on Windows.)
>> >
>> > I'll keep you posted with our progress, (and I'll also be at the next
>> > ClueCon conference - if you're going then perhaps we can have a little
>> > brainstorm then.)
>> >
>> > Daniel
>> >
>> >
>> > -----Original Message-----
>> > From: freeswitch-dev-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anton VG
>> > Sent: 07 May 2011 08:13
>> > To: freeswitch-dev at lists.freeswitch.org
>> > Subject: [Freeswitch-dev] is there any info abount which git tag/
>> > Freeswitch release is stable?
>> >
>> > Guys, here is a little bit of critics,
>> >
>> > There is no info - what version of freeswitch should be used in
>> > production.
>> > the wiki pages
>> >
>> > http://wiki.freeswitch.org/wiki/Installation_Guide
>> > http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix
>> >
>> > says that latest GIT is "very" stable, but in 5 pulls i did, 2 times I
>> > immediately hit regressions:
>> > first time - non working DNS resolver in mod_sofia second - broken ESL
>> > outbound (as of this night)
>> >
>> > at the same time, it's mentioned that 1.0.6 is not the "latest stable"
>> > anymore, but in files, there is only 1.0.6
>> >
>> > so it would be very helpful if there is some way to find out, what is
>> > the latest GIT tag or release could be downloaded to be usable, it's just
>> > not possible while in development, to have GIT always stable. None can do
>> > it, so there really should be atleast "testing" tag/brunch, which is known
>> > working over serveral weeks (debian testing can be an example)
>> >
>> > Hope this piece of critics not insults anyone, FS is a great stuff,
>> > Regards, Anton.
>> >
>> > _______________________________________________
>> > 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
>> >
>> >
>> > The information transmitted is intended only for the person or entity
>> > to which it is addressed and may contain confidential and/or privileged
>> > material. Any review, retransmission, dissemination or other use of, or
>> > taking of any action in reliance upon, this information by persons or
>> > entities other than the intended recipient is prohibited. If you
>> > received
>> > this in error, please contact the sender and delete the material from
>> > your computer.
>> >
>> > _______________________________________________
>> > 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
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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
>
>

_______________________________________________
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110508/a66547c9/attachment.html 

From anton.vazir at gmail.com  Sun May  8 16:09:06 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sun, 8 May 2011 17:09:06 +0500
Subject: [Freeswitch-dev] Antw.: is there any info abount which git tag/
 Freeswitch release is stable?
In-Reply-To: <4dc67292.223fec0a.55e0.20e3SMTPIN_ADDED@mx.google.com>
References: <4dc67292.223fec0a.55e0.20e3SMTPIN_ADDED@mx.google.com>
Message-ID: <BANLkTi=Hme8-hJs7WJ-TTqy0n7-5o5yj9Q@mail.gmail.com>

hm, If I'm not wrong, there was a number of packages in debian, which
was updated from svn
previous stable had ffmpeg or mplayer from svn

in current stable i see libeina-svn-03 and 05

so it seems it accepts

2011/5/8 Michal Bielicki <michal.bielicki at seventhsignal.de>:
> Debian dies Not accept packages without versions. Git releases are not
> accepted
>
> Gesendet mit meinem HTC
>
> ----- Reply message -----
> Von: "Anton VG" <anton.vazir at gmail.com>
> An: <freeswitch-dev at lists.freeswitch.org>
> Betreff: [Freeswitch-dev] is there any info abount which git tag/ Freeswitch
> release is stable?
> Datum: So., Mai. 8, 2011 10:38
>
>
> Jo?o,
> your point is clear, and for sure it is very impressive, what work can
> be done by that modest
> number of developers. Don't get me wrong, there was no intention to
> sound "demanding", just thoughts.
> I myself watching FS progress from the moment Anthony have announced it.
> FS do not have backing company as some other, this makes some things a
> little more difficult
> But I trust it would be in a great benefit for FS to gain bigger
> community, and little more organization would not bother...
> Though it requires men power, which seems is just not enough for the moment.
>
> BTW, somehow strange, FS is not included to Debian distro, not even in
> UNSTABLE, although, FS support a debian build out of GIT with no extra
> work.
> Just Debian is one of the most recognized distro for servers, and
> having FS included would add to it's popularity quite sufficiently.
> Is there no willing maintainer or some other reason?
>
> BR,
> Anton.
>
> 2011/5/8 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> Anton, I consider myself a part of this community as well as an active
>> contributor. Whoever knows me personally or has spoken to me for more then
>> 5
>> minutes is able to tell the passion that I hold for OSS and, of course,
>> FreeSWITCH.
>> Maybe this passion is too blindly but I believe that there are some
>> importants points being overseen here by most of those who write these
>> types
>> of email I see pop up on the mailing list every now and again.
>> 1. FreeSWITCH's core is developed mainly by 3 or 4 guys. We are talking
>> about less then 5 people where one of them is the person who has made over
>> 60% of the code in there.
>> 2. FreeSWITCH does not generate direct profit for anyone who has been
>> actively contributing that I know of. We all add a twist to the product
>> ?and
>> that is where our value added lies on.
>> 3. No one is working for no one, this is one of the things that makes it a
>> community instead of a company.
>> My point is that it can be a little offending when I see those comments
>> that
>> are almost demanding and that doesn't have a proposition to go with.
>> I don't want to make any enemies and don't intend to offend either, but
>> instead of making a short response to this critic, I would think that we
>> can
>> all benefit from more ppl helping on the project. Maybe the right question
>> could have been, who is the tester on the project and how can I help test
>> to
>> get a stable branch?
>>
>> I am really sorry if I seem aggressive, it is not the intention. I simply
>> took the opportunity of this email to write about my feelings towards
>> emails
>> like those and not yours exclusively.
>>
>> I hope my critics do not offend anyone. :-)
>> Regards,
>> Jo?o Mesquita
>>
>>
>>
>> On Sat, May 7, 2011 at 6:57 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>>
>>> current HEAD is a candidate for a stable branch.
>>>
>>> On Sat, May 7, 2011 at 4:47 PM, Daniel Bryars
>>> <daniel.bryars at aeriandi.com> wrote:
>>> > I don't think it's impossible - just difficult. My view is based on my
>>> > experience where we "almost" continuously deploy our web sites to our
>>> > live
>>> > environment; we continuously deploy after every check in (waiting 15
>>> > minutes
>>> > for a quiescent period) to stage and then manually push to live weekly
>>> > (it
>>> > used to be ad hoc, now it's every week, I hope soon it will be every
>>> > day).
>>> > To do this we've needed key unit tests (not full coverage by any means)
>>> > and
>>> > automated regression testing. ?And, most importantly, a change in
>>> > developer
>>> > mind set (we now know that as soon as we check in some code it's going
>>> > to
>>> > wind itself up on a live machine with live customers without any QA
>>> > team to
>>> > rely on).
>>> >
>>> > We're building a new solution with FS (on Windows) and ideally would
>>> > like to adopt the same strategy. Ideally I don't want a "stable
>>> > release" I
>>> > want to always pull the latest version and run that in production.
>>> >
>>> > Right now for our FS project we've got the build process deploying to
>>> > our stage environment (no live customers yet) using some Powershell
>>> > scripts
>>> > and Microsoft Web Deploy. We intend to build on this over the next 4 to
>>> > 6
>>> > weeks. The plan is to pull from GIT, apply our own regression tests and
>>> > if
>>> > they pass push out to live (automatically).
>>> >
>>> > Currently we have plans to test only the functionality which we
>>> > actually
>>> > use - but depending on how things go I'm up for increasing the coverage
>>> > and
>>> > providing the results of our tests publicly. Perhaps this will provide
>>> > a
>>> > useful guide to the current state of the build and help provide what
>>> > you're
>>> > asking for (but I can only do this on Windows.)
>>> >
>>> > I'll keep you posted with our progress, (and I'll also be at the next
>>> > ClueCon conference - if you're going then perhaps we can have a little
>>> > brainstorm then.)
>>> >
>>> > Daniel
>>> >
>>> >
>>> > -----Original Message-----
>>> > From: freeswitch-dev-bounces at lists.freeswitch.org
>>> > [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anton
>>> > VG
>>> > Sent: 07 May 2011 08:13
>>> > To: freeswitch-dev at lists.freeswitch.org
>>> > Subject: [Freeswitch-dev] is there any info abount which git tag/
>>> > Freeswitch release is stable?
>>> >
>>> > Guys, here is a little bit of critics,
>>> >
>>> > There is no info - what version of freeswitch should be used in
>>> > production.
>>> > the wiki pages
>>> >
>>> > http://wiki.freeswitch.org/wiki/Installation_Guide
>>> > http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix
>>> >
>>> > says that latest GIT is "very" stable, but in 5 pulls i did, 2 times I
>>> > immediately hit regressions:
>>> > first time - non working DNS resolver in mod_sofia second - broken ESL
>>> > outbound (as of this night)
>>> >
>>> > at the same time, it's mentioned that 1.0.6 is not the "latest stable"
>>> > anymore, but in files, there is only 1.0.6
>>> >
>>> > so it would be very helpful if there is some way to find out, what is
>>> > the latest GIT tag or release could be downloaded to be usable, it's
>>> > just
>>> > not possible while in development, to have GIT always stable. None can
>>> > do
>>> > it, so there really should be atleast "testing" tag/brunch, which is
>>> > known
>>> > working over serveral weeks (debian testing can be an example)
>>> >
>>> > Hope this piece of critics not insults anyone, FS is a great stuff,
>>> > Regards, Anton.
>>> >
>>> > _______________________________________________
>>> > 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
>>> >
>>> >
>>> > The information transmitted is intended only for the person or entity
>>> > to which it is addressed and may contain confidential and/or privileged
>>> > material. Any review, retransmission, dissemination or other use of, or
>>> > taking of any action in reliance upon, this information by persons or
>>> > entities other than the intended recipient is prohibited. If you
>>> > received
>>> > this in error, please contact the sender and delete the material from
>>> > your computer.
>>> >
>>> > _______________________________________________
>>> > 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
>>>
>>> _______________________________________________
>>> 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
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>
> _______________________________________________
> 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 paul at cupis.co.uk  Sun May  8 21:04:13 2011
From: paul at cupis.co.uk (Paul Cupis)
Date: Sun, 08 May 2011 18:04:13 +0100
Subject: [Freeswitch-dev] Antw.: is there any info abount which git tag/
 Freeswitch release is stable?
In-Reply-To: <0MKaX9-1QHeGz10jf-001xMk@mx.kundenserver.de>
References: <0MKaX9-1QHeGz10jf-001xMk@mx.kundenserver.de>
Message-ID: <4DC6CD0D.6020108@cupis.co.uk>

On 08/05/11 11:36, Michal Bielicki wrote:
> Debian dies Not accept packages without versions. Git releases are not accepted

This is not true. There have been efforts to get FreeSWITCH packages
inside Debian, but I think the biggest issues are how to built it
against separately packaged libraries rather than using in-tree libraries.

http://bugs.debian.org/389591

Regards,


From gmaruzz at gmail.com  Sun May  8 21:17:32 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 8 May 2011 19:17:32 +0200
Subject: [Freeswitch-dev] skypopen.ko compile failed for 2.6.36.4
In-Reply-To: <BANLkTi=tDH7yHQQ1M4Xx_Ocy10KrxphEFA@mail.gmail.com>
References: <BANLkTikh2+DmFMiOYAwbZNF+9JnGQbpeBw@mail.gmail.com>
	<BANLkTi=fQ1DdfNB6cnw6HJ+Xm=GK55Bv+A@mail.gmail.com>
	<BANLkTi=PO8zYTKbhDgNq6kJ8wVeiTShexg@mail.gmail.com>
	<BANLkTikLiUqhhCyU8JX0u2OabEkcXD5wMQ@mail.gmail.com>
	<BANLkTi=tDH7yHQQ1M4Xx_Ocy10KrxphEFA@mail.gmail.com>
Message-ID: <BANLkTimSnAcuj0myQsONXgg=TTgp0Q6ESw@mail.gmail.com>

Hi Anton,

I would not like to add support for kernels that are not adopted by
the major distro FS is used on.

Adding that support would mean (for me) to compile, install, and use
those newer kernels, and maintain the driver under them (and
subsequents) and this is a burden I cannot afford.

If you are in the mood, please make a patch, test it very deeply, then
make it available somewhere (with a minimal of documentation, if
needed) and I happily add to the mod_skypopen wikipage a link to it
making it clear that's under your responsibility.

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From krice at freeswitch.org  Sun May  8 21:20:28 2011
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 08 May 2011 12:20:28 -0500
Subject: [Freeswitch-dev] Antw.: is there any info abount which git tag/
 Freeswitch release is stable?
In-Reply-To: <4DC6CD0D.6020108@cupis.co.uk>
Message-ID: <C9EC3B0C.16E5D%krice@freeswitch.org>

The Libraries that are included inline with freeswitch are done so for a
reason. Just because there is a similar or later version in debians package
tree does not mean it will work properly with FreeSWITCH. A perfect example
of this is libcurl... While included on many platforms its either broken, or
'fixed' in a way that changes the API and its not compatible w/ FreeSWITCH.


I guys over at sun put out a very good white paper on this subject a couple
of yars ago I wish I could find a link to include with this post.

K


On 5/8/11 12:04 PM, "Paul Cupis" <paul at cupis.co.uk> wrote:

> On 08/05/11 11:36, Michal Bielicki wrote:
>> Debian dies Not accept packages without versions. Git releases are not
>> accepted
> 
> This is not true. There have been efforts to get FreeSWITCH packages
> inside Debian, but I think the biggest issues are how to built it
> against separately packaged libraries rather than using in-tree libraries.
> 
> http://bugs.debian.org/389591
> 
> Regards,
> 
> _______________________________________________
> 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 michal.bielicki at seventhsignal.de  Sun May  8 23:04:14 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Sun, 08 May 2011 21:04:14 +0200
Subject: [Freeswitch-dev] Antw.: is there any info abount which git tag/
 Freeswitch release is stable?
In-Reply-To: <C9EC3B0C.16E5D%krice@freeswitch.org>
References: <C9EC3B0C.16E5D%krice@freeswitch.org>
Message-ID: <4DC6E92E.9010904@seventhsignal.de>


http://blogs.oracle.com/rvs/entry/what_does_dynamic_linking_and


Am 08.05.2011 19:20, schrieb Ken Rice:
> The Libraries that are included inline with freeswitch are done so for a
> reason. Just because there is a similar or later version in debians package
> tree does not mean it will work properly with FreeSWITCH. A perfect example
> of this is libcurl... While included on many platforms its either broken, or
> 'fixed' in a way that changes the API and its not compatible w/ FreeSWITCH.
>
>
> I guys over at sun put out a very good white paper on this subject a couple
> of yars ago I wish I could find a link to include with this post.
>
> K
>
>
> On 5/8/11 12:04 PM, "Paul Cupis" <paul at cupis.co.uk> wrote:
>
>> On 08/05/11 11:36, Michal Bielicki wrote:
>>> Debian dies Not accept packages without versions. Git releases are not
>>> accepted
>> This is not true. There have been efforts to get FreeSWITCH packages
>> inside Debian, but I think the biggest issues are how to built it
>> against separately packaged libraries rather than using in-tree libraries.
>>
>> http://bugs.debian.org/389591
>>
>> Regards,
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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 krice at freeswitch.org  Mon May  9 01:09:23 2011
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 08 May 2011 16:09:23 -0500
Subject: [Freeswitch-dev] Antw.: is there any info abount which git tag/
 Freeswitch release is stable?
In-Reply-To: <4DC6E92E.9010904@seventhsignal.de>
Message-ID: <C9EC70B3.16E69%krice@freeswitch.org>

Don't forget Tony's response to Roman Shaposhnik's blog entry also... Man I
dunno how I didn't find this on FreeSWITCH.org

http://www.freeswitch.org/node/56

K


On 5/8/11 2:04 PM, "Michal Bielicki" <michal.bielicki at seventhsignal.de>
wrote:

> 
> http://blogs.oracle.com/rvs/entry/what_does_dynamic_linking_and
> 
> 
> Am 08.05.2011 19:20, schrieb Ken Rice:
>> The Libraries that are included inline with freeswitch are done so for a
>> reason. Just because there is a similar or later version in debians package
>> tree does not mean it will work properly with FreeSWITCH. A perfect example
>> of this is libcurl... While included on many platforms its either broken, or
>> 'fixed' in a way that changes the API and its not compatible w/ FreeSWITCH.
>> 
>> 
>> I guys over at sun put out a very good white paper on this subject a couple
>> of yars ago I wish I could find a link to include with this post.
>> 
>> K
>> 
>> 
>> On 5/8/11 12:04 PM, "Paul Cupis" <paul at cupis.co.uk> wrote:
>> 
>>> On 08/05/11 11:36, Michal Bielicki wrote:
>>>> Debian dies Not accept packages without versions. Git releases are not
>>>> accepted
>>> This is not true. There have been efforts to get FreeSWITCH packages
>>> inside Debian, but I think the biggest issues are how to built it
>>> against separately packaged libraries rather than using in-tree libraries.
>>> 
>>> http://bugs.debian.org/389591
>>> 
>>> Regards,
>>> 
>>> _______________________________________________
>>> 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
>> 
>> 
>> _______________________________________________
>> 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
> 
> 
> _______________________________________________
> 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 michal.bielicki at seventhsignal.de  Mon May  9 01:37:27 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Sun, 08 May 2011 23:37:27 +0200
Subject: [Freeswitch-dev] Antw.: is there any info abount which git tag/
 Freeswitch release is stable?
In-Reply-To: <C9EC70B3.16E69%krice@freeswitch.org>
References: <C9EC70B3.16E69%krice@freeswitch.org>
Message-ID: <4DC70D17.8010302@seventhsignal.de>

No chance with all the distro package bla bla bla to ever forget it :D

Am 08.05.2011 23:09, schrieb Ken Rice:
> Don't forget Tony's response to Roman Shaposhnik's blog entry also... Man I
> dunno how I didn't find this on FreeSWITCH.org
>
> http://www.freeswitch.org/node/56
>
> K
>
>
> On 5/8/11 2:04 PM, "Michal Bielicki" <michal.bielicki at seventhsignal.de>
> wrote:
>
>> http://blogs.oracle.com/rvs/entry/what_does_dynamic_linking_and
>>
>>
>> Am 08.05.2011 19:20, schrieb Ken Rice:
>>> The Libraries that are included inline with freeswitch are done so for a
>>> reason. Just because there is a similar or later version in debians package
>>> tree does not mean it will work properly with FreeSWITCH. A perfect example
>>> of this is libcurl... While included on many platforms its either broken, or
>>> 'fixed' in a way that changes the API and its not compatible w/ FreeSWITCH.
>>>
>>>
>>> I guys over at sun put out a very good white paper on this subject a couple
>>> of yars ago I wish I could find a link to include with this post.
>>>
>>> K
>>>
>>>
>>> On 5/8/11 12:04 PM, "Paul Cupis" <paul at cupis.co.uk> wrote:
>>>
>>>> On 08/05/11 11:36, Michal Bielicki wrote:
>>>>> Debian dies Not accept packages without versions. Git releases are not
>>>>> accepted
>>>> This is not true. There have been efforts to get FreeSWITCH packages
>>>> inside Debian, but I think the biggest issues are how to built it
>>>> against separately packaged libraries rather than using in-tree libraries.
>>>>
>>>> http://bugs.debian.org/389591
>>>>
>>>> Regards,
>>>>
>>>> _______________________________________________
>>>> 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
>>>
>>> _______________________________________________
>>> 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
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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 anton.vazir at gmail.com  Mon May  9 11:06:44 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Mon, 9 May 2011 12:06:44 +0500
Subject: [Freeswitch-dev] skypopen.ko compile failed for 2.6.36.4
In-Reply-To: <BANLkTimSnAcuj0myQsONXgg=TTgp0Q6ESw@mail.gmail.com>
References: <BANLkTikh2+DmFMiOYAwbZNF+9JnGQbpeBw@mail.gmail.com>
	<BANLkTi=fQ1DdfNB6cnw6HJ+Xm=GK55Bv+A@mail.gmail.com>
	<BANLkTi=PO8zYTKbhDgNq6kJ8wVeiTShexg@mail.gmail.com>
	<BANLkTikLiUqhhCyU8JX0u2OabEkcXD5wMQ@mail.gmail.com>
	<BANLkTi=tDH7yHQQ1M4Xx_Ocy10KrxphEFA@mail.gmail.com>
	<BANLkTimSnAcuj0myQsONXgg=TTgp0Q6ESw@mail.gmail.com>
Message-ID: <BANLkTikTAcuvgHB4Z=e9HOXNkcv9ggOWpA@mail.gmail.com>

;) Understandable.
Those who needs to run that under new kernel will find this mail
thread sufficient to fix it.
Though, it have a little use, since my test showed that there is some
problem with timing, and sound becomes itchy -
right now I have no time to look for solution,
maybe it's because of getting rid of BKL, maybe not.
Personally I've chosen to stick with the ALSA version with patched
dummy.c for the time being, that gives no trouble with newer kernels.

2011/5/8 Giovanni Maruzzelli <gmaruzz at gmail.com>:
> Hi Anton,
>
> I would not like to add support for kernels that are not adopted by
> the major distro FS is used on.
>
> Adding that support would mean (for me) to compile, install, and use
> those newer kernels, and maintain the driver under them (and
> subsequents) and this is a burden I cannot afford.
>
> If you are in the mood, please make a patch, test it very deeply, then
> make it available somewhere (with a minimal of documentation, if
> needed) and I happily add to the mod_skypopen wikipage a link to it
> making it clear that's under your responsibility.
>
> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>


From msc at freeswitch.org  Mon May  9 20:17:56 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 9 May 2011 09:17:56 -0700
Subject: [Freeswitch-dev] is there any info abount which git tag/
 Freeswitch release is stable?
In-Reply-To: <BANLkTin5O-DtV0Be6ttTtVZK+2AYVpV8JQ@mail.gmail.com>
References: <BANLkTinNo1JvE+Lf=nH8FyCRPD1DSEHoMA@mail.gmail.com>
	<2CC6B62FEE3A634CAD6B5D44FC9315E24FA6693F55@aeramsdc1.AERIANDI.LAN>
	<BANLkTikNbA1hgon8OCnK0cD+55p4MXY8Og@mail.gmail.com>
	<BANLkTintPOGStozBV7NhCV5v9Op4SZUqSw@mail.gmail.com>
	<BANLkTin5O-DtV0Be6ttTtVZK+2AYVpV8JQ@mail.gmail.com>
Message-ID: <BANLkTinmrrLbpaHf+YQyrUDTHqfg585cew@mail.gmail.com>

>
>
> BTW, somehow strange, FS is not included to Debian distro, not even in
> UNSTABLE, although, FS support a debian build out of GIT with no extra
> work.
> Just Debian is one of the most recognized distro for servers, and
> having FS included would add to it's popularity quite sufficiently.
> Is there no willing maintainer or some other reason?
>
> Bingo! We need help for this kind of thing. If you or someone you know is
in a position to be the maintainer then that would be a great help for the
project.

Thanks,
MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110509/0b47112b/attachment.html 

From singhujjwal at gmail.com  Tue May 10 11:08:04 2011
From: singhujjwal at gmail.com (Ujjwal SIngh)
Date: Tue, 10 May 2011 12:38:04 +0530
Subject: [Freeswitch-dev] Noise heard when a user holds a conference in
 SRTP mode
In-Reply-To: <BANLkTin_mvk0MCR0n+gjYKQ3PnqTBUKuXg@mail.gmail.com>
References: <BANLkTinp3JQkJw3ze4_dEHOT+BzDsEMPAQ@mail.gmail.com>
	<FE25D14D-D8C2-495E-A283-E4E2B8D0A150@freeswitch.org>
	<BANLkTikKvA8YaH676JsErZ3g6dcQfuNmNA@mail.gmail.com>
	<F64CEF9B-2292-45C5-8D3A-9AC2230D30E9@freeswitch.org>
	<BANLkTin_mvk0MCR0n+gjYKQ3PnqTBUKuXg@mail.gmail.com>
Message-ID: <BANLkTik4rwEOsrZmcG=a5YrtnW5At4+QEQ@mail.gmail.com>

Hi Brian,
>
>
> I feel that the key exchange is fine in the way the hold is taking place.
>
>                                          (FreeSwitch)
> A                                CONF                IVR
>   |ReINVITE F1(w/o SDP)|                         |
>   |------------------------------->|                         |
>
>   |  200 (with SDP) (K1)  |                         |
>   |<-------------------------------|                         |
>   |     INVITE(K1) F3       |                         |
>   |---------------------------------|----------------------->|
>   |     200 (K2)  F4          |                         |
>
> |<-------------------------------|-------------------------|
>   |     ACK  F5               |                          |
>   |---------------------------------|----------------------->|
>   |     ACK (SDP)(K2) F6 |                         |
>   |-------------------------------->|                        |
>   |                                  |                        |
>   |                                  |                        |
>   |                                  |                        |
>   |                                  |                        |
>                                   (SRTP key K1)      (SRTP key K2)
>
> As you can see in the above call flow :
>
> The local key of conference generated in the offer(200 OK) of the ReINVITE
> is used as the local key by the endpoint for the INVITE sent to MoH, and the
> key generated by the MoH in the answer(200 OK) is used as the key in the
> answer of the ReINVITE(ACK).
>
> I have removed the wireshark trace, the mail wasn't getting through :'(
>
>
> F1 :  (ReINVITE w/o SDP)
>
> F2 :  (200 OK with SDP)        Key K1
>
> F3 :  (INVITE with SDP)          Key K1
>
> F4 :  (200 OK with SDP)        Key K2
>
> F5 :  ACK
>
> F6:  (ACK with SDP)              Key K2
> Please let me know if I am missing something here.
>
> Regards,
> Ujjwal
>
> On Thu, Apr 21, 2011 at 7:38 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Yah SRTP might NEVER work with that method of hold because now the MOH
>> server and the endpoint have to exchange SRTP keys when you refer to them
>> and they probably are NOT doing that right now which is why you get white
>> noise.
>>
>> /b
>>
>> On Apr 21, 2011, at 6:26 AM, Ujjwal SIngh wrote:
>>
>> Yes Brian its a white noise, but the hold method works perfectly fine when
>> used with RTP, the hold is
>>
>> implemented  according to the draft
>>
>> http://datatracker.ietf.org/doc/draft-worley-service-example/?include_text=1
>>
>>
>> Regards,
>> Ujjwal
>>
>>
>>
>>
>> On Thu, Apr 21, 2011 at 3:13 AM, Brian West <brian at freeswitch.org> wrote:
>>
>> I'm going to guess your device method of placing you on hold is wrong.
>>  And
>>
>> possibly doesn't encrypt the data or doesn't signal a key change... is it
>>
>> white noise?
>>
>>
>> /b
>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110510/54d611f9/attachment.html 

From math.parent at gmail.com  Wed May 11 00:19:22 2011
From: math.parent at gmail.com (Mathieu Parent)
Date: Tue, 10 May 2011 22:19:22 +0200
Subject: [Freeswitch-dev] Implementing Forward-{All, Busy, NoAnswer},
 ring policies and DND in FS
In-Reply-To: <BANLkTinV7ww4hjs-XXUxffxRk0vMDf4Aqg@mail.gmail.com>
References: <BANLkTinV7ww4hjs-XXUxffxRk0vMDf4Aqg@mail.gmail.com>
Message-ID: <BANLkTim3yoLC-c4q=VZKfqO-ztzpHL645g@mail.gmail.com>

Hi,

Back on this topic.

2011/4/14 Mathieu Parent <math.parent at gmail.com>:
> Hello,
>
> *Context*:
> ---------
>
> I want to implement the following in mod_skinny (FS-3048 and FS-3047)
> in a way shared with other endpoints modules:
> - call forwarding[cfw],
> - DND and ring policies

I will focus on cfw first: as Marc Olivier suggested, this is more
common in SIP.

>
> *Call forwarding*:
> ---------
> There are three main kinds of forwarding:
> - forward all (immediately)
> - forward when line is busy
> - forward when no answer
>
> We should also store for each the state of the forwarding (enabled or not).
>
> I think a reasonable default here would be:
> - to forward "no answer" to voicemail (and enabled)
> - to forward busy to voicemail (but disabled)
> - to forward all to "" (and disabled)
>
> We also should define what a busy line is, especially on shared lines
> (perhaps define a busy threshold, defaulting to 1).
>
> Call forwarding should be implemented in dialplan, but endpoint
> modules should know when a call is forwarded (the bridge event is
> probably sufficient).
>
(snip)
>
> *Proposal*:
> ---------
>
> I propose to use hashes to store the various data:
>
> ? ? ? ?<action application="hash" data="insert/${realm}/${extension}/${data}"/>
>
> Where ${realm} is one of:
> - ${domain_name}-forward-{all,busy,noanswer}-destination: a destination number
> - ${domain_name}-forward-{all,busy,noanswer}-status: true/false
> - ${domain_name}-busy-threshold: [0-9]+
(...)
>
> The dialplan should be enhanced to manage forwarding.
> The endpoint modules should be enhanced to manage forwarding (notify
> that the call has been forwarded) and ring policies.
>
> Does this makes sense? How can we improve it?

I have doubt about mod_hash because it doesn't propagate hashes on
remote servers (it does for limits only). Should I use mod_db instead?

>
> *Next steps*:
> ---------
> - proposal discussion <-- we are still here ;-)
> - forward-* dialplan implementation (I need help here, I am not an
> dialplan expert)
> - mod_skinny forward-* implementation FS-3048
> - mod_skinny ring policies implementation FS-3047
> - other endpoints implementation
>
> Regards
>
> Mathieu
>
> [cfw]: http://en.wikipedia.org/wiki/Call_forwarding
> [dnd]: http://en.wikipedia.org/wiki/Do_Not_Disturb_%28telecommunications%29
>



-- 
Mathieu


From rupa at rupa.com  Wed May 11 00:46:33 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 10 May 2011 15:46:33 -0500
Subject: [Freeswitch-dev] Implementing Forward-{All, Busy, NoAnswer},
 ring policies and DND in FS
In-Reply-To: <BANLkTim3yoLC-c4q=VZKfqO-ztzpHL645g@mail.gmail.com>
References: <BANLkTinV7ww4hjs-XXUxffxRk0vMDf4Aqg@mail.gmail.com>
	<BANLkTim3yoLC-c4q=VZKfqO-ztzpHL645g@mail.gmail.com>
Message-ID: <BANLkTinhMzGO=btnm2dVtMN97yD4wqN+oA@mail.gmail.com>

On Tue, May 10, 2011 at 3:19 PM, Mathieu Parent <math.parent at gmail.com> wrote:
> I have doubt about mod_hash because it doesn't propagate hashes on
> remote servers (it does for limits only). Should I use mod_db instead?
>

+1

-- 
-Rupa


From msc at freeswitch.org  Wed May 11 19:21:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 11 May 2011 08:21:02 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <BANLkTik2xP+7h56TJZYV5oFD8wx6wk+cJg@mail.gmail.com>

Hello gang!

We have a light agenda again today, however I still want everyone to call in
so that we can discuss a few topics. The agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_05_11

Be sure to add any items that you want to discuss today.

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110511/5a729e3f/attachment.html 

From tayeb.meftah at gmail.com  Tue May  3 15:44:38 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 03 May 2011 13:44:38 +0200
Subject: [Freeswitch-dev] FreeSWITCH Cookbook: resources
In-Reply-To: <BANLkTi=sJdtc-00KbxAzomtiyq_8+_K6hw@mail.gmail.com>
References: <BANLkTi=sJdtc-00KbxAzomtiyq_8+_K6hw@mail.gmail.com>
Message-ID: <4DBFEAA6.8010300@gmail.com>

ready to ofer you a quad core Del Server with 4GB of ram and a did of 
your choice
please contact me this evening.
anything else needed?
thank you.
On 02/05/2011 21:04, Michael Collins wrote:
> Hello all!
>
> The response to my request for reviewers is overwhelming - thank you 
> so much! We have more than enough reviewers. However, any of you who 
> wish to read the draft and offer feedback are still welcome to do so. 
> You just won't be "official" Packt reviewers.
>
> I'd like to focus attention on testing resources. For those of you who 
> have data center resources and such I would like to ask for your 
> assistance. It would be helpful to have a spare server (low power == 
> okay, cuz it's only for a few simultaneous calls max) with a DID & 
> public IP address. This will help us test various scenarios and make 
> sure that we can do box2box and NAT scenarios.
>
> Please contact me off list if you have any servers, DIDs, etc. that 
> could be used for this purpose.
>
> Thanks again to everyone who has volunteered to help! I would much 
> rather have to sift through dozens of emails from people who want to 
> help than have to keep begging and have no one step up. ;)  You guys 
> are awesome - keep up the good work.
>
> -MC
>
>
> _______________________________________________
> 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
>    


-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110503/d0269157/attachment.html 

From tomp at tomp.co.uk  Mon May  9 22:37:32 2011
From: tomp at tomp.co.uk (Tom Parrott)
Date: Mon, 09 May 2011 19:37:32 +0100
Subject: [Freeswitch-dev] is there any info abount which git tag/ Freeswitch
 release is stable?
Message-ID: <4DC8346C.8000803@tomp.co.uk>

Hi,

I work for a company selling call tracking services, and we have 
recently put into production a 3 node Freeswitch cluster to use features 
such as call recording and call announce.

We are using the event socket feature to control freeswitch from our 
custom written service.

I have found Freeswitch a pleasure to use, and much easier to get along 
with than Asterisk.

However one thing that has concerned me is the lack of regular 
point-in-time releases.

When putting services in production, we use RPMs to allow us to easily 
upgrade (and if need be rollback) software versions.

The 1.07 version available at http://latest.freeswitch.org seems to be 
rebuilt each night, does this mean that a 1.07 built yesterday is 
different from one built today?

If I can offer any assistance with the automation of point-in-time 
tagged releases I would be happy to help, as I feel this would make 
Freeswitch more attractive to commercial operations as it would easily 
fit into well defined release processes.

Thanks
Tom




From tomp at tomp.co.uk  Tue May 10 22:39:00 2011
From: tomp at tomp.co.uk (Tom Parrott)
Date: Tue, 10 May 2011 19:39:00 +0100
Subject: [Freeswitch-dev] is there any info abount which git tag/ Freeswitch
	release is stable?
Message-ID: <4DC98644.3030306@tomp.co.uk>

Hi,

I work for a company selling call tracking services, and we have 
recently put into production a 3 node Freeswitch cluster to use features 
such as call recording and call announce.

We are using the event socket feature to control freeswitch from our 
custom written service.

I have found Freeswitch a pleasure to use, and much easier to get along 
with than Asterisk.

However one thing that has concerned me is the lack of regular 
point-in-time releases.

When putting services in production, we use RPMs to allow us to easily 
upgrade (and if need be rollback) software versions.

The 1.07 version available at http://latest.freeswitch.org seems to be 
rebuilt each night, does this mean that a 1.07 built yesterday is 
different from one built today?

If I can offer any assistance with the automation of point-in-time 
tagged releases I would be happy to help, as I feel this would make 
Freeswitch more attractive to commercial operations as it would easily 
fit into well defined release processes.

Thanks
Tom


From prashant.lamba at gmail.com  Thu May 12 09:41:31 2011
From: prashant.lamba at gmail.com (Prashant Lamba)
Date: Thu, 12 May 2011 11:11:31 +0530
Subject: [Freeswitch-dev] using freeswich as for SIP signalling and
 programming an external HW.
In-Reply-To: <BANLkTim4ntPv0CbM62UF5+JL-1RzbQpb7Q@mail.gmail.com>
References: <BANLkTim4ntPv0CbM62UF5+JL-1RzbQpb7Q@mail.gmail.com>
Message-ID: <BANLkTikJdFPeyHqOGjBq7Sd4p0C=53_vSQ@mail.gmail.com>

On Wed, Apr 27, 2011 at 10:05 AM, Narendra Sirugudi <nsirugudi at gmail.com>wrote:

> Hi All,
>
> I want to use freeswich for SIP signalling and program an external HW for
> RTP processing.
>
> Does freeswitch provide any mechanism/hooks to program an external HW for
> RTP processing ?
>
> thanks,
> --kumar
>
> _______________________________________________
> 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
>
>

I do believe you need to use Freeswitch in Media Bypass mode. We did the
same and it works like a bomb!
Ofcourse you need to find the way to divert the audio to your HW for RTP
capture. Enjoy.

Prashant
--
Phonologies (India)

To save our tigers, save their habitat. Think before you print this email. <
www.saveourtigers.com>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110512/89b9218d/attachment.html 

From steveayre at gmail.com  Thu May 12 11:45:48 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 12 May 2011 08:45:48 +0100
Subject: [Freeswitch-dev] is there any info abount which git tag/
 Freeswitch release is stable?
In-Reply-To: <4DC98644.3030306@tomp.co.uk>
References: <4DC98644.3030306@tomp.co.uk>
Message-ID: <BANLkTin0J8jpeNe0NOfU88ktuTdS0ng-9w@mail.gmail.com>

>
> The 1.07 version available at http://latest.freeswitch.org seems to be rebuilt
> each night, does this mean that a 1.07 built yesterday is different from
> one built today?


Yes. Think of it as a release candidate for 1.0.8

The git head is pretty much the most stable. Development tends to move too
fast to make more regular 'stable' releases. You should of course test
releases work as expected before putting them into production.

If you find a Git version that you're comfortable with, then you can roll
out that version across your whole network even if there's a newer Git head
by running:
> git clone git://git.freeswitch.org/freeswitch.git
> cd freeswitch.git
> git checkout GIT_HASH

I believe the developers have been discussing making a 1.0 stable branch
soon with only bugfix and stability improvements and moving new development
onto a new 1.2 branch. If that happens then perhaps you'll be more
comfortable with that branch. Not sure when that's planned for though.

-Steve


On 10 May 2011 19:39, Tom Parrott <tomp at tomp.co.uk> wrote:

> Hi,
>
> I work for a company selling call tracking services, and we have
> recently put into production a 3 node Freeswitch cluster to use features
> such as call recording and call announce.
>
> We are using the event socket feature to control freeswitch from our
> custom written service.
>
> I have found Freeswitch a pleasure to use, and much easier to get along
> with than Asterisk.
>
> However one thing that has concerned me is the lack of regular
> point-in-time releases.
>
> When putting services in production, we use RPMs to allow us to easily
> upgrade (and if need be rollback) software versions.
>
> The 1.07 version available at http://latest.freeswitch.org seems to be
> rebuilt each night, does this mean that a 1.07 built yesterday is
> different from one built today?
>
> If I can offer any assistance with the automation of point-in-time
> tagged releases I would be happy to help, as I feel this would make
> Freeswitch more attractive to commercial operations as it would easily
> fit into well defined release processes.
>
> Thanks
> Tom
>
> _______________________________________________
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110512/4a3c1dd3/attachment.html 

From steveayre at gmail.com  Thu May 12 11:49:56 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 12 May 2011 08:49:56 +0100
Subject: [Freeswitch-dev] is there any info abount which git tag/
 Freeswitch release is stable?
In-Reply-To: <BANLkTin0J8jpeNe0NOfU88ktuTdS0ng-9w@mail.gmail.com>
References: <4DC98644.3030306@tomp.co.uk>
	<BANLkTin0J8jpeNe0NOfU88ktuTdS0ng-9w@mail.gmail.com>
Message-ID: <BANLkTikByQjFh0c8J+CuB3K6B7MBSACf5w@mail.gmail.com>

>
> I believe the developers have been discussing making a 1.0 stable branch
> soon with only bugfix and stability improvements and moving new development
> onto a new 1.2 branch. If that happens then perhaps you'll be more
> comfortable with that branch. Not sure when that's planned for though.


The issue by the way is with developers. While there's a lot of developers
and plenty of people using FS in the wild testing and making bug reports on
Jira and plenty of people volunteering support on the list and IRC,
something like 90% of the code is still written by half a dozen people. It's
damn good code though. :) That's too much to make it easy for them to both
maintain a stable branch and work on the newer branch, so up to this point
it has been better to focus on making FS complete, feature rich and stable
all in the same branch. Once the branches diverge, someone is going to have
to both handle bug reports for 2 different codebases and also look at all
1.2 changes to decide whether any are work backporting to 1.0 which won't
always be trivial if the code diverges significantly.

-Steve


On 12 May 2011 08:45, Steven Ayre <steveayre at gmail.com> wrote:

>  The 1.07 version available at http://latest.freeswitch.org seems to be rebuilt
>> each night, does this mean that a 1.07 built yesterday is different from
>> one built today?
>
>
> Yes. Think of it as a release candidate for 1.0.8
>
> The git head is pretty much the most stable. Development tends to move too
> fast to make more regular 'stable' releases. You should of course test
> releases work as expected before putting them into production.
>
> If you find a Git version that you're comfortable with, then you can roll
> out that version across your whole network even if there's a newer Git head
> by running:
> > git clone git://git.freeswitch.org/freeswitch.git
> > cd freeswitch.git
> > git checkout GIT_HASH
>
> I believe the developers have been discussing making a 1.0 stable branch
> soon with only bugfix and stability improvements and moving new development
> onto a new 1.2 branch. If that happens then perhaps you'll be more
> comfortable with that branch. Not sure when that's planned for though.
>
> -Steve
>
>
> On 10 May 2011 19:39, Tom Parrott <tomp at tomp.co.uk> wrote:
>
>> Hi,
>>
>> I work for a company selling call tracking services, and we have
>> recently put into production a 3 node Freeswitch cluster to use features
>> such as call recording and call announce.
>>
>> We are using the event socket feature to control freeswitch from our
>> custom written service.
>>
>> I have found Freeswitch a pleasure to use, and much easier to get along
>> with than Asterisk.
>>
>> However one thing that has concerned me is the lack of regular
>> point-in-time releases.
>>
>> When putting services in production, we use RPMs to allow us to easily
>> upgrade (and if need be rollback) software versions.
>>
>> The 1.07 version available at http://latest.freeswitch.org seems to be
>> rebuilt each night, does this mean that a 1.07 built yesterday is
>> different from one built today?
>>
>> If I can offer any assistance with the automation of point-in-time
>> tagged releases I would be happy to help, as I feel this would make
>> Freeswitch more attractive to commercial operations as it would easily
>> fit into well defined release processes.
>>
>> Thanks
>> Tom
>>
>> _______________________________________________
>> 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
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110512/49d231c8/attachment-0001.html 

From tomp at tomp.co.uk  Thu May 12 12:02:56 2011
From: tomp at tomp.co.uk (Tom Parrott)
Date: Thu, 12 May 2011 09:02:56 +0100
Subject: [Freeswitch-dev] is there any info abount which git tag/
 Freeswitch release is stable?
In-Reply-To: <mailman.8860.1305186658.24555.freeswitch-dev@lists.freeswitch.org>
References: <mailman.8860.1305186658.24555.freeswitch-dev@lists.freeswitch.org>
Message-ID: <83e5d123d690d463ea41168ab73dfdde.squirrel@my.tomp.co.uk>

Hi,

Does the file at http://latest.freeswitch.org change each day to represent
the current state of HEAD?

If so I could version that as 1.0.7-yyyymmdd to make it more suitable for
RPMs.

Thanks
Tom

>>
>> I believe the developers have been discussing making a 1.0 stable branch
>> soon with only bugfix and stability improvements and moving new
>> development
>> onto a new 1.2 branch. If that happens then perhaps you'll be more
>> comfortable with that branch. Not sure when that's planned for though.
>
>
> The issue by the way is with developers. While there's a lot of developers
> and plenty of people using FS in the wild testing and making bug reports
> on
> Jira and plenty of people volunteering support on the list and IRC,
> something like 90% of the code is still written by half a dozen people.
> It's
> damn good code though. :) That's too much to make it easy for them to both
> maintain a stable branch and work on the newer branch, so up to this point
> it has been better to focus on making FS complete, feature rich and stable
> all in the same branch. Once the branches diverge, someone is going to
> have
> to both handle bug reports for 2 different codebases and also look at all
> 1.2 changes to decide whether any are work backporting to 1.0 which won't
> always be trivial if the code diverges significantly.
>
> -Steve
>
>
> On 12 May 2011 08:45, Steven Ayre <steveayre at gmail.com> wrote:
>
>>  The 1.07 version available at http://latest.freeswitch.org seems to be
>> rebuilt
>>> each night, does this mean that a 1.07 built yesterday is different
>>> from
>>> one built today?
>>
>>
>> Yes. Think of it as a release candidate for 1.0.8
>>
>> The git head is pretty much the most stable. Development tends to move
>> too
>> fast to make more regular 'stable' releases. You should of course test
>> releases work as expected before putting them into production.
>>
>> If you find a Git version that you're comfortable with, then you can
>> roll
>> out that version across your whole network even if there's a newer Git
>> head
>> by running:
>> > git clone git://git.freeswitch.org/freeswitch.git
>> > cd freeswitch.git
>> > git checkout GIT_HASH
>>
>> I believe the developers have been discussing making a 1.0 stable branch
>> soon with only bugfix and stability improvements and moving new
>> development
>> onto a new 1.2 branch. If that happens then perhaps you'll be more
>> comfortable with that branch. Not sure when that's planned for though.
>>
>> -Steve
>>
>>
>> On 10 May 2011 19:39, Tom Parrott <tomp at tomp.co.uk> wrote:
>>
>>> Hi,
>>>
>>> I work for a company selling call tracking services, and we have
>>> recently put into production a 3 node Freeswitch cluster to use
>>> features
>>> such as call recording and call announce.
>>>
>>> We are using the event socket feature to control freeswitch from our
>>> custom written service.
>>>
>>> I have found Freeswitch a pleasure to use, and much easier to get along
>>> with than Asterisk.
>>>
>>> However one thing that has concerned me is the lack of regular
>>> point-in-time releases.
>>>
>>> When putting services in production, we use RPMs to allow us to easily
>>> upgrade (and if need be rollback) software versions.
>>>
>>> The 1.07 version available at http://latest.freeswitch.org seems to be
>>> rebuilt each night, does this mean that a 1.07 built yesterday is
>>> different from one built today?
>>>
>>> If I can offer any assistance with the automation of point-in-time
>>> tagged releases I would be happy to help, as I feel this would make
>>> Freeswitch more attractive to commercial operations as it would easily
>>> fit into well defined release processes.
>>>
>>> Thanks
>>> Tom
>>>
>>> _______________________________________________
>>> 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 msc at freeswitch.org  Thu May 12 19:35:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 12 May 2011 08:35:11 -0700
Subject: [Freeswitch-dev] is there any info abount which git tag/
 Freeswitch release is stable?
In-Reply-To: <83e5d123d690d463ea41168ab73dfdde.squirrel@my.tomp.co.uk>
References: <mailman.8860.1305186658.24555.freeswitch-dev@lists.freeswitch.org>
	<83e5d123d690d463ea41168ab73dfdde.squirrel@my.tomp.co.uk>
Message-ID: <BANLkTi=MPGT5OMbEoNQXf9xCUCpJ--oM7A@mail.gmail.com>

On Thu, May 12, 2011 at 1:02 AM, Tom Parrott <tomp at tomp.co.uk> wrote:

> Hi,
>
> Does the file at http://latest.freeswitch.org change each day to represent
> the current state of HEAD?
>
Correct.
-MC


>
> If so I could version that as 1.0.7-yyyymmdd to make it more suitable for
> RPMs.
>
> Thanks
> Tom
>
> >>
> >> I believe the developers have been discussing making a 1.0 stable branch
> >> soon with only bugfix and stability improvements and moving new
> >> development
> >> onto a new 1.2 branch. If that happens then perhaps you'll be more
> >> comfortable with that branch. Not sure when that's planned for though.
> >
> >
> > The issue by the way is with developers. While there's a lot of
> developers
> > and plenty of people using FS in the wild testing and making bug reports
> > on
> > Jira and plenty of people volunteering support on the list and IRC,
> > something like 90% of the code is still written by half a dozen people.
> > It's
> > damn good code though. :) That's too much to make it easy for them to
> both
> > maintain a stable branch and work on the newer branch, so up to this
> point
> > it has been better to focus on making FS complete, feature rich and
> stable
> > all in the same branch. Once the branches diverge, someone is going to
> > have
> > to both handle bug reports for 2 different codebases and also look at all
> > 1.2 changes to decide whether any are work backporting to 1.0 which won't
> > always be trivial if the code diverges significantly.
> >
> > -Steve
> >
> >
> > On 12 May 2011 08:45, Steven Ayre <steveayre at gmail.com> wrote:
> >
> >>  The 1.07 version available at http://latest.freeswitch.org seems to be
> >> rebuilt
> >>> each night, does this mean that a 1.07 built yesterday is different
> >>> from
> >>> one built today?
> >>
> >>
> >> Yes. Think of it as a release candidate for 1.0.8
> >>
> >> The git head is pretty much the most stable. Development tends to move
> >> too
> >> fast to make more regular 'stable' releases. You should of course test
> >> releases work as expected before putting them into production.
> >>
> >> If you find a Git version that you're comfortable with, then you can
> >> roll
> >> out that version across your whole network even if there's a newer Git
> >> head
> >> by running:
> >> > git clone git://git.freeswitch.org/freeswitch.git
> >> > cd freeswitch.git
> >> > git checkout GIT_HASH
> >>
> >> I believe the developers have been discussing making a 1.0 stable branch
> >> soon with only bugfix and stability improvements and moving new
> >> development
> >> onto a new 1.2 branch. If that happens then perhaps you'll be more
> >> comfortable with that branch. Not sure when that's planned for though.
> >>
> >> -Steve
> >>
> >>
> >> On 10 May 2011 19:39, Tom Parrott <tomp at tomp.co.uk> wrote:
> >>
> >>> Hi,
> >>>
> >>> I work for a company selling call tracking services, and we have
> >>> recently put into production a 3 node Freeswitch cluster to use
> >>> features
> >>> such as call recording and call announce.
> >>>
> >>> We are using the event socket feature to control freeswitch from our
> >>> custom written service.
> >>>
> >>> I have found Freeswitch a pleasure to use, and much easier to get along
> >>> with than Asterisk.
> >>>
> >>> However one thing that has concerned me is the lack of regular
> >>> point-in-time releases.
> >>>
> >>> When putting services in production, we use RPMs to allow us to easily
> >>> upgrade (and if need be rollback) software versions.
> >>>
> >>> The 1.07 version available at http://latest.freeswitch.org seems to be
> >>> rebuilt each night, does this mean that a 1.07 built yesterday is
> >>> different from one built today?
> >>>
> >>> If I can offer any assistance with the automation of point-in-time
> >>> tagged releases I would be happy to help, as I feel this would make
> >>> Freeswitch more attractive to commercial operations as it would easily
> >>> fit into well defined release processes.
> >>>
> >>> Thanks
> >>> Tom
> >>>
> >>> _______________________________________________
> >>> 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
> >>>
> >>
> >>
> >
>
>
>
> _______________________________________________
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110512/5d5791c7/attachment.html 

From anton.vazir at gmail.com  Thu May 12 21:13:21 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Thu, 12 May 2011 22:13:21 +0500
Subject: [Freeswitch-dev] start_dtmf detects dtmf as may times per dtmf as
 many times start_dtmf issued.
Message-ID: <BANLkTikb-TfAy91MgVM_YRAD3yQtUza=rQ@mail.gmail.com>

Just noted that my ESL get as may DTMF events per single keypress as,
many times I issued start_dtmf on the uuid.

it means if I do:

execute('start_dtmf')
execute('start_dtmf')

I get the same DTMF event twice.

Is that expected behavior? Logically, it supposed to to trigger detection on/off


From steveayre at gmail.com  Thu May 12 21:18:50 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 12 May 2011 18:18:50 +0100
Subject: [Freeswitch-dev] start_dtmf detects dtmf as may times per dtmf
 as many times start_dtmf issued.
In-Reply-To: <BANLkTikb-TfAy91MgVM_YRAD3yQtUza=rQ@mail.gmail.com>
References: <BANLkTikb-TfAy91MgVM_YRAD3yQtUza=rQ@mail.gmail.com>
Message-ID: <BANLkTimGducvQn6ikgArFQyVaVzVZM=Qug@mail.gmail.com>

I think so.

Is that expected behavior? Logically, it supposed to to trigger detection
> on/off

No, it's start_dtmf not toggle_dtmf.

It's an application that adds a media bug (callback) that gets given the
audio. Multiple media bugs can attach to a call, so I can definitely see it
adding the callback multiple times so it generates duplicates if you run it
more than once.

That's not to say it couldn't be patched to only attach once though.

-Steve



On 12 May 2011 18:13, Anton VG <anton.vazir at gmail.com> wrote:

> Just noted that my ESL get as may DTMF events per single keypress as,
> many times I issued start_dtmf on the uuid.
>
> it means if I do:
>
> execute('start_dtmf')
> execute('start_dtmf')
>
> I get the same DTMF event twice.
>
> Is that expected behavior? Logically, it supposed to to trigger detection
> on/off
>
> _______________________________________________
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110512/823212d8/attachment.html 

From anton.vazir at gmail.com  Thu May 12 21:16:15 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Thu, 12 May 2011 22:16:15 +0500
Subject: [Freeswitch-dev] start_dtmf detects dtmf as may times per dtmf
 as many times start_dtmf issued.
In-Reply-To: <BANLkTikb-TfAy91MgVM_YRAD3yQtUza=rQ@mail.gmail.com>
References: <BANLkTikb-TfAy91MgVM_YRAD3yQtUza=rQ@mail.gmail.com>
Message-ID: <BANLkTimRgCX+PpZpYHmE33O8zzwLjNGy1g@mail.gmail.com>

Update, for SIP-INFO and (supposelly) rfc2833 it doesn't. But does for INBAND

2011/5/12 Anton VG <anton.vazir at gmail.com>:
> Just noted that my ESL get as may DTMF events per single keypress as,
> many times I issued start_dtmf on the uuid.
>
> it means if I do:
>
> execute('start_dtmf')
> execute('start_dtmf')
>
> I get the same DTMF event twice.
>
> Is that expected behavior? Logically, it supposed to to trigger detection on/off
>


From steveayre at gmail.com  Thu May 12 23:09:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 12 May 2011 20:09:01 +0100
Subject: [Freeswitch-dev] start_dtmf detects dtmf as may times per dtmf
 as many times start_dtmf issued.
In-Reply-To: <BANLkTimRgCX+PpZpYHmE33O8zzwLjNGy1g@mail.gmail.com>
References: <BANLkTikb-TfAy91MgVM_YRAD3yQtUza=rQ@mail.gmail.com>
	<BANLkTimRgCX+PpZpYHmE33O8zzwLjNGy1g@mail.gmail.com>
Message-ID: <BANLkTinmdNSi3TXqcM1yC3a0yKpOwe22gw@mail.gmail.com>

start_dtmf is only used for detecting inband DTMF

On 12 May 2011 18:16, Anton VG <anton.vazir at gmail.com> wrote:

> Update, for SIP-INFO and (supposelly) rfc2833 it doesn't. But does for
> INBAND
>
> 2011/5/12 Anton VG <anton.vazir at gmail.com>:
> > Just noted that my ESL get as may DTMF events per single keypress as,
> > many times I issued start_dtmf on the uuid.
> >
> > it means if I do:
> >
> > execute('start_dtmf')
> > execute('start_dtmf')
> >
> > I get the same DTMF event twice.
> >
> > Is that expected behavior? Logically, it supposed to to trigger detection
> on/off
> >
>
> _______________________________________________
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110512/a177548d/attachment-0001.html 

From anton.vazir at gmail.com  Fri May 13 08:18:00 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Fri, 13 May 2011 09:18:00 +0500
Subject: [Freeswitch-dev] start_dtmf detects dtmf as may times per dtmf
 as many times start_dtmf issued.
In-Reply-To: <BANLkTinmdNSi3TXqcM1yC3a0yKpOwe22gw@mail.gmail.com>
References: <BANLkTikb-TfAy91MgVM_YRAD3yQtUza=rQ@mail.gmail.com>
	<BANLkTimRgCX+PpZpYHmE33O8zzwLjNGy1g@mail.gmail.com>
	<BANLkTinmdNSi3TXqcM1yC3a0yKpOwe22gw@mail.gmail.com>
Message-ID: <BANLkTi=xLA-_twXdOV+5BQPuS-ZU7p9Y=A@mail.gmail.com>

Understand, so if nobody minds, I'll note that on freeswitch wiki.

2011/5/13 Steven Ayre <steveayre at gmail.com>:
> start_dtmf is only used for detecting inband DTMF
>
> On 12 May 2011 18:16, Anton VG <anton.vazir at gmail.com> wrote:
>>
>> Update, for SIP-INFO and (supposelly) rfc2833 it doesn't. But does for
>> INBAND
>>
>> 2011/5/12 Anton VG <anton.vazir at gmail.com>:
>> > Just noted that my ESL get as may DTMF events per single keypress as,
>> > many times I issued start_dtmf on the uuid.
>> >
>> > it means if I do:
>> >
>> > execute('start_dtmf')
>> > execute('start_dtmf')
>> >
>> > I get the same DTMF event twice.
>> >
>> > Is that expected behavior? Logically, it supposed to to trigger
>> > detection on/off
>> >
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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 msc at freeswitch.org  Fri May 13 11:31:56 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 13 May 2011 00:31:56 -0700
Subject: [Freeswitch-dev] start_dtmf detects dtmf as may times per dtmf
 as many times start_dtmf issued.
In-Reply-To: <BANLkTi=xLA-_twXdOV+5BQPuS-ZU7p9Y=A@mail.gmail.com>
References: <BANLkTikb-TfAy91MgVM_YRAD3yQtUza=rQ@mail.gmail.com>
	<BANLkTimRgCX+PpZpYHmE33O8zzwLjNGy1g@mail.gmail.com>
	<BANLkTinmdNSi3TXqcM1yC3a0yKpOwe22gw@mail.gmail.com>
	<BANLkTi=xLA-_twXdOV+5BQPuS-ZU7p9Y=A@mail.gmail.com>
Message-ID: <BANLkTikbJ_CCJB4mD=5eak2+S7-Z48og-Q@mail.gmail.com>

On Thu, May 12, 2011 at 9:18 PM, Anton VG <anton.vazir at gmail.com> wrote:

> Understand, so if nobody minds, I'll note that on freeswitch wiki.
>
Please do.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110513/aa957f25/attachment.html 

From m.sobkow at marketelsystems.com  Sat May 14 00:43:27 2011
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Fri, 13 May 2011 14:43:27 -0600
Subject: [Freeswitch-dev] MEDIA_BUG_START
Message-ID: <4DCD97EF.4050703@marketelsystems.com>

I'm getting a MEDIA_BUG_START event.  I think there's a correlation 
between this event and some dropped calls we're experiencing.

I haven't found any way to force the event to occur, so I haven't been 
able to capture a log file of it happening in fs_cli.

Our FreeSwitch box is configured to use a SIP Trunk provided by our 
production Asterisk box, with T1 lines from thereon out.  For the most 
part it works, but one user in particular has a serious problem with 
dropped calls and I'm trying to figure it out.

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com

Visit our Blog for industry related information.
http://marketel-systems.blogspot.com/



From m.sobkow at marketelsystems.com  Sat May 14 00:59:06 2011
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Fri, 13 May 2011 14:59:06 -0600
Subject: [Freeswitch-dev] Can UUID's change?
Message-ID: <4DCD9B9A.6060609@marketelsystems.com>

Here's our scenario:

The operator logs in to our system, and has their UUID parked.

The system calls the customer when requested, which creates a UUID.

The operator and customer UUIDs are bridged.

We watch for events from the customer UUID to determine when the 
customer has hung up their end of the call, which causes the operator to 
be parked again.

However, I'm seeing some unrecognized UUIDs after bridging the call 
(i.e. that weren't returned by the place call code.)

Are there any events which might indicate that a UUID is being changed 
by the system, and that I should be watching for a new UUID instead?

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com

Visit our Blog for industry related information.
http://marketel-systems.blogspot.com/



From gabe at gundy.org  Sat May 14 01:03:37 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 May 2011 15:03:37 -0600
Subject: [Freeswitch-dev] Can UUID's change?
In-Reply-To: <4DCD9B9A.6060609@marketelsystems.com>
References: <4DCD9B9A.6060609@marketelsystems.com>
Message-ID: <BANLkTikNHUym9FKZF0Jshw3dTW7ufSpu7g@mail.gmail.com>

On Fri, May 13, 2011 at 2:59 PM, Mark Sobkow
<m.sobkow at marketelsystems.com> wrote:
> Are there any events which might indicate that a UUID is being changed
> by the system, and that I should be watching for a new UUID instead?

These look like questions that should go the the regular FreeSWITCH users list:
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

Best,
Gabe


From gabe at gundy.org  Sat May 14 02:20:58 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 May 2011 16:20:58 -0600
Subject: [Freeswitch-dev] MEDIA_BUG_START
In-Reply-To: <4DCD97EF.4050703@marketelsystems.com>
References: <4DCD97EF.4050703@marketelsystems.com>
Message-ID: <BANLkTinyqkiRFeUyCNmuzxVAr0Zup89pYA@mail.gmail.com>

On Fri, May 13, 2011 at 2:43 PM, Mark Sobkow
<m.sobkow at marketelsystems.com> wrote:
> I'm getting a MEDIA_BUG_START event. ?I think there's a correlation
> between this event and some dropped calls we're experiencing.

Please re-post this on the other list where regular users will be able
to benefit from it and I'll share my thoughts... I don't know that
they'll be worth much, but I do have some input to share :)

Best,
Gabe


From msc at freeswitch.org  Sat May 14 05:02:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 13 May 2011 18:02:52 -0700
Subject: [Freeswitch-dev] Can UUID's change?
In-Reply-To: <4DCD9B9A.6060609@marketelsystems.com>
References: <4DCD9B9A.6060609@marketelsystems.com>
Message-ID: <BANLkTikm=saVoii9pUTYSe4dnROGD4=8rA@mail.gmail.com>

Also, pastebin a debug log of this scenario. Be sure to use "FreeSWITCH log"
as the syntax highlighting. Paste the link in this thread.

-MC

On Fri, May 13, 2011 at 1:59 PM, Mark Sobkow
<m.sobkow at marketelsystems.com>wrote:

> Here's our scenario:
>
> The operator logs in to our system, and has their UUID parked.
>
> The system calls the customer when requested, which creates a UUID.
>
> The operator and customer UUIDs are bridged.
>
> We watch for events from the customer UUID to determine when the
> customer has hung up their end of the call, which causes the operator to
> be parked again.
>
> However, I'm seeing some unrecognized UUIDs after bridging the call
> (i.e. that weren't returned by the place call code.)
>
> Are there any events which might indicate that a UUID is being changed
> by the system, and that I should be watching for a new UUID instead?
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
> Visit our Blog for industry related information.
> http://marketel-systems.blogspot.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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110513/e39e4da3/attachment.html 

From math.parent at gmail.com  Sat May 14 13:15:26 2011
From: math.parent at gmail.com (Mathieu Parent)
Date: Sat, 14 May 2011 11:15:26 +0200
Subject: [Freeswitch-dev] Implementing Forward-{All, Busy, NoAnswer},
 ring policies and DND in FS
In-Reply-To: <BANLkTinhMzGO=btnm2dVtMN97yD4wqN+oA@mail.gmail.com>
References: <BANLkTinV7ww4hjs-XXUxffxRk0vMDf4Aqg@mail.gmail.com>
	<BANLkTim3yoLC-c4q=VZKfqO-ztzpHL645g@mail.gmail.com>
	<BANLkTinhMzGO=btnm2dVtMN97yD4wqN+oA@mail.gmail.com>
Message-ID: <BANLkTikOSShRLEK1Ft=g7jWhiJa=Ce4JyQ@mail.gmail.com>

2011/5/10 Rupa Schomaker <rupa at rupa.com>:
> On Tue, May 10, 2011 at 3:19 PM, Mathieu Parent <math.parent at gmail.com> wrote:
>> I have doubt about mod_hash because it doesn't propagate hashes on
>> remote servers (it does for limits only). Should I use mod_db instead?
>>
>
> +1


Thinking wider, what we need is an updatable directory. The directory
is by default stored as readonly XML files. Mod_voicemail currently
workaround this (IMO) by creating an SQL table to store vm password.
Couldn't we provide a more generic handler ?

Proposal: add a new xml_int binding: "directory_update".
-------------

With mod_xml_curl, the POST request will look like this:

[hostname] => testmachine
[section] => directory_update
[tag_name] => domain
[key_name] => name
[key_value] => domain1.awesomevoipdomain.faketld
[key] => id
[user] => 1004
[domain] => domain1.awesomevoipdomain.faketld
[set-param-forward-all-destination] => 1234
[set-param-forward-all-status] => 1

(POST variables prefixed with "set-param-" or "set-variable-" modify
the relevant XML attribute)

The xml_int should return either:
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="result">
    <result status="success" />
  </section>
</document>
or
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="result">
    <result status="not found" />
  </section>
</document>

A new function is needed:
SWITCH_DECLARE(switch_status_t) switch_xml_update_user(const char *key,

                                    const char *user_name,

                                    const char *domain_name,

                                    switch_event_t *params)

Where params is an event containing the set-* variables.

As mod_xml_curl is not configured by default, we need something to
update XML files: any idea?

I will start to implement the suggested proposal.

-- 
Mathieu


From anthony.minessale at gmail.com  Sat May 14 22:09:09 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 14 May 2011 13:09:09 -0500
Subject: [Freeswitch-dev] Implementing Forward-{All, Busy, NoAnswer},
 ring policies and DND in FS
In-Reply-To: <BANLkTikOSShRLEK1Ft=g7jWhiJa=Ce4JyQ@mail.gmail.com>
References: <BANLkTinV7ww4hjs-XXUxffxRk0vMDf4Aqg@mail.gmail.com>
	<BANLkTim3yoLC-c4q=VZKfqO-ztzpHL645g@mail.gmail.com>
	<BANLkTinhMzGO=btnm2dVtMN97yD4wqN+oA@mail.gmail.com>
	<BANLkTikOSShRLEK1Ft=g7jWhiJa=Ce4JyQ@mail.gmail.com>
Message-ID: <BANLkTi=KeeO0Q8GQAiSrjJsuZy71GTh-OA@mail.gmail.com>

It is important to not violate scope.  Do not let one problem define your
path.  The goal is not to make the default a certain way.  The goal is to
scale from static configs to dynamic as seamlessly as possible.
On May 14, 2011 4:20 AM, "Mathieu Parent" <math.parent at gmail.com> wrote:
> 2011/5/10 Rupa Schomaker <rupa at rupa.com>:
>> On Tue, May 10, 2011 at 3:19 PM, Mathieu Parent <math.parent at gmail.com>
wrote:
>>> I have doubt about mod_hash because it doesn't propagate hashes on
>>> remote servers (it does for limits only). Should I use mod_db instead?
>>>
>>
>> +1
>
>
> Thinking wider, what we need is an updatable directory. The directory
> is by default stored as readonly XML files. Mod_voicemail currently
> workaround this (IMO) by creating an SQL table to store vm password.
> Couldn't we provide a more generic handler ?
>
> Proposal: add a new xml_int binding: "directory_update".
> -------------
>
> With mod_xml_curl, the POST request will look like this:
>
> [hostname] => testmachine
> [section] => directory_update
> [tag_name] => domain
> [key_name] => name
> [key_value] => domain1.awesomevoipdomain.faketld
> [key] => id
> [user] => 1004
> [domain] => domain1.awesomevoipdomain.faketld
> [set-param-forward-all-destination] => 1234
> [set-param-forward-all-status] => 1
>
> (POST variables prefixed with "set-param-" or "set-variable-" modify
> the relevant XML attribute)
>
> The xml_int should return either:
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="result">
> <result status="success" />
> </section>
> </document>
> or
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="result">
> <result status="not found" />
> </section>
> </document>
>
> A new function is needed:
> SWITCH_DECLARE(switch_status_t) switch_xml_update_user(const char *key,
>
> const char *user_name,
>
> const char *domain_name,
>
> switch_event_t *params)
>
> Where params is an event containing the set-* variables.
>
> As mod_xml_curl is not configured by default, we need something to
> update XML files: any idea?
>
> I will start to implement the suggested proposal.
>
> --
> Mathieu
>
> _______________________________________________
> 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110514/fd8e2f57/attachment-0001.html 

From anton.vazir at gmail.com  Mon May 16 17:44:34 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Mon, 16 May 2011 18:44:34 +0500
Subject: [Freeswitch-dev] unable to access jira
Message-ID: <BANLkTi=u6=5syuqrYPnY8BgUDBh9NRx1fA@mail.gmail.com>

Curios, I'm unable to access Jira for several days, and direct
connection failed. I have tried through iranprox.com and anonymizer.ru
to make sure it's not my local routing problem - the same result -
timeout.

Is it down?

---------

ERROR

The requested URL could not be retrieved

While trying to retrieve the URL: http://jira.freeswitch.org/

The following error was encountered:

Read Error
The system returned:

    (104) Connection reset by peer
An error condition occurred while reading data from the network.
Please retry your request.


From steveayre at gmail.com  Mon May 16 18:07:25 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 16 May 2011 15:07:25 +0100
Subject: [Freeswitch-dev] unable to access jira
In-Reply-To: <BANLkTi=u6=5syuqrYPnY8BgUDBh9NRx1fA@mail.gmail.com>
References: <BANLkTi=u6=5syuqrYPnY8BgUDBh9NRx1fA@mail.gmail.com>
Message-ID: <BANLkTik-D4JMosLv=QUrw+F19qLfmMVr9g@mail.gmail.com>

Same here for me.

bkw_ mentioned in the channel the other day that it has problems
because Jira/Fisheye use so much memory, thanks to Java. Currently
it's running in a VM but he's got some dedicated hardware on the way
that should sort it out.

-Steve



On 16 May 2011 14:44, Anton VG <anton.vazir at gmail.com> wrote:
> Curios, I'm unable to access Jira for several days, and direct
> connection failed. I have tried through iranprox.com and anonymizer.ru
> to make sure it's not my local routing problem - the same result -
> timeout.
>
> Is it down?
>
> ---------
>
> ERROR
>
> The requested URL could not be retrieved
>
> While trying to retrieve the URL: http://jira.freeswitch.org/
>
> The following error was encountered:
>
> Read Error
> The system returned:
>
> ? ?(104) Connection reset by peer
> An error condition occurred while reading data from the network.
> Please retry your request.
>
> _______________________________________________
> 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 msc at freeswitch.org  Mon May 16 23:28:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 May 2011 12:28:15 -0700
Subject: [Freeswitch-dev] unable to access jira
In-Reply-To: <BANLkTik-D4JMosLv=QUrw+F19qLfmMVr9g@mail.gmail.com>
References: <BANLkTi=u6=5syuqrYPnY8BgUDBh9NRx1fA@mail.gmail.com>
	<BANLkTik-D4JMosLv=QUrw+F19qLfmMVr9g@mail.gmail.com>
Message-ID: <BANLkTin-Yb7xyeOER5jOFyihmi6mQ8ikTQ@mail.gmail.com>

I just logged into jira a few minutes ago. Can you guys all try again and
let me know if you have any issues?
-MC

On Mon, May 16, 2011 at 7:07 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Same here for me.
>
> bkw_ mentioned in the channel the other day that it has problems
> because Jira/Fisheye use so much memory, thanks to Java. Currently
> it's running in a VM but he's got some dedicated hardware on the way
> that should sort it out.
>
> -Steve
>
>
>
> On 16 May 2011 14:44, Anton VG <anton.vazir at gmail.com> wrote:
> > Curios, I'm unable to access Jira for several days, and direct
> > connection failed. I have tried through iranprox.com and anonymizer.ru
> > to make sure it's not my local routing problem - the same result -
> > timeout.
> >
> > Is it down?
> >
> > ---------
> >
> > ERROR
> >
> > The requested URL could not be retrieved
> >
> > While trying to retrieve the URL: http://jira.freeswitch.org/
> >
> > The following error was encountered:
> >
> > Read Error
> > The system returned:
> >
> >    (104) Connection reset by peer
> > An error condition occurred while reading data from the network.
> > Please retry your request.
> >
> > _______________________________________________
> > 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
> >
>
> _______________________________________________
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110516/beda11f9/attachment.html 

From steveayre at gmail.com  Tue May 17 02:37:36 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 16 May 2011 23:37:36 +0100
Subject: [Freeswitch-dev] unable to access jira
In-Reply-To: <BANLkTin-Yb7xyeOER5jOFyihmi6mQ8ikTQ@mail.gmail.com>
References: <BANLkTi=u6=5syuqrYPnY8BgUDBh9NRx1fA@mail.gmail.com>
	<BANLkTik-D4JMosLv=QUrw+F19qLfmMVr9g@mail.gmail.com>
	<BANLkTin-Yb7xyeOER5jOFyihmi6mQ8ikTQ@mail.gmail.com>
Message-ID: <BANLkTinswHLnvT96jucxe-6K_aHJoscpkg@mail.gmail.com>

Loads for me now. :)


On 16 May 2011 20:28, Michael Collins <msc at freeswitch.org> wrote:
> I just logged into jira a few minutes ago. Can you guys all try again and
> let me know if you have any issues?
> -MC
>
> On Mon, May 16, 2011 at 7:07 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Same here for me.
>>
>> bkw_ mentioned in the channel the other day that it has problems
>> because Jira/Fisheye use so much memory, thanks to Java. Currently
>> it's running in a VM but he's got some dedicated hardware on the way
>> that should sort it out.
>>
>> -Steve
>>
>>
>>
>> On 16 May 2011 14:44, Anton VG <anton.vazir at gmail.com> wrote:
>> > Curios, I'm unable to access Jira for several days, and direct
>> > connection failed. I have tried through iranprox.com and anonymizer.ru
>> > to make sure it's not my local routing problem - the same result -
>> > timeout.
>> >
>> > Is it down?
>> >
>> > ---------
>> >
>> > ERROR
>> >
>> > The requested URL could not be retrieved
>> >
>> > While trying to retrieve the URL: http://jira.freeswitch.org/
>> >
>> > The following error was encountered:
>> >
>> > Read Error
>> > The system returned:
>> >
>> > ? ?(104) Connection reset by peer
>> > An error condition occurred while reading data from the network.
>> > Please retry your request.
>> >
>> > _______________________________________________
>> > 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
>> >
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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 anton.vazir at gmail.com  Tue May 17 13:03:50 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Tue, 17 May 2011 14:03:50 +0500
Subject: [Freeswitch-dev] unable to access jira
In-Reply-To: <BANLkTinswHLnvT96jucxe-6K_aHJoscpkg@mail.gmail.com>
References: <BANLkTi=u6=5syuqrYPnY8BgUDBh9NRx1fA@mail.gmail.com>
	<BANLkTik-D4JMosLv=QUrw+F19qLfmMVr9g@mail.gmail.com>
	<BANLkTin-Yb7xyeOER5jOFyihmi6mQ8ikTQ@mail.gmail.com>
	<BANLkTinswHLnvT96jucxe-6K_aHJoscpkg@mail.gmail.com>
Message-ID: <BANLkTimQ5haEOcfa7yqkrJXRBJOcq4SP1g@mail.gmail.com>

Works!

2011/5/17 Steven Ayre <steveayre at gmail.com>:
> Loads for me now. :)
>
>
> On 16 May 2011 20:28, Michael Collins <msc at freeswitch.org> wrote:
>> I just logged into jira a few minutes ago. Can you guys all try again and
>> let me know if you have any issues?
>> -MC
>>
>> On Mon, May 16, 2011 at 7:07 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> Same here for me.
>>>
>>> bkw_ mentioned in the channel the other day that it has problems
>>> because Jira/Fisheye use so much memory, thanks to Java. Currently
>>> it's running in a VM but he's got some dedicated hardware on the way
>>> that should sort it out.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 16 May 2011 14:44, Anton VG <anton.vazir at gmail.com> wrote:
>>> > Curios, I'm unable to access Jira for several days, and direct
>>> > connection failed. I have tried through iranprox.com and anonymizer.ru
>>> > to make sure it's not my local routing problem - the same result -
>>> > timeout.
>>> >
>>> > Is it down?
>>> >
>>> > ---------
>>> >
>>> > ERROR
>>> >
>>> > The requested URL could not be retrieved
>>> >
>>> > While trying to retrieve the URL: http://jira.freeswitch.org/
>>> >
>>> > The following error was encountered:
>>> >
>>> > Read Error
>>> > The system returned:
>>> >
>>> > ? ?(104) Connection reset by peer
>>> > An error condition occurred while reading data from the network.
>>> > Please retry your request.
>>> >
>>> > _______________________________________________
>>> > 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
>>> >
>>>
>>> _______________________________________________
>>> 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
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 anton.vazir at gmail.com  Wed May 18 12:40:47 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 18 May 2011 13:40:47 +0500
Subject: [Freeswitch-dev] unable to access jira
In-Reply-To: <BANLkTimQ5haEOcfa7yqkrJXRBJOcq4SP1g@mail.gmail.com>
References: <BANLkTi=u6=5syuqrYPnY8BgUDBh9NRx1fA@mail.gmail.com>
	<BANLkTik-D4JMosLv=QUrw+F19qLfmMVr9g@mail.gmail.com>
	<BANLkTin-Yb7xyeOER5jOFyihmi6mQ8ikTQ@mail.gmail.com>
	<BANLkTinswHLnvT96jucxe-6K_aHJoscpkg@mail.gmail.com>
	<BANLkTimQ5haEOcfa7yqkrJXRBJOcq4SP1g@mail.gmail.com>
Message-ID: <BANLkTikgyqnftWtErKZV4tULgrVp1zbZbg@mail.gmail.com>

down again for me. Java suxx

2011/5/17 Anton VG <anton.vazir at gmail.com>:
> Works!
>
> 2011/5/17 Steven Ayre <steveayre at gmail.com>:
>> Loads for me now. :)
>>
>>
>> On 16 May 2011 20:28, Michael Collins <msc at freeswitch.org> wrote:
>>> I just logged into jira a few minutes ago. Can you guys all try again and
>>> let me know if you have any issues?
>>> -MC
>>>
>>> On Mon, May 16, 2011 at 7:07 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>>>
>>>> Same here for me.
>>>>
>>>> bkw_ mentioned in the channel the other day that it has problems
>>>> because Jira/Fisheye use so much memory, thanks to Java. Currently
>>>> it's running in a VM but he's got some dedicated hardware on the way
>>>> that should sort it out.
>>>>
>>>> -Steve
>>>>
>>>>
>>>>
>>>> On 16 May 2011 14:44, Anton VG <anton.vazir at gmail.com> wrote:
>>>> > Curios, I'm unable to access Jira for several days, and direct
>>>> > connection failed. I have tried through iranprox.com and anonymizer.ru
>>>> > to make sure it's not my local routing problem - the same result -
>>>> > timeout.
>>>> >
>>>> > Is it down?
>>>> >
>>>> > ---------
>>>> >
>>>> > ERROR
>>>> >
>>>> > The requested URL could not be retrieved
>>>> >
>>>> > While trying to retrieve the URL: http://jira.freeswitch.org/
>>>> >
>>>> > The following error was encountered:
>>>> >
>>>> > Read Error
>>>> > The system returned:
>>>> >
>>>> > ? ?(104) Connection reset by peer
>>>> > An error condition occurred while reading data from the network.
>>>> > Please retry your request.
>>>> >
>>>> > _______________________________________________
>>>> > 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
>>>> >
>>>>
>>>> _______________________________________________
>>>> 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
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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 msc at freeswitch.org  Wed May 18 19:46:47 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 18 May 2011 08:46:47 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <BANLkTimYj3BueYFKnUuAVz7BwUE17F6r=Q@mail.gmail.com>

Hello all!

Today's agenda is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2011_05_18

It's a bit light but there are a few things to talk about. Please bring your
questions for discussion.

Thanks,
MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110518/d1ca4e32/attachment.html 

From anton.vazir at gmail.com  Thu May 19 17:11:14 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Thu, 19 May 2011 18:11:14 +0500
Subject: [Freeswitch-dev] SIP endpoint can be registered several times
Message-ID: <BANLkTinpN7fZxiLNv0bt=D8cTwP2hakRzg@mail.gmail.com>

Just noted an issue (or feature :) - I ve been switching on/off my sip
phone by powering it on/off from power plug, and now it's registered
several time with the same properties. I suppose that is kind of
strange.
I found that whle looking for a reason, why there is more than a
single UUID on originate, while originating a single end point.

Should I report the given on JIRA?

freeswitch at lab3> sofia_contact sip3779100

sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
freeswitch at lab3>


From anton.vazir at gmail.com  Thu May 19 17:26:01 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Thu, 19 May 2011 18:26:01 +0500
Subject: [Freeswitch-dev] SIP endpoint can be registered several times
In-Reply-To: <BANLkTinpN7fZxiLNv0bt=D8cTwP2hakRzg@mail.gmail.com>
References: <BANLkTinpN7fZxiLNv0bt=D8cTwP2hakRzg@mail.gmail.com>
Message-ID: <BANLkTimuGAE+iRSrz8ONjJ23h9iqw2jy-Q@mail.gmail.com>

forgot to mention: multiple registrations are enabled. So it should be
able to register several times, but since it's a single IP phone - it
should not be listed a many times.

2011/5/19 Anton VG <anton.vazir at gmail.com>:
> Just noted an issue (or feature :) - I ve been switching on/off my sip
> phone by powering it on/off from power plug, and now it's registered
> several time with the same properties. I suppose that is kind of
> strange.
> I found that whle looking for a reason, why there is more than a
> single UUID on originate, while originating a single end point.
>
> Should I report the given on JIRA?
>
> freeswitch at lab3> sofia_contact sip3779100
>
> sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
> freeswitch at lab3>
>


From steveayre at gmail.com  Thu May 19 18:29:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 19 May 2011 15:29:14 +0100
Subject: [Freeswitch-dev] SIP endpoint can be registered several times
In-Reply-To: <BANLkTinpN7fZxiLNv0bt=D8cTwP2hakRzg@mail.gmail.com>
References: <BANLkTinpN7fZxiLNv0bt=D8cTwP2hakRzg@mail.gmail.com>
Message-ID: <BANLkTinT=uifmr1Y5eg3s8TF4rYkmsVYMg@mail.gmail.com>

Registrations expire after a certain amount of time, until then you'll have
more than one registration.

-Steve


On 19 May 2011 14:11, Anton VG <anton.vazir at gmail.com> wrote:

> Just noted an issue (or feature :) - I ve been switching on/off my sip
> phone by powering it on/off from power plug, and now it's registered
> several time with the same properties. I suppose that is kind of
> strange.
> I found that whle looking for a reason, why there is more than a
> single UUID on originate, while originating a single end point.
>
> Should I report the given on JIRA?
>
> freeswitch at lab3> sofia_contact sip3779100
>
> sofia/internal/sip:sip3779100 at 192.168.5.21:5062
> ;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062
> ;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062
> ;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062
> ;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
> freeswitch at lab3>
>
> _______________________________________________
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110519/bd0dfbb7/attachment.html 

From anton.vazir at gmail.com  Thu May 19 18:38:45 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Thu, 19 May 2011 19:38:45 +0500
Subject: [Freeswitch-dev] SIP endpoint can be registered several times
In-Reply-To: <BANLkTinT=uifmr1Y5eg3s8TF4rYkmsVYMg@mail.gmail.com>
References: <BANLkTinpN7fZxiLNv0bt=D8cTwP2hakRzg@mail.gmail.com>
	<BANLkTinT=uifmr1Y5eg3s8TF4rYkmsVYMg@mail.gmail.com>
Message-ID: <BANLkTikHmNqNoAxR2ASJi8u+F4NGGcR39A@mail.gmail.com>

I understand that. If you look at the credentials, there are exactly
the same, so what the point of keeping several exact registrations?

2011/5/19 Steven Ayre <steveayre at gmail.com>:
> Registrations expire after a certain amount of time, until then you'll have
> more than one registration.
>
> -Steve
>
>
> On 19 May 2011 14:11, Anton VG <anton.vazir at gmail.com> wrote:
>>
>> Just noted an issue (or feature :) - I ve been switching on/off my sip
>> phone by powering it on/off from power plug, and now it's registered
>> several time with the same properties. I suppose that is kind of
>> strange.
>> I found that whle looking for a reason, why there is more than a
>> single UUID on originate, while originating a single end point.
>>
>> Should I report the given on JIRA?
>>
>> freeswitch at lab3> sofia_contact sip3779100
>>
>>
>> sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
>> freeswitch at lab3>
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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 peter.olsson at visionutveckling.se  Thu May 19 18:49:10 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 19 May 2011 16:49:10 +0200
Subject: [Freeswitch-dev] SIP endpoint can be registered several times
In-Reply-To: <BANLkTikHmNqNoAxR2ASJi8u+F4NGGcR39A@mail.gmail.com>
References: <BANLkTinpN7fZxiLNv0bt=D8cTwP2hakRzg@mail.gmail.com>
	<BANLkTinT=uifmr1Y5eg3s8TF4rYkmsVYMg@mail.gmail.com>
	<BANLkTikHmNqNoAxR2ASJi8u+F4NGGcR39A@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58F79DBEDC@cooper>

Try "sofia status profile internal" (or whatever profile you are using), I bet you'll see that the registrations are not identical...

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Anton VG
Skickat: den 19 maj 2011 16:39
Till: freeswitch-dev at lists.freeswitch.org
?mne: Re: [Freeswitch-dev] SIP endpoint can be registered several times

I understand that. If you look at the credentials, there are exactly
the same, so what the point of keeping several exact registrations?

2011/5/19 Steven Ayre <steveayre at gmail.com>:
> Registrations expire after a certain amount of time, until then you'll have
> more than one registration.
>
> -Steve
>
>
> On 19 May 2011 14:11, Anton VG <anton.vazir at gmail.com> wrote:
>>
>> Just noted an issue (or feature :) - I ve been switching on/off my sip
>> phone by powering it on/off from power plug, and now it's registered
>> several time with the same properties. I suppose that is kind of
>> strange.
>> I found that whle looking for a reason, why there is more than a
>> single UUID on originate, while originating a single end point.
>>
>> Should I report the given on JIRA?
>>
>> freeswitch at lab3> sofia_contact sip3779100
>>
>>
>> sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
>> freeswitch at lab3>
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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
>
>

_______________________________________________
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

!DSPAM:4dd52b9532769872813097!



From anton.vazir at gmail.com  Thu May 19 19:10:06 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Thu, 19 May 2011 20:10:06 +0500
Subject: [Freeswitch-dev] SIP fantom calls,
 when calling disconnected SIP endpoint. EP called when plugged in.
 suppose bug?
Message-ID: <BANLkTi=X+sa4O-gjU=h=qYeWWaGPRAUNOg@mail.gmail.com>

While emulating end-user behavior, encountered a problem. When there
is a call to the SIP endpoint, which have disappeared (power
unplugged), SOFIA queues the call and calls that endpoint, if it's
plugged IN back, within
origination_timeout interval. So it means, that there is NO job UUID
(I kill all job uuid's by UUID_kill, no show calls, no show channels.
But still - I do plug my phone back, it registers to FS, and in 10
seconds I'm receiving a call to it!

Reproducibility 100%

How to reproduce:

Register an endpoint.
Unplug the power, so registration persists in FS
I do 'originate' a call to a SIP endpoint and &park as shown

originate {ringback='%(2000,4000,440.0,480.0)',origination_caller_id_number=907388397,originate_timeout=60,origination_uuid=f84558f4-46e9-4f8a-a546-7edfd6fd0d5e,leg_timeout=60}[continue_on_fail=false,leg_a_uuid=bb744c31-92cf-47d1-aaeb-1c984cb4d599]sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
&park

than System attempts to call that endpoint, wait for 4 seconds,
disconnect. Kill all job UUIDs, which was originating the call.

than plug in the phone, and in 10 seconds there is an incoming call.

Just discovered:
If i do not kill the JOB UUIDs - than, when phantom call happens, I
can see the debug:

2011-05-19 19:59:56.662932 [WARNING] sofia_reg.c:1241 SIP auth
challenge (REGISTER) on sofia profile 'internal' for
[sip3779100 at 192.168.100.11] from ip 192.168.5.21
2011-05-19 20:00:08.182927 [INFO] sofia.c:748
sofia/internal/sip:sip3779100 at 192.168.5.21:5062 Update Callee ID to
"Outbound Call" <sip3779100>
2011-05-19 20:00:08.252929 [DEBUG] sofia.c:4770 Channel
sofia/internal/sip:sip3779100 at 192.168.5.21:5062 entering state
[proceeding][180]
2011-05-19 20:00:08.252929 [NOTICE] sofia.c:4848 Ring-Ready
sofia/internal/sip:sip3779100 at 192.168.5.21:5062!
2011-05-19 20:04:24.822927 [DEBUG] switch_channel.c:2592
(sofia/internal/sip:sip3779100 at 192.168.5.21:5062) Callstate Change
RINGING -> HANGUP
2011-05-19 20:04:24.822927 [NOTICE] mod_commands.c:2117 Hangup
sofia/internal/sip:sip3779100 at 192.168.5.21:5062 [CS_CONSUME_MEDIA]
[NORMAL_CLEARING]
2011-05-19 20:04:24.822927 [DEBUG] switch_channel.c:2608 Send signal
sofia/internal/sip:sip3779100 at 192.168.5.21:5062 [KILL]
2011-05-19 20:04:24.822927 [DEBUG] switch_core_session.c:1114 Send
signal sofia/internal/sip:sip3779100 at 192.168.5.21:5062 [BREAK]
2011-05-19 20:04:24.822927 [CONSOLE] switch_cpp.cpp:1197 API cmd
result: [+OK
]2011-05-19 20:04:24.822927 [DEBUG] mod_python.c:186 Finished calling
python script
2011-05-19 20:04:24.822927 [DEBUG] switch_cpp.cpp:988
sofia/external/907388397 at 10.34.0.4 destroy/unlink session from object
2011-05-19 20:04:24.822927 [DEBUG] switch_core_state_machine.c:497
Hangup Command with Session python(cleanup
f84558f4-46e9-4f8a-a546-7edfd6fd0d5e):

If I KILL the uuids by api hangup hook - there is no DEBUG, but still
the call...

2011-05-19 20:04:24.822927 [CONSOLE] switch_cpp.cpp:1197 hangup hook,
UUIDS: f84558f4-46e9-4f8a-a546-7edfd6fd0d5e
2011-05-19 20:04:24.822927 [CONSOLE] switch_cpp.cpp:1197 API cmd:
[uuid_setvar f84558f4-46e9-4f8a-a546-7edfd6fd0d5e disposition
ORIGINATOR_CANCEL]
2011-05-19 20:04:24.822927 [CONSOLE] switch_cpp.cpp:1197 API cmd result: [+OK
]2011-05-19 20:04:24.822927 [CONSOLE] switch_cpp.cpp:1197 API cmd:
[uuid_kill f84558f4-46e9-4f8a-a546-7edfd6fd0d5e]

show version
FreeSWITCH Version 1.0.head (git-fccbba5 2011-05-18 19-00-42 -0400)

Seems JOB is detached from UUID - but still not killed itself.


From anton.vazir at gmail.com  Thu May 19 19:13:37 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Thu, 19 May 2011 20:13:37 +0500
Subject: [Freeswitch-dev] SIP endpoint can be registered several times
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58F79DBEDC@cooper>
References: <BANLkTinpN7fZxiLNv0bt=D8cTwP2hakRzg@mail.gmail.com>
	<BANLkTinT=uifmr1Y5eg3s8TF4rYkmsVYMg@mail.gmail.com>
	<BANLkTikHmNqNoAxR2ASJi8u+F4NGGcR39A@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F79DBEDC@cooper>
Message-ID: <BANLkTikEEWJZqfvJ2NPHB3fxWMdxFhDmrA@mail.gmail.com>

The only difference is Call-ID line - suppose that is not enough.


Call-ID:        b7e81237-d749ef63 at 192.168.5.21
User:           sip3779100 at 192.168.100.11
Contact:        "sip3779100"
<sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062>
Agent:          Linksys/SPA942-6.1.3(a)
Status:         Registered(UDP-NAT)(unknown) EXP(2011-05-19 21:05:40)
EXPSECS(3288)
Host:           lab3
IP:             192.168.5.21
Port:           5062
Auth-User:      sip3779100
Auth-Realm:     192.168.100.11
MWI-Account:    sip3779100 at 192.168.100.11


Call-ID:        ab03086b-91d64ebf at 192.168.5.21
User:           sip3779100 at 192.168.100.11
Contact:        "sip3779100"
<sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062>
Agent:          Linksys/SPA942-6.1.3(a)
Status:         Registered(UDP-NAT)(unknown) EXP(2011-05-19 21:00:56)
EXPSECS(3004)
Host:           lab3
IP:             192.168.5.21
Port:           5062
Auth-User:      sip3779100
Auth-Realm:     192.168.100.11
MWI-Account:    sip3779100 at 192.168.100.11


2011/5/19 Peter Olsson <peter.olsson at visionutveckling.se>:
> Try "sofia status profile internal" (or whatever profile you are using), I bet you'll see that the registrations are not identical...
>
> /Peter
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Anton VG
> Skickat: den 19 maj 2011 16:39
> Till: freeswitch-dev at lists.freeswitch.org
> ?mne: Re: [Freeswitch-dev] SIP endpoint can be registered several times
>
> I understand that. If you look at the credentials, there are exactly
> the same, so what the point of keeping several exact registrations?
>
> 2011/5/19 Steven Ayre <steveayre at gmail.com>:
>> Registrations expire after a certain amount of time, until then you'll have
>> more than one registration.
>>
>> -Steve
>>
>>
>> On 19 May 2011 14:11, Anton VG <anton.vazir at gmail.com> wrote:
>>>
>>> Just noted an issue (or feature :) - I ve been switching on/off my sip
>>> phone by powering it on/off from power plug, and now it's registered
>>> several time with the same properties. I suppose that is kind of
>>> strange.
>>> I found that whle looking for a reason, why there is more than a
>>> single UUID on originate, while originating a single end point.
>>>
>>> Should I report the given on JIRA?
>>>
>>> freeswitch at lab3> sofia_contact sip3779100
>>>
>>>
>>> sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
>>> freeswitch at lab3>
>>>
>>> _______________________________________________
>>> 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
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>
> !DSPAM:4dd52b9532769872813097!
>
>
> _______________________________________________
> 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 anton.vazir at gmail.com  Thu May 19 19:19:55 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Thu, 19 May 2011 20:19:55 +0500
Subject: [Freeswitch-dev] SIP fantom calls,
 when calling disconnected SIP endpoint. EP called when plugged in.
 suppose bug?
In-Reply-To: <BANLkTi=X+sa4O-gjU=h=qYeWWaGPRAUNOg@mail.gmail.com>
References: <BANLkTi=X+sa4O-gjU=h=qYeWWaGPRAUNOg@mail.gmail.com>
Message-ID: <BANLkTinjfQaqZUi=ccr+ueXCo5MM8k+OYQ@mail.gmail.com>

Yea, there is siptrace, when enabled.

sofia global siptrace on

freeswitch at lab3> send 1181 bytes to udp/[192.168.5.21]:5062 at 15:16:40.952027:
   ------------------------------------------------------------------------
   INVITE sip:sip3779100 at 192.168.5.21:5062 SIP/2.0
   Via: SIP/2.0/UDP 192.168.100.11;rport;branch=z9hG4bKN2QvX33eK72rr
   Route: <sip:sip3779100 at 192.168.5.21:5062>
   Max-Forwards: 70
   From: "" <sip:907388397 at 192.168.100.11>;tag=KXXUQ57gS66ZS
   To: <sip:sip3779100 at 192.168.5.21:5062>
   Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2
   CSeq: 12573020 INVITE
   Contact: <sip:mod_sofia at 192.168.100.11:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fccbba5 2011-05-18
19-00-42 -0400
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
sla, include-session-description, presence.winfo, message-summary,
refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 209
   X-FS-Support: update_display
   Remote-Party-ID:
<sip:907388397 at 192.168.100.11>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1305790653 1305790654 IN IP4 192.168.100.11
   s=FreeSWITCH
   c=IN IP4 192.168.100.11
   t=0 0
   m=audio 27516 RTP/AVP 0 8 3 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 304 bytes from udp/[192.168.5.21]:5062 at 15:16:40.968185:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   To: <sip:sip3779100 at 192.168.5.21:5062>
   From: "" <sip:907388397 at 192.168.100.11>;tag=KXXUQ57gS66ZS
   Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2
   CSeq: 12573020 INVITE
   Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKN2QvX33eK72rr
   Server: Linksys/SPA942-6.1.3(a)
   Content-Length: 0

   ------------------------------------------------------------------------
send 414 bytes to udp/[192.168.5.21]:5062 at 15:16:40.968378:
   ------------------------------------------------------------------------
   CANCEL sip:sip3779100 at 192.168.5.21:5062 SIP/2.0
   Via: SIP/2.0/UDP 192.168.100.11;rport;branch=z9hG4bKN2QvX33eK72rr
   Route: <sip:sip3779100 at 192.168.5.21:5062>
   Max-Forwards: 70
   From: "" <sip:907388397 at 192.168.100.11>;tag=KXXUQ57gS66ZS
   To: <sip:sip3779100 at 192.168.5.21:5062>
   Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2
   CSeq: 12573020 CANCEL
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 356 bytes from udp/[192.168.5.21]:5062 at 15:16:40.984188:
   ------------------------------------------------------------------------
   SIP/2.0 481 Call Leg/Transaction Does Not Exist
   To: <sip:sip3779100 at 192.168.5.21:5062>;tag=1f405cc57aec781ei2
   From: "" <sip:907388397 at 192.168.100.11>;tag=KXXUQ57gS66ZS
   Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2
   CSeq: 12573020 CANCEL
   Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKN2QvX33eK72rr
   Server: Linksys/SPA942-6.1.3(a)
   Content-Length: 0

   ------------------------------------------------------------------------
recv 386 bytes from udp/[192.168.5.21]:5062 at 15:16:40.994546:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   To: <sip:sip3779100 at 192.168.5.21:5062>;tag=4482dc0be7df9455i2
   From: "" <sip:907388397 at 192.168.100.11>;tag=KXXUQ57gS66ZS
   Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2
   CSeq: 12573020 INVITE
   Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKN2QvX33eK72rr
   Contact: "sip3779100" <sip:sip3779100 at 192.168.5.21:5062>
   Server: Linksys/SPA942-6.1.3(a)
   Content-Length: 0


From anton.vazir at gmail.com  Thu May 19 19:38:44 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Thu, 19 May 2011 20:38:44 +0500
Subject: [Freeswitch-dev] SIP fantom calls,
 when calling disconnected SIP endpoint. EP called when plugged in.
 suppose bug?
In-Reply-To: <BANLkTinjfQaqZUi=ccr+ueXCo5MM8k+OYQ@mail.gmail.com>
References: <BANLkTi=X+sa4O-gjU=h=qYeWWaGPRAUNOg@mail.gmail.com>
	<BANLkTinjfQaqZUi=ccr+ueXCo5MM8k+OYQ@mail.gmail.com>
Message-ID: <BANLkTi=1gZVDTJ1Zu0ZpZVtSE+4DzEonVA@mail.gmail.com>

Adding progress_timeout and leg_progress_timeout did not changed it behavior

originate {originate_retries=0,originate_timeout=60,progress_timeout=5,leg_timeout=60,origination_caller_id_number=907388397,ringback='%(2000,4000,440.0,480.0)',origination_uuid=f7f2d66d-d5e0-4b77-9fda-bae5556b6c8f}[continue_on_fail=false,leg_progress_timeout=7,leg_a_uuid=9bf89608-dd3f-4b28-9ab5-7f71bdd2fbd1]sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
&park


From steveayre at gmail.com  Thu May 19 20:19:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 19 May 2011 17:19:19 +0100
Subject: [Freeswitch-dev] SIP endpoint can be registered several times
In-Reply-To: <BANLkTikEEWJZqfvJ2NPHB3fxWMdxFhDmrA@mail.gmail.com>
References: <BANLkTinpN7fZxiLNv0bt=D8cTwP2hakRzg@mail.gmail.com>
	<BANLkTinT=uifmr1Y5eg3s8TF4rYkmsVYMg@mail.gmail.com>
	<BANLkTikHmNqNoAxR2ASJi8u+F4NGGcR39A@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F79DBEDC@cooper>
	<BANLkTikEEWJZqfvJ2NPHB3fxWMdxFhDmrA@mail.gmail.com>
Message-ID: <BANLkTin=sQ0fgTeLtTJzZQh08LKDC7XhZw@mail.gmail.com>

That makes them separate SIP dialogs, so they are separate registrations.

-Steve



On 19 May 2011 16:13, Anton VG <anton.vazir at gmail.com> wrote:

> The only difference is Call-ID line - suppose that is not enough.
>
>
> Call-ID:        b7e81237-d749ef63 at 192.168.5.21
> User:           sip3779100 at 192.168.100.11
> Contact:        "sip3779100"
> <sip:sip3779100 at 192.168.5.21:5062
> ;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062>
> Agent:          Linksys/SPA942-6.1.3(a)
> Status:         Registered(UDP-NAT)(unknown) EXP(2011-05-19 21:05:40)
> EXPSECS(3288)
> Host:           lab3
> IP:             192.168.5.21
> Port:           5062
> Auth-User:      sip3779100
> Auth-Realm:     192.168.100.11
> MWI-Account:    sip3779100 at 192.168.100.11
>
>
> Call-ID:        ab03086b-91d64ebf at 192.168.5.21
> User:           sip3779100 at 192.168.100.11
> Contact:        "sip3779100"
> <sip:sip3779100 at 192.168.5.21:5062
> ;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062>
> Agent:          Linksys/SPA942-6.1.3(a)
> Status:         Registered(UDP-NAT)(unknown) EXP(2011-05-19 21:00:56)
> EXPSECS(3004)
> Host:           lab3
> IP:             192.168.5.21
> Port:           5062
> Auth-User:      sip3779100
> Auth-Realm:     192.168.100.11
> MWI-Account:    sip3779100 at 192.168.100.11
>
>
> 2011/5/19 Peter Olsson <peter.olsson at visionutveckling.se>:
> > Try "sofia status profile internal" (or whatever profile you are using),
> I bet you'll see that the registrations are not identical...
> >
> > /Peter
> >
> > -----Ursprungligt meddelande-----
> > Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] F?r Anton VG
> > Skickat: den 19 maj 2011 16:39
> > Till: freeswitch-dev at lists.freeswitch.org
> > ?mne: Re: [Freeswitch-dev] SIP endpoint can be registered several times
> >
> > I understand that. If you look at the credentials, there are exactly
> > the same, so what the point of keeping several exact registrations?
> >
> > 2011/5/19 Steven Ayre <steveayre at gmail.com>:
> >> Registrations expire after a certain amount of time, until then you'll
> have
> >> more than one registration.
> >>
> >> -Steve
> >>
> >>
> >> On 19 May 2011 14:11, Anton VG <anton.vazir at gmail.com> wrote:
> >>>
> >>> Just noted an issue (or feature :) - I ve been switching on/off my sip
> >>> phone by powering it on/off from power plug, and now it's registered
> >>> several time with the same properties. I suppose that is kind of
> >>> strange.
> >>> I found that whle looking for a reason, why there is more than a
> >>> single UUID on originate, while originating a single end point.
> >>>
> >>> Should I report the given on JIRA?
> >>>
> >>> freeswitch at lab3> sofia_contact sip3779100
> >>>
> >>>
> >>> sofia/internal/sip:sip3779100 at 192.168.5.21:5062
> ;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062
> ;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062
> ;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062
> ;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
> >>> freeswitch at lab3>
> >>>
> >>> _______________________________________________
> >>> 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
> >>
> >>
> >> _______________________________________________
> >> 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
> >>
> >>
> >
> > _______________________________________________
> > 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
> >
> > !DSPAM:4dd52b9532769872813097!
> >
> >
> > _______________________________________________
> > 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
> >
>
> _______________________________________________
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110519/366dda11/attachment.html 

From anthony.minessale at gmail.com  Thu May 19 21:01:21 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 19 May 2011 12:01:21 -0500
Subject: [Freeswitch-dev] SIP endpoint can be registered several times
In-Reply-To: <BANLkTin=sQ0fgTeLtTJzZQh08LKDC7XhZw@mail.gmail.com>
References: <BANLkTinpN7fZxiLNv0bt=D8cTwP2hakRzg@mail.gmail.com>
	<BANLkTinT=uifmr1Y5eg3s8TF4rYkmsVYMg@mail.gmail.com>
	<BANLkTikHmNqNoAxR2ASJi8u+F4NGGcR39A@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F79DBEDC@cooper>
	<BANLkTikEEWJZqfvJ2NPHB3fxWMdxFhDmrA@mail.gmail.com>
	<BANLkTin=sQ0fgTeLtTJzZQh08LKDC7XhZw@mail.gmail.com>
Message-ID: <BANLkTi=r3D62YmhMrYZJH8RzpfO0gsAU_A@mail.gmail.com>

you are using a phone that does not properly support a multiple
registrations environment.
Disable it in sofia or change the phone.


On Thu, May 19, 2011 at 11:19 AM, Steven Ayre <steveayre at gmail.com> wrote:
> That makes them separate SIP dialogs, so they are separate registrations.
>
> -Steve
>
>
>
> On 19 May 2011 16:13, Anton VG <anton.vazir at gmail.com> wrote:
>>
>> The only difference is Call-ID line - suppose that is not enough.
>>
>>
>> Call-ID: ? ? ? ?b7e81237-d749ef63 at 192.168.5.21
>> User: ? ? ? ? ? sip3779100 at 192.168.100.11
>> Contact: ? ? ? ?"sip3779100"
>>
>> <sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062>
>> Agent: ? ? ? ? ?Linksys/SPA942-6.1.3(a)
>> Status: ? ? ? ? Registered(UDP-NAT)(unknown) EXP(2011-05-19 21:05:40)
>> EXPSECS(3288)
>> Host: ? ? ? ? ? lab3
>> IP: ? ? ? ? ? ? 192.168.5.21
>> Port: ? ? ? ? ? 5062
>> Auth-User: ? ? ?sip3779100
>> Auth-Realm: ? ? 192.168.100.11
>> MWI-Account: ? ?sip3779100 at 192.168.100.11
>>
>>
>> Call-ID: ? ? ? ?ab03086b-91d64ebf at 192.168.5.21
>> User: ? ? ? ? ? sip3779100 at 192.168.100.11
>> Contact: ? ? ? ?"sip3779100"
>>
>> <sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062>
>> Agent: ? ? ? ? ?Linksys/SPA942-6.1.3(a)
>> Status: ? ? ? ? Registered(UDP-NAT)(unknown) EXP(2011-05-19 21:00:56)
>> EXPSECS(3004)
>> Host: ? ? ? ? ? lab3
>> IP: ? ? ? ? ? ? 192.168.5.21
>> Port: ? ? ? ? ? 5062
>> Auth-User: ? ? ?sip3779100
>> Auth-Realm: ? ? 192.168.100.11
>> MWI-Account: ? ?sip3779100 at 192.168.100.11
>>
>>
>> 2011/5/19 Peter Olsson <peter.olsson at visionutveckling.se>:
>> > Try "sofia status profile internal" (or whatever profile you are using),
>> > I bet you'll see that the registrations are not identical...
>> >
>> > /Peter
>> >
>> > -----Ursprungligt meddelande-----
>> > Fr?n: freeswitch-dev-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Anton VG
>> > Skickat: den 19 maj 2011 16:39
>> > Till: freeswitch-dev at lists.freeswitch.org
>> > ?mne: Re: [Freeswitch-dev] SIP endpoint can be registered several times
>> >
>> > I understand that. If you look at the credentials, there are exactly
>> > the same, so what the point of keeping several exact registrations?
>> >
>> > 2011/5/19 Steven Ayre <steveayre at gmail.com>:
>> >> Registrations expire after a certain amount of time, until then you'll
>> >> have
>> >> more than one registration.
>> >>
>> >> -Steve
>> >>
>> >>
>> >> On 19 May 2011 14:11, Anton VG <anton.vazir at gmail.com> wrote:
>> >>>
>> >>> Just noted an issue (or feature :) - I ve been switching on/off my sip
>> >>> phone by powering it on/off from power plug, and now it's registered
>> >>> several time with the same properties. I suppose that is kind of
>> >>> strange.
>> >>> I found that whle looking for a reason, why there is more than a
>> >>> single UUID on originate, while originating a single end point.
>> >>>
>> >>> Should I report the given on JIRA?
>> >>>
>> >>> freeswitch at lab3> sofia_contact sip3779100
>> >>>
>> >>>
>> >>>
>> >>> sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
>> >>> freeswitch at lab3>
>> >>>
>> >>> _______________________________________________
>> >>> 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
>> >>
>> >>
>> >> _______________________________________________
>> >> 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
>> >>
>> >>
>> >
>> > _______________________________________________
>> > 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
>> >
>> > !DSPAM:4dd52b9532769872813097!
>> >
>> >
>> > _______________________________________________
>> > 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
>> >
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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


From anthony.minessale at gmail.com  Thu May 19 21:13:02 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 19 May 2011 12:13:02 -0500
Subject: [Freeswitch-dev] SIP fantom calls,
 when calling disconnected SIP endpoint. EP called when plugged in.
 suppose bug?
In-Reply-To: <BANLkTi=1gZVDTJ1Zu0ZpZVtSE+4DzEonVA@mail.gmail.com>
References: <BANLkTi=X+sa4O-gjU=h=qYeWWaGPRAUNOg@mail.gmail.com>
	<BANLkTinjfQaqZUi=ccr+ueXCo5MM8k+OYQ@mail.gmail.com>
	<BANLkTi=1gZVDTJ1Zu0ZpZVtSE+4DzEonVA@mail.gmail.com>
Message-ID: <BANLkTinvi90dpUEz=KL7KkPB8G6bM0G4UQ@mail.gmail.com>

I am trying to understand your explanation but based on what you say
if you think you can call uuid_kill on a job uuid you are totally
wrong.

If you want to know the uuid before originate is over you would need
to provide it as origination_uuid

bgapi originate {origination_uuid=cluecon}sofia/foo/foo at bar.com 9999

now you can do

uuid_kill cluecon at will

job uuid is only the uuid the result will be in.

Please stop assuming everything is a bug.





On Thu, May 19, 2011 at 10:38 AM, Anton VG <anton.vazir at gmail.com> wrote:
> Adding progress_timeout and leg_progress_timeout did not changed it behavior
>
> originate {originate_retries=0,originate_timeout=60,progress_timeout=5,leg_timeout=60,origination_caller_id_number=907388397,ringback='%(2000,4000,440.0,480.0)',origination_uuid=f7f2d66d-d5e0-4b77-9fda-bae5556b6c8f}[continue_on_fail=false,leg_progress_timeout=7,leg_a_uuid=9bf89608-dd3f-4b28-9ab5-7f71bdd2fbd1]sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
> &park
>
> _______________________________________________
> 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


From steveayre at gmail.com  Thu May 19 21:30:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 19 May 2011 18:30:44 +0100
Subject: [Freeswitch-dev] SIP fantom calls,
 when calling disconnected SIP endpoint. EP called when plugged in.
 suppose bug?
In-Reply-To: <BANLkTinvi90dpUEz=KL7KkPB8G6bM0G4UQ@mail.gmail.com>
References: <BANLkTi=X+sa4O-gjU=h=qYeWWaGPRAUNOg@mail.gmail.com>
	<BANLkTinjfQaqZUi=ccr+ueXCo5MM8k+OYQ@mail.gmail.com>
	<BANLkTi=1gZVDTJ1Zu0ZpZVtSE+4DzEonVA@mail.gmail.com>
	<BANLkTinvi90dpUEz=KL7KkPB8G6bM0G4UQ@mail.gmail.com>
Message-ID: <BANLkTimYxgke_qqAP-Fk7ofu8QEfpvETfA@mail.gmail.com>

Also, there's a handy API call to create a UUID so you don't need to
reinvent the wheel:

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

-Steve


On 19 May 2011 18:13, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> I am trying to understand your explanation but based on what you say
> if you think you can call uuid_kill on a job uuid you are totally
> wrong.
>
> If you want to know the uuid before originate is over you would need
> to provide it as origination_uuid
>
> bgapi originate {origination_uuid=cluecon}sofia/foo/foo at bar.com 9999
>
> now you can do
>
> uuid_kill cluecon at will
>
> job uuid is only the uuid the result will be in.
>
> Please stop assuming everything is a bug.
>
>
>
>
>
> On Thu, May 19, 2011 at 10:38 AM, Anton VG <anton.vazir at gmail.com> wrote:
> > Adding progress_timeout and leg_progress_timeout did not changed it
> behavior
> >
> > originate
> {originate_retries=0,originate_timeout=60,progress_timeout=5,leg_timeout=60,origination_caller_id_number=907388397,ringback='%(2000,4000,440.0,480.0)',origination_uuid=f7f2d66d-d5e0-4b77-9fda-bae5556b6c8f}[continue_on_fail=false,leg_progress_timeout=7,leg_a_uuid=9bf89608-dd3f-4b28-9ab5-7f71bdd2fbd1]sofia/internal/sip:sip3779100 at 192.168.5.21:5062
> ;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
> > &park
> >
> > _______________________________________________
> > 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
>
> _______________________________________________
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110519/61a4ce4b/attachment-0001.html 

From anton.vazir at gmail.com  Thu May 19 23:49:12 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Fri, 20 May 2011 00:49:12 +0500
Subject: [Freeswitch-dev] SIP fantom calls,
 when calling disconnected SIP endpoint. EP called when plugged in.
 suppose bug?
In-Reply-To: <BANLkTinvi90dpUEz=KL7KkPB8G6bM0G4UQ@mail.gmail.com>
References: <BANLkTi=X+sa4O-gjU=h=qYeWWaGPRAUNOg@mail.gmail.com>
	<BANLkTinjfQaqZUi=ccr+ueXCo5MM8k+OYQ@mail.gmail.com>
	<BANLkTi=1gZVDTJ1Zu0ZpZVtSE+4DzEonVA@mail.gmail.com>
	<BANLkTinvi90dpUEz=KL7KkPB8G6bM0G4UQ@mail.gmail.com>
Message-ID: <BANLkTi=j7LnRMLqiSArdUyUVe6WnX72HwQ@mail.gmail.com>

I was killing not job uuid, I expressed it wrong. I'm killing call
UUID, previously created by create_uuid

Except this, the meaning is the same, and phantom calls persits ;)

2011/5/19 Anthony Minessale <anthony.minessale at gmail.com>:
> I am trying to understand your explanation but based on what you say
> if you think you can call uuid_kill on a job uuid you are totally
> wrong.
>
> If you want to know the uuid before originate is over you would need
> to provide it as origination_uuid
>
> bgapi originate {origination_uuid=cluecon}sofia/foo/foo at bar.com 9999
>
> now you can do
>
> uuid_kill cluecon at will
>
> job uuid is only the uuid the result will be in.
>
> Please stop assuming everything is a bug.
>
>
>
>
>
> On Thu, May 19, 2011 at 10:38 AM, Anton VG <anton.vazir at gmail.com> wrote:
>> Adding progress_timeout and leg_progress_timeout did not changed it behavior
>>
>> originate {originate_retries=0,originate_timeout=60,progress_timeout=5,leg_timeout=60,origination_caller_id_number=907388397,ringback='%(2000,4000,440.0,480.0)',origination_uuid=f7f2d66d-d5e0-4b77-9fda-bae5556b6c8f}[continue_on_fail=false,leg_progress_timeout=7,leg_a_uuid=9bf89608-dd3f-4b28-9ab5-7f71bdd2fbd1]sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
>> &park
>>
>> _______________________________________________
>> 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
>
> _______________________________________________
> 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 anton.vazir at gmail.com  Thu May 19 23:57:22 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Fri, 20 May 2011 00:57:22 +0500
Subject: [Freeswitch-dev] SIP endpoint can be registered several times
In-Reply-To: <BANLkTi=r3D62YmhMrYZJH8RzpfO0gsAU_A@mail.gmail.com>
References: <BANLkTinpN7fZxiLNv0bt=D8cTwP2hakRzg@mail.gmail.com>
	<BANLkTinT=uifmr1Y5eg3s8TF4rYkmsVYMg@mail.gmail.com>
	<BANLkTikHmNqNoAxR2ASJi8u+F4NGGcR39A@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F79DBEDC@cooper>
	<BANLkTikEEWJZqfvJ2NPHB3fxWMdxFhDmrA@mail.gmail.com>
	<BANLkTin=sQ0fgTeLtTJzZQh08LKDC7XhZw@mail.gmail.com>
	<BANLkTi=r3D62YmhMrYZJH8RzpfO0gsAU_A@mail.gmail.com>
Message-ID: <BANLkTikRV9G6SSj0GnPt4AznQCfhgp-38Q@mail.gmail.com>

I'll do the same test on my Nokia N900, which is using the same sofia
stack :) will write back he result

hm... should phone know anything about multiple registrations?

Actually, I have filtered duplicate values in ESL handler, Everything
works, my concern if you guys wish to know that such things happens
with FS, or not, you decide.

2011/5/19 Anthony Minessale <anthony.minessale at gmail.com>:
> you are using a phone that does not properly support a multiple
> registrations environment.
> Disable it in sofia or change the phone.
>
>
> On Thu, May 19, 2011 at 11:19 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> That makes them separate SIP dialogs, so they are separate registrations.
>>
>> -Steve
>>
>>
>>
>> On 19 May 2011 16:13, Anton VG <anton.vazir at gmail.com> wrote:
>>>
>>> The only difference is Call-ID line - suppose that is not enough.
>>>
>>>
>>> Call-ID: ? ? ? ?b7e81237-d749ef63 at 192.168.5.21
>>> User: ? ? ? ? ? sip3779100 at 192.168.100.11
>>> Contact: ? ? ? ?"sip3779100"
>>>
>>> <sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062>
>>> Agent: ? ? ? ? ?Linksys/SPA942-6.1.3(a)
>>> Status: ? ? ? ? Registered(UDP-NAT)(unknown) EXP(2011-05-19 21:05:40)
>>> EXPSECS(3288)
>>> Host: ? ? ? ? ? lab3
>>> IP: ? ? ? ? ? ? 192.168.5.21
>>> Port: ? ? ? ? ? 5062
>>> Auth-User: ? ? ?sip3779100
>>> Auth-Realm: ? ? 192.168.100.11
>>> MWI-Account: ? ?sip3779100 at 192.168.100.11
>>>
>>>
>>> Call-ID: ? ? ? ?ab03086b-91d64ebf at 192.168.5.21
>>> User: ? ? ? ? ? sip3779100 at 192.168.100.11
>>> Contact: ? ? ? ?"sip3779100"
>>>
>>> <sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062>
>>> Agent: ? ? ? ? ?Linksys/SPA942-6.1.3(a)
>>> Status: ? ? ? ? Registered(UDP-NAT)(unknown) EXP(2011-05-19 21:00:56)
>>> EXPSECS(3004)
>>> Host: ? ? ? ? ? lab3
>>> IP: ? ? ? ? ? ? 192.168.5.21
>>> Port: ? ? ? ? ? 5062
>>> Auth-User: ? ? ?sip3779100
>>> Auth-Realm: ? ? 192.168.100.11
>>> MWI-Account: ? ?sip3779100 at 192.168.100.11
>>>
>>>
>>> 2011/5/19 Peter Olsson <peter.olsson at visionutveckling.se>:
>>> > Try "sofia status profile internal" (or whatever profile you are using),
>>> > I bet you'll see that the registrations are not identical...
>>> >
>>> > /Peter
>>> >
>>> > -----Ursprungligt meddelande-----
>>> > Fr?n: freeswitch-dev-bounces at lists.freeswitch.org
>>> > [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Anton VG
>>> > Skickat: den 19 maj 2011 16:39
>>> > Till: freeswitch-dev at lists.freeswitch.org
>>> > ?mne: Re: [Freeswitch-dev] SIP endpoint can be registered several times
>>> >
>>> > I understand that. If you look at the credentials, there are exactly
>>> > the same, so what the point of keeping several exact registrations?
>>> >
>>> > 2011/5/19 Steven Ayre <steveayre at gmail.com>:
>>> >> Registrations expire after a certain amount of time, until then you'll
>>> >> have
>>> >> more than one registration.
>>> >>
>>> >> -Steve
>>> >>
>>> >>
>>> >> On 19 May 2011 14:11, Anton VG <anton.vazir at gmail.com> wrote:
>>> >>>
>>> >>> Just noted an issue (or feature :) - I ve been switching on/off my sip
>>> >>> phone by powering it on/off from power plug, and now it's registered
>>> >>> several time with the same properties. I suppose that is kind of
>>> >>> strange.
>>> >>> I found that whle looking for a reason, why there is more than a
>>> >>> single UUID on originate, while originating a single end point.
>>> >>>
>>> >>> Should I report the given on JIRA?
>>> >>>
>>> >>> freeswitch at lab3> sofia_contact sip3779100
>>> >>>
>>> >>>
>>> >>>
>>> >>> sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062,sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
>>> >>> freeswitch at lab3>
>>> >>>
>>> >>> _______________________________________________
>>> >>> 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
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> 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
>>> >>
>>> >>
>>> >
>>> > _______________________________________________
>>> > 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
>>> >
>>> > !DSPAM:4dd52b9532769872813097!
>>> >
>>> >
>>> > _______________________________________________
>>> > 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
>>> >
>>>
>>> _______________________________________________
>>> 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
>>
>>
>> _______________________________________________
>> 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
>
> _______________________________________________
> 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 anton.vazir at gmail.com  Fri May 20 00:37:44 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Fri, 20 May 2011 01:37:44 +0500
Subject: [Freeswitch-dev] SIP endpoint can be registered several times
In-Reply-To: <BANLkTikRV9G6SSj0GnPt4AznQCfhgp-38Q@mail.gmail.com>
References: <BANLkTinpN7fZxiLNv0bt=D8cTwP2hakRzg@mail.gmail.com>
	<BANLkTinT=uifmr1Y5eg3s8TF4rYkmsVYMg@mail.gmail.com>
	<BANLkTikHmNqNoAxR2ASJi8u+F4NGGcR39A@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F79DBEDC@cooper>
	<BANLkTikEEWJZqfvJ2NPHB3fxWMdxFhDmrA@mail.gmail.com>
	<BANLkTin=sQ0fgTeLtTJzZQh08LKDC7XhZw@mail.gmail.com>
	<BANLkTi=r3D62YmhMrYZJH8RzpfO0gsAU_A@mail.gmail.com>
	<BANLkTikRV9G6SSj0GnPt4AznQCfhgp-38Q@mail.gmail.com>
Message-ID: <BANLkTim99e4GkMZvSWOamTih10iFS4X_pA@mail.gmail.com>

Similar stuff happens with SOFIA base sip endpoint (Nokia N900 cell
phone, integrated sip client) - only difference that it has
registration expiration MUCH shorted than
CISCO/LinkSys SPA942 - business class telephone. Since it's behind NAT
- contact line differs, so only one instance will be contacted. I
suppose If I will use it with no NAT it will be the same as for
SPA942. Anyway that stuff is not critical in any sence.

freeswitch at default> sofia_contact sip3779100
sofia/internal/sip:sip3779100 at 192.168.1.78:62216;transport=udp;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.1.78%3A62216%3Btransport%3Dudp,sofia/internal/sip:sip3779100 at 192.168.1.78:57180;transport=udp;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.1.78%3A57180%3Btransport%3Dudp
freeswitch at default>

Registrations:
=================================================================================================
Call-ID:        b48531de-fcf9-122e-f08a-5c57c8780000
User:           sip3779100 at 192.168.100.11
Contact:        "user"
<sip:sip3779100 at 192.168.1.78:57180;transport=udp;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.1.78%3A57180%3Btransport%3Dudp>
Agent:          Telepathy-SofiaSIP/0.6.2 sofia-sip/1.12.10devel
Status:         Registered(UDP-NAT)(unknown) EXP(2011-05-20 01:31:22)
EXPSECS(40)
Host:           lab3
IP:             192.168.1.78
Port:           57180
Auth-User:      sip3779100
Auth-Realm:     192.168.100.11
MWI-Account:    sip3779100 at 192.168.100.11

Call-ID:        07021fba-fcfa-122e-7e8d-5c57c8780000
User:           sip3779100 at 192.168.100.11
Contact:        "user"
<sip:sip3779100 at 192.168.1.78:49438;transport=udp;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.1.78%3A49438%3Btransport%3Dudp>
Agent:          Telepathy-SofiaSIP/0.6.2 sofia-sip/1.12.10devel
Status:         Registered(UDP-NAT)(unknown) EXP(2011-05-20 01:32:03)
EXPSECS(81)
Host:           lab3
IP:             192.168.1.78
Port:           49438
Auth-User:      sip3779100
Auth-Realm:     192.168.100.11
MWI-Account:    sip3779100 at 192.168.100.11

Total items returned: 2
=================================================================================================


From anton.vazir at gmail.com  Fri May 20 01:10:41 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Fri, 20 May 2011 02:10:41 +0500
Subject: [Freeswitch-dev] SIP fantom calls,
 when calling disconnected SIP endpoint. EP called when plugged in.
 suppose bug?
In-Reply-To: <BANLkTi=j7LnRMLqiSArdUyUVe6WnX72HwQ@mail.gmail.com>
References: <BANLkTi=X+sa4O-gjU=h=qYeWWaGPRAUNOg@mail.gmail.com>
	<BANLkTinjfQaqZUi=ccr+ueXCo5MM8k+OYQ@mail.gmail.com>
	<BANLkTi=1gZVDTJ1Zu0ZpZVtSE+4DzEonVA@mail.gmail.com>
	<BANLkTinvi90dpUEz=KL7KkPB8G6bM0G4UQ@mail.gmail.com>
	<BANLkTi=j7LnRMLqiSArdUyUVe6WnX72HwQ@mail.gmail.com>
Message-ID: <BANLkTi=m-q1bKR_ew5hAYqu3njLiaY4+Lg@mail.gmail.com>

originate_timeout does not play role.
Reproducibility is somewhere around 100%-30% (sometimes with every
call, sometimes i have to do that 2-3 times to catch it).

- Switch off the phone,
- make call,
- switch it on,
- wait a little to call appear.

but looks like the phone should be plugged in fast, otherwise this not happens.

2011/5/20 Anton VG <anton.vazir at gmail.com>:
> I was killing not job uuid, I expressed it wrong. I'm killing call
> UUID, previously created by create_uuid
>
> Except this, the meaning is the same, and phantom calls persits ;)
>
> 2011/5/19 Anthony Minessale <anthony.minessale at gmail.com>:
>> I am trying to understand your explanation but based on what you say
>> if you think you can call uuid_kill on a job uuid you are totally
>> wrong.
>>
>> If you want to know the uuid before originate is over you would need
>> to provide it as origination_uuid
>>
>> bgapi originate {origination_uuid=cluecon}sofia/foo/foo at bar.com 9999
>>
>> now you can do
>>
>> uuid_kill cluecon at will
>>
>> job uuid is only the uuid the result will be in.
>>
>> Please stop assuming everything is a bug.
>>
>>
>>
>>
>>
>> On Thu, May 19, 2011 at 10:38 AM, Anton VG <anton.vazir at gmail.com> wrote:
>>> Adding progress_timeout and leg_progress_timeout did not changed it behavior
>>>
>>> originate {originate_retries=0,originate_timeout=60,progress_timeout=5,leg_timeout=60,origination_caller_id_number=907388397,ringback='%(2000,4000,440.0,480.0)',origination_uuid=f7f2d66d-d5e0-4b77-9fda-bae5556b6c8f}[continue_on_fail=false,leg_progress_timeout=7,leg_a_uuid=9bf89608-dd3f-4b28-9ab5-7f71bdd2fbd1]sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
>>> &park
>>>
>>> _______________________________________________
>>> 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
>>
>> _______________________________________________
>> 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 anthony.minessale at gmail.com  Fri May 20 21:37:24 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 20 May 2011 12:37:24 -0500
Subject: [Freeswitch-dev] SIP fantom calls,
 when calling disconnected SIP endpoint. EP called when plugged in.
 suppose bug?
In-Reply-To: <BANLkTi=m-q1bKR_ew5hAYqu3njLiaY4+Lg@mail.gmail.com>
References: <BANLkTi=X+sa4O-gjU=h=qYeWWaGPRAUNOg@mail.gmail.com>
	<BANLkTinjfQaqZUi=ccr+ueXCo5MM8k+OYQ@mail.gmail.com>
	<BANLkTi=1gZVDTJ1Zu0ZpZVtSE+4DzEonVA@mail.gmail.com>
	<BANLkTinvi90dpUEz=KL7KkPB8G6bM0G4UQ@mail.gmail.com>
	<BANLkTi=j7LnRMLqiSArdUyUVe6WnX72HwQ@mail.gmail.com>
	<BANLkTi=m-q1bKR_ew5hAYqu3njLiaY4+Lg@mail.gmail.com>
Message-ID: <BANLkTimpexgNmg64a=9Dt76i-6f2kujHSA@mail.gmail.com>

you would need to provide a log.

sofia global siptrace on
console loglevel debug

reproduce and attach to pastebin.freeswitch.org


On Thu, May 19, 2011 at 4:10 PM, Anton VG <anton.vazir at gmail.com> wrote:
> originate_timeout does not play role.
> Reproducibility is somewhere around 100%-30% (sometimes with every
> call, sometimes i have to do that 2-3 times to catch it).
>
> - Switch off the phone,
> - make call,
> - switch it on,
> - wait a little to call appear.
>
> but looks like the phone should be plugged in fast, otherwise this not happens.
>
> 2011/5/20 Anton VG <anton.vazir at gmail.com>:
>> I was killing not job uuid, I expressed it wrong. I'm killing call
>> UUID, previously created by create_uuid
>>
>> Except this, the meaning is the same, and phantom calls persits ;)
>>
>> 2011/5/19 Anthony Minessale <anthony.minessale at gmail.com>:
>>> I am trying to understand your explanation but based on what you say
>>> if you think you can call uuid_kill on a job uuid you are totally
>>> wrong.
>>>
>>> If you want to know the uuid before originate is over you would need
>>> to provide it as origination_uuid
>>>
>>> bgapi originate {origination_uuid=cluecon}sofia/foo/foo at bar.com 9999
>>>
>>> now you can do
>>>
>>> uuid_kill cluecon at will
>>>
>>> job uuid is only the uuid the result will be in.
>>>
>>> Please stop assuming everything is a bug.
>>>
>>>
>>>
>>>
>>>
>>> On Thu, May 19, 2011 at 10:38 AM, Anton VG <anton.vazir at gmail.com> wrote:
>>>> Adding progress_timeout and leg_progress_timeout did not changed it behavior
>>>>
>>>> originate {originate_retries=0,originate_timeout=60,progress_timeout=5,leg_timeout=60,origination_caller_id_number=907388397,ringback='%(2000,4000,440.0,480.0)',origination_uuid=f7f2d66d-d5e0-4b77-9fda-bae5556b6c8f}[continue_on_fail=false,leg_progress_timeout=7,leg_a_uuid=9bf89608-dd3f-4b28-9ab5-7f71bdd2fbd1]sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
>>>> &park
>>>>
>>>> _______________________________________________
>>>> 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
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>
> _______________________________________________
> 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


From anton.vazir at gmail.com  Fri May 20 23:02:13 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 21 May 2011 00:02:13 +0500
Subject: [Freeswitch-dev] SIP fantom calls,
 when calling disconnected SIP endpoint. EP called when plugged in.
 suppose bug?
In-Reply-To: <BANLkTimpexgNmg64a=9Dt76i-6f2kujHSA@mail.gmail.com>
References: <BANLkTi=X+sa4O-gjU=h=qYeWWaGPRAUNOg@mail.gmail.com>
	<BANLkTinjfQaqZUi=ccr+ueXCo5MM8k+OYQ@mail.gmail.com>
	<BANLkTi=1gZVDTJ1Zu0ZpZVtSE+4DzEonVA@mail.gmail.com>
	<BANLkTinvi90dpUEz=KL7KkPB8G6bM0G4UQ@mail.gmail.com>
	<BANLkTi=j7LnRMLqiSArdUyUVe6WnX72HwQ@mail.gmail.com>
	<BANLkTi=m-q1bKR_ew5hAYqu3njLiaY4+Lg@mail.gmail.com>
	<BANLkTimpexgNmg64a=9Dt76i-6f2kujHSA@mail.gmail.com>
Message-ID: <BANLkTi=Jw0VnBERrcw3bqs0UYLj=D33j5w@mail.gmail.com>

Anthony, believe or not: There is no debug log. Only SIPTRACE, but
it's given already above.
Debug log appears if I do not kill a UUID, if UUID killed, no log
(except siptrace)

2011/5/20 Anthony Minessale <anthony.minessale at gmail.com>:
> you would need to provide a log.
>
> sofia global siptrace on
> console loglevel debug
>
> reproduce and attach to pastebin.freeswitch.org
>
>
> On Thu, May 19, 2011 at 4:10 PM, Anton VG <anton.vazir at gmail.com> wrote:
>> originate_timeout does not play role.
>> Reproducibility is somewhere around 100%-30% (sometimes with every
>> call, sometimes i have to do that 2-3 times to catch it).
>>
>> - Switch off the phone,
>> - make call,
>> - switch it on,
>> - wait a little to call appear.
>>
>> but looks like the phone should be plugged in fast, otherwise this not happens.
>>
>> 2011/5/20 Anton VG <anton.vazir at gmail.com>:
>>> I was killing not job uuid, I expressed it wrong. I'm killing call
>>> UUID, previously created by create_uuid
>>>
>>> Except this, the meaning is the same, and phantom calls persits ;)
>>>
>>> 2011/5/19 Anthony Minessale <anthony.minessale at gmail.com>:
>>>> I am trying to understand your explanation but based on what you say
>>>> if you think you can call uuid_kill on a job uuid you are totally
>>>> wrong.
>>>>
>>>> If you want to know the uuid before originate is over you would need
>>>> to provide it as origination_uuid
>>>>
>>>> bgapi originate {origination_uuid=cluecon}sofia/foo/foo at bar.com 9999
>>>>
>>>> now you can do
>>>>
>>>> uuid_kill cluecon at will
>>>>
>>>> job uuid is only the uuid the result will be in.
>>>>
>>>> Please stop assuming everything is a bug.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, May 19, 2011 at 10:38 AM, Anton VG <anton.vazir at gmail.com> wrote:
>>>>> Adding progress_timeout and leg_progress_timeout did not changed it behavior
>>>>>
>>>>> originate {originate_retries=0,originate_timeout=60,progress_timeout=5,leg_timeout=60,origination_caller_id_number=907388397,ringback='%(2000,4000,440.0,480.0)',origination_uuid=f7f2d66d-d5e0-4b77-9fda-bae5556b6c8f}[continue_on_fail=false,leg_progress_timeout=7,leg_a_uuid=9bf89608-dd3f-4b28-9ab5-7f71bdd2fbd1]sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
>>>>> &park
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>
>>
>> _______________________________________________
>> 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
>
> _______________________________________________
> 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 anton.vazir at gmail.com  Fri May 20 23:13:49 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 21 May 2011 00:13:49 +0500
Subject: [Freeswitch-dev] SIP fantom calls,
 when calling disconnected SIP endpoint. EP called when plugged in.
 suppose bug?
In-Reply-To: <BANLkTi=Jw0VnBERrcw3bqs0UYLj=D33j5w@mail.gmail.com>
References: <BANLkTi=X+sa4O-gjU=h=qYeWWaGPRAUNOg@mail.gmail.com>
	<BANLkTinjfQaqZUi=ccr+ueXCo5MM8k+OYQ@mail.gmail.com>
	<BANLkTi=1gZVDTJ1Zu0ZpZVtSE+4DzEonVA@mail.gmail.com>
	<BANLkTinvi90dpUEz=KL7KkPB8G6bM0G4UQ@mail.gmail.com>
	<BANLkTi=j7LnRMLqiSArdUyUVe6WnX72HwQ@mail.gmail.com>
	<BANLkTi=m-q1bKR_ew5hAYqu3njLiaY4+Lg@mail.gmail.com>
	<BANLkTimpexgNmg64a=9Dt76i-6f2kujHSA@mail.gmail.com>
	<BANLkTi=Jw0VnBERrcw3bqs0UYLj=D33j5w@mail.gmail.com>
Message-ID: <BANLkTin+x8xn6qcuJ8vTgbeitOXrAvCDtw@mail.gmail.com>

Sending debug to you directly as attachment.

and siptrace, which appeared exactly when SIP fantom call comes is below:

freeswitch at lab3> send 1185 bytes to udp/[192.168.5.21]:5062 at 19:10:19.075023:
   ------------------------------------------------------------------------
   INVITE sip:sip3779100 at 192.168.5.21:5062 SIP/2.0
   Via: SIP/2.0/UDP 192.168.100.11;rport;branch=z9hG4bKFgg44amt4Z8BN
   Route: <sip:sip3779100 at 192.168.5.21:5062>
   Max-Forwards: 70
   From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
   To: <sip:sip3779100 at 192.168.5.21:5062>
   Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
   CSeq: 12623230 INVITE
   Contact: <sip:mod_sofia at 192.168.100.11:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fccbba5 2011-05-18
19-00-42 -0400
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
sla, include-session-description, presence.winfo, message-summary,
refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 209
   X-MT-Context: None
   X-FS-Support: update_display
   Remote-Party-ID:
<sip:0 at 192.168.100.11>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1305894626 1305894627 IN IP4 192.168.100.11
   s=FreeSWITCH
   c=IN IP4 192.168.100.11
   t=0 0
   m=audio 23962 RTP/AVP 0 8 3 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 296 bytes from udp/[192.168.5.21]:5062 at 19:10:19.088162:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   To: <sip:sip3779100 at 192.168.5.21:5062>
   From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
   Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
   CSeq: 12623230 INVITE
   Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKFgg44amt4Z8BN
   Server: Linksys/SPA942-6.1.3(a)
   Content-Length: 0

   ------------------------------------------------------------------------
send 406 bytes to udp/[192.168.5.21]:5062 at 19:10:19.088319:
   ------------------------------------------------------------------------
   CANCEL sip:sip3779100 at 192.168.5.21:5062 SIP/2.0
   Via: SIP/2.0/UDP 192.168.100.11;rport;branch=z9hG4bKFgg44amt4Z8BN
   Route: <sip:sip3779100 at 192.168.5.21:5062>
   Max-Forwards: 70
   From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
   To: <sip:sip3779100 at 192.168.5.21:5062>
   Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
   CSeq: 12623230 CANCEL
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 348 bytes from udp/[192.168.5.21]:5062 at 19:10:19.101833:
   ------------------------------------------------------------------------
   SIP/2.0 481 Call Leg/Transaction Does Not Exist
   To: <sip:sip3779100 at 192.168.5.21:5062>;tag=ae5521c6a6820f1ai2
   From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
   Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
   CSeq: 12623230 CANCEL
   Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKFgg44amt4Z8BN
   Server: Linksys/SPA942-6.1.3(a)
   Content-Length: 0

   ------------------------------------------------------------------------
recv 378 bytes from udp/[192.168.5.21]:5062 at 19:10:19.109452:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   To: <sip:sip3779100 at 192.168.5.21:5062>;tag=53489ed715c1d150i2
   From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
   Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
   CSeq: 12623230 INVITE
   Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKFgg44amt4Z8BN
   Contact: "sip3779100" <sip:sip3779100 at 192.168.5.21:5062>
   Server: Linksys/SPA942-6.1.3(a)
   Content-Length: 0

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


2011/5/21 Anton VG <anton.vazir at gmail.com>:
> Anthony, believe or not: There is no debug log. Only SIPTRACE, but
> it's given already above.
> Debug log appears if I do not kill a UUID, if UUID killed, no log
> (except siptrace)
>
> 2011/5/20 Anthony Minessale <anthony.minessale at gmail.com>:
>> you would need to provide a log.
>>
>> sofia global siptrace on
>> console loglevel debug
>>
>> reproduce and attach to pastebin.freeswitch.org
>>
>>
>> On Thu, May 19, 2011 at 4:10 PM, Anton VG <anton.vazir at gmail.com> wrote:
>>> originate_timeout does not play role.
>>> Reproducibility is somewhere around 100%-30% (sometimes with every
>>> call, sometimes i have to do that 2-3 times to catch it).
>>>
>>> - Switch off the phone,
>>> - make call,
>>> - switch it on,
>>> - wait a little to call appear.
>>>
>>> but looks like the phone should be plugged in fast, otherwise this not happens.
>>>
>>> 2011/5/20 Anton VG <anton.vazir at gmail.com>:
>>>> I was killing not job uuid, I expressed it wrong. I'm killing call
>>>> UUID, previously created by create_uuid
>>>>
>>>> Except this, the meaning is the same, and phantom calls persits ;)
>>>>
>>>> 2011/5/19 Anthony Minessale <anthony.minessale at gmail.com>:
>>>>> I am trying to understand your explanation but based on what you say
>>>>> if you think you can call uuid_kill on a job uuid you are totally
>>>>> wrong.
>>>>>
>>>>> If you want to know the uuid before originate is over you would need
>>>>> to provide it as origination_uuid
>>>>>
>>>>> bgapi originate {origination_uuid=cluecon}sofia/foo/foo at bar.com 9999
>>>>>
>>>>> now you can do
>>>>>
>>>>> uuid_kill cluecon at will
>>>>>
>>>>> job uuid is only the uuid the result will be in.
>>>>>
>>>>> Please stop assuming everything is a bug.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, May 19, 2011 at 10:38 AM, Anton VG <anton.vazir at gmail.com> wrote:
>>>>>> Adding progress_timeout and leg_progress_timeout did not changed it behavior
>>>>>>
>>>>>> originate {originate_retries=0,originate_timeout=60,progress_timeout=5,leg_timeout=60,origination_caller_id_number=907388397,ringback='%(2000,4000,440.0,480.0)',origination_uuid=f7f2d66d-d5e0-4b77-9fda-bae5556b6c8f}[continue_on_fail=false,leg_progress_timeout=7,leg_a_uuid=9bf89608-dd3f-4b28-9ab5-7f71bdd2fbd1]sofia/internal/sip:sip3779100 at 192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
>>>>>> &park
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>
>> _______________________________________________
>> 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 steveayre at gmail.com  Sat May 21 08:18:52 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 21 May 2011 05:18:52 +0100
Subject: [Freeswitch-dev] SIP fantom calls,
 when calling disconnected SIP endpoint. EP called when plugged in.
 suppose bug?
In-Reply-To: <BANLkTin+x8xn6qcuJ8vTgbeitOXrAvCDtw@mail.gmail.com>
References: <BANLkTi=X+sa4O-gjU=h=qYeWWaGPRAUNOg@mail.gmail.com>
	<BANLkTinjfQaqZUi=ccr+ueXCo5MM8k+OYQ@mail.gmail.com>
	<BANLkTi=1gZVDTJ1Zu0ZpZVtSE+4DzEonVA@mail.gmail.com>
	<BANLkTinvi90dpUEz=KL7KkPB8G6bM0G4UQ@mail.gmail.com>
	<BANLkTi=j7LnRMLqiSArdUyUVe6WnX72HwQ@mail.gmail.com>
	<BANLkTi=m-q1bKR_ew5hAYqu3njLiaY4+Lg@mail.gmail.com>
	<BANLkTimpexgNmg64a=9Dt76i-6f2kujHSA@mail.gmail.com>
	<BANLkTi=Jw0VnBERrcw3bqs0UYLj=D33j5w@mail.gmail.com>
	<BANLkTin+x8xn6qcuJ8vTgbeitOXrAvCDtw@mail.gmail.com>
Message-ID: <BANLkTin0rK7Wf-30m5rrn8v5CwixEr3dGQ@mail.gmail.com>

' console loglevel debug' only controls the log level of mod_console (i.e.
when you start FS in foreground with -c)

fs_cli uses its own command "/log debug"

Connect using fs_cli and run:
> /log debug
> sofia global siptrace on

And you should then get debug output.

-Steve


On 20 May 2011 20:13, Anton VG <anton.vazir at gmail.com> wrote:

> Sending debug to you directly as attachment.
>
> and siptrace, which appeared exactly when SIP fantom call comes is below:
>
> freeswitch at lab3> send 1185 bytes to udp/[192.168.5.21]:5062 at
> 19:10:19.075023:
>    ------------------------------------------------------------------------
>   INVITE sip:sip3779100 at 192.168.5.21:5062 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.100.11;rport;branch=z9hG4bKFgg44amt4Z8BN
>    Route: <sip:sip3779100 at 192.168.5.21:5062>
>   Max-Forwards: 70
>    From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
>   To: <sip:sip3779100 at 192.168.5.21:5062>
>   Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
>   CSeq: 12623230 INVITE
>    Contact: <sip:mod_sofia at 192.168.100.11:5060>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fccbba5 2011-05-18
> 19-00-42 -0400
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>   Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> sla, include-session-description, presence.winfo, message-summary,
> refer
>   Content-Type: application/sdp
>   Content-Disposition: session
>   Content-Length: 209
>    X-MT-Context: None
>    X-FS-Support: update_display
>   Remote-Party-ID:
> <sip:0 at 192.168.100.11>;party=calling;screen=yes;privacy=off
>
>   v=0
>   o=FreeSWITCH 1305894626 1305894627 IN IP4 192.168.100.11
>    s=FreeSWITCH
>   c=IN IP4 192.168.100.11
>   t=0 0
>    m=audio 23962 RTP/AVP 0 8 3 101 13
>    a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-16
>   a=ptime:20
>   ------------------------------------------------------------------------
> recv 296 bytes from udp/[192.168.5.21]:5062 at 19:10:19.088162:
>    ------------------------------------------------------------------------
>   SIP/2.0 100 Trying
>    To: <sip:sip3779100 at 192.168.5.21:5062>
>   From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
>   Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
>   CSeq: 12623230 INVITE
>   Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKFgg44amt4Z8BN
>    Server: Linksys/SPA942-6.1.3(a)
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
> send 406 bytes to udp/[192.168.5.21]:5062 at 19:10:19.088319:
>   ------------------------------------------------------------------------
>   CANCEL sip:sip3779100 at 192.168.5.21:5062 SIP/2.0
>   Via: SIP/2.0/UDP 192.168.100.11;rport;branch=z9hG4bKFgg44amt4Z8BN
>    Route: <sip:sip3779100 at 192.168.5.21:5062>
>   Max-Forwards: 70
>    From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
>   To: <sip:sip3779100 at 192.168.5.21:5062>
>   Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
>   CSeq: 12623230 CANCEL
>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
> recv 348 bytes from udp/[192.168.5.21]:5062 at 19:10:19.101833:
>    ------------------------------------------------------------------------
>   SIP/2.0 481 Call Leg/Transaction Does Not Exist
>    To: <sip:sip3779100 at 192.168.5.21:5062>;tag=ae5521c6a6820f1ai2
>   From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
>   Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
>   CSeq: 12623230 CANCEL
>   Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKFgg44amt4Z8BN
>    Server: Linksys/SPA942-6.1.3(a)
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
> recv 378 bytes from udp/[192.168.5.21]:5062 at 19:10:19.109452:
>    ------------------------------------------------------------------------
>   SIP/2.0 180 Ringing
>    To: <sip:sip3779100 at 192.168.5.21:5062>;tag=53489ed715c1d150i2
>   From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
>   Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
>   CSeq: 12623230 INVITE
>   Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKFgg44amt4Z8BN
>   Contact: "sip3779100" <sip:sip3779100 at 192.168.5.21:5062>
>    Server: Linksys/SPA942-6.1.3(a)
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
>
>
> 2011/5/21 Anton VG <anton.vazir at gmail.com>:
> > Anthony, believe or not: There is no debug log. Only SIPTRACE, but
> > it's given already above.
> > Debug log appears if I do not kill a UUID, if UUID killed, no log
> > (except siptrace)
> >
> > 2011/5/20 Anthony Minessale <anthony.minessale at gmail.com>:
> >> you would need to provide a log.
> >>
> >> sofia global siptrace on
> >> console loglevel debug
> >>
> >> reproduce and attach to pastebin.freeswitch.org
> >>
> >>
> >> On Thu, May 19, 2011 at 4:10 PM, Anton VG <anton.vazir at gmail.com>
> wrote:
> >>> originate_timeout does not play role.
> >>> Reproducibility is somewhere around 100%-30% (sometimes with every
> >>> call, sometimes i have to do that 2-3 times to catch it).
> >>>
> >>> - Switch off the phone,
> >>> - make call,
> >>> - switch it on,
> >>> - wait a little to call appear.
> >>>
> >>> but looks like the phone should be plugged in fast, otherwise this not
> happens.
> >>>
> >>> 2011/5/20 Anton VG <anton.vazir at gmail.com>:
> >>>> I was killing not job uuid, I expressed it wrong. I'm killing call
> >>>> UUID, previously created by create_uuid
> >>>>
> >>>> Except this, the meaning is the same, and phantom calls persits ;)
> >>>>
> >>>> 2011/5/19 Anthony Minessale <anthony.minessale at gmail.com>:
> >>>>> I am trying to understand your explanation but based on what you say
> >>>>> if you think you can call uuid_kill on a job uuid you are totally
> >>>>> wrong.
> >>>>>
> >>>>> If you want to know the uuid before originate is over you would need
> >>>>> to provide it as origination_uuid
> >>>>>
> >>>>> bgapi originate {origination_uuid=cluecon}sofia/foo/foo at bar.com 9999
> >>>>>
> >>>>> now you can do
> >>>>>
> >>>>> uuid_kill cluecon at will
> >>>>>
> >>>>> job uuid is only the uuid the result will be in.
> >>>>>
> >>>>> Please stop assuming everything is a bug.
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>> On Thu, May 19, 2011 at 10:38 AM, Anton VG <anton.vazir at gmail.com>
> wrote:
> >>>>>> Adding progress_timeout and leg_progress_timeout did not changed it
> behavior
> >>>>>>
> >>>>>> originate
> {originate_retries=0,originate_timeout=60,progress_timeout=5,leg_timeout=60,origination_caller_id_number=907388397,ringback='%(2000,4000,440.0,480.0)',origination_uuid=f7f2d66d-d5e0-4b77-9fda-bae5556b6c8f}[continue_on_fail=false,leg_progress_timeout=7,leg_a_uuid=9bf89608-dd3f-4b28-9ab5-7f71bdd2fbd1]sofia/internal/sip:sip3779100 at 192.168.5.21:5062
> ;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
> >>>>>> &park
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> 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
> >>>>>
> >>>>> _______________________________________________
> >>>>> 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
> >>>>>
> >>>>
> >>>
> >>> _______________________________________________
> >>> 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
> >>
> >> _______________________________________________
> >> 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
> >>
> >
>
> _______________________________________________
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110521/ad4bdb8a/attachment-0001.html 

From jaybinks at gmail.com  Sat May 21 09:15:27 2011
From: jaybinks at gmail.com (jay binks)
Date: Sat, 21 May 2011 15:15:27 +1000
Subject: [Freeswitch-dev] RTCP patch to send source reports in the
 RECV_RTCP_MESSAGE event
In-Reply-To: <AANLkTim830YXB0NF9KJxt11VKeC4--oboJWy2SU39GlE@mail.gmail.com>
References: <1F5B435178035E45A495F36C97955C39B62641FDA0@domainserver.HMBEINT.LOCAL>
	<AANLkTinFxSgty-22g3LFN5LDUNy8l6NAX3Z24Rm3Q4NS@mail.gmail.com>
	<AANLkTim830YXB0NF9KJxt11VKeC4--oboJWy2SU39GlE@mail.gmail.com>
Message-ID: <BANLkTi=OqC3hfM+RbkKVMEGp666X52n0dw@mail.gmail.com>

did this patch ever make it to jira / git ?

Jay

On Wed, Jun 30, 2010 at 5:08 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> yes thanks for the patch but can you submit it to jira
> http://jira.freeswitch.org so we can track the workflow.
>
>
> 2010/6/29 Jo?o Mesquita <jmesquita at freeswitch.org>
>
>> This belongs to Jira, not here.
>>
>> Regards,
>> Jo?o Mesquita
>>
>>
>> On Tue, Jun 29, 2010 at 7:16 AM, Guillaume Binet <
>> G.Binet at mondialtelecom.be> wrote:
>>
>>> Hi all,
>>>
>>> Please find attached a patch that parses and add RTCP source reports and
>>> added the info to the corresponding the switch event.
>>>
>>> It adds the following headers, numbered by source :
>>>
>>> Event arrived : RECV_RTCP_MESSAGE
>>>     Event Header : Event-Name -> RECV_RTCP_MESSAGE
>>>     Event Header : Core-UUID -> 649cbe4e-7bb8-4edd-a0f5-2662246966f8
>>>     Event Header : FreeSWITCH-Hostname -> sal
>>>     Event Header : FreeSWITCH-IPv4 -> 192.168.0.89
>>>     Event Header : FreeSWITCH-IPv6 -> ::1
>>>     Event Header : Event-Date-Local -> 2010-06-29 12:11:00
>>>     Event Header : Event-Date-GMT -> Tue, 29 Jun 2010 10:11:00 GMT
>>>     Event Header : Event-Date-Timestamp -> 1277806260737911
>>>     Event Header : Event-Calling-File -> mod_sofia.c
>>>     Event Header : Event-Calling-Function -> sofia_read_frame
>>>     Event Header : Event-Calling-Line-Number -> 907
>>>     Event Header : Unique-ID -> 4fa7d00b-64f2-47ff-b2db-266943ee5fe2
>>>     Event Header : SSRC -> 402076d6
>>>     Event Header : NTP-Most-Significant-Word -> 3486795060
>>>     Event Header : NTP-Least-Significant-Word -> 3111085330
>>>     Event Header : RTP-Timestamp -> 40160
>>>     Event Header : Sender-Packet-Count -> 246
>>>     Event Header : Octect-Packet-Count -> 39360
>>>     Event Header : Last-RTP-Timestamp -> 39840
>>>     Event Header : RTP-Rate -> 8000
>>>     Event Header : Capture-Time -> 1277806260738059
>>> --> HERE
>>>     Event Header : Source0-SSRC -> 18320497
>>>     Event Header : Source0-Fraction -> 0
>>>     Event Header : Source0-Lost -> 0
>>>     Event Header : Source0-Highest-Sequence-Number-Received -> 40032
>>>     Event Header : Source0-Jitter -> 5
>>>     Event Header : Source0-LSR -> 0
>>>     Event Header : Source0-DLSR -> 0
>>>
>>> Guillaume BINET.
>>> Guillaume Binet
>>> Mobile: +32 487 57 35 81
>>> Fax: +32 2 223 01 15
>>>
>>> Ch. de la Hulpe 181 Terhulpsesteenweg
>>> Bruxelles          B-1170            Brussel
>>> www.mondialtelecom.be
>>>
>>> To discover the innovation of Mondial Telecom,
>>> becherry.be
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>
> _______________________________________________
> 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
>
>


-- 
Sincerely

Jay
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110521/c908dda5/attachment.html 

From anton.vazir at gmail.com  Sat May 21 11:12:10 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 21 May 2011 12:12:10 +0500
Subject: [Freeswitch-dev] Skypopen segfault
Message-ID: <BANLkTik2xHU4ZXgPkHEDGdyQs+=tejec9A@mail.gmail.com>

Giovanny, I've got several segfaults of FS,
seems they belongs to SKYPOPEN


Core was generated by `./freeswitch -core -nc'.
Program terminated with signal 11, Segmentation fault.
#0  0x00007f53e7234028 in switch_core_session_get_channel
(session=0x0) at src/switch_core_session.c:1082
1082            switch_assert(session->channel);
(gdb) bt full
#0  0x00007f53e7234028 in switch_core_session_get_channel
(session=0x0) at src/switch_core_session.c:1082
        __PRETTY_FUNCTION__ = "switch_core_session_get_channel"
#1  0x00007f53ce23dd0a in dtmf_received (tech_pvt=0x7f53ce470af0,
value=0x7f53cd8c6560 "2") at mod_skypopen.c:2138
        session = 0x0
        channel = 0x0
        __func__ = "dtmf_received"
#2  0x00007f53ce24a1dd in skypopen_signaling_read
(tech_pvt=0x7f53ce470af0) at skypopen_protocol.c:537
        read_from_pipe = "CALL 41977 DTMF 2", '\000' <repeats 4078 times>
        message = "CALL\000\064\061\071\067\067\000DTMF\000\062",
'\000' <repeats 4078 times>
        message_2 = "CALL 41977 DTMF 2", '\000' <repeats 4078 times>
        buf = 0x0
        obj = "CALL", '\000' <repeats 507 times>
        id = "41977", '\000' <repeats 506 times>
        prop = "DTMF", '\000' <repeats 507 times>
        value = "2", '\000' <repeats 510 times>
        where = 0x0
        stringp = 0x7f53cd8c6d68
        a = 18
        howmany = 18
        i = 17
        __func__ = "skypopen_signaling_read"
#3  0x00007f53ce236c65 in skypopen_signaling_thread_func
(thread=0x7f53d80a6378, obj=0x7f53ce470af0) at mod_skypopen.c:1399
        tech_pvt = 0x7f53ce470af0
        res = 0
        forever = 1
        event = 0x0
        __func__ = "skypopen_signaling_thread_func"
#4  0x00007f53e72e1317 in dummy_worker (opaque=0x7f53d80a6378) at
threadproc/unix/thread.c:138
        thread = 0x7f53d80a6378
#5  0x00007f53e67628ba in start_thread () from /lib/libpthread.so.0
No symbol table info available.
#6  0x00007f53e5c8e02d in clone () from /lib/libc.so.6
No symbol table info available.
#7  0x0000000000000000 in ?? ()
No symbol table info available.
(gdb)


From gmaruzz at gmail.com  Sat May 21 11:27:57 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 21 May 2011 09:27:57 +0200
Subject: [Freeswitch-dev] Skypopen segfault
In-Reply-To: <BANLkTik2xHU4ZXgPkHEDGdyQs+=tejec9A@mail.gmail.com>
References: <BANLkTik2xHU4ZXgPkHEDGdyQs+=tejec9A@mail.gmail.com>
Message-ID: <BANLkTim52Jk_2OyopT1kkuBsrPzY0AQK4g@mail.gmail.com>

Please open a Jira issue with all related info, particularly how to
reproduce this bug and a debug listing.

Important is that you give a clear way to reproduce the bug, ie: a
step by step way to have the same result (crash in this case).

-giovanni

On Sat, May 21, 2011 at 9:12 AM, Anton VG <anton.vazir at gmail.com> wrote:
> Giovanny, I've got several segfaults of FS,
> seems they belongs to SKYPOPEN
>
>
> Core was generated by `./freeswitch -core -nc'.
> Program terminated with signal 11, Segmentation fault.
> #0 ?0x00007f53e7234028 in switch_core_session_get_channel
> (session=0x0) at src/switch_core_session.c:1082
> 1082 ? ? ? ? ? ?switch_assert(session->channel);
> (gdb) bt full
> #0 ?0x00007f53e7234028 in switch_core_session_get_channel
> (session=0x0) at src/switch_core_session.c:1082
> ? ? ? ?__PRETTY_FUNCTION__ = "switch_core_session_get_channel"
> #1 ?0x00007f53ce23dd0a in dtmf_received (tech_pvt=0x7f53ce470af0,
> value=0x7f53cd8c6560 "2") at mod_skypopen.c:2138
> ? ? ? ?session = 0x0
> ? ? ? ?channel = 0x0
> ? ? ? ?__func__ = "dtmf_received"
> #2 ?0x00007f53ce24a1dd in skypopen_signaling_read
> (tech_pvt=0x7f53ce470af0) at skypopen_protocol.c:537
> ? ? ? ?read_from_pipe = "CALL 41977 DTMF 2", '\000' <repeats 4078 times>
> ? ? ? ?message = "CALL\000\064\061\071\067\067\000DTMF\000\062",
> '\000' <repeats 4078 times>
> ? ? ? ?message_2 = "CALL 41977 DTMF 2", '\000' <repeats 4078 times>
> ? ? ? ?buf = 0x0
> ? ? ? ?obj = "CALL", '\000' <repeats 507 times>
> ? ? ? ?id = "41977", '\000' <repeats 506 times>
> ? ? ? ?prop = "DTMF", '\000' <repeats 507 times>
> ? ? ? ?value = "2", '\000' <repeats 510 times>
> ? ? ? ?where = 0x0
> ? ? ? ?stringp = 0x7f53cd8c6d68
> ? ? ? ?a = 18
> ? ? ? ?howmany = 18
> ? ? ? ?i = 17
> ? ? ? ?__func__ = "skypopen_signaling_read"
> #3 ?0x00007f53ce236c65 in skypopen_signaling_thread_func
> (thread=0x7f53d80a6378, obj=0x7f53ce470af0) at mod_skypopen.c:1399
> ? ? ? ?tech_pvt = 0x7f53ce470af0
> ? ? ? ?res = 0
> ? ? ? ?forever = 1
> ? ? ? ?event = 0x0
> ? ? ? ?__func__ = "skypopen_signaling_thread_func"
> #4 ?0x00007f53e72e1317 in dummy_worker (opaque=0x7f53d80a6378) at
> threadproc/unix/thread.c:138
> ? ? ? ?thread = 0x7f53d80a6378
> #5 ?0x00007f53e67628ba in start_thread () from /lib/libpthread.so.0
> No symbol table info available.
> #6 ?0x00007f53e5c8e02d in clone () from /lib/libc.so.6
> No symbol table info available.
> #7 ?0x0000000000000000 in ?? ()
> No symbol table info available.
> (gdb)
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at gmail.com  Sat May 21 12:00:06 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 21 May 2011 10:00:06 +0200
Subject: [Freeswitch-dev] Skypopen segfault
In-Reply-To: <BANLkTim52Jk_2OyopT1kkuBsrPzY0AQK4g@mail.gmail.com>
References: <BANLkTik2xHU4ZXgPkHEDGdyQs+=tejec9A@mail.gmail.com>
	<BANLkTim52Jk_2OyopT1kkuBsrPzY0AQK4g@mail.gmail.com>
Message-ID: <BANLkTimb-=fci3tT5qGQc1nO4PA5SsLOYA@mail.gmail.com>

Anyway, I just added some guards that will avoid the crashes, but I
would still very interested in the procedure to replicate the problem.

Please, let me know how you got this problem, also if it does not bite
you again.

commit 2146583663a5067f6ff71df1e48c3bb900e89db8
Author: Giovanni Maruzzelli <gmaruzz at gmail.com>
Date:   Sat May 21 02:45:39 2011 -0500

    skypopen: adding some guards against NULL sessions and channels

commit 7fa3f7f3dd484a0f9688b9df1fc55e093e6d27f0
Author: Giovanni Maruzzelli <gmaruzz at gmail.com>
Date:   Sat May 21 02:36:51 2011 -0500

    skypopen: fixing bug from Anton VG, adding some guards against
NULL sessions and channels


-giovanni

On Sat, May 21, 2011 at 9:27 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> Please open a Jira issue with all related info, particularly how to
> reproduce this bug and a debug listing.
>
> Important is that you give a clear way to reproduce the bug, ie: a
> step by step way to have the same result (crash in this case).
>
> -giovanni
>
> On Sat, May 21, 2011 at 9:12 AM, Anton VG <anton.vazir at gmail.com> wrote:
>> Giovanny, I've got several segfaults of FS,
>> seems they belongs to SKYPOPEN
>>
>>
>> Core was generated by `./freeswitch -core -nc'.
>> Program terminated with signal 11, Segmentation fault.
>> #0 ?0x00007f53e7234028 in switch_core_session_get_channel
>> (session=0x0) at src/switch_core_session.c:1082
>> 1082 ? ? ? ? ? ?switch_assert(session->channel);
>> (gdb) bt full
>> #0 ?0x00007f53e7234028 in switch_core_session_get_channel
>> (session=0x0) at src/switch_core_session.c:1082
>> ? ? ? ?__PRETTY_FUNCTION__ = "switch_core_session_get_channel"
>> #1 ?0x00007f53ce23dd0a in dtmf_received (tech_pvt=0x7f53ce470af0,
>> value=0x7f53cd8c6560 "2") at mod_skypopen.c:2138
>> ? ? ? ?session = 0x0
>> ? ? ? ?channel = 0x0
>> ? ? ? ?__func__ = "dtmf_received"
>> #2 ?0x00007f53ce24a1dd in skypopen_signaling_read
>> (tech_pvt=0x7f53ce470af0) at skypopen_protocol.c:537
>> ? ? ? ?read_from_pipe = "CALL 41977 DTMF 2", '\000' <repeats 4078 times>
>> ? ? ? ?message = "CALL\000\064\061\071\067\067\000DTMF\000\062",
>> '\000' <repeats 4078 times>
>> ? ? ? ?message_2 = "CALL 41977 DTMF 2", '\000' <repeats 4078 times>
>> ? ? ? ?buf = 0x0
>> ? ? ? ?obj = "CALL", '\000' <repeats 507 times>
>> ? ? ? ?id = "41977", '\000' <repeats 506 times>
>> ? ? ? ?prop = "DTMF", '\000' <repeats 507 times>
>> ? ? ? ?value = "2", '\000' <repeats 510 times>
>> ? ? ? ?where = 0x0
>> ? ? ? ?stringp = 0x7f53cd8c6d68
>> ? ? ? ?a = 18
>> ? ? ? ?howmany = 18
>> ? ? ? ?i = 17
>> ? ? ? ?__func__ = "skypopen_signaling_read"
>> #3 ?0x00007f53ce236c65 in skypopen_signaling_thread_func
>> (thread=0x7f53d80a6378, obj=0x7f53ce470af0) at mod_skypopen.c:1399
>> ? ? ? ?tech_pvt = 0x7f53ce470af0
>> ? ? ? ?res = 0
>> ? ? ? ?forever = 1
>> ? ? ? ?event = 0x0
>> ? ? ? ?__func__ = "skypopen_signaling_thread_func"
>> #4 ?0x00007f53e72e1317 in dummy_worker (opaque=0x7f53d80a6378) at
>> threadproc/unix/thread.c:138
>> ? ? ? ?thread = 0x7f53d80a6378
>> #5 ?0x00007f53e67628ba in start_thread () from /lib/libpthread.so.0
>> No symbol table info available.
>> #6 ?0x00007f53e5c8e02d in clone () from /lib/libc.so.6
>> No symbol table info available.
>> #7 ?0x0000000000000000 in ?? ()
>> No symbol table info available.
>> (gdb)
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From anton.vazir at gmail.com  Sat May 21 13:41:13 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 21 May 2011 14:41:13 +0500
Subject: [Freeswitch-dev] Skypopen segfault
In-Reply-To: <BANLkTimb-=fci3tT5qGQc1nO4PA5SsLOYA@mail.gmail.com>
References: <BANLkTik2xHU4ZXgPkHEDGdyQs+=tejec9A@mail.gmail.com>
	<BANLkTim52Jk_2OyopT1kkuBsrPzY0AQK4g@mail.gmail.com>
	<BANLkTimb-=fci3tT5qGQc1nO4PA5SsLOYA@mail.gmail.com>
Message-ID: <BANLkTi=xJbnrUn4Qc2H=kMUtfMTsWvp5=Q@mail.gmail.com>

Problem it's not easy to reproduce that. I't working on production PC,
as a skype gateway to an asterisk host, serving a bunch of users.
As per recommendations, after the first segfault I've recompiled the
FS with no optimization, and dumping the core. Not sure That I can
produce guidelines to reproduce the given.
Since I have the core, we can trace it deeper, though. I can add that
it happen already 5-6 times, and maybe related to fail of the network
link to neighboring ASTERISK box, while in conversation. But anyway
not sure though. Atleast once it failed without any network failure,
with similar backtrace. Seems it's necessary to analyze the
segfaulting function.

2011/5/21 Giovanni Maruzzelli <gmaruzz at gmail.com>:
> Anyway, I just added some guards that will avoid the crashes, but I
> would still very interested in the procedure to replicate the problem.
>
> Please, let me know how you got this problem, also if it does not bite
> you again.
>
> commit 2146583663a5067f6ff71df1e48c3bb900e89db8
> Author: Giovanni Maruzzelli <gmaruzz at gmail.com>
> Date: ? Sat May 21 02:45:39 2011 -0500
>
> ? ?skypopen: adding some guards against NULL sessions and channels
>
> commit 7fa3f7f3dd484a0f9688b9df1fc55e093e6d27f0
> Author: Giovanni Maruzzelli <gmaruzz at gmail.com>
> Date: ? Sat May 21 02:36:51 2011 -0500
>
> ? ?skypopen: fixing bug from Anton VG, adding some guards against
> NULL sessions and channels
>
>
> -giovanni
>
> On Sat, May 21, 2011 at 9:27 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>> Please open a Jira issue with all related info, particularly how to
>> reproduce this bug and a debug listing.
>>
>> Important is that you give a clear way to reproduce the bug, ie: a
>> step by step way to have the same result (crash in this case).
>>
>> -giovanni
>>
>> On Sat, May 21, 2011 at 9:12 AM, Anton VG <anton.vazir at gmail.com> wrote:
>>> Giovanny, I've got several segfaults of FS,
>>> seems they belongs to SKYPOPEN
>>>
>>>
>>> Core was generated by `./freeswitch -core -nc'.
>>> Program terminated with signal 11, Segmentation fault.
>>> #0 ?0x00007f53e7234028 in switch_core_session_get_channel
>>> (session=0x0) at src/switch_core_session.c:1082
>>> 1082 ? ? ? ? ? ?switch_assert(session->channel);
>>> (gdb) bt full
>>> #0 ?0x00007f53e7234028 in switch_core_session_get_channel
>>> (session=0x0) at src/switch_core_session.c:1082
>>> ? ? ? ?__PRETTY_FUNCTION__ = "switch_core_session_get_channel"
>>> #1 ?0x00007f53ce23dd0a in dtmf_received (tech_pvt=0x7f53ce470af0,
>>> value=0x7f53cd8c6560 "2") at mod_skypopen.c:2138
>>> ? ? ? ?session = 0x0
>>> ? ? ? ?channel = 0x0
>>> ? ? ? ?__func__ = "dtmf_received"
>>> #2 ?0x00007f53ce24a1dd in skypopen_signaling_read
>>> (tech_pvt=0x7f53ce470af0) at skypopen_protocol.c:537
>>> ? ? ? ?read_from_pipe = "CALL 41977 DTMF 2", '\000' <repeats 4078 times>
>>> ? ? ? ?message = "CALL\000\064\061\071\067\067\000DTMF\000\062",
>>> '\000' <repeats 4078 times>
>>> ? ? ? ?message_2 = "CALL 41977 DTMF 2", '\000' <repeats 4078 times>
>>> ? ? ? ?buf = 0x0
>>> ? ? ? ?obj = "CALL", '\000' <repeats 507 times>
>>> ? ? ? ?id = "41977", '\000' <repeats 506 times>
>>> ? ? ? ?prop = "DTMF", '\000' <repeats 507 times>
>>> ? ? ? ?value = "2", '\000' <repeats 510 times>
>>> ? ? ? ?where = 0x0
>>> ? ? ? ?stringp = 0x7f53cd8c6d68
>>> ? ? ? ?a = 18
>>> ? ? ? ?howmany = 18
>>> ? ? ? ?i = 17
>>> ? ? ? ?__func__ = "skypopen_signaling_read"
>>> #3 ?0x00007f53ce236c65 in skypopen_signaling_thread_func
>>> (thread=0x7f53d80a6378, obj=0x7f53ce470af0) at mod_skypopen.c:1399
>>> ? ? ? ?tech_pvt = 0x7f53ce470af0
>>> ? ? ? ?res = 0
>>> ? ? ? ?forever = 1
>>> ? ? ? ?event = 0x0
>>> ? ? ? ?__func__ = "skypopen_signaling_thread_func"
>>> #4 ?0x00007f53e72e1317 in dummy_worker (opaque=0x7f53d80a6378) at
>>> threadproc/unix/thread.c:138
>>> ? ? ? ?thread = 0x7f53d80a6378
>>> #5 ?0x00007f53e67628ba in start_thread () from /lib/libpthread.so.0
>>> No symbol table info available.
>>> #6 ?0x00007f53e5c8e02d in clone () from /lib/libc.so.6
>>> No symbol table info available.
>>> #7 ?0x0000000000000000 in ?? ()
>>> No symbol table info available.
>>> (gdb)
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>


From anton.vazir at gmail.com  Sat May 21 13:43:27 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 21 May 2011 14:43:27 +0500
Subject: [Freeswitch-dev] Skypopen segfault
In-Reply-To: <BANLkTi=xJbnrUn4Qc2H=kMUtfMTsWvp5=Q@mail.gmail.com>
References: <BANLkTik2xHU4ZXgPkHEDGdyQs+=tejec9A@mail.gmail.com>
	<BANLkTim52Jk_2OyopT1kkuBsrPzY0AQK4g@mail.gmail.com>
	<BANLkTimb-=fci3tT5qGQc1nO4PA5SsLOYA@mail.gmail.com>
	<BANLkTi=xJbnrUn4Qc2H=kMUtfMTsWvp5=Q@mail.gmail.com>
Message-ID: <BANLkTik632eiyP5Rq6sZ-Pki_aUQA2-vfQ@mail.gmail.com>

Do you still want me to post a JIRA issue, regardles absence of
gudelines of it's reproduction?

2011/5/21 Anton VG <anton.vazir at gmail.com>:
> Problem it's not easy to reproduce that. I't working on production PC,
> as a skype gateway to an asterisk host, serving a bunch of users.
> As per recommendations, after the first segfault I've recompiled the
> FS with no optimization, and dumping the core. Not sure That I can
> produce guidelines to reproduce the given.
> Since I have the core, we can trace it deeper, though. I can add that
> it happen already 5-6 times, and maybe related to fail of the network
> link to neighboring ASTERISK box, while in conversation. But anyway
> not sure though. Atleast once it failed without any network failure,
> with similar backtrace. Seems it's necessary to analyze the
> segfaulting function.
>
> 2011/5/21 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>> Anyway, I just added some guards that will avoid the crashes, but I
>> would still very interested in the procedure to replicate the problem.
>>
>> Please, let me know how you got this problem, also if it does not bite
>> you again.
>>
>> commit 2146583663a5067f6ff71df1e48c3bb900e89db8
>> Author: Giovanni Maruzzelli <gmaruzz at gmail.com>
>> Date: ? Sat May 21 02:45:39 2011 -0500
>>
>> ? ?skypopen: adding some guards against NULL sessions and channels
>>
>> commit 7fa3f7f3dd484a0f9688b9df1fc55e093e6d27f0
>> Author: Giovanni Maruzzelli <gmaruzz at gmail.com>
>> Date: ? Sat May 21 02:36:51 2011 -0500
>>
>> ? ?skypopen: fixing bug from Anton VG, adding some guards against
>> NULL sessions and channels
>>
>>
>> -giovanni
>>
>> On Sat, May 21, 2011 at 9:27 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>>> Please open a Jira issue with all related info, particularly how to
>>> reproduce this bug and a debug listing.
>>>
>>> Important is that you give a clear way to reproduce the bug, ie: a
>>> step by step way to have the same result (crash in this case).
>>>
>>> -giovanni
>>>
>>> On Sat, May 21, 2011 at 9:12 AM, Anton VG <anton.vazir at gmail.com> wrote:
>>>> Giovanny, I've got several segfaults of FS,
>>>> seems they belongs to SKYPOPEN
>>>>
>>>>
>>>> Core was generated by `./freeswitch -core -nc'.
>>>> Program terminated with signal 11, Segmentation fault.
>>>> #0 ?0x00007f53e7234028 in switch_core_session_get_channel
>>>> (session=0x0) at src/switch_core_session.c:1082
>>>> 1082 ? ? ? ? ? ?switch_assert(session->channel);
>>>> (gdb) bt full
>>>> #0 ?0x00007f53e7234028 in switch_core_session_get_channel
>>>> (session=0x0) at src/switch_core_session.c:1082
>>>> ? ? ? ?__PRETTY_FUNCTION__ = "switch_core_session_get_channel"
>>>> #1 ?0x00007f53ce23dd0a in dtmf_received (tech_pvt=0x7f53ce470af0,
>>>> value=0x7f53cd8c6560 "2") at mod_skypopen.c:2138
>>>> ? ? ? ?session = 0x0
>>>> ? ? ? ?channel = 0x0
>>>> ? ? ? ?__func__ = "dtmf_received"
>>>> #2 ?0x00007f53ce24a1dd in skypopen_signaling_read
>>>> (tech_pvt=0x7f53ce470af0) at skypopen_protocol.c:537
>>>> ? ? ? ?read_from_pipe = "CALL 41977 DTMF 2", '\000' <repeats 4078 times>
>>>> ? ? ? ?message = "CALL\000\064\061\071\067\067\000DTMF\000\062",
>>>> '\000' <repeats 4078 times>
>>>> ? ? ? ?message_2 = "CALL 41977 DTMF 2", '\000' <repeats 4078 times>
>>>> ? ? ? ?buf = 0x0
>>>> ? ? ? ?obj = "CALL", '\000' <repeats 507 times>
>>>> ? ? ? ?id = "41977", '\000' <repeats 506 times>
>>>> ? ? ? ?prop = "DTMF", '\000' <repeats 507 times>
>>>> ? ? ? ?value = "2", '\000' <repeats 510 times>
>>>> ? ? ? ?where = 0x0
>>>> ? ? ? ?stringp = 0x7f53cd8c6d68
>>>> ? ? ? ?a = 18
>>>> ? ? ? ?howmany = 18
>>>> ? ? ? ?i = 17
>>>> ? ? ? ?__func__ = "skypopen_signaling_read"
>>>> #3 ?0x00007f53ce236c65 in skypopen_signaling_thread_func
>>>> (thread=0x7f53d80a6378, obj=0x7f53ce470af0) at mod_skypopen.c:1399
>>>> ? ? ? ?tech_pvt = 0x7f53ce470af0
>>>> ? ? ? ?res = 0
>>>> ? ? ? ?forever = 1
>>>> ? ? ? ?event = 0x0
>>>> ? ? ? ?__func__ = "skypopen_signaling_thread_func"
>>>> #4 ?0x00007f53e72e1317 in dummy_worker (opaque=0x7f53d80a6378) at
>>>> threadproc/unix/thread.c:138
>>>> ? ? ? ?thread = 0x7f53d80a6378
>>>> #5 ?0x00007f53e67628ba in start_thread () from /lib/libpthread.so.0
>>>> No symbol table info available.
>>>> #6 ?0x00007f53e5c8e02d in clone () from /lib/libc.so.6
>>>> No symbol table info available.
>>>> #7 ?0x0000000000000000 in ?? ()
>>>> No symbol table info available.
>>>> (gdb)
>>>>
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>


From gmaruzz at gmail.com  Sat May 21 14:13:29 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 21 May 2011 12:13:29 +0200
Subject: [Freeswitch-dev] Skypopen segfault
In-Reply-To: <BANLkTik632eiyP5Rq6sZ-Pki_aUQA2-vfQ@mail.gmail.com>
References: <BANLkTik2xHU4ZXgPkHEDGdyQs+=tejec9A@mail.gmail.com>
	<BANLkTim52Jk_2OyopT1kkuBsrPzY0AQK4g@mail.gmail.com>
	<BANLkTimb-=fci3tT5qGQc1nO4PA5SsLOYA@mail.gmail.com>
	<BANLkTi=xJbnrUn4Qc2H=kMUtfMTsWvp5=Q@mail.gmail.com>
	<BANLkTik632eiyP5Rq6sZ-Pki_aUQA2-vfQ@mail.gmail.com>
Message-ID: <BANLkTimrcB_UhonzuwUmj_vAKfP2FhPMUg@mail.gmail.com>

No need for a Jira, because I fixed it in git.

It's not related to network.

A skype interface was receiving a DTMF without a channel on it. That
was causing a segfault (my code expected a channel).

Now it will not segfault anymore, it will just give a warning.

I was curious why there was no channel. Maybe a channel was destroyed
(hangup, cancel, etc) at the same time the skype client was getting
the dtmf...

Anyway, is fixed now. Please update to latest git.

Please keep up reporting bugs and issues, that's the way to have an
ever improving software.

Thanks,
-giovanni


On 5/21/11, Anton VG <anton.vazir at gmail.com> wrote:
> Do you still want me to post a JIRA issue, regardles absence of
> gudelines of it's reproduction?
>
> 2011/5/21 Anton VG <anton.vazir at gmail.com>:
>> Problem it's not easy to reproduce that. I't working on production PC,
>> as a skype gateway to an asterisk host, serving a bunch of users.
>> As per recommendations, after the first segfault I've recompiled the
>> FS with no optimization, and dumping the core. Not sure That I can
>> produce guidelines to reproduce the given.
>> Since I have the core, we can trace it deeper, though. I can add that
>> it happen already 5-6 times, and maybe related to fail of the network
>> link to neighboring ASTERISK box, while in conversation. But anyway
>> not sure though. Atleast once it failed without any network failure,
>> with similar backtrace. Seems it's necessary to analyze the
>> segfaulting function.
>>
>> 2011/5/21 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>>> Anyway, I just added some guards that will avoid the crashes, but I
>>> would still very interested in the procedure to replicate the problem.
>>>
>>> Please, let me know how you got this problem, also if it does not bite
>>> you again.
>>>
>>> commit 2146583663a5067f6ff71df1e48c3bb900e89db8
>>> Author: Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> Date: ? Sat May 21 02:45:39 2011 -0500
>>>
>>> ? ?skypopen: adding some guards against NULL sessions and channels
>>>
>>> commit 7fa3f7f3dd484a0f9688b9df1fc55e093e6d27f0
>>> Author: Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> Date: ? Sat May 21 02:36:51 2011 -0500
>>>
>>> ? ?skypopen: fixing bug from Anton VG, adding some guards against
>>> NULL sessions and channels
>>>
>>>
>>> -giovanni
>>>
>>> On Sat, May 21, 2011 at 9:27 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>> Please open a Jira issue with all related info, particularly how to
>>>> reproduce this bug and a debug listing.
>>>>
>>>> Important is that you give a clear way to reproduce the bug, ie: a
>>>> step by step way to have the same result (crash in this case).
>>>>
>>>> -giovanni
>>>>
>>>> On Sat, May 21, 2011 at 9:12 AM, Anton VG <anton.vazir at gmail.com> wrote:
>>>>> Giovanny, I've got several segfaults of FS,
>>>>> seems they belongs to SKYPOPEN
>>>>>
>>>>>
>>>>> Core was generated by `./freeswitch -core -nc'.
>>>>> Program terminated with signal 11, Segmentation fault.
>>>>> #0 ?0x00007f53e7234028 in switch_core_session_get_channel
>>>>> (session=0x0) at src/switch_core_session.c:1082
>>>>> 1082 ? ? ? ? ? ?switch_assert(session->channel);
>>>>> (gdb) bt full
>>>>> #0 ?0x00007f53e7234028 in switch_core_session_get_channel
>>>>> (session=0x0) at src/switch_core_session.c:1082
>>>>> ? ? ? ?__PRETTY_FUNCTION__ = "switch_core_session_get_channel"
>>>>> #1 ?0x00007f53ce23dd0a in dtmf_received (tech_pvt=0x7f53ce470af0,
>>>>> value=0x7f53cd8c6560 "2") at mod_skypopen.c:2138
>>>>> ? ? ? ?session = 0x0
>>>>> ? ? ? ?channel = 0x0
>>>>> ? ? ? ?__func__ = "dtmf_received"
>>>>> #2 ?0x00007f53ce24a1dd in skypopen_signaling_read
>>>>> (tech_pvt=0x7f53ce470af0) at skypopen_protocol.c:537
>>>>> ? ? ? ?read_from_pipe = "CALL 41977 DTMF 2", '\000' <repeats 4078
>>>>> times>
>>>>> ? ? ? ?message = "CALL\000\064\061\071\067\067\000DTMF\000\062",
>>>>> '\000' <repeats 4078 times>
>>>>> ? ? ? ?message_2 = "CALL 41977 DTMF 2", '\000' <repeats 4078 times>
>>>>> ? ? ? ?buf = 0x0
>>>>> ? ? ? ?obj = "CALL", '\000' <repeats 507 times>
>>>>> ? ? ? ?id = "41977", '\000' <repeats 506 times>
>>>>> ? ? ? ?prop = "DTMF", '\000' <repeats 507 times>
>>>>> ? ? ? ?value = "2", '\000' <repeats 510 times>
>>>>> ? ? ? ?where = 0x0
>>>>> ? ? ? ?stringp = 0x7f53cd8c6d68
>>>>> ? ? ? ?a = 18
>>>>> ? ? ? ?howmany = 18
>>>>> ? ? ? ?i = 17
>>>>> ? ? ? ?__func__ = "skypopen_signaling_read"
>>>>> #3 ?0x00007f53ce236c65 in skypopen_signaling_thread_func
>>>>> (thread=0x7f53d80a6378, obj=0x7f53ce470af0) at mod_skypopen.c:1399
>>>>> ? ? ? ?tech_pvt = 0x7f53ce470af0
>>>>> ? ? ? ?res = 0
>>>>> ? ? ? ?forever = 1
>>>>> ? ? ? ?event = 0x0
>>>>> ? ? ? ?__func__ = "skypopen_signaling_thread_func"
>>>>> #4 ?0x00007f53e72e1317 in dummy_worker (opaque=0x7f53d80a6378) at
>>>>> threadproc/unix/thread.c:138
>>>>> ? ? ? ?thread = 0x7f53d80a6378
>>>>> #5 ?0x00007f53e67628ba in start_thread () from /lib/libpthread.so.0
>>>>> No symbol table info available.
>>>>> #6 ?0x00007f53e5c8e02d in clone () from /lib/libc.so.6
>>>>> No symbol table info available.
>>>>> #7 ?0x0000000000000000 in ?? ()
>>>>> No symbol table info available.
>>>>> (gdb)
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From anton.vazir at gmail.com  Sat May 21 14:40:27 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 21 May 2011 15:40:27 +0500
Subject: [Freeswitch-dev] Skypopen segfault
In-Reply-To: <BANLkTimrcB_UhonzuwUmj_vAKfP2FhPMUg@mail.gmail.com>
References: <BANLkTik2xHU4ZXgPkHEDGdyQs+=tejec9A@mail.gmail.com>
	<BANLkTim52Jk_2OyopT1kkuBsrPzY0AQK4g@mail.gmail.com>
	<BANLkTimb-=fci3tT5qGQc1nO4PA5SsLOYA@mail.gmail.com>
	<BANLkTi=xJbnrUn4Qc2H=kMUtfMTsWvp5=Q@mail.gmail.com>
	<BANLkTik632eiyP5Rq6sZ-Pki_aUQA2-vfQ@mail.gmail.com>
	<BANLkTimrcB_UhonzuwUmj_vAKfP2FhPMUg@mail.gmail.com>
Message-ID: <BANLkTi=hpPhkV=S-HHhteSz1=xc3xBi_aQ@mail.gmail.com>

Be sure I will :)
Thanks Man!

2011/5/21 Giovanni Maruzzelli <gmaruzz at gmail.com>:
> No need for a Jira, because I fixed it in git.
>
> It's not related to network.
>
> A skype interface was receiving a DTMF without a channel on it. That
> was causing a segfault (my code expected a channel).
>
> Now it will not segfault anymore, it will just give a warning.
>
> I was curious why there was no channel. Maybe a channel was destroyed
> (hangup, cancel, etc) at the same time the skype client was getting
> the dtmf...
>
> Anyway, is fixed now. Please update to latest git.
>
> Please keep up reporting bugs and issues, that's the way to have an
> ever improving software.
>
> Thanks,
> -giovanni
>
>
> On 5/21/11, Anton VG <anton.vazir at gmail.com> wrote:
>> Do you still want me to post a JIRA issue, regardles absence of
>> gudelines of it's reproduction?
>>
>> 2011/5/21 Anton VG <anton.vazir at gmail.com>:
>>> Problem it's not easy to reproduce that. I't working on production PC,
>>> as a skype gateway to an asterisk host, serving a bunch of users.
>>> As per recommendations, after the first segfault I've recompiled the
>>> FS with no optimization, and dumping the core. Not sure That I can
>>> produce guidelines to reproduce the given.
>>> Since I have the core, we can trace it deeper, though. I can add that
>>> it happen already 5-6 times, and maybe related to fail of the network
>>> link to neighboring ASTERISK box, while in conversation. But anyway
>>> not sure though. Atleast once it failed without any network failure,
>>> with similar backtrace. Seems it's necessary to analyze the
>>> segfaulting function.
>>>
>>> 2011/5/21 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>>>> Anyway, I just added some guards that will avoid the crashes, but I
>>>> would still very interested in the procedure to replicate the problem.
>>>>
>>>> Please, let me know how you got this problem, also if it does not bite
>>>> you again.
>>>>
>>>> commit 2146583663a5067f6ff71df1e48c3bb900e89db8
>>>> Author: Giovanni Maruzzelli <gmaruzz at gmail.com>
>>>> Date: ? Sat May 21 02:45:39 2011 -0500
>>>>
>>>> ? ?skypopen: adding some guards against NULL sessions and channels
>>>>
>>>> commit 7fa3f7f3dd484a0f9688b9df1fc55e093e6d27f0
>>>> Author: Giovanni Maruzzelli <gmaruzz at gmail.com>
>>>> Date: ? Sat May 21 02:36:51 2011 -0500
>>>>
>>>> ? ?skypopen: fixing bug from Anton VG, adding some guards against
>>>> NULL sessions and channels
>>>>
>>>>
>>>> -giovanni
>>>>
>>>> On Sat, May 21, 2011 at 9:27 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>>> wrote:
>>>>> Please open a Jira issue with all related info, particularly how to
>>>>> reproduce this bug and a debug listing.
>>>>>
>>>>> Important is that you give a clear way to reproduce the bug, ie: a
>>>>> step by step way to have the same result (crash in this case).
>>>>>
>>>>> -giovanni
>>>>>
>>>>> On Sat, May 21, 2011 at 9:12 AM, Anton VG <anton.vazir at gmail.com> wrote:
>>>>>> Giovanny, I've got several segfaults of FS,
>>>>>> seems they belongs to SKYPOPEN
>>>>>>
>>>>>>
>>>>>> Core was generated by `./freeswitch -core -nc'.
>>>>>> Program terminated with signal 11, Segmentation fault.
>>>>>> #0 ?0x00007f53e7234028 in switch_core_session_get_channel
>>>>>> (session=0x0) at src/switch_core_session.c:1082
>>>>>> 1082 ? ? ? ? ? ?switch_assert(session->channel);
>>>>>> (gdb) bt full
>>>>>> #0 ?0x00007f53e7234028 in switch_core_session_get_channel
>>>>>> (session=0x0) at src/switch_core_session.c:1082
>>>>>> ? ? ? ?__PRETTY_FUNCTION__ = "switch_core_session_get_channel"
>>>>>> #1 ?0x00007f53ce23dd0a in dtmf_received (tech_pvt=0x7f53ce470af0,
>>>>>> value=0x7f53cd8c6560 "2") at mod_skypopen.c:2138
>>>>>> ? ? ? ?session = 0x0
>>>>>> ? ? ? ?channel = 0x0
>>>>>> ? ? ? ?__func__ = "dtmf_received"
>>>>>> #2 ?0x00007f53ce24a1dd in skypopen_signaling_read
>>>>>> (tech_pvt=0x7f53ce470af0) at skypopen_protocol.c:537
>>>>>> ? ? ? ?read_from_pipe = "CALL 41977 DTMF 2", '\000' <repeats 4078
>>>>>> times>
>>>>>> ? ? ? ?message = "CALL\000\064\061\071\067\067\000DTMF\000\062",
>>>>>> '\000' <repeats 4078 times>
>>>>>> ? ? ? ?message_2 = "CALL 41977 DTMF 2", '\000' <repeats 4078 times>
>>>>>> ? ? ? ?buf = 0x0
>>>>>> ? ? ? ?obj = "CALL", '\000' <repeats 507 times>
>>>>>> ? ? ? ?id = "41977", '\000' <repeats 506 times>
>>>>>> ? ? ? ?prop = "DTMF", '\000' <repeats 507 times>
>>>>>> ? ? ? ?value = "2", '\000' <repeats 510 times>
>>>>>> ? ? ? ?where = 0x0
>>>>>> ? ? ? ?stringp = 0x7f53cd8c6d68
>>>>>> ? ? ? ?a = 18
>>>>>> ? ? ? ?howmany = 18
>>>>>> ? ? ? ?i = 17
>>>>>> ? ? ? ?__func__ = "skypopen_signaling_read"
>>>>>> #3 ?0x00007f53ce236c65 in skypopen_signaling_thread_func
>>>>>> (thread=0x7f53d80a6378, obj=0x7f53ce470af0) at mod_skypopen.c:1399
>>>>>> ? ? ? ?tech_pvt = 0x7f53ce470af0
>>>>>> ? ? ? ?res = 0
>>>>>> ? ? ? ?forever = 1
>>>>>> ? ? ? ?event = 0x0
>>>>>> ? ? ? ?__func__ = "skypopen_signaling_thread_func"
>>>>>> #4 ?0x00007f53e72e1317 in dummy_worker (opaque=0x7f53d80a6378) at
>>>>>> threadproc/unix/thread.c:138
>>>>>> ? ? ? ?thread = 0x7f53d80a6378
>>>>>> #5 ?0x00007f53e67628ba in start_thread () from /lib/libpthread.so.0
>>>>>> No symbol table info available.
>>>>>> #6 ?0x00007f53e5c8e02d in clone () from /lib/libc.so.6
>>>>>> No symbol table info available.
>>>>>> #7 ?0x0000000000000000 in ?? ()
>>>>>> No symbol table info available.
>>>>>> (gdb)
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> Cell : +39-347-2665618
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>
>>
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>


From andyw999 at hotmail.com  Fri May 13 17:58:22 2011
From: andyw999 at hotmail.com (Andy Wood)
Date: Fri, 13 May 2011 14:58:22 +0100
Subject: [Freeswitch-dev] Speex rtpmap negotiation
Message-ID: <BLU151-w2CCCDDC5157893EF1F1E993880@phx.gbl>


I have client software (Win32 and ARM Linux) which uses an old eXosip and is not doing SDP negotiation properly.

The clients use rtpmap:103 and 110 for Speex WB and NB respectively. 

If the clients call each other then everything is OK as FreeSwitch behaves very well and passes the received rtpmap values forward.

Where my difficulty lies is when I use bgapi originate user xxxx yyyy.

What happens then is that FreeSwitch then uses 98 and 99 as the rtpmap values, my client does not like it and the call fails with a BYE Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION".

I have tried to alter the ianacodes in mod_speex to what I want but it does not work.

Please could somebody let me know how to do this as I can't really use G711 due to bandwidth constraints.

Thanks

Andy.
 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110513/dc109fdf/attachment.html 

From barisyanar at gmail.com  Mon May 16 17:12:01 2011
From: barisyanar at gmail.com (barisyanar)
Date: Mon, 16 May 2011 16:12:01 +0300
Subject: [Freeswitch-dev] Sipx TLS connections make Freeswitch crash
Message-ID: <BANLkTinSVup=JqM7zPoBh8tadYkzxJcTBA@mail.gmail.com>

Hi all,

I am using Freeswitch 1.0.7 with sipx release-4.4.
Whenever I try to make an IVR call with TLS, Freeswitch crashes dumping the
attached core file. The output bt command is:

#0  __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131
#1  0x00007f9c0b4373c3 in sofia_glue_get_url_from_contact (buf=0x0,
    to_dup=0 '\000') at sofia_glue.c:4943
#2  0x00007f9c0b438e1b in sofia_overcome_sip_uri_weakness (
    session=0x7f9c04017108, uri=<value optimized out>,
    transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE, params=0x0)
    at sofia_glue.c:1296
#3  0x00007f9c0b4182b0 in sofia_handle_sip_i_invite (nua=0x7f9c04008890,
    profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=<value optimized
out>,
    sip=<value optimized out>, tags=<value optimized out>) at sofia.c:6793
#4  0x00007f9c0b433aca in sofia_event_callback (event=nua_i_invite,
    status=100, phrase=0x7f9c0000ab40 "Trying", nua=0x7f9c04008890,
    profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=0x0,
    sip=0x7f9c00006658, tags=0x7f9c0000ab30) at sofia.c:942
#5  0x00007f9c0b4a5f53 in nua_application_event (dummy=<value optimized
out>,
    sumsg=<value optimized out>, ee=0x7f9c0000ab08) at nua_stack.c:393
#6  0x00007f9c0b503974 in su_base_port_execute_msgs (queue=0x0)
    at su_base_port.c:280
#7  0x00007f9c0b503ec1 in su_base_port_step (self=0x7f9c040010f0, tout=0)
    at su_base_port.c:473
#8  0x00007f9c0b42d4ea in sofia_profile_thread_run (
    thread=<value optimized out>, obj=0x148dad0) at sofia.c:1623
#9  0x0000003efd2068e0 in start_thread (arg=0x7f9c18116710)
---Type <return> to continue, or q <return> to quit---
    at pthread_create.c:297
#10 0x0000003efcee0c9d in clone ()

And sipx's freeswitch log prints the line:

2011-05-16 15:04:51.530426 [WARNING] switch_core.c:1120 Cannot locate domain
deneme.deneme.karelarge.com,

where deneme.deneme.karelarge.com is my fqdn.
No problem occurs during TCP connection.
Problem is possibly caused by sipx, but i'll be glad to hear FS community's
ideas.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110516/960bf0d3/attachment-0001.html 

From kheimerl at cs.berkeley.edu  Thu May 19 04:59:30 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Wed, 18 May 2011 17:59:30 -0700
Subject: [Freeswitch-dev] Project Announcement and Request for Direction
In-Reply-To: <BANLkTimuCyEEkRN8ftJaS+0XCNXoFgd=Jg@mail.gmail.com>
References: <BANLkTimuCyEEkRN8ftJaS+0XCNXoFgd=Jg@mail.gmail.com>
Message-ID: <BANLkTi=tHBL5tZxqE7ObZJ8UrBsZY5UzPQ@mail.gmail.com>

Hello Freeswitch-dev! I originally sent this email to
freeswitch-users, and didn't get a lot of responses. I thought this
might be a better list for this discussion.

My name is Kurtis Heimerl, and I'm a graduate researcher at the
University of California, Berkeley in the Technology and
Infrastructure for Emerging Regions (TIER) group under Eric Brewer.
We're currently investigating the use of OpenBTS
(http://openbts.sourceforge.net/) for providing cellular coverage via
low-cost base stations in low-density parts of the world. This project
is called The Village Base Station. In support of that project, we're
also investigating the use of Freeswitch to support OpenBTS and
provide a flexible, extensible, and simple platform for deploying
voice/sms/data applications on the basestation itself. Towards this
end, I'm asking you, the freeswitch community, for advice and
direction on some of our research goals.

The basic story is simple. OpenBTS uses a software radio and generic
PC to provide cellular service. As a byproduct of this architecture,
we also gain the ability to run freeswitch applications "locally"
(concurrently with OpenBTS), and take advantage of the benefits this
tight coupling provides us. These benefits are numerous; calls/SMS
between BTS users can be much cheaper, as they do not use any backhaul
bandwidth. Applications can query the system for more information
about users, such as location or status. As an example, unlike
traditional GSM telephony applications, we are able to query if users
are currently available on the network. This could be used to create
voice "chat lists", which tell participants which of their friends are
currently within cellular range.

We foresee 6 features required to support these "local" freeswitch
applications on our OpenBTS system. I'm very curious how the
freeswitch community feels about these possible additions, as well how
they might be implemented.

There are a few that are already available in freeswitch, but may be
rougher than we would like. These include:

1) Identity: The ability to query for user's status, numbers, etc.
This seems simple enough in the existing system. However, we'd like to
provide hooks for applications to act on these sign-ons or offs. For
instance, an app may hold messages until a phone logs onto the system
and push them then. My understanding is that this should be simple,
probably hooking onto "SIP presence" events?
2) Storage: Freeswitch currently seems to support only per-application
storage, with limited support for cross-application storage (mostly
user directories). This is occasionally problematic: one issue we've
heard is that it is difficult to place messages into voice mailboxes
from other apps. We'd like a more unified storage framework. Or...
3) "Pipes": This is the ability to pass messages between freeswitch
apps. This seems pretty well supported though simple dialplan
interactions, though the modules themselves may not provide enough
functionality. Is there a way to do this inside of apps? I consider
this an alternative to the storage framework discussed in #2

Lastly, there are three functions we don't believe are well-supported
in freeswitch. These are...
1) Privacy: We expect our BTS to be used in politically sensitive
areas. Given this, freeswitch could provide an anonymity layer,
providing short term phone/SMS numbers, or directing communications
through more secure layers (e.g., Tor).
2) Asynchrony: While freeswitch seems to support basic asynchrony
though its event system, I couldn't find any way to delay events for
indeterminate times. For instance, we may want to schedule a traffic
warning for 1PM every Wednesday to every phone currently on the
system. Is there a way to do that currently?
3) SMS: Freeswitch seems to currently support SIP chat messages (using
SIMPLE?). We need to either extend OpenBTS to speak SIMPLE, or extend
freeswitch to speak OpenBTS's SMS protocol. Neither seems particularly
difficult. This will allow our apps to send and receive both voice and
SMS messages from users.

We believe these core functions will enable a wide variety of BTS
applications. I have a laundry list of those, but I'll omit them for
sake of space.

If any members of the community (that means you!) have any directions,
ideas, projects, or thoughts, please pass them on! We're just
beginning this part of the project, and getting the lay of the land.
Feedback is critical at this point.

Thanks!


From rhuddleston at gmail.com  Thu May 12 16:25:14 2011
From: rhuddleston at gmail.com (Robert-iPhone)
Date: Thu, 12 May 2011 08:25:14 -0400
Subject: [Freeswitch-dev] using freeswich as for SIP signalling and
	programming an external HW.
In-Reply-To: <BANLkTikJdFPeyHqOGjBq7Sd4p0C=53_vSQ@mail.gmail.com>
References: <BANLkTim4ntPv0CbM62UF5+JL-1RzbQpb7Q@mail.gmail.com>
	<BANLkTikJdFPeyHqOGjBq7Sd4p0C=53_vSQ@mail.gmail.com>
Message-ID: <7F6EBD6C-997F-4BFD-9BE6-DD7DC2716CA2@gmail.com>

i think you mean "da bomb" :)


Sent from my iPhone

On May 12, 2011, at 1:41 AM, Prashant Lamba <prashant.lamba at gmail.com> wrote:

> On Wed, Apr 27, 2011 at 10:05 AM, Narendra Sirugudi <nsirugudi at gmail.com> wrote:
> Hi All,
>  
> I want to use freeswich for SIP signalling and program an external HW for RTP processing.
>  
> Does freeswitch provide any mechanism/hooks to program an external HW for RTP processing ?
>  
> thanks,
> --kumar
> 
> _______________________________________________
> 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
> 
> 
> 
> I do believe you need to use Freeswitch in Media Bypass mode. We did the same and it works like a bomb!
> Ofcourse you need to find the way to divert the audio to your HW for RTP capture. Enjoy.
> 
> Prashant
> --
> Phonologies (India)
> 
> To save our tigers, save their habitat. Think before you print this email. <www.saveourtigers.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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110512/ade1fcba/attachment.html 

From steveayre at gmail.com  Sun May 22 01:45:51 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 21 May 2011 22:45:51 +0100
Subject: [Freeswitch-dev] Speex rtpmap negotiation
In-Reply-To: <BLU151-w2CCCDDC5157893EF1F1E993880@phx.gbl>
References: <BLU151-w2CCCDDC5157893EF1F1E993880@phx.gbl>
Message-ID: <48279975-B937-4978-8865-7732230D6076@gmail.com>

Sounds like your phone's sip implementation is broken then... They are dynamic payload types - the phone should accept any number in 96..127 if the rtpmap says it is speex.

Steve on iPhone

On 13 May 2011, at 14:58, Andy Wood <andyw999 at hotmail.com> wrote:

> I have client software (Win32 and ARM Linux) which uses an old eXosip and is not doing SDP negotiation properly.
> 
> The clients use rtpmap:103 and 110 for Speex WB and NB respectively. 
> 
> If the clients call each other then everything is OK as FreeSwitch behaves very well and passes the received rtpmap values forward.
> 
> Where my difficulty lies is when I use bgapi originate user xxxx yyyy.
> 
> What happens then is that FreeSwitch then uses 98 and 99 as the rtpmap values, my client does not like it and the call fails with a BYE Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION".
> 
> I have tried to alter the ianacodes in mod_speex to what I want but it does not work.
> 
> Please could somebody let me know how to do this as I can't really use G711 due to bandwidth constraints.
> 
> Thanks
> 
> Andy.
> _______________________________________________
> 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 anton.vazir at gmail.com  Sun May 22 13:06:26 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sun, 22 May 2011 14:06:26 +0500
Subject: [Freeswitch-dev] Sipx TLS connections make Freeswitch crash
In-Reply-To: <BANLkTinSVup=JqM7zPoBh8tadYkzxJcTBA@mail.gmail.com>
References: <BANLkTinSVup=JqM7zPoBh8tadYkzxJcTBA@mail.gmail.com>
Message-ID: <BANLkTin2B8m7X=yQU2HhfCv9QBQpzce_oQ@mail.gmail.com>

for valuable backtrace, you should compile FS with no optimization
(-O0), running devel-bootstrap.sh will do it for you.

2011/5/16 barisyanar <barisyanar at gmail.com>:
> Hi all,
> I am using Freeswitch 1.0.7 with sipx release-4.4.
> Whenever I try to make an IVR call with TLS, Freeswitch crashes dumping the
> attached core file. The output bt command is:
> #0 ?__strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131
> #1 ?0x00007f9c0b4373c3 in sofia_glue_get_url_from_contact (buf=0x0,
> ?? ?to_dup=0 '\000') at sofia_glue.c:4943
> #2 ?0x00007f9c0b438e1b in sofia_overcome_sip_uri_weakness (
> ?? ?session=0x7f9c04017108, uri=<value optimized out>,
> ?? ?transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE, params=0x0)
> ?? ?at sofia_glue.c:1296
> #3 ?0x00007f9c0b4182b0 in sofia_handle_sip_i_invite (nua=0x7f9c04008890,
> ?? ?profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=<value optimized
> out>,
> ?? ?sip=<value optimized out>, tags=<value optimized out>) at sofia.c:6793
> #4 ?0x00007f9c0b433aca in sofia_event_callback (event=nua_i_invite,
> ?? ?status=100, phrase=0x7f9c0000ab40 "Trying", nua=0x7f9c04008890,
> ?? ?profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=0x0,
> ?? ?sip=0x7f9c00006658, tags=0x7f9c0000ab30) at sofia.c:942
> #5 ?0x00007f9c0b4a5f53 in nua_application_event (dummy=<value optimized
> out>,
> ?? ?sumsg=<value optimized out>, ee=0x7f9c0000ab08) at nua_stack.c:393
> #6 ?0x00007f9c0b503974 in su_base_port_execute_msgs (queue=0x0)
> ?? ?at su_base_port.c:280
> #7 ?0x00007f9c0b503ec1 in su_base_port_step (self=0x7f9c040010f0, tout=0)
> ?? ?at su_base_port.c:473
> #8 ?0x00007f9c0b42d4ea in sofia_profile_thread_run (
> ?? ?thread=<value optimized out>, obj=0x148dad0) at sofia.c:1623
> #9 ?0x0000003efd2068e0 in start_thread (arg=0x7f9c18116710)
> ---Type <return> to continue, or q <return> to quit---
> ?? ?at pthread_create.c:297
> #10 0x0000003efcee0c9d in clone ()
> And sipx's freeswitch log prints the line:
> 2011-05-16 15:04:51.530426 [WARNING] switch_core.c:1120 Cannot locate domain
> deneme.deneme.karelarge.com,
> where?deneme.deneme.karelarge.com is my fqdn.
> No problem occurs during TCP connection.
> Problem is possibly caused by sipx, but i'll be glad to hear FS community's
> ideas.
> _______________________________________________
> 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 anton.vazir at gmail.com  Wed May 25 02:08:29 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 25 May 2011 03:08:29 +0500
Subject: [Freeswitch-dev] No Unique-ID field in BACKGROUND JOB events
Message-ID: <BANLkTinJ1jBk7NoC925N3vGqjn7_pcY8YA@mail.gmail.com>

In a while there was a thread (cand find it) where one of the core FS
developers mentioned that there always should be an Unique-ID field in
the event, so if there is no Unique ID field it can be added. Below is
the event with no Unique-ID

Event received
Event-Name: BACKGROUND_JOB
Core-UUID: bae130be-6719-4854-a0ed-2fdbb2477a12
FreeSWITCH-Hostname: lab3
FreeSWITCH-Switchname: lab3
FreeSWITCH-IPv4: 192.168.100.11
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2011-05-25%2002%3A56%3A20
Event-Date-GMT: Tue,%2024%20May%202011%2021%3A56%3A20%20GMT
Event-Date-Timestamp: 1306274180052931
Event-Calling-File: mod_event_socket.c
Event-Calling-Function: api_exec
Event-Calling-Line-Number: 1424
Job-UUID: 26266606-d9b4-4c9d-9642-792f6912d160
Job-Command: originate
Job-Command-Arg:
%7BMT-fwd_dst_type%3D'Phone%20No',progress_timeout%3D15,MT-fwd_condition%3DALWAYS,originate_timeout%3D90,MT-fwd_oclid%3DTrue,MT-fwd_dst_tech%3DNUM,MT-
fwd_timeout%3DNone,leg_timeout%3D60,origination_caller_id_number%3D433777100,MT-fwd_dst%3D901055490,origination_uuid%3D67829d64-918f-4ecf-8ece-e0b391d05172%7D%5Bsip_h_
X-MT-fwd_dst_tech%3DNUM,sip_h_X-MT-fwd_timeout%3DNone,sip_h_X-MT-fwd_condition%3DALWAYS,continue_on_fail%3Dfalse,leg_a_uuid%3D9c3e5ca3-4081-4cde-8316-6dbd4a1d021c,sip_
h_X-MT-fwd_dst_type%3D'Phone%20No',sip_h_X-MT-fwd_dst%3D901055490,sip_h_X-MT-fwd_oclid%3DTrue%5Dsofia/gateway/fs_local/sip3779050%20%26park
Content-Length: 27

-ERR INVALID_NUMBER_FORMAT


From steveayre at gmail.com  Wed May 25 10:15:48 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 25 May 2011 07:15:48 +0100
Subject: [Freeswitch-dev] No Unique-ID field in BACKGROUND JOB events
In-Reply-To: <BANLkTinJ1jBk7NoC925N3vGqjn7_pcY8YA@mail.gmail.com>
References: <BANLkTinJ1jBk7NoC925N3vGqjn7_pcY8YA@mail.gmail.com>
Message-ID: <BANLkTiniLjDXsVDTvf7kn0VdVSqpbC4acg@mail.gmail.com>

I think that only applies to channel events.

-Steve


On 24 May 2011 23:08, Anton VG <anton.vazir at gmail.com> wrote:

> In a while there was a thread (cand find it) where one of the core FS
> developers mentioned that there always should be an Unique-ID field in
> the event, so if there is no Unique ID field it can be added. Below is
> the event with no Unique-ID
>
> Event received
> Event-Name: BACKGROUND_JOB
> Core-UUID: bae130be-6719-4854-a0ed-2fdbb2477a12
> FreeSWITCH-Hostname: lab3
> FreeSWITCH-Switchname: lab3
> FreeSWITCH-IPv4: 192.168.100.11
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2011-05-25%2002%3A56%3A20
> Event-Date-GMT: Tue,%2024%20May%202011%2021%3A56%3A20%20GMT
> Event-Date-Timestamp: 1306274180052931
> Event-Calling-File: mod_event_socket.c
> Event-Calling-Function: api_exec
> Event-Calling-Line-Number: 1424
> Job-UUID: 26266606-d9b4-4c9d-9642-792f6912d160
> Job-Command: originate
> Job-Command-Arg:
>
> %7BMT-fwd_dst_type%3D'Phone%20No',progress_timeout%3D15,MT-fwd_condition%3DALWAYS,originate_timeout%3D90,MT-fwd_oclid%3DTrue,MT-fwd_dst_tech%3DNUM,MT-
>
> fwd_timeout%3DNone,leg_timeout%3D60,origination_caller_id_number%3D433777100,MT-fwd_dst%3D901055490,origination_uuid%3D67829d64-918f-4ecf-8ece-e0b391d05172%7D%5Bsip_h_
>
> X-MT-fwd_dst_tech%3DNUM,sip_h_X-MT-fwd_timeout%3DNone,sip_h_X-MT-fwd_condition%3DALWAYS,continue_on_fail%3Dfalse,leg_a_uuid%3D9c3e5ca3-4081-4cde-8316-6dbd4a1d021c,sip_
>
> h_X-MT-fwd_dst_type%3D'Phone%20No',sip_h_X-MT-fwd_dst%3D901055490,sip_h_X-MT-fwd_oclid%3DTrue%5Dsofia/gateway/fs_local/sip3779050%20%26park
> Content-Length: 27
>
> -ERR INVALID_NUMBER_FORMAT
>
> _______________________________________________
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110525/a1f881e5/attachment-0001.html 

From peter.olsson at visionutveckling.se  Wed May 25 11:40:43 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 25 May 2011 09:40:43 +0200
Subject: [Freeswitch-dev] No Unique-ID field in BACKGROUND JOB events
In-Reply-To: <BANLkTiniLjDXsVDTvf7kn0VdVSqpbC4acg@mail.gmail.com>
References: <BANLkTinJ1jBk7NoC925N3vGqjn7_pcY8YA@mail.gmail.com>,
	<BANLkTiniLjDXsVDTvf7kn0VdVSqpbC4acg@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DE2B62F7@cooper>

Yes, Steve is right.

BACKGROUND_JOB is not a channel event, the only ID the applies to this event is the "Job-UUID". And in this case, if originate had suncceeded, it would also return "+OK <uuid>" in the body.

No channel == no Unique-ID

/Peter
________________________________________
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] f&#246;r Steven Ayre [steveayre at gmail.com]
Skickat: den 25 maj 2011 08:15
Till: freeswitch-dev at lists.freeswitch.org
?mne: Re: [Freeswitch-dev] No Unique-ID field in BACKGROUND JOB events

I think that only applies to channel events.

-Steve


On 24 May 2011 23:08, Anton VG <anton.vazir at gmail.com<mailto:anton.vazir at gmail.com>> wrote:
In a while there was a thread (cand find it) where one of the core FS
developers mentioned that there always should be an Unique-ID field in
the event, so if there is no Unique ID field it can be added. Below is
the event with no Unique-ID

Event received
Event-Name: BACKGROUND_JOB
Core-UUID: bae130be-6719-4854-a0ed-2fdbb2477a12
FreeSWITCH-Hostname: lab3
FreeSWITCH-Switchname: lab3
FreeSWITCH-IPv4: 192.168.100.11
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2011-05-25%2002%3A56%3A20
Event-Date-GMT: Tue,%2024%20May%202011%2021%3A56%3A20%20GMT
Event-Date-Timestamp: 1306274180052931
Event-Calling-File: mod_event_socket.c
Event-Calling-Function: api_exec
Event-Calling-Line-Number: 1424
Job-UUID: 26266606-d9b4-4c9d-9642-792f6912d160
Job-Command: originate
Job-Command-Arg:
%7BMT-fwd_dst_type%3D'Phone%20No',progress_timeout%3D15,MT-fwd_condition%3DALWAYS,originate_timeout%3D90,MT-fwd_oclid%3DTrue,MT-fwd_dst_tech%3DNUM,MT-
fwd_timeout%3DNone,leg_timeout%3D60,origination_caller_id_number%3D433777100,MT-fwd_dst%3D901055490,origination_uuid%3D67829d64-918f-4ecf-8ece-e0b391d05172%7D%5Bsip_h_
X-MT-fwd_dst_tech%3DNUM,sip_h_X-MT-fwd_timeout%3DNone,sip_h_X-MT-fwd_condition%3DALWAYS,continue_on_fail%3Dfalse,leg_a_uuid%3D9c3e5ca3-4081-4cde-8316-6dbd4a1d021c,sip_
h_X-MT-fwd_dst_type%3D'Phone%20No',sip_h_X-MT-fwd_dst%3D901055490,sip_h_X-MT-fwd_oclid%3DTrue%5Dsofia/gateway/fs_local/sip3779050%20%26park
Content-Length: 27

-ERR INVALID_NUMBER_FORMAT

_______________________________________________
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:4ddc9efd32761110212216!


From anton.vazir at gmail.com  Wed May 25 13:23:35 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 25 May 2011 14:23:35 +0500
Subject: [Freeswitch-dev] No Unique-ID field in BACKGROUND JOB events
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59DE2B62F7@cooper>
References: <BANLkTinJ1jBk7NoC925N3vGqjn7_pcY8YA@mail.gmail.com>
	<BANLkTiniLjDXsVDTvf7kn0VdVSqpbC4acg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DE2B62F7@cooper>
Message-ID: <BANLkTimw-Au0htTDAgfqNMsLhGGtpAQneg@mail.gmail.com>

Thanks guys, it's clear now

2011/5/25 Peter Olsson <peter.olsson at visionutveckling.se>:
> Yes, Steve is right.
>
> BACKGROUND_JOB is not a channel event, the only ID the applies to this event is the "Job-UUID". And in this case, if originate had suncceeded, it would also return "+OK <uuid>" in the body.
>
> No channel == no Unique-ID
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] f&#246;r Steven Ayre [steveayre at gmail.com]
> Skickat: den 25 maj 2011 08:15
> Till: freeswitch-dev at lists.freeswitch.org
> ?mne: Re: [Freeswitch-dev] No Unique-ID field in BACKGROUND JOB events
>
> I think that only applies to channel events.
>
> -Steve
>
>
> On 24 May 2011 23:08, Anton VG <anton.vazir at gmail.com<mailto:anton.vazir at gmail.com>> wrote:
> In a while there was a thread (cand find it) where one of the core FS
> developers mentioned that there always should be an Unique-ID field in
> the event, so if there is no Unique ID field it can be added. Below is
> the event with no Unique-ID
>
> Event received
> Event-Name: BACKGROUND_JOB
> Core-UUID: bae130be-6719-4854-a0ed-2fdbb2477a12
> FreeSWITCH-Hostname: lab3
> FreeSWITCH-Switchname: lab3
> FreeSWITCH-IPv4: 192.168.100.11
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2011-05-25%2002%3A56%3A20
> Event-Date-GMT: Tue,%2024%20May%202011%2021%3A56%3A20%20GMT
> Event-Date-Timestamp: 1306274180052931
> Event-Calling-File: mod_event_socket.c
> Event-Calling-Function: api_exec
> Event-Calling-Line-Number: 1424
> Job-UUID: 26266606-d9b4-4c9d-9642-792f6912d160
> Job-Command: originate
> Job-Command-Arg:
> %7BMT-fwd_dst_type%3D'Phone%20No',progress_timeout%3D15,MT-fwd_condition%3DALWAYS,originate_timeout%3D90,MT-fwd_oclid%3DTrue,MT-fwd_dst_tech%3DNUM,MT-
> fwd_timeout%3DNone,leg_timeout%3D60,origination_caller_id_number%3D433777100,MT-fwd_dst%3D901055490,origination_uuid%3D67829d64-918f-4ecf-8ece-e0b391d05172%7D%5Bsip_h_
> X-MT-fwd_dst_tech%3DNUM,sip_h_X-MT-fwd_timeout%3DNone,sip_h_X-MT-fwd_condition%3DALWAYS,continue_on_fail%3Dfalse,leg_a_uuid%3D9c3e5ca3-4081-4cde-8316-6dbd4a1d021c,sip_
> h_X-MT-fwd_dst_type%3D'Phone%20No',sip_h_X-MT-fwd_dst%3D901055490,sip_h_X-MT-fwd_oclid%3DTrue%5Dsofia/gateway/fs_local/sip3779050%20%26park
> Content-Length: 27
>
> -ERR INVALID_NUMBER_FORMAT
>
> _______________________________________________
> 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:4ddc9efd32761110212216!
>
> _______________________________________________
> 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 anton.vazir at gmail.com  Wed May 25 16:42:08 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 25 May 2011 17:42:08 +0500
Subject: [Freeswitch-dev] event filter add order
Message-ID: <BANLkTik4vGee4Ezt_bjzvsNCUwVO58GdJw@mail.gmail.com>

I am noticed that I do not receive the BACKGROUND_JOB event if filter
for the given event is added after filter on Unique-ID filter.

e.g. if I do

filter Unique-ID XXXXX
filter Unique-ID YYYY
filter Event-Name BACKGROUND_JOB

I do not receive the BACKGROUND_JOB event at all

but if I do

filter Event-Name BACKGROUND_JOB
filter Unique-ID XXXXX
filter Unique-ID YYYY

I receive it

Seems a little odd, since described that it's a filter in
Is the given intended behavior or a bug?


From msc at freeswitch.org  Wed May 25 20:38:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 25 May 2011 09:38:38 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <BANLkTikSN0hNbvjhYM9QFMte_Ah6s1RAGg@mail.gmail.com>

Hi folks!

Today's agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_05_25

We have a few news items to discuss and if there's time I will share with
you the technique I demonstrated for using bind_digit_action in conferences.

Talk to you soon,
Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110525/32fa4ab2/attachment.html 

From moises.silva at gmail.com  Thu May 26 09:20:45 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 26 May 2011 01:20:45 -0400
Subject: [Freeswitch-dev] mod_portaudio multiple endpoints support
Message-ID: <BANLkTim5jOf70cX52n4VJeqNrDWoxf2iHg@mail.gmail.com>

Hello,

I've been doing some changes to mod_portaudio sponsored by Comrex
Corporation. The changes involve adding support for using multiple audio
devices simultaneously, including devices with multiple channels (note that
is not exactly the same as stereo).

The changes are available at git://
github.com/moises-silva/mod_portaudio-endpoints.git

Complete documentation on the new functionality in portaudio can be found in
the sample configuration in the form of comments
(conf/autoload_configs/portaudio.conf.xml).

The portaudio changes are totally backwards compatible, existing
configurations should keep working as expected.

In addition to the new portaudio functionality there are a few other changes
that were necessary to use the new functionality. All the changes described
below:

1. mod_portaudio.c and pablio.c modifications to allow usage of multiple
streams, channels and devices.

2. uuid_outgoing_answer API added to mod_commands.c, this API allows to
answer outgoing channels. This may only be useful for portaudio endpoints
since most, if not all, endpoints receive the answer notification using some
sort of signaling message (200 OK for example).

3. Changes to switch_generate_sln_silence() in switch_resample.c to allow
specifying -1 as a special divisor value meaning to zero-out the buffer
completely. This option was required by Comrex too since the equipment they
have seems to be amplifying the random low-level noise that
switch_generate_sln_silence() currently generates by default and they don't
like the "hissing noise" on the background while the call is answered and
there is no ringback tone.

I'd appreciate anyone who can review and comment. I just merged a few
moments ago with latest git therefore reviewing should be easy with git diff
adding the github repo as a remote.

I'd like to get approval from the core FreeSWITCH team to merge this into
mainstream. Needless to say I'll be available in the case any bugs are
introduced in the portaudio code, virtually impossible, but just in case
:-)

Thanks,

-
Moy
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110526/3821428a/attachment.html 

From gabe at gundy.org  Thu May 26 09:47:04 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 25 May 2011 23:47:04 -0600
Subject: [Freeswitch-dev] mod_portaudio multiple endpoints support
In-Reply-To: <BANLkTim5jOf70cX52n4VJeqNrDWoxf2iHg@mail.gmail.com>
References: <BANLkTim5jOf70cX52n4VJeqNrDWoxf2iHg@mail.gmail.com>
Message-ID: <BANLkTinzSB8KDx5c2a7fi2TV_cwNKJ-S2g@mail.gmail.com>

On Wed, May 25, 2011 at 11:20 PM, Moises Silva <moises.silva at gmail.com> wrote:
> I've been doing some changes to mod_portaudio sponsored by Comrex
> Corporation. The changes involve adding support for using multiple audio
> devices simultaneously, including devices with multiple channels (note that
> is not exactly the same as stereo).

All these changes sound *very* cool. I can't wait to try them out.
Thanks Comrex for your willingness to get them into FS :)

Gabe


From mbrancaleoni at voismart.it  Thu May 26 16:09:19 2011
From: mbrancaleoni at voismart.it (Matteo)
Date: Thu, 26 May 2011 14:09:19 +0200 (CEST)
Subject: [Freeswitch-dev] extra sdp is always added in sofia_answer_channel ?
In-Reply-To: <a8802e76-2fca-4300-8ca6-88dfa1003c93@mx.voismart.com>
Message-ID: <27b9b2d6-a560-49f7-8143-ed34d91649a3@mx.voismart.com>

Hi,  

don't know if is a bug/typo or is correct, so I'm posting here.

I'm noticed that in sofia_answer_channel (@mod_sofia.c:738, latest git)
NUTAG_INCLUDE_EXTRA_SDP(1) is always added.

Since we're into soa mode, maybe it should be added only 
by testing if 100rel is disabled, so it should be:

--- mod_sofia.c.orig	2011-05-26 14:16:34.000000000 +0200
+++ mod_sofia.c	2011-05-26 14:16:47.000000000 +0200
@@ -734,7 +734,8 @@
 						SIPTAG_CONTACT_STR(tech_pvt->reply_contact),
 						SIPTAG_CALL_INFO_STR(switch_channel_get_variable(tech_pvt->channel, SOFIA_SIP_HEADER_PREFIX "call_info")),
 						SOATAG_USER_SDP_STR(tech_pvt->local_sdp_str),
-						SOATAG_REUSE_REJECTED(1), SOATAG_ORDERED_USER(1), SOATAG_AUDIO_AUX("cn telephone-event"),NUTAG_INCLUDE_EXTRA_SDP(1),
+						SOATAG_REUSE_REJECTED(1), SOATAG_ORDERED_USER(1), SOATAG_AUDIO_AUX("cn telephone-event"),
+						TAG_IF(sofia_test_pflag(tech_pvt->profile, PFLAG_DISABLE_100REL), NUTAG_INCLUDE_EXTRA_SDP(1)),
 						TAG_IF(!zstr(extra_headers), SIPTAG_HEADER_STR(extra_headers)),
 						TAG_IF(switch_stristr("update_display", tech_pvt->x_freeswitch_support_remote),
 							   SIPTAG_HEADER_STR("X-FS-Support: " FREESWITCH_SUPPORT)), TAG_END());

Or is correct to add it always?

I've had to fix that because my upstream provider requires 100rel
and does not want sdp in 200OK if is not changed from last progress message.

Regards,
Matteo


From anton.vazir at gmail.com  Thu May 26 18:41:00 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Thu, 26 May 2011 19:41:00 +0500
Subject: [Freeswitch-dev] event filter add order
In-Reply-To: <BANLkTik4vGee4Ezt_bjzvsNCUwVO58GdJw@mail.gmail.com>
References: <BANLkTik4vGee4Ezt_bjzvsNCUwVO58GdJw@mail.gmail.com>
Message-ID: <BANLkTikfUXrDTyXW5+3m5CaBGS4j704xmA@mail.gmail.com>

Hello guys!


From m.sobkow at marketelsystems.com  Thu May 26 23:40:07 2011
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 26 May 2011 13:40:07 -0600
Subject: [Freeswitch-dev] Late response from Erlang to dialplan request
 causing Freeswitch to shut down
Message-ID: <4DDEAC97.5000604@marketelsystems.com>

Platform: Ubuntu 10.04 amd64, 8GB RAM, 500GB HDD, quad-core CPU
Configured to use SIP trunking as a slave off an Asterisk server with a 
T1 connection to the outside world

We're using Erlang and PostgreSQL to provide responses to Freeswitch 
configuration requests and to manage our call queues, operator logins, etc.

With one operator, the system runs fine, so we know the basic coding 
we've done works.

However, with only a couple or three operators on the line and 
concurrent dialing going on, we're getting a consistent problem where 
the responses to requests for dialplan configurations are getting 
seriously delayed -- by well over a minute (an eternity in software terms.)

I had suspected that there may be some sort of blocking IO going on when 
we issue the originate command (which doesn't return control to Erlang 
for up to 30 seconds), but I'm seeing other Freeswitch requests for 
directory information getting handled during this same time period, so 
the communications between Freeswitch and Erlang do support concurrent 
requests.

The only thing I've been able to find in the mail list logs or wiki that 
seem relevant is a comment someone made today about fixing the TCP 
transmissions in Freeswitch so that they'll retry the transmission if 
the IP stream is clogged and would result in a blocking call.  (i.e. FS 
used to be unreliable about sending out messages before this patch was 
done.)

It's worth noting that Erlang threads are quite busy during the minute 
between FS issueing the request and Erlang responding to it.  However, 
the thread/Erlang process that handles the configuration requests is 
idle, so it _should_ be able to respond in a much timelier fashion.

It looks to me like there is some sort of delay between FS issuing the 
request and Erlang getting the message into it's processing queue.

Any suggestions/ideas as to what might be causing this?  And more 
importantly -- what can we do to fix or workaround the issue?

And no, I have not downloaded and installed the latest version of 
Freeswitch from git.  I'm trying to avoid that if possible because it'll 
cost me two weeks of regression testing and I need to have this out to a 
beta site by the 31st, which doesn't allow enough time to do the 
regression tests if we do an upgrade (and only a fool ships untested 
software to paying customers.)  If someone can confirm that there was a 
fix checked into git for this issue or a related one, then I can justify 
the need for an upgrade and regression testing to my boss (though he 
won't be happy about the delay.)

Here's the relevent snippet of the fs_cli log:

2011-05-26 11:45:50.648334 [INFO] mod_dialplan_xml.c:331 Processing 
Gotham <5311>->219 in context public
2011-05-26 11:45:50.648334 [ERR] mod_dialplan_xml.c:353 Open of dialplan 
failed
2011-05-26 11:45:50.648334 [INFO] switch_core_state_machine.c:142 No 
Route, Aborting
2011-05-26 11:45:50.648334 [DEBUG] switch_channel.c:2535 
(sofia/external/5311 at 10.77.0.12:6080) Callstate Change RINGING -> HANGUP
2011-05-26 11:45:50.657061 [NOTICE] switch_core_state_machine.c:143 
Hangup sofia/external/5311 at 10.77.0.12:6080 [CS_ROUTING] 
[NO_ROUTE_DESTINATION]


And here's the log from where our response is finally being issued to 
the request, long after FS has decided to shut down because it's not 
getting responses from the configuration services:

2011-05-26 11:47:45 DEBUG <0.28104.15>: pbx_bind_cfg:fetch() Invoking 
xml_fetch()
2011-05-26 11:47:45 DEBUG <0.28104.15>: pbx_bind_cfg:xml_fetch( {fetch, 
dialplan, Tag=undefined, Key=undefined, Value=undefined, Params=...} )
2011-05-26 11:47:46 DEBUG <0.28104.15>: pbx_bind_cfg:fetch() Response XML is
<document type="freeswitch/xml">
<section name="dialplan" description="Dial Plan For FreeSwitch">
<context name="default">
<extension required_name="ConferenceDialer">
<condition field="destination_number" expression="^8(\d\d\d)$">
<action application="conference" data="conf$0 at dialer"/>
</condition>
</extension>
<extension required_name="dialer_park">
<condition field="destination_number" expression="^park$">
<action application="fifo" data="park@${domain_name} in" />
</condition>
</extension>
<extension required_name="park">
<condition field="destination_number" expression="^7000$">
<action application="set" data="fifo_music=$${hold_music}" />
<action application="fifo" data="7000@${domain_name} in" />
</condition>
</extension>
<extension required_name="unpark">
<condition field="destination_number" expression="^7001$">
<action application="answer" />
<action application="fifo" data="7000@${domain_name} out nowait" />
</condition>
</extension>
<extension required_name="InternalFS">
<condition field="destination_number" expression="^(\d\d\d\d)$">
<action application="bridge" data="sofia/external/$0%pursuit.marketel"/>
</condition>
</extension>
<extension required_name="RatsExtension">
<condition field="destination_number" expression="^(\d\d\d)$">
<action application="bridge" data="sofia/external/$0 at rats.marketel"/>
</condition>
</extension>
<extension required_name="LocalCall">
<condition field="destination_number" expression="^(9\d\d\d\d\d\d\d)$">
<action application="bridge" data="sofia/external/$0 at rats.marketel"/>
</condition>
</extension>
<extension required_name="LongDistanceCall">
<condition field="destination_number" 
expression="^91(\d\d\d\d\d\d\d\d\d\d)$">
<action application="bridge" data="sofia/external/$0 at rats.marketel"/>
</condition>
</extension>
</context>
<context name="public">
<extension required_name="Operator">
<condition field="destination_number" expression="^(0)$">
<action application="bridge" data="sofia/external/160 at rats.marketel"/>
</condition>
</extension>
<extension required_name="OperatorLogin">
<condition field="destination_number" expression="^620(\d)$">
<action application="answer" />
<action application="set" data="chain_id=$1" />
<action application="set" data="hangup_after_bridge=false" />
<action application="play_and_get_digits" data="4 8 1 30000 # 
/usr/lib/pursuit/lib/pbx-1.2/priv/welcome_please_enter_pin-8000.wav 
/usr/lib/pursuit/lib/pbx-1.2/priv/invalid_pin_try_again-8000.wav 
operator_pin \d+" />
<action application="log" data="INFO operator_pin is ${operator_pin}, 
chain_id is ${chain_id}" />
<action application="erlang" data="pbx_op_logged_in pursuit at pursuit" />
</condition>
</extension>

<extension required_name="OperatorLogin2">
<condition field="destination_number" expression="^62(\d\d)$">
<action application="answer" />
<action application="set" data="chain_id=$1" />
<action application="set" data="hangup_after_bridge=false" />
<action application="play_and_get_digits" data="4 8 1 30000 # 
/usr/lib/pursuit/lib/pbx-1.2/priv/welcome_please_enter_pin-8000.wav 
/usr/lib/pursuit/lib/pbx-1.2/priv/invalid_pin_try_again-8000.wav 
operator_pin \d+" />
<action application="log" data="INFO operator_pin is ${operator_pin}, 
chain_id is ${chain_id}" />
<action application="erlang" data="pbx_op_logged_in pursuit at pursuit" />
</condition>
</extension>
<extension required_name="CollectCallResult">
<condition field="destination_number" expression="^63(\d\d)$">
<action application="answer" />
<action application="set" data="hangup_after_bridge=false" />
<action application="play_and_get_digits" data="1 2 1 30000 # 
/usr/lib/pursuit/lib/pbx-1.2/priv/please_enter_result_code-8000.wav 
/usr/lib/pursuit/lib/pbx-1.2/priv/invalid_result_code-8000.wav 
call_result \d+" />
<action application="log" data="INFO call_result is ${call_result}" />
<action application="erlang" data="pbx_call_result pursuit at pursuit" />
</condition>
</extension>

<extension required_name="DummyCallResult">
<condition field="destination_number" expression="^6400$">
<action application="answer" />
<action application="erlang" data="pbx_dummy_call_result pursuit at pursuit" />
</condition>
</extension>

<extension required_name="CustomerCallin">
<condition field="destination_number" expression="^650(\d)$">
<action application="answer" />
<action application="set" data="chain_id=$1" />
<action application="erlang" data="pbx_cust_called pursuit at pursuit" />
</condition>
</extension>
<extension required_name="CustomerCallin2">
<condition field="destination_number" expression="^65(\d\d)$">
<action application="answer" />
<action application="set" data="chain_id=$1" />
<action application="erlang" data="pbx_cust_called pursuit at pursuit" />
</condition>
</extension>

<extension required_name="TestIVR">
<condition field="destination_number" expression="^6600$">
<action application="answer" />
<action application="ivr" data="demo_ivr" />
</condition>
</extension>

<extension required_name="eavesdrop">
<condition field="destination_number" expression="^67(\d\d\d\d)$">
<action application="answer"/>
<action application="set" data="eavesdrop_extension=$1" />
<action application="erlang" data="pbx_eavesdrop pursuit at pursuit" />
</condition>
</extension>

<extension required_name="park">
<condition field="destination_number" expression="^6003$">
<action application="fifo" data="6003@${domain_name} in" />
</condition>
</extension>
<extension required_name="unpark">
<condition field="destination_number" expression="^6004$">
<action application="answer" />
<action application="fifo" data="6003@${domain_name} out wait" />
</condition>
</extension>

<extension required_name="ConferenceDialer">
<condition field="destination_number" expression="^8(\d\d\d)$">
<action application="conference" data="conf$0 at dialer"/>
</condition>
</extension>
<extension required_name="InternalFS">
<condition field="destination_number" expression="^(\d\d\d\d)$">
<action application="bridge" data="sofia/external/$0%10.77.0.12"/>
</condition>
</extension>
<extension required_name="RatsExtension">
<condition field="destination_number" expression="^(\d\d\d)$">
<action application="bridge" data="sofia/external/$0 at rats.marketel"/>
</condition>
</extension>
<extension required_name="LocalCall">
<condition field="destination_number" expression="^(9\d\d\d\d\d\d\d)$">
<action application="bridge" data="sofia/external/$0 at rats.marketel"/>
</condition>
</extension>
<extension required_name="LongDistanceCall">
<condition field="destination_number" 
expression="^91(\d\d\d\d\d\d\d\d\d\d)$">
<action application="bridge" data="sofia/external/$0 at rats.marketel"/>
</condition>
</extension>
</context>
</section>
</document>


-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com

Visit our Blog for industry related information.
http://marketel-systems.blogspot.com/



From m.sobkow at marketelsystems.com  Thu May 26 23:49:31 2011
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 26 May 2011 13:49:31 -0600
Subject: [Freeswitch-dev] Late response from Erlang to dialplan request
 causing Freeswitch to shut down
In-Reply-To: <4DDEAC97.5000604@marketelsystems.com>
References: <4DDEAC97.5000604@marketelsystems.com>
Message-ID: <4DDEAECB.1010209@marketelsystems.com>

On further review of the log and matching up notices about FS requests 
to Erlang responses, it looks like the request that's timing out is 
getting lost completely and is never processed by Erlang.

On 26/05/2011 1:40 PM, Mark Sobkow wrote:
> Platform: Ubuntu 10.04 amd64, 8GB RAM, 500GB HDD, quad-core CPU
> Configured to use SIP trunking as a slave off an Asterisk server with 
> a T1 connection to the outside world
>
> We're using Erlang and PostgreSQL to provide responses to Freeswitch 
> configuration requests and to manage our call queues, operator logins, 
> etc.
>
> With one operator, the system runs fine, so we know the basic coding 
> we've done works.
>
> However, with only a couple or three operators on the line and 
> concurrent dialing going on, we're getting a consistent problem where 
> the responses to requests for dialplan configurations are getting 
> seriously delayed -- by well over a minute (an eternity in software 
> terms.)
>
> I had suspected that there may be some sort of blocking IO going on 
> when we issue the originate command (which doesn't return control to 
> Erlang for up to 30 seconds), but I'm seeing other Freeswitch requests 
> for directory information getting handled during this same time 
> period, so the communications between Freeswitch and Erlang do support 
> concurrent requests.
>
> The only thing I've been able to find in the mail list logs or wiki 
> that seem relevant is a comment someone made today about fixing the 
> TCP transmissions in Freeswitch so that they'll retry the transmission 
> if the IP stream is clogged and would result in a blocking call.  
> (i.e. FS used to be unreliable about sending out messages before this 
> patch was done.)
>
> It's worth noting that Erlang threads are quite busy during the minute 
> between FS issueing the request and Erlang responding to it.  However, 
> the thread/Erlang process that handles the configuration requests is 
> idle, so it _should_ be able to respond in a much timelier fashion.
>
> It looks to me like there is some sort of delay between FS issuing the 
> request and Erlang getting the message into it's processing queue.
>
> Any suggestions/ideas as to what might be causing this?  And more 
> importantly -- what can we do to fix or workaround the issue?
>
> And no, I have not downloaded and installed the latest version of 
> Freeswitch from git.  I'm trying to avoid that if possible because 
> it'll cost me two weeks of regression testing and I need to have this 
> out to a beta site by the 31st, which doesn't allow enough time to do 
> the regression tests if we do an upgrade (and only a fool ships 
> untested software to paying customers.)  If someone can confirm that 
> there was a fix checked into git for this issue or a related one, then 
> I can justify the need for an upgrade and regression testing to my 
> boss (though he won't be happy about the delay.)
>
> Here's the relevent snippet of the fs_cli log:
>
> 2011-05-26 11:45:50.648334 [INFO] mod_dialplan_xml.c:331 Processing 
> Gotham <5311>->219 in context public
> 2011-05-26 11:45:50.648334 [ERR] mod_dialplan_xml.c:353 Open of 
> dialplan failed
> 2011-05-26 11:45:50.648334 [INFO] switch_core_state_machine.c:142 No 
> Route, Aborting
> 2011-05-26 11:45:50.648334 [DEBUG] switch_channel.c:2535 
> (sofia/external/5311 at 10.77.0.12:6080) Callstate Change RINGING -> HANGUP
> 2011-05-26 11:45:50.657061 [NOTICE] switch_core_state_machine.c:143 
> Hangup sofia/external/5311 at 10.77.0.12:6080 [CS_ROUTING] 
> [NO_ROUTE_DESTINATION]
>
>
> And here's the log from where our response is finally being issued to 
> the request, long after FS has decided to shut down because it's not 
> getting responses from the configuration services:
>
> 2011-05-26 11:47:45 DEBUG <0.28104.15>: pbx_bind_cfg:fetch() Invoking 
> xml_fetch()
> 2011-05-26 11:47:45 DEBUG <0.28104.15>: pbx_bind_cfg:xml_fetch( 
> {fetch, dialplan, Tag=undefined, Key=undefined, Value=undefined, 
> Params=...} )
> 2011-05-26 11:47:46 DEBUG <0.28104.15>: pbx_bind_cfg:fetch() Response 
> XML is
> <document type="freeswitch/xml">
> <section name="dialplan" description="Dial Plan For FreeSwitch">
> <context name="default">
> <extension required_name="ConferenceDialer">
> <condition field="destination_number" expression="^8(\d\d\d)$">
> <action application="conference" data="conf$0 at dialer"/>
> </condition>
> </extension>
> <extension required_name="dialer_park">
> <condition field="destination_number" expression="^park$">
> <action application="fifo" data="park@${domain_name} in" />
> </condition>
> </extension>
> <extension required_name="park">
> <condition field="destination_number" expression="^7000$">
> <action application="set" data="fifo_music=$${hold_music}" />
> <action application="fifo" data="7000@${domain_name} in" />
> </condition>
> </extension>
> <extension required_name="unpark">
> <condition field="destination_number" expression="^7001$">
> <action application="answer" />
> <action application="fifo" data="7000@${domain_name} out nowait" />
> </condition>
> </extension>
> <extension required_name="InternalFS">
> <condition field="destination_number" expression="^(\d\d\d\d)$">
> <action application="bridge" data="sofia/external/$0%pursuit.marketel"/>
> </condition>
> </extension>
> <extension required_name="RatsExtension">
> <condition field="destination_number" expression="^(\d\d\d)$">
> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
> </condition>
> </extension>
> <extension required_name="LocalCall">
> <condition field="destination_number" expression="^(9\d\d\d\d\d\d\d)$">
> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
> </condition>
> </extension>
> <extension required_name="LongDistanceCall">
> <condition field="destination_number" 
> expression="^91(\d\d\d\d\d\d\d\d\d\d)$">
> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
> </condition>
> </extension>
> </context>
> <context name="public">
> <extension required_name="Operator">
> <condition field="destination_number" expression="^(0)$">
> <action application="bridge" data="sofia/external/160 at rats.marketel"/>
> </condition>
> </extension>
> <extension required_name="OperatorLogin">
> <condition field="destination_number" expression="^620(\d)$">
> <action application="answer" />
> <action application="set" data="chain_id=$1" />
> <action application="set" data="hangup_after_bridge=false" />
> <action application="play_and_get_digits" data="4 8 1 30000 # 
> /usr/lib/pursuit/lib/pbx-1.2/priv/welcome_please_enter_pin-8000.wav 
> /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_pin_try_again-8000.wav 
> operator_pin \d+" />
> <action application="log" data="INFO operator_pin is ${operator_pin}, 
> chain_id is ${chain_id}" />
> <action application="erlang" data="pbx_op_logged_in pursuit at pursuit" />
> </condition>
> </extension>
>
> <extension required_name="OperatorLogin2">
> <condition field="destination_number" expression="^62(\d\d)$">
> <action application="answer" />
> <action application="set" data="chain_id=$1" />
> <action application="set" data="hangup_after_bridge=false" />
> <action application="play_and_get_digits" data="4 8 1 30000 # 
> /usr/lib/pursuit/lib/pbx-1.2/priv/welcome_please_enter_pin-8000.wav 
> /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_pin_try_again-8000.wav 
> operator_pin \d+" />
> <action application="log" data="INFO operator_pin is ${operator_pin}, 
> chain_id is ${chain_id}" />
> <action application="erlang" data="pbx_op_logged_in pursuit at pursuit" />
> </condition>
> </extension>
> <extension required_name="CollectCallResult">
> <condition field="destination_number" expression="^63(\d\d)$">
> <action application="answer" />
> <action application="set" data="hangup_after_bridge=false" />
> <action application="play_and_get_digits" data="1 2 1 30000 # 
> /usr/lib/pursuit/lib/pbx-1.2/priv/please_enter_result_code-8000.wav 
> /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_result_code-8000.wav 
> call_result \d+" />
> <action application="log" data="INFO call_result is ${call_result}" />
> <action application="erlang" data="pbx_call_result pursuit at pursuit" />
> </condition>
> </extension>
>
> <extension required_name="DummyCallResult">
> <condition field="destination_number" expression="^6400$">
> <action application="answer" />
> <action application="erlang" data="pbx_dummy_call_result 
> pursuit at pursuit" />
> </condition>
> </extension>
>
> <extension required_name="CustomerCallin">
> <condition field="destination_number" expression="^650(\d)$">
> <action application="answer" />
> <action application="set" data="chain_id=$1" />
> <action application="erlang" data="pbx_cust_called pursuit at pursuit" />
> </condition>
> </extension>
> <extension required_name="CustomerCallin2">
> <condition field="destination_number" expression="^65(\d\d)$">
> <action application="answer" />
> <action application="set" data="chain_id=$1" />
> <action application="erlang" data="pbx_cust_called pursuit at pursuit" />
> </condition>
> </extension>
>
> <extension required_name="TestIVR">
> <condition field="destination_number" expression="^6600$">
> <action application="answer" />
> <action application="ivr" data="demo_ivr" />
> </condition>
> </extension>
>
> <extension required_name="eavesdrop">
> <condition field="destination_number" expression="^67(\d\d\d\d)$">
> <action application="answer"/>
> <action application="set" data="eavesdrop_extension=$1" />
> <action application="erlang" data="pbx_eavesdrop pursuit at pursuit" />
> </condition>
> </extension>
>
> <extension required_name="park">
> <condition field="destination_number" expression="^6003$">
> <action application="fifo" data="6003@${domain_name} in" />
> </condition>
> </extension>
> <extension required_name="unpark">
> <condition field="destination_number" expression="^6004$">
> <action application="answer" />
> <action application="fifo" data="6003@${domain_name} out wait" />
> </condition>
> </extension>
>
> <extension required_name="ConferenceDialer">
> <condition field="destination_number" expression="^8(\d\d\d)$">
> <action application="conference" data="conf$0 at dialer"/>
> </condition>
> </extension>
> <extension required_name="InternalFS">
> <condition field="destination_number" expression="^(\d\d\d\d)$">
> <action application="bridge" data="sofia/external/$0%10.77.0.12"/>
> </condition>
> </extension>
> <extension required_name="RatsExtension">
> <condition field="destination_number" expression="^(\d\d\d)$">
> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
> </condition>
> </extension>
> <extension required_name="LocalCall">
> <condition field="destination_number" expression="^(9\d\d\d\d\d\d\d)$">
> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
> </condition>
> </extension>
> <extension required_name="LongDistanceCall">
> <condition field="destination_number" 
> expression="^91(\d\d\d\d\d\d\d\d\d\d)$">
> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
> </condition>
> </extension>
> </context>
> </section>
> </document>
>
>


-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com

Visit our Blog for industry related information.
http://marketel-systems.blogspot.com/



From m.sobkow at marketelsystems.com  Thu May 26 23:54:51 2011
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 26 May 2011 13:54:51 -0600
Subject: [Freeswitch-dev] Late response from Erlang to dialplan request
 causing Freeswitch to shut down
In-Reply-To: <4DDEAECB.1010209@marketelsystems.com>
References: <4DDEAC97.5000604@marketelsystems.com>
	<4DDEAECB.1010209@marketelsystems.com>
Message-ID: <4DDEB00B.40907@marketelsystems.com>

On 3:59 PM, Peter Olsson wrote:

> >>  Did you try latest git? I recently submitted a patch to handle resend on the socket, if the socket is really busy (and returns "would block" state). It's in latest git.
Would this fix affect the sending of messages to Erlang nodes as well?  
(I suspect it might as it would only make sense to reuse the network IO 
APIs rather than having seperate ones for each module.  Andrew Thompson 
would know for sure.)

If it does affect Erlang communications, I'll have to do a git-pull and 
schedule some regression testing.

On 26/05/2011 1:49 PM, Mark Sobkow wrote:
> On further review of the log and matching up notices about FS requests 
> to Erlang responses, it looks like the request that's timing out is 
> getting lost completely and is never processed by Erlang.
>
> On 26/05/2011 1:40 PM, Mark Sobkow wrote:
>> Platform: Ubuntu 10.04 amd64, 8GB RAM, 500GB HDD, quad-core CPU
>> Configured to use SIP trunking as a slave off an Asterisk server with 
>> a T1 connection to the outside world
>>
>> We're using Erlang and PostgreSQL to provide responses to Freeswitch 
>> configuration requests and to manage our call queues, operator 
>> logins, etc.
>>
>> With one operator, the system runs fine, so we know the basic coding 
>> we've done works.
>>
>> However, with only a couple or three operators on the line and 
>> concurrent dialing going on, we're getting a consistent problem where 
>> the responses to requests for dialplan configurations are getting 
>> seriously delayed -- by well over a minute (an eternity in software 
>> terms.)
>>
>> I had suspected that there may be some sort of blocking IO going on 
>> when we issue the originate command (which doesn't return control to 
>> Erlang for up to 30 seconds), but I'm seeing other Freeswitch 
>> requests for directory information getting handled during this same 
>> time period, so the communications between Freeswitch and Erlang do 
>> support concurrent requests.
>>
>> The only thing I've been able to find in the mail list logs or wiki 
>> that seem relevant is a comment someone made today about fixing the 
>> TCP transmissions in Freeswitch so that they'll retry the 
>> transmission if the IP stream is clogged and would result in a 
>> blocking call.  (i.e. FS used to be unreliable about sending out 
>> messages before this patch was done.)
>>
>> It's worth noting that Erlang threads are quite busy during the 
>> minute between FS issueing the request and Erlang responding to it.  
>> However, the thread/Erlang process that handles the configuration 
>> requests is idle, so it _should_ be able to respond in a much 
>> timelier fashion.
>>
>> It looks to me like there is some sort of delay between FS issuing 
>> the request and Erlang getting the message into it's processing queue.
>>
>> Any suggestions/ideas as to what might be causing this?  And more 
>> importantly -- what can we do to fix or workaround the issue?
>>
>> And no, I have not downloaded and installed the latest version of 
>> Freeswitch from git.  I'm trying to avoid that if possible because 
>> it'll cost me two weeks of regression testing and I need to have this 
>> out to a beta site by the 31st, which doesn't allow enough time to do 
>> the regression tests if we do an upgrade (and only a fool ships 
>> untested software to paying customers.)  If someone can confirm that 
>> there was a fix checked into git for this issue or a related one, 
>> then I can justify the need for an upgrade and regression testing to 
>> my boss (though he won't be happy about the delay.)
>>
>> Here's the relevent snippet of the fs_cli log:
>>
>> 2011-05-26 11:45:50.648334 [INFO] mod_dialplan_xml.c:331 Processing 
>> Gotham <5311>->219 in context public
>> 2011-05-26 11:45:50.648334 [ERR] mod_dialplan_xml.c:353 Open of 
>> dialplan failed
>> 2011-05-26 11:45:50.648334 [INFO] switch_core_state_machine.c:142 No 
>> Route, Aborting
>> 2011-05-26 11:45:50.648334 [DEBUG] switch_channel.c:2535 
>> (sofia/external/5311 at 10.77.0.12:6080) Callstate Change RINGING -> HANGUP
>> 2011-05-26 11:45:50.657061 [NOTICE] switch_core_state_machine.c:143 
>> Hangup sofia/external/5311 at 10.77.0.12:6080 [CS_ROUTING] 
>> [NO_ROUTE_DESTINATION]
>>
>>
>> And here's the log from where our response is finally being issued to 
>> the request, long after FS has decided to shut down because it's not 
>> getting responses from the configuration services:
>>
>> 2011-05-26 11:47:45 DEBUG <0.28104.15>: pbx_bind_cfg:fetch() Invoking 
>> xml_fetch()
>> 2011-05-26 11:47:45 DEBUG <0.28104.15>: pbx_bind_cfg:xml_fetch( 
>> {fetch, dialplan, Tag=undefined, Key=undefined, Value=undefined, 
>> Params=...} )
>> 2011-05-26 11:47:46 DEBUG <0.28104.15>: pbx_bind_cfg:fetch() Response 
>> XML is
>> <document type="freeswitch/xml">
>> <section name="dialplan" description="Dial Plan For FreeSwitch">
>> <context name="default">
>> <extension required_name="ConferenceDialer">
>> <condition field="destination_number" expression="^8(\d\d\d)$">
>> <action application="conference" data="conf$0 at dialer"/>
>> </condition>
>> </extension>
>> <extension required_name="dialer_park">
>> <condition field="destination_number" expression="^park$">
>> <action application="fifo" data="park@${domain_name} in" />
>> </condition>
>> </extension>
>> <extension required_name="park">
>> <condition field="destination_number" expression="^7000$">
>> <action application="set" data="fifo_music=$${hold_music}" />
>> <action application="fifo" data="7000@${domain_name} in" />
>> </condition>
>> </extension>
>> <extension required_name="unpark">
>> <condition field="destination_number" expression="^7001$">
>> <action application="answer" />
>> <action application="fifo" data="7000@${domain_name} out nowait" />
>> </condition>
>> </extension>
>> <extension required_name="InternalFS">
>> <condition field="destination_number" expression="^(\d\d\d\d)$">
>> <action application="bridge" data="sofia/external/$0%pursuit.marketel"/>
>> </condition>
>> </extension>
>> <extension required_name="RatsExtension">
>> <condition field="destination_number" expression="^(\d\d\d)$">
>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>> </condition>
>> </extension>
>> <extension required_name="LocalCall">
>> <condition field="destination_number" expression="^(9\d\d\d\d\d\d\d)$">
>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>> </condition>
>> </extension>
>> <extension required_name="LongDistanceCall">
>> <condition field="destination_number" 
>> expression="^91(\d\d\d\d\d\d\d\d\d\d)$">
>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>> </condition>
>> </extension>
>> </context>
>> <context name="public">
>> <extension required_name="Operator">
>> <condition field="destination_number" expression="^(0)$">
>> <action application="bridge" data="sofia/external/160 at rats.marketel"/>
>> </condition>
>> </extension>
>> <extension required_name="OperatorLogin">
>> <condition field="destination_number" expression="^620(\d)$">
>> <action application="answer" />
>> <action application="set" data="chain_id=$1" />
>> <action application="set" data="hangup_after_bridge=false" />
>> <action application="play_and_get_digits" data="4 8 1 30000 # 
>> /usr/lib/pursuit/lib/pbx-1.2/priv/welcome_please_enter_pin-8000.wav 
>> /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_pin_try_again-8000.wav 
>> operator_pin \d+" />
>> <action application="log" data="INFO operator_pin is ${operator_pin}, 
>> chain_id is ${chain_id}" />
>> <action application="erlang" data="pbx_op_logged_in pursuit at pursuit" />
>> </condition>
>> </extension>
>>
>> <extension required_name="OperatorLogin2">
>> <condition field="destination_number" expression="^62(\d\d)$">
>> <action application="answer" />
>> <action application="set" data="chain_id=$1" />
>> <action application="set" data="hangup_after_bridge=false" />
>> <action application="play_and_get_digits" data="4 8 1 30000 # 
>> /usr/lib/pursuit/lib/pbx-1.2/priv/welcome_please_enter_pin-8000.wav 
>> /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_pin_try_again-8000.wav 
>> operator_pin \d+" />
>> <action application="log" data="INFO operator_pin is ${operator_pin}, 
>> chain_id is ${chain_id}" />
>> <action application="erlang" data="pbx_op_logged_in pursuit at pursuit" />
>> </condition>
>> </extension>
>> <extension required_name="CollectCallResult">
>> <condition field="destination_number" expression="^63(\d\d)$">
>> <action application="answer" />
>> <action application="set" data="hangup_after_bridge=false" />
>> <action application="play_and_get_digits" data="1 2 1 30000 # 
>> /usr/lib/pursuit/lib/pbx-1.2/priv/please_enter_result_code-8000.wav 
>> /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_result_code-8000.wav 
>> call_result \d+" />
>> <action application="log" data="INFO call_result is ${call_result}" />
>> <action application="erlang" data="pbx_call_result pursuit at pursuit" />
>> </condition>
>> </extension>
>>
>> <extension required_name="DummyCallResult">
>> <condition field="destination_number" expression="^6400$">
>> <action application="answer" />
>> <action application="erlang" data="pbx_dummy_call_result 
>> pursuit at pursuit" />
>> </condition>
>> </extension>
>>
>> <extension required_name="CustomerCallin">
>> <condition field="destination_number" expression="^650(\d)$">
>> <action application="answer" />
>> <action application="set" data="chain_id=$1" />
>> <action application="erlang" data="pbx_cust_called pursuit at pursuit" />
>> </condition>
>> </extension>
>> <extension required_name="CustomerCallin2">
>> <condition field="destination_number" expression="^65(\d\d)$">
>> <action application="answer" />
>> <action application="set" data="chain_id=$1" />
>> <action application="erlang" data="pbx_cust_called pursuit at pursuit" />
>> </condition>
>> </extension>
>>
>> <extension required_name="TestIVR">
>> <condition field="destination_number" expression="^6600$">
>> <action application="answer" />
>> <action application="ivr" data="demo_ivr" />
>> </condition>
>> </extension>
>>
>> <extension required_name="eavesdrop">
>> <condition field="destination_number" expression="^67(\d\d\d\d)$">
>> <action application="answer"/>
>> <action application="set" data="eavesdrop_extension=$1" />
>> <action application="erlang" data="pbx_eavesdrop pursuit at pursuit" />
>> </condition>
>> </extension>
>>
>> <extension required_name="park">
>> <condition field="destination_number" expression="^6003$">
>> <action application="fifo" data="6003@${domain_name} in" />
>> </condition>
>> </extension>
>> <extension required_name="unpark">
>> <condition field="destination_number" expression="^6004$">
>> <action application="answer" />
>> <action application="fifo" data="6003@${domain_name} out wait" />
>> </condition>
>> </extension>
>>
>> <extension required_name="ConferenceDialer">
>> <condition field="destination_number" expression="^8(\d\d\d)$">
>> <action application="conference" data="conf$0 at dialer"/>
>> </condition>
>> </extension>
>> <extension required_name="InternalFS">
>> <condition field="destination_number" expression="^(\d\d\d\d)$">
>> <action application="bridge" data="sofia/external/$0%10.77.0.12"/>
>> </condition>
>> </extension>
>> <extension required_name="RatsExtension">
>> <condition field="destination_number" expression="^(\d\d\d)$">
>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>> </condition>
>> </extension>
>> <extension required_name="LocalCall">
>> <condition field="destination_number" expression="^(9\d\d\d\d\d\d\d)$">
>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>> </condition>
>> </extension>
>> <extension required_name="LongDistanceCall">
>> <condition field="destination_number" 
>> expression="^91(\d\d\d\d\d\d\d\d\d\d)$">
>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>> </condition>
>> </extension>
>> </context>
>> </section>
>> </document>
>>
>>
>
>


-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com

Visit our Blog for industry related information.
http://marketel-systems.blogspot.com/



From rhuddleston at gmail.com  Sat May 21 23:36:43 2011
From: rhuddleston at gmail.com (Robert-iPhone)
Date: Sat, 21 May 2011 15:36:43 -0400
Subject: [Freeswitch-dev] Project Announcement and Request for Direction
In-Reply-To: <BANLkTi=tHBL5tZxqE7ObZJ8UrBsZY5UzPQ@mail.gmail.com>
References: <BANLkTimuCyEEkRN8ftJaS+0XCNXoFgd=Jg@mail.gmail.com>
	<BANLkTi=tHBL5tZxqE7ObZJ8UrBsZY5UzPQ@mail.gmail.com>
Message-ID: <F7C708E1-92CF-4E7B-9717-190F5C817461@gmail.com>

I would be willing to contribute

Sent from my iPhone

On May 18, 2011, at 8:59 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:

> Hello Freeswitch-dev! I originally sent this email to
> freeswitch-users, and didn't get a lot of responses. I thought this
> might be a better list for this discussion.
> 
> My name is Kurtis Heimerl, and I'm a graduate researcher at the
> University of California, Berkeley in the Technology and
> Infrastructure for Emerging Regions (TIER) group under Eric Brewer.
> We're currently investigating the use of OpenBTS
> (http://openbts.sourceforge.net/) for providing cellular coverage via
> low-cost base stations in low-density parts of the world. This project
> is called The Village Base Station. In support of that project, we're
> also investigating the use of Freeswitch to support OpenBTS and
> provide a flexible, extensible, and simple platform for deploying
> voice/sms/data applications on the basestation itself. Towards this
> end, I'm asking you, the freeswitch community, for advice and
> direction on some of our research goals.
> 
> The basic story is simple. OpenBTS uses a software radio and generic
> PC to provide cellular service. As a byproduct of this architecture,
> we also gain the ability to run freeswitch applications "locally"
> (concurrently with OpenBTS), and take advantage of the benefits this
> tight coupling provides us. These benefits are numerous; calls/SMS
> between BTS users can be much cheaper, as they do not use any backhaul
> bandwidth. Applications can query the system for more information
> about users, such as location or status. As an example, unlike
> traditional GSM telephony applications, we are able to query if users
> are currently available on the network. This could be used to create
> voice "chat lists", which tell participants which of their friends are
> currently within cellular range.
> 
> We foresee 6 features required to support these "local" freeswitch
> applications on our OpenBTS system. I'm very curious how the
> freeswitch community feels about these possible additions, as well how
> they might be implemented.
> 
> There are a few that are already available in freeswitch, but may be
> rougher than we would like. These include:
> 
> 1) Identity: The ability to query for user's status, numbers, etc.
> This seems simple enough in the existing system. However, we'd like to
> provide hooks for applications to act on these sign-ons or offs. For
> instance, an app may hold messages until a phone logs onto the system
> and push them then. My understanding is that this should be simple,
> probably hooking onto "SIP presence" events?
> 2) Storage: Freeswitch currently seems to support only per-application
> storage, with limited support for cross-application storage (mostly
> user directories). This is occasionally problematic: one issue we've
> heard is that it is difficult to place messages into voice mailboxes
> from other apps. We'd like a more unified storage framework. Or...
> 3) "Pipes": This is the ability to pass messages between freeswitch
> apps. This seems pretty well supported though simple dialplan
> interactions, though the modules themselves may not provide enough
> functionality. Is there a way to do this inside of apps? I consider
> this an alternative to the storage framework discussed in #2
> 
> Lastly, there are three functions we don't believe are well-supported
> in freeswitch. These are...
> 1) Privacy: We expect our BTS to be used in politically sensitive
> areas. Given this, freeswitch could provide an anonymity layer,
> providing short term phone/SMS numbers, or directing communications
> through more secure layers (e.g., Tor).
> 2) Asynchrony: While freeswitch seems to support basic asynchrony
> though its event system, I couldn't find any way to delay events for
> indeterminate times. For instance, we may want to schedule a traffic
> warning for 1PM every Wednesday to every phone currently on the
> system. Is there a way to do that currently?
> 3) SMS: Freeswitch seems to currently support SIP chat messages (using
> SIMPLE?). We need to either extend OpenBTS to speak SIMPLE, or extend
> freeswitch to speak OpenBTS's SMS protocol. Neither seems particularly
> difficult. This will allow our apps to send and receive both voice and
> SMS messages from users.
> 
> We believe these core functions will enable a wide variety of BTS
> applications. I have a laundry list of those, but I'll omit them for
> sake of space.
> 
> If any members of the community (that means you!) have any directions,
> ideas, projects, or thoughts, please pass them on! We're just
> beginning this part of the project, and getting the lay of the land.
> Feedback is critical at this point.
> 
> Thanks!
> 
> _______________________________________________
> 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 avni.mahajan at eng.knowlarity.com  Tue May 24 08:28:42 2011
From: avni.mahajan at eng.knowlarity.com (Avni Mahajan)
Date: Tue, 24 May 2011 09:58:42 +0530
Subject: [Freeswitch-dev] VXML parser
In-Reply-To: <BANLkTikGk5TR_TAW8UiF9WRb+f5-ua73+A@mail.gmail.com>
References: <BANLkTikGk5TR_TAW8UiF9WRb+f5-ua73+A@mail.gmail.com>
Message-ID: <BANLkTimo17oWXftLamK6RuBD4TqOwzh-Kw@mail.gmail.com>

>
> Hi,
>
> I am looking for a vxml (voicexml) parser in python language. Need to use
> the parsed vxml tags and interact with freeswitch to run IVR. Can anyone
> help me with any kind of opensource vxml parsers?
>
> Regards,
> Avni Mahajan
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110524/a3bd04e9/attachment.html 

From gustavkoller at googlemail.com  Tue May 24 17:27:23 2011
From: gustavkoller at googlemail.com (Gustav Koller)
Date: Tue, 24 May 2011 15:27:23 +0200
Subject: [Freeswitch-dev] Accessing rtp_sessions (switch_rtp) from modules
Message-ID: <BANLkTikB_kM4GGh20VEDj11TsR=q0Oj0HA@mail.gmail.com>

Hello,

I'm new to FreeSwitch (developement).

I'm currently writing a FreeSwitch module where I would like to access
the zrtp_stream property of rtp streams (located in the switch_rtp
struct/switch_rtp.c) from a module.

Example:

switch_rtp_t *rtp_session;
rtp_session = switch_channel_get_private(channel, "__zrtp_audio_rtp_session");

zrtp_stream_t *zrtp_stream;
zrtp_stream = rtp_session->zrtp_stream;

The last line fails with "error: dereferencing pointer to incomplete type"

It something I expected. The struct is defined in switch_rtp.c,
which can't be included. So I can access the properties only from withing
switch_rtp.c.

Am I missing something?
Is there another way to access the active zrtp_streams ?

Best Regards,
Gustav Koller


From mitch.capper at gmail.com  Thu May 26 21:37:51 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 26 May 2011 10:37:51 -0700
Subject: [Freeswitch-dev] mod_portaudio multiple endpoints support
In-Reply-To: <BANLkTim5jOf70cX52n4VJeqNrDWoxf2iHg@mail.gmail.com>
References: <BANLkTim5jOf70cX52n4VJeqNrDWoxf2iHg@mail.gmail.com>
Message-ID: <BANLkTi=8dt5_9n4LQX2Dj6fdA6Dc8yb4eg@mail.gmail.com>

Hi Moy,
First of all I wanted to thank you and Comrex for the new improvements
and code you have done I think a lot of people will find these
features very useful!
Thanks for clarifying some things on IRC.   The changes look good my
main concerns with the current code are how well it reacts to dynamic
changes (usb devices plugged or unplugged)  and the fact all streams
and endpoints have to be specified in the xml config itself.
I am not sure if there is a large reason to have endpoints that are
defined separately from streams.  An endpoint is just a name, a the
indev, the outdev, and the channel for each.  Streams already define
an indev and outdev which makes things a bit confusing.   If channel
selection was moved to the stream definition it could all be defined
there.    The only downside to this that I see is when you have an
audio device with multiple channels you are going to end up
duplicating the stream section partially for each but I am not sure if
that is enough to warrant to require the double configuration of an
audio device before you can start using it.   I would say that if the
device selection was expanded so that rather than just a device ID
/name to also include the channel I think it would be an improved
interface.

Short of that I think before it is merged into trunk the changes
should be merged in a bit better with the existing commands and the
existing stream code should be merged with the shared stream code as
right now some functionality is being duplicated.   While the way the
changes were made looks like relatively few new code paths are
followed if you don't use the new shared streams/endpoints, it makes
for a confusing interface for someone trying to figure out how things
work.

My suggested list of changes would be:
-Merge the existing stream code with the new stream code, this should
eliminate some of the redundancies and allow the new code to be used
everywhere, streams would be expanded to have the naming, channels,
rates, and other features that shared streams allow
-Ability to create streams through the API (with the merge of the
above code its just expanding the API to take the additional params)
-Remove endpoints and allow bridging directly to a stream by using the
stream's name
-Do not validate streams until you go to use them, right now if a
device is not ready when the xml config is read you can't use it
-pa answer, pa call, pa play  are expanded to optionally take the
streamname to act on
-pa switchstream takes the call to switch the stream on
-pa switch this is a tougher one but would need to figure something else out
-Checks to make sure a specific device input is not already in use
prior to use it for input or output by a stream, this needs to go a
step further than just making sure a specific stream isn't used
multiple times as multiple streams could use the same device and I
have found this rarely works out


Many of the changes are just to make portaudio a bit more aware and
capable of multi-active call multi-device support and allowing
anything to be specified dynamically.   It will be important to also
do it in such a way that someone can still use the functions without
inadvertently enabling multiple active calls when they are only
looking to use one device.

I think it will probably be a decent bit of work to do these things
but should allow for the great new features you have added without
causing confusion for other users.   By all means I am not asking you
do this work any time someone contributes code back to FS its great
and something much appreciated and certainly others can work to fully
integrate it into freeswitch.  I figured I would just give some other
feedback about it as you asked for comments.   I will try and test it
on windows sometime shortly also for compatibility there but again
thanks to you and Comrex for the new feature and I am sure it took
some real work to ensure full backwards compatibility.

My 2c.

~Mitch


On Wed, May 25, 2011 at 10:20 PM, Moises Silva <moises.silva at gmail.com> wrote:
> Hello,
> I've been doing some changes to mod_portaudio sponsored by Comrex
> Corporation. The changes involve adding support for using multiple audio
> devices simultaneously, including devices with multiple channels (note that
> is not exactly the same as stereo).


From anthony.minessale at gmail.com  Fri May 27 00:10:15 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 26 May 2011 15:10:15 -0500
Subject: [Freeswitch-dev] Late response from Erlang to dialplan request
 causing Freeswitch to shut down
In-Reply-To: <4DDEB00B.40907@marketelsystems.com>
References: <4DDEAC97.5000604@marketelsystems.com>
	<4DDEAECB.1010209@marketelsystems.com>
	<4DDEB00B.40907@marketelsystems.com>
Message-ID: <BANLkTikrrzyh=iN=_rD1iyxGLEOmsGVCLg@mail.gmail.com>

That TCP fix has nothing to do with anything you are describing.
Its a windows specific fix that was an edge case.



The erlang module is a 3-rd party addition and you would have to take it up wit
On Thu, May 26, 2011 at 2:54 PM, Mark Sobkow
<m.sobkow at marketelsystems.com> wrote:
> On 3:59 PM, Peter Olsson wrote:
>
>> >> ?Did you try latest git? I recently submitted a patch to handle resend on the socket, if the socket is really busy (and returns "would block" state). It's in latest git.
> Would this fix affect the sending of messages to Erlang nodes as well?
> (I suspect it might as it would only make sense to reuse the network IO
> APIs rather than having seperate ones for each module. ?Andrew Thompson
> would know for sure.)
>
> If it does affect Erlang communications, I'll have to do a git-pull and
> schedule some regression testing.
>
> On 26/05/2011 1:49 PM, Mark Sobkow wrote:
>> On further review of the log and matching up notices about FS requests
>> to Erlang responses, it looks like the request that's timing out is
>> getting lost completely and is never processed by Erlang.
>>
>> On 26/05/2011 1:40 PM, Mark Sobkow wrote:
>>> Platform: Ubuntu 10.04 amd64, 8GB RAM, 500GB HDD, quad-core CPU
>>> Configured to use SIP trunking as a slave off an Asterisk server with
>>> a T1 connection to the outside world
>>>
>>> We're using Erlang and PostgreSQL to provide responses to Freeswitch
>>> configuration requests and to manage our call queues, operator
>>> logins, etc.
>>>
>>> With one operator, the system runs fine, so we know the basic coding
>>> we've done works.
>>>
>>> However, with only a couple or three operators on the line and
>>> concurrent dialing going on, we're getting a consistent problem where
>>> the responses to requests for dialplan configurations are getting
>>> seriously delayed -- by well over a minute (an eternity in software
>>> terms.)
>>>
>>> I had suspected that there may be some sort of blocking IO going on
>>> when we issue the originate command (which doesn't return control to
>>> Erlang for up to 30 seconds), but I'm seeing other Freeswitch
>>> requests for directory information getting handled during this same
>>> time period, so the communications between Freeswitch and Erlang do
>>> support concurrent requests.
>>>
>>> The only thing I've been able to find in the mail list logs or wiki
>>> that seem relevant is a comment someone made today about fixing the
>>> TCP transmissions in Freeswitch so that they'll retry the
>>> transmission if the IP stream is clogged and would result in a
>>> blocking call. ?(i.e. FS used to be unreliable about sending out
>>> messages before this patch was done.)
>>>
>>> It's worth noting that Erlang threads are quite busy during the
>>> minute between FS issueing the request and Erlang responding to it.
>>> However, the thread/Erlang process that handles the configuration
>>> requests is idle, so it _should_ be able to respond in a much
>>> timelier fashion.
>>>
>>> It looks to me like there is some sort of delay between FS issuing
>>> the request and Erlang getting the message into it's processing queue.
>>>
>>> Any suggestions/ideas as to what might be causing this? ?And more
>>> importantly -- what can we do to fix or workaround the issue?
>>>
>>> And no, I have not downloaded and installed the latest version of
>>> Freeswitch from git. ?I'm trying to avoid that if possible because
>>> it'll cost me two weeks of regression testing and I need to have this
>>> out to a beta site by the 31st, which doesn't allow enough time to do
>>> the regression tests if we do an upgrade (and only a fool ships
>>> untested software to paying customers.) ?If someone can confirm that
>>> there was a fix checked into git for this issue or a related one,
>>> then I can justify the need for an upgrade and regression testing to
>>> my boss (though he won't be happy about the delay.)
>>>
>>> Here's the relevent snippet of the fs_cli log:
>>>
>>> 2011-05-26 11:45:50.648334 [INFO] mod_dialplan_xml.c:331 Processing
>>> Gotham <5311>->219 in context public
>>> 2011-05-26 11:45:50.648334 [ERR] mod_dialplan_xml.c:353 Open of
>>> dialplan failed
>>> 2011-05-26 11:45:50.648334 [INFO] switch_core_state_machine.c:142 No
>>> Route, Aborting
>>> 2011-05-26 11:45:50.648334 [DEBUG] switch_channel.c:2535
>>> (sofia/external/5311 at 10.77.0.12:6080) Callstate Change RINGING -> HANGUP
>>> 2011-05-26 11:45:50.657061 [NOTICE] switch_core_state_machine.c:143
>>> Hangup sofia/external/5311 at 10.77.0.12:6080 [CS_ROUTING]
>>> [NO_ROUTE_DESTINATION]
>>>
>>>
>>> And here's the log from where our response is finally being issued to
>>> the request, long after FS has decided to shut down because it's not
>>> getting responses from the configuration services:
>>>
>>> 2011-05-26 11:47:45 DEBUG <0.28104.15>: pbx_bind_cfg:fetch() Invoking
>>> xml_fetch()
>>> 2011-05-26 11:47:45 DEBUG <0.28104.15>: pbx_bind_cfg:xml_fetch(
>>> {fetch, dialplan, Tag=undefined, Key=undefined, Value=undefined,
>>> Params=...} )
>>> 2011-05-26 11:47:46 DEBUG <0.28104.15>: pbx_bind_cfg:fetch() Response
>>> XML is
>>> <document type="freeswitch/xml">
>>> <section name="dialplan" description="Dial Plan For FreeSwitch">
>>> <context name="default">
>>> <extension required_name="ConferenceDialer">
>>> <condition field="destination_number" expression="^8(\d\d\d)$">
>>> <action application="conference" data="conf$0 at dialer"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="dialer_park">
>>> <condition field="destination_number" expression="^park$">
>>> <action application="fifo" data="park@${domain_name} in" />
>>> </condition>
>>> </extension>
>>> <extension required_name="park">
>>> <condition field="destination_number" expression="^7000$">
>>> <action application="set" data="fifo_music=$${hold_music}" />
>>> <action application="fifo" data="7000@${domain_name} in" />
>>> </condition>
>>> </extension>
>>> <extension required_name="unpark">
>>> <condition field="destination_number" expression="^7001$">
>>> <action application="answer" />
>>> <action application="fifo" data="7000@${domain_name} out nowait" />
>>> </condition>
>>> </extension>
>>> <extension required_name="InternalFS">
>>> <condition field="destination_number" expression="^(\d\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0%pursuit.marketel"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="RatsExtension">
>>> <condition field="destination_number" expression="^(\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="LocalCall">
>>> <condition field="destination_number" expression="^(9\d\d\d\d\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="LongDistanceCall">
>>> <condition field="destination_number"
>>> expression="^91(\d\d\d\d\d\d\d\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>>> </condition>
>>> </extension>
>>> </context>
>>> <context name="public">
>>> <extension required_name="Operator">
>>> <condition field="destination_number" expression="^(0)$">
>>> <action application="bridge" data="sofia/external/160 at rats.marketel"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="OperatorLogin">
>>> <condition field="destination_number" expression="^620(\d)$">
>>> <action application="answer" />
>>> <action application="set" data="chain_id=$1" />
>>> <action application="set" data="hangup_after_bridge=false" />
>>> <action application="play_and_get_digits" data="4 8 1 30000 #
>>> /usr/lib/pursuit/lib/pbx-1.2/priv/welcome_please_enter_pin-8000.wav
>>> /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_pin_try_again-8000.wav
>>> operator_pin \d+" />
>>> <action application="log" data="INFO operator_pin is ${operator_pin},
>>> chain_id is ${chain_id}" />
>>> <action application="erlang" data="pbx_op_logged_in pursuit at pursuit" />
>>> </condition>
>>> </extension>
>>>
>>> <extension required_name="OperatorLogin2">
>>> <condition field="destination_number" expression="^62(\d\d)$">
>>> <action application="answer" />
>>> <action application="set" data="chain_id=$1" />
>>> <action application="set" data="hangup_after_bridge=false" />
>>> <action application="play_and_get_digits" data="4 8 1 30000 #
>>> /usr/lib/pursuit/lib/pbx-1.2/priv/welcome_please_enter_pin-8000.wav
>>> /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_pin_try_again-8000.wav
>>> operator_pin \d+" />
>>> <action application="log" data="INFO operator_pin is ${operator_pin},
>>> chain_id is ${chain_id}" />
>>> <action application="erlang" data="pbx_op_logged_in pursuit at pursuit" />
>>> </condition>
>>> </extension>
>>> <extension required_name="CollectCallResult">
>>> <condition field="destination_number" expression="^63(\d\d)$">
>>> <action application="answer" />
>>> <action application="set" data="hangup_after_bridge=false" />
>>> <action application="play_and_get_digits" data="1 2 1 30000 #
>>> /usr/lib/pursuit/lib/pbx-1.2/priv/please_enter_result_code-8000.wav
>>> /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_result_code-8000.wav
>>> call_result \d+" />
>>> <action application="log" data="INFO call_result is ${call_result}" />
>>> <action application="erlang" data="pbx_call_result pursuit at pursuit" />
>>> </condition>
>>> </extension>
>>>
>>> <extension required_name="DummyCallResult">
>>> <condition field="destination_number" expression="^6400$">
>>> <action application="answer" />
>>> <action application="erlang" data="pbx_dummy_call_result
>>> pursuit at pursuit" />
>>> </condition>
>>> </extension>
>>>
>>> <extension required_name="CustomerCallin">
>>> <condition field="destination_number" expression="^650(\d)$">
>>> <action application="answer" />
>>> <action application="set" data="chain_id=$1" />
>>> <action application="erlang" data="pbx_cust_called pursuit at pursuit" />
>>> </condition>
>>> </extension>
>>> <extension required_name="CustomerCallin2">
>>> <condition field="destination_number" expression="^65(\d\d)$">
>>> <action application="answer" />
>>> <action application="set" data="chain_id=$1" />
>>> <action application="erlang" data="pbx_cust_called pursuit at pursuit" />
>>> </condition>
>>> </extension>
>>>
>>> <extension required_name="TestIVR">
>>> <condition field="destination_number" expression="^6600$">
>>> <action application="answer" />
>>> <action application="ivr" data="demo_ivr" />
>>> </condition>
>>> </extension>
>>>
>>> <extension required_name="eavesdrop">
>>> <condition field="destination_number" expression="^67(\d\d\d\d)$">
>>> <action application="answer"/>
>>> <action application="set" data="eavesdrop_extension=$1" />
>>> <action application="erlang" data="pbx_eavesdrop pursuit at pursuit" />
>>> </condition>
>>> </extension>
>>>
>>> <extension required_name="park">
>>> <condition field="destination_number" expression="^6003$">
>>> <action application="fifo" data="6003@${domain_name} in" />
>>> </condition>
>>> </extension>
>>> <extension required_name="unpark">
>>> <condition field="destination_number" expression="^6004$">
>>> <action application="answer" />
>>> <action application="fifo" data="6003@${domain_name} out wait" />
>>> </condition>
>>> </extension>
>>>
>>> <extension required_name="ConferenceDialer">
>>> <condition field="destination_number" expression="^8(\d\d\d)$">
>>> <action application="conference" data="conf$0 at dialer"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="InternalFS">
>>> <condition field="destination_number" expression="^(\d\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0%10.77.0.12"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="RatsExtension">
>>> <condition field="destination_number" expression="^(\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="LocalCall">
>>> <condition field="destination_number" expression="^(9\d\d\d\d\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="LongDistanceCall">
>>> <condition field="destination_number"
>>> expression="^91(\d\d\d\d\d\d\d\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>>> </condition>
>>> </extension>
>>> </context>
>>> </section>
>>> </document>
>>>
>>>
>>
>>
>
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
> Visit our Blog for industry related information.
> http://marketel-systems.blogspot.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


From anthony.minessale at gmail.com  Fri May 27 00:12:38 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 26 May 2011 15:12:38 -0500
Subject: [Freeswitch-dev] Accessing rtp_sessions (switch_rtp) from
	modules
In-Reply-To: <BANLkTikB_kM4GGh20VEDj11TsR=q0Oj0HA@mail.gmail.com>
References: <BANLkTikB_kM4GGh20VEDj11TsR=q0Oj0HA@mail.gmail.com>
Message-ID: <BANLkTikWX_zSx8p9Amh+zDLg8SXJNgmLdg@mail.gmail.com>

you would have to propose a patch to switch_rtp.c to add a pointer to
the zrtp stream using the switch_channel_set_private
so you could get it from your module.

On Tue, May 24, 2011 at 8:27 AM, Gustav Koller
<gustavkoller at googlemail.com> wrote:
> Hello,
>
> I'm new to FreeSwitch (developement).
>
> I'm currently writing a FreeSwitch module where I would like to access
> the zrtp_stream property of rtp streams (located in the switch_rtp
> struct/switch_rtp.c) from a module.
>
> Example:
>
> switch_rtp_t *rtp_session;
> rtp_session = switch_channel_get_private(channel, "__zrtp_audio_rtp_session");
>
> zrtp_stream_t *zrtp_stream;
> zrtp_stream = rtp_session->zrtp_stream;
>
> The last line fails with "error: dereferencing pointer to incomplete type"
>
> It something I expected. The struct is defined in switch_rtp.c,
> which can't be included. So I can access the properties only from withing
> switch_rtp.c.
>
> Am I missing something?
> Is there another way to access the active zrtp_streams ?
>
> Best Regards,
> Gustav Koller
>
> _______________________________________________
> 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


From peter.olsson at visionutveckling.se  Fri May 27 01:02:11 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 26 May 2011 23:02:11 +0200
Subject: [Freeswitch-dev] Late response from Erlang to dialplan request
 causing Freeswitch to shut down
In-Reply-To: <BANLkTikrrzyh=iN=_rD1iyxGLEOmsGVCLg@mail.gmail.com>
References: <4DDEAC97.5000604@marketelsystems.com>
	<4DDEAECB.1010209@marketelsystems.com>
	<4DDEB00B.40907@marketelsystems.com>,
	<BANLkTikrrzyh=iN=_rD1iyxGLEOmsGVCLg@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DE2B6305@cooper>

Tony, actually it was Linux specific (it was already handled for Windows) :) But yes, it was for a quite specific case, and the only time I could reproduce that specific problem was when sending much more events/data over a WAN connection (slow one) then it could actually swallow - which is usually not a problem for anyone these days...

/Peter
________________________________________
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] f&#246;r Anthony Minessale [anthony.minessale at gmail.com]
Skickat: den 26 maj 2011 22:10
Till: freeswitch-dev at lists.freeswitch.org
?mne: Re: [Freeswitch-dev] Late response from Erlang to dialplan request causing Freeswitch to shut down

That TCP fix has nothing to do with anything you are describing.
Its a windows specific fix that was an edge case.



The erlang module is a 3-rd party addition and you would have to take it up wit
On Thu, May 26, 2011 at 2:54 PM, Mark Sobkow
<m.sobkow at marketelsystems.com> wrote:
> On 3:59 PM, Peter Olsson wrote:
>
>> >>  Did you try latest git? I recently submitted a patch to handle resend on the socket, if the socket is really busy (and returns "would block" state). It's in latest git.
> Would this fix affect the sending of messages to Erlang nodes as well?
> (I suspect it might as it would only make sense to reuse the network IO
> APIs rather than having seperate ones for each module.  Andrew Thompson
> would know for sure.)
>
> If it does affect Erlang communications, I'll have to do a git-pull and
> schedule some regression testing.
>
> On 26/05/2011 1:49 PM, Mark Sobkow wrote:
>> On further review of the log and matching up notices about FS requests
>> to Erlang responses, it looks like the request that's timing out is
>> getting lost completely and is never processed by Erlang.
>>
>> On 26/05/2011 1:40 PM, Mark Sobkow wrote:
>>> Platform: Ubuntu 10.04 amd64, 8GB RAM, 500GB HDD, quad-core CPU
>>> Configured to use SIP trunking as a slave off an Asterisk server with
>>> a T1 connection to the outside world
>>>
>>> We're using Erlang and PostgreSQL to provide responses to Freeswitch
>>> configuration requests and to manage our call queues, operator
>>> logins, etc.
>>>
>>> With one operator, the system runs fine, so we know the basic coding
>>> we've done works.
>>>
>>> However, with only a couple or three operators on the line and
>>> concurrent dialing going on, we're getting a consistent problem where
>>> the responses to requests for dialplan configurations are getting
>>> seriously delayed -- by well over a minute (an eternity in software
>>> terms.)
>>>
>>> I had suspected that there may be some sort of blocking IO going on
>>> when we issue the originate command (which doesn't return control to
>>> Erlang for up to 30 seconds), but I'm seeing other Freeswitch
>>> requests for directory information getting handled during this same
>>> time period, so the communications between Freeswitch and Erlang do
>>> support concurrent requests.
>>>
>>> The only thing I've been able to find in the mail list logs or wiki
>>> that seem relevant is a comment someone made today about fixing the
>>> TCP transmissions in Freeswitch so that they'll retry the
>>> transmission if the IP stream is clogged and would result in a
>>> blocking call.  (i.e. FS used to be unreliable about sending out
>>> messages before this patch was done.)
>>>
>>> It's worth noting that Erlang threads are quite busy during the
>>> minute between FS issueing the request and Erlang responding to it.
>>> However, the thread/Erlang process that handles the configuration
>>> requests is idle, so it _should_ be able to respond in a much
>>> timelier fashion.
>>>
>>> It looks to me like there is some sort of delay between FS issuing
>>> the request and Erlang getting the message into it's processing queue.
>>>
>>> Any suggestions/ideas as to what might be causing this?  And more
>>> importantly -- what can we do to fix or workaround the issue?
>>>
>>> And no, I have not downloaded and installed the latest version of
>>> Freeswitch from git.  I'm trying to avoid that if possible because
>>> it'll cost me two weeks of regression testing and I need to have this
>>> out to a beta site by the 31st, which doesn't allow enough time to do
>>> the regression tests if we do an upgrade (and only a fool ships
>>> untested software to paying customers.)  If someone can confirm that
>>> there was a fix checked into git for this issue or a related one,
>>> then I can justify the need for an upgrade and regression testing to
>>> my boss (though he won't be happy about the delay.)
>>>
>>> Here's the relevent snippet of the fs_cli log:
>>>
>>> 2011-05-26 11:45:50.648334 [INFO] mod_dialplan_xml.c:331 Processing
>>> Gotham <5311>->219 in context public
>>> 2011-05-26 11:45:50.648334 [ERR] mod_dialplan_xml.c:353 Open of
>>> dialplan failed
>>> 2011-05-26 11:45:50.648334 [INFO] switch_core_state_machine.c:142 No
>>> Route, Aborting
>>> 2011-05-26 11:45:50.648334 [DEBUG] switch_channel.c:2535
>>> (sofia/external/5311 at 10.77.0.12:6080) Callstate Change RINGING -> HANGUP
>>> 2011-05-26 11:45:50.657061 [NOTICE] switch_core_state_machine.c:143
>>> Hangup sofia/external/5311 at 10.77.0.12:6080 [CS_ROUTING]
>>> [NO_ROUTE_DESTINATION]
>>>
>>>
>>> And here's the log from where our response is finally being issued to
>>> the request, long after FS has decided to shut down because it's not
>>> getting responses from the configuration services:
>>>
>>> 2011-05-26 11:47:45 DEBUG <0.28104.15>: pbx_bind_cfg:fetch() Invoking
>>> xml_fetch()
>>> 2011-05-26 11:47:45 DEBUG <0.28104.15>: pbx_bind_cfg:xml_fetch(
>>> {fetch, dialplan, Tag=undefined, Key=undefined, Value=undefined,
>>> Params=...} )
>>> 2011-05-26 11:47:46 DEBUG <0.28104.15>: pbx_bind_cfg:fetch() Response
>>> XML is
>>> <document type="freeswitch/xml">
>>> <section name="dialplan" description="Dial Plan For FreeSwitch">
>>> <context name="default">
>>> <extension required_name="ConferenceDialer">
>>> <condition field="destination_number" expression="^8(\d\d\d)$">
>>> <action application="conference" data="conf$0 at dialer"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="dialer_park">
>>> <condition field="destination_number" expression="^park$">
>>> <action application="fifo" data="park@${domain_name} in" />
>>> </condition>
>>> </extension>
>>> <extension required_name="park">
>>> <condition field="destination_number" expression="^7000$">
>>> <action application="set" data="fifo_music=$${hold_music}" />
>>> <action application="fifo" data="7000@${domain_name} in" />
>>> </condition>
>>> </extension>
>>> <extension required_name="unpark">
>>> <condition field="destination_number" expression="^7001$">
>>> <action application="answer" />
>>> <action application="fifo" data="7000@${domain_name} out nowait" />
>>> </condition>
>>> </extension>
>>> <extension required_name="InternalFS">
>>> <condition field="destination_number" expression="^(\d\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0%pursuit.marketel"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="RatsExtension">
>>> <condition field="destination_number" expression="^(\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="LocalCall">
>>> <condition field="destination_number" expression="^(9\d\d\d\d\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="LongDistanceCall">
>>> <condition field="destination_number"
>>> expression="^91(\d\d\d\d\d\d\d\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>>> </condition>
>>> </extension>
>>> </context>
>>> <context name="public">
>>> <extension required_name="Operator">
>>> <condition field="destination_number" expression="^(0)$">
>>> <action application="bridge" data="sofia/external/160 at rats.marketel"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="OperatorLogin">
>>> <condition field="destination_number" expression="^620(\d)$">
>>> <action application="answer" />
>>> <action application="set" data="chain_id=$1" />
>>> <action application="set" data="hangup_after_bridge=false" />
>>> <action application="play_and_get_digits" data="4 8 1 30000 #
>>> /usr/lib/pursuit/lib/pbx-1.2/priv/welcome_please_enter_pin-8000.wav
>>> /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_pin_try_again-8000.wav
>>> operator_pin \d+" />
>>> <action application="log" data="INFO operator_pin is ${operator_pin},
>>> chain_id is ${chain_id}" />
>>> <action application="erlang" data="pbx_op_logged_in pursuit at pursuit" />
>>> </condition>
>>> </extension>
>>>
>>> <extension required_name="OperatorLogin2">
>>> <condition field="destination_number" expression="^62(\d\d)$">
>>> <action application="answer" />
>>> <action application="set" data="chain_id=$1" />
>>> <action application="set" data="hangup_after_bridge=false" />
>>> <action application="play_and_get_digits" data="4 8 1 30000 #
>>> /usr/lib/pursuit/lib/pbx-1.2/priv/welcome_please_enter_pin-8000.wav
>>> /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_pin_try_again-8000.wav
>>> operator_pin \d+" />
>>> <action application="log" data="INFO operator_pin is ${operator_pin},
>>> chain_id is ${chain_id}" />
>>> <action application="erlang" data="pbx_op_logged_in pursuit at pursuit" />
>>> </condition>
>>> </extension>
>>> <extension required_name="CollectCallResult">
>>> <condition field="destination_number" expression="^63(\d\d)$">
>>> <action application="answer" />
>>> <action application="set" data="hangup_after_bridge=false" />
>>> <action application="play_and_get_digits" data="1 2 1 30000 #
>>> /usr/lib/pursuit/lib/pbx-1.2/priv/please_enter_result_code-8000.wav
>>> /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_result_code-8000.wav
>>> call_result \d+" />
>>> <action application="log" data="INFO call_result is ${call_result}" />
>>> <action application="erlang" data="pbx_call_result pursuit at pursuit" />
>>> </condition>
>>> </extension>
>>>
>>> <extension required_name="DummyCallResult">
>>> <condition field="destination_number" expression="^6400$">
>>> <action application="answer" />
>>> <action application="erlang" data="pbx_dummy_call_result
>>> pursuit at pursuit" />
>>> </condition>
>>> </extension>
>>>
>>> <extension required_name="CustomerCallin">
>>> <condition field="destination_number" expression="^650(\d)$">
>>> <action application="answer" />
>>> <action application="set" data="chain_id=$1" />
>>> <action application="erlang" data="pbx_cust_called pursuit at pursuit" />
>>> </condition>
>>> </extension>
>>> <extension required_name="CustomerCallin2">
>>> <condition field="destination_number" expression="^65(\d\d)$">
>>> <action application="answer" />
>>> <action application="set" data="chain_id=$1" />
>>> <action application="erlang" data="pbx_cust_called pursuit at pursuit" />
>>> </condition>
>>> </extension>
>>>
>>> <extension required_name="TestIVR">
>>> <condition field="destination_number" expression="^6600$">
>>> <action application="answer" />
>>> <action application="ivr" data="demo_ivr" />
>>> </condition>
>>> </extension>
>>>
>>> <extension required_name="eavesdrop">
>>> <condition field="destination_number" expression="^67(\d\d\d\d)$">
>>> <action application="answer"/>
>>> <action application="set" data="eavesdrop_extension=$1" />
>>> <action application="erlang" data="pbx_eavesdrop pursuit at pursuit" />
>>> </condition>
>>> </extension>
>>>
>>> <extension required_name="park">
>>> <condition field="destination_number" expression="^6003$">
>>> <action application="fifo" data="6003@${domain_name} in" />
>>> </condition>
>>> </extension>
>>> <extension required_name="unpark">
>>> <condition field="destination_number" expression="^6004$">
>>> <action application="answer" />
>>> <action application="fifo" data="6003@${domain_name} out wait" />
>>> </condition>
>>> </extension>
>>>
>>> <extension required_name="ConferenceDialer">
>>> <condition field="destination_number" expression="^8(\d\d\d)$">
>>> <action application="conference" data="conf$0 at dialer"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="InternalFS">
>>> <condition field="destination_number" expression="^(\d\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0%10.77.0.12"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="RatsExtension">
>>> <condition field="destination_number" expression="^(\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="LocalCall">
>>> <condition field="destination_number" expression="^(9\d\d\d\d\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>>> </condition>
>>> </extension>
>>> <extension required_name="LongDistanceCall">
>>> <condition field="destination_number"
>>> expression="^91(\d\d\d\d\d\d\d\d\d\d)$">
>>> <action application="bridge" data="sofia/external/$0 at rats.marketel"/>
>>> </condition>
>>> </extension>
>>> </context>
>>> </section>
>>> </document>
>>>
>>>
>>
>>
>
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
> Visit our Blog for industry related information.
> http://marketel-systems.blogspot.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

_______________________________________________
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

!DSPAM:4ddeb38d32764268878487!



From brian at freeswitch.org  Fri May 27 01:35:45 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 26 May 2011 16:35:45 -0500
Subject: [Freeswitch-dev] extra sdp is always added in
	sofia_answer_channel ?
In-Reply-To: <27b9b2d6-a560-49f7-8143-ed34d91649a3@mx.voismart.com>
References: <27b9b2d6-a560-49f7-8143-ed34d91649a3@mx.voismart.com>
Message-ID: <CDD0FC7D-77E6-4865-B83D-4FE6BBEC91AE@freeswitch.org>

Please post to jira.freeswitch.org

/b
On May 26, 2011, at 7:09 AM, Matteo wrote:

> Hi,  
> 
> don't know if is a bug/typo or is correct, so I'm posting here.
> 
> I'm noticed that in sofia_answer_channel (@mod_sofia.c:738, latest git)
> NUTAG_INCLUDE_EXTRA_SDP(1) is always added.
> 
> Since we're into soa mode, maybe it should be added only 
> by testing if 100rel is disabled, so it should be:
> 
> --- mod_sofia.c.orig	2011-05-26 14:16:34.000000000 +0200
> +++ mod_sofia.c	2011-05-26 14:16:47.000000000 +0200
> @@ -734,7 +734,8 @@
> 						SIPTAG_CONTACT_STR(tech_pvt->reply_contact),
> 						SIPTAG_CALL_INFO_STR(switch_channel_get_variable(tech_pvt->channel, SOFIA_SIP_HEADER_PREFIX "call_info")),
> 						SOATAG_USER_SDP_STR(tech_pvt->local_sdp_str),
> -						SOATAG_REUSE_REJECTED(1), SOATAG_ORDERED_USER(1), SOATAG_AUDIO_AUX("cn telephone-event"),NUTAG_INCLUDE_EXTRA_SDP(1),
> +						SOATAG_REUSE_REJECTED(1), SOATAG_ORDERED_USER(1), SOATAG_AUDIO_AUX("cn telephone-event"),
> +						TAG_IF(sofia_test_pflag(tech_pvt->profile, PFLAG_DISABLE_100REL), NUTAG_INCLUDE_EXTRA_SDP(1)),
> 						TAG_IF(!zstr(extra_headers), SIPTAG_HEADER_STR(extra_headers)),
> 						TAG_IF(switch_stristr("update_display", tech_pvt->x_freeswitch_support_remote),
> 							   SIPTAG_HEADER_STR("X-FS-Support: " FREESWITCH_SUPPORT)), TAG_END());
> 
> Or is correct to add it always?
> 
> I've had to fix that because my upstream provider requires 100rel
> and does not want sdp in 200OK if is not changed from last progress message.
> 
> Regards,
> Matteo
> 
> _______________________________________________
> 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 brian at freeswitch.org  Fri May 27 01:36:36 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 26 May 2011 16:36:36 -0500
Subject: [Freeswitch-dev] Accessing rtp_sessions (switch_rtp) from
	modules
In-Reply-To: <BANLkTikWX_zSx8p9Amh+zDLg8SXJNgmLdg@mail.gmail.com>
References: <BANLkTikB_kM4GGh20VEDj11TsR=q0Oj0HA@mail.gmail.com>
	<BANLkTikWX_zSx8p9Amh+zDLg8SXJNgmLdg@mail.gmail.com>
Message-ID: <6587AD8D-FD52-4E7E-B842-DFDF2FD688C5@freeswitch.org>

I'm going to guess you're working on the app level to toggle things around ?

/b

On May 26, 2011, at 3:12 PM, Anthony Minessale wrote:

> you would have to propose a patch to switch_rtp.c to add a pointer to
> the zrtp stream using the switch_channel_set_private
> so you could get it from your module.



From moises.silva at gmail.com  Fri May 27 05:47:16 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 26 May 2011 21:47:16 -0400
Subject: [Freeswitch-dev] mod_portaudio multiple endpoints support
In-Reply-To: <BANLkTi=8dt5_9n4LQX2Dj6fdA6Dc8yb4eg@mail.gmail.com>
References: <BANLkTim5jOf70cX52n4VJeqNrDWoxf2iHg@mail.gmail.com>
	<BANLkTi=8dt5_9n4LQX2Dj6fdA6Dc8yb4eg@mail.gmail.com>
Message-ID: <BANLkTimqq6XnKQcK3cx2aZTRN0hZ-Hx09w@mail.gmail.com>

Hello Mitch,

Thanks for taking your time to review this feature and share your thoughts.

On Thu, May 26, 2011 at 1:37 PM, Mitch Capper <mitch.capper at gmail.com>wrote:

> Hi Moy,
> First of all I wanted to thank you and Comrex for the new improvements
> and code you have done I think a lot of people will find these
> features very useful!
>

You're certainly welcomed, it is nice to work with companies like Comrex
that realize is better to contribute and be part of the community.


> Thanks for clarifying some things on IRC.   The changes look good my
> main concerns with the current code are how well it reacts to dynamic
> changes (usb devices plugged or unplugged)  and the fact all streams
> and endpoints have to be specified in the xml config itself.
>

Where else should be specified if not in the XML config? doing it
dynamically would have created extra complexity and I did not feel added
much value for the current user.


> I am not sure if there is a large reason to have endpoints that are
> defined separately from streams.  An endpoint is just a name, a the
> indev, the outdev, and the channel for each.  Streams already define
> an indev and outdev which makes things a bit confusing.   If channel
> selection was moved to the stream definition it could all be defined
> there.    The only downside to this that I see is when you have an
> audio device with multiple channels you are going to end up
> duplicating the stream section partially for each but I am not sure if
> that is enough to warrant to require the double configuration of an
> audio device before you can start using it.   I would say that if the
> device selection was expanded so that rather than just a device ID
> /name to also include the channel I think it would be an improved
> interface.
>

Well, you have also the sampling rate and IO sizes (codec-ms) which belong
to the stream. The main reason for the separation is that they are meant to
be used separate. You can define 10 different endpoints using the same
stream. As long as they don't conflict in the channel they use and the
direction. You can define an endpoint that only uses the input of a given
stream and channel, and does not use the output, and another endpoint doing
the opposite, both at the same time having a call using the same stream.


> Short of that I think before it is merged into trunk the changes
> should be merged in a bit better with the existing commands and the
> existing stream code should be merged with the shared stream code as
> right now some functionality is being duplicated.   While the way the
> changes were made looks like relatively few new code paths are
> followed if you don't use the new shared streams/endpoints, it makes
> for a confusing interface for someone trying to figure out how things
> work.
>

The decision here had to do mostly with backwards compatibility and the
amount of things I'd most likely break if I touched the original code paths
and the amount of testing required to make sure the previous functionality
wasn't affected. I'd rather slowly start moving chunks from the old code
compatible with the new one as the need arises.


> My suggested list of changes would be:
> -Merge the existing stream code with the new stream code, this should
> eliminate some of the redundancies and allow the new code to be used
> everywhere, streams would be expanded to have the naming, channels,
> rates, and other features that shared streams allow
>

I don't think I'd be able to do that at this point.


> -Ability to create streams through the API (with the merge of the
> above code its just expanding the API to take the additional params)
> -Remove endpoints and allow bridging directly to a stream by using the
> stream's name
>

For the reasons I explained before (the same stream being used by multiple
endpoints), I can't see how this would work.


> -Do not validate streams until you go to use them, right now if a
> device is not ready when the xml config is read you can't use it


I like failing early, is easier to spot mistakes in the configuration. I
understand this might be a problem if you're planing to plug/unplug USB
devices, but that is not our case yet :-)


> -pa answer, pa call, pa play  are expanded to optionally take the
> streamname to act on
> -pa switchstream takes the call to switch the stream on
> -pa switch this is a tougher one but would need to figure something else
> out


pa answer, pa call are replaced by uuid_outgoing_answer, originate from the
command line respectively.


> I think it will probably be a decent bit of work to do these things
> but should allow for the great new features you have added without
> causing confusion for other users.


The current state of things is exactly to avoid confusing existing users.
They can go ahead and keep being happy with their current monodevice
configuration and usage. If, and only if they want to move to multi devices
they will have to learn something new. I found too complicated expanding the
current API's to support the new architecture without changing the usage of
the commands.


> By all means I am not asking you
> do this work any time someone contributes code back to FS its great
> and something much appreciated and certainly others can work to fully
> integrate it into freeswitch.  I figured I would just give some other
> feedback about it as you asked for comments.


Your input is very much appreciated. Although I can certainly tell you,
given that my own needs and Comrex needs are completely fulfilled with the
current feature set, I am looking mostly for bugs or small improvements.
Most the things you described make the interface nicer and consistent, but
will require a reasonable amount of work and a high chance of breaking
existing users (either monodevice users or Comrex themselves).

Once the code is merged is easier to start improving one thing at the time.

-
Moy
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110526/abf7cbe8/attachment-0001.html 

From mitch.capper at gmail.com  Fri May 27 07:10:48 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 26 May 2011 20:10:48 -0700
Subject: [Freeswitch-dev] mod_portaudio multiple endpoints support
In-Reply-To: <BANLkTimqq6XnKQcK3cx2aZTRN0hZ-Hx09w@mail.gmail.com>
References: <BANLkTim5jOf70cX52n4VJeqNrDWoxf2iHg@mail.gmail.com>
	<BANLkTi=8dt5_9n4LQX2Dj6fdA6Dc8yb4eg@mail.gmail.com>
	<BANLkTimqq6XnKQcK3cx2aZTRN0hZ-Hx09w@mail.gmail.com>
Message-ID: <BANLkTim7d7omgr0dpN0de8USvSzxO35aGQ@mail.gmail.com>

> Where else should be specified if not in the XML config? doing it
> dynamically would have created extra complexity and I did not feel added
> much value for the current user.
Sorry I was not saying not in the xml config, just the ability to
specify it through the API is useful.  Freeswitch can be interfaced
through things in a lot of ways and many users configure portaudio
through the command line or web interface so it makes sense just to do
it both ways I figure.

> Well, you have also the sampling rate and IO sizes (codec-ms) which belong
> to the stream. The main reason for the separation is that they are meant to
> be used separate. You can define 10 different endpoints using the same
> stream. As long as they don't conflict in the channel they use and the
> direction. You can define an endpoint that only uses the input of a given
> stream and channel, and does not use the output, and another endpoint doing
> the opposite, both at the same time having a call using the same stream.
Sure, but you could just as easily define multiple <stream> entries
that use the same device but different channels.

> Your input is very much appreciated. Although I can certainly tell you,
> given that my own needs and Comrex needs are completely fulfilled with the
> current feature set, I am looking mostly for bugs or small improvements.
Completely understandable my suggestions were a bit less of what you
should do and more of what should be done for integration into trunk.
  Your code works well, obviously accomplishes what you need, and
doesn't break existing stuff all which are good.  The downside is
anyone trying to understand portaudio and use portaudio now has two
very different interfaces for doing so depending on what they want to
do.   In addition any code that wants to take advantage of
multi-channel portaudio has to be rewritten.

~Mitch


On Thu, May 26, 2011 at 6:47 PM, Moises Silva <moises.silva at gmail.com> wrote:
> Hello Mitch,
> Thanks for taking your time to review this feature and share your thoughts.
>
> On Thu, May 26, 2011 at 1:37 PM, Mitch Capper <mitch.capper at gmail.com>
> wrote:
>>
>> Hi Moy,
>> First of all I wanted to thank you and Comrex for the new improvements
>> and code you have done I think a lot of people will find these
>> features very useful!
>
> You're certainly welcomed, it is nice to work with companies like Comrex
> that realize is better to contribute and be part of the community.
>
>>
>> Thanks for clarifying some things on IRC. ? The changes look good my
>> main concerns with the current code are how well it reacts to dynamic
>> changes (usb devices plugged or unplugged) ?and the fact all streams
>> and endpoints have to be specified in the xml config itself.
>
> Where else should be specified if not in the XML config? doing it
> dynamically would have created extra complexity and I did not feel added
> much value for the current user.
>
>>
>> I am not sure if there is a large reason to have endpoints that are
>> defined separately from streams. ?An endpoint is just a name, a the
>> indev, the outdev, and the channel for each. ?Streams already define
>> an indev and outdev which makes things a bit confusing. ? If channel
>> selection was moved to the stream definition it could all be defined
>> there. ? ?The only downside to this that I see is when you have an
>> audio device with multiple channels you are going to end up
>> duplicating the stream section partially for each but I am not sure if
>> that is enough to warrant to require the double configuration of an
>> audio device before you can start using it. ? I would say that if the
>> device selection was expanded so that rather than just a device ID
>> /name to also include the channel I think it would be an improved
>> interface.
>
> Well, you have also the sampling rate and IO sizes (codec-ms) which belong
> to the stream. The main reason for the separation is that they are meant to
> be used separate. You can define 10 different endpoints using the same
> stream. As long as they don't conflict in the channel they use and the
> direction. You can define an endpoint that only uses the input of a given
> stream and channel, and does not use the output, and another endpoint doing
> the opposite, both at the same time having a call using the same stream.
>
>>
>> Short of that I think before it is merged into trunk the changes
>> should be merged in a bit better with the existing commands and the
>> existing stream code should be merged with the shared stream code as
>> right now some functionality is being duplicated. ? While the way the
>> changes were made looks like relatively few new code paths are
>> followed if you don't use the new shared streams/endpoints, it makes
>> for a confusing interface for someone trying to figure out how things
>> work.
>
> The decision here had to do mostly with backwards compatibility and the
> amount of things I'd most likely break if I touched the original code paths
> and the amount of testing required to make sure the previous functionality
> wasn't affected. I'd rather slowly start moving chunks from the old code
> compatible with the new one as the need arises.
>
>>
>> My suggested list of changes would be:
>> -Merge the existing stream code with the new stream code, this should
>> eliminate some of the redundancies and allow the new code to be used
>> everywhere, streams would be expanded to have the naming, channels,
>> rates, and other features that shared streams allow
>
> I don't think I'd be able to do that at this point.
>
>>
>> -Ability to create streams through the API (with the merge of the
>> above code its just expanding the API to take the additional params)
>> -Remove endpoints and allow bridging directly to a stream by using the
>> stream's name
>
> For the reasons I explained before (the same stream being used by multiple
> endpoints), I can't see how this would work.
>
>>
>> -Do not validate streams until you go to use them, right now if a
>> device is not ready when the xml config is read you can't use it
>
> I like failing early, is easier to spot mistakes in the configuration. I
> understand this might be a problem if you're planing to plug/unplug USB
> devices, but that is not our case yet :-)
>
>>
>> -pa answer, pa call, pa play ?are expanded to optionally take the
>> streamname to act on
>> -pa switchstream takes the call to switch the stream on
>> -pa switch this is a tougher one but would need to figure something else
>> out
>
> pa answer, pa call are replaced by uuid_outgoing_answer, originate from the
> command line respectively.
>
>>
>> I think it will probably be a decent bit of work to do these things
>> but should allow for the great new features you have added without
>> causing confusion for other users.
>
> The current state of things is exactly to avoid confusing existing users.
> They can go ahead and keep being happy with their current monodevice
> configuration and usage. If, and only if they want to move to multi devices
> they will have to learn something new. I found too complicated expanding the
> current API's to support the new architecture without changing the usage of
> the commands.
>
>>
>> By all means I am not asking you
>> do this work any time someone contributes code back to FS its great
>> and something much appreciated and certainly others can work to fully
>> integrate it into freeswitch. ?I figured I would just give some other
>> feedback about it as you asked for comments.
>
> Your input is very much appreciated. Although I can certainly tell you,
> given that my own needs and Comrex needs are completely?fulfilled with the
> current feature set, I am looking mostly for bugs or small improvements.
> Most the things you described make the interface nicer and consistent, but
> will require a reasonable amount of work and a high chance of breaking
> existing users (either monodevice users or Comrex themselves).
> Once the code is merged is easier to start improving one thing at the time.
> -
> Moy
>
> _______________________________________________
> 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 moises.silva at gmail.com  Fri May 27 08:01:45 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 27 May 2011 00:01:45 -0400
Subject: [Freeswitch-dev] mod_portaudio multiple endpoints support
In-Reply-To: <BANLkTim7d7omgr0dpN0de8USvSzxO35aGQ@mail.gmail.com>
References: <BANLkTim5jOf70cX52n4VJeqNrDWoxf2iHg@mail.gmail.com>
	<BANLkTi=8dt5_9n4LQX2Dj6fdA6Dc8yb4eg@mail.gmail.com>
	<BANLkTimqq6XnKQcK3cx2aZTRN0hZ-Hx09w@mail.gmail.com>
	<BANLkTim7d7omgr0dpN0de8USvSzxO35aGQ@mail.gmail.com>
Message-ID: <BANLkTi=aEc-7FvVhH5Ajxtb=mp9YuuLK8Q@mail.gmail.com>

>
> Completely understandable my suggestions were a bit less of what you
> should do and more of what should be done for integration into trunk.
>  Your code works well, obviously accomplishes what you need, and
> doesn't break existing stuff all which are good.  The downside is
> anyone trying to understand portaudio and use portaudio now has two
> very different interfaces for doing so depending on what they want to
> do.   In addition any code that wants to take advantage of
> multi-channel portaudio has to be rewritten.


And may be that's the reason writing a mod_portaudio2 would make more sense
where we would not be worried about breaking current code.

There was a guy back in January that mentioned having something ready and
about to release, but I could not find anything in the tree.

-
Moy
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110527/dc2d0524/attachment.html 

From williams.owen at gmail.com  Fri May 27 17:57:42 2011
From: williams.owen at gmail.com (Owen Williams)
Date: Fri, 27 May 2011 14:57:42 +0100
Subject: [Freeswitch-dev] VXML parser
In-Reply-To: <BANLkTimo17oWXftLamK6RuBD4TqOwzh-Kw@mail.gmail.com>
References: <BANLkTikGk5TR_TAW8UiF9WRb+f5-ua73+A@mail.gmail.com>
	<BANLkTimo17oWXftLamK6RuBD4TqOwzh-Kw@mail.gmail.com>
Message-ID: <BANLkTi=7Xd+Ek0TyhXq9M9VOJE8-hAoCqg@mail.gmail.com>

There is no VoiceXML parser for freeswitch.  It is however possible to write
a remote application that controls a freeswitch IVR dialog (e.g. using the
freeswitch event socket layer).

What is it that you are trying to do? Must you use VoiceXML?

Owen

On 24 May 2011 05:28, Avni Mahajan <avni.mahajan at eng.knowlarity.com> wrote:

> Hi,
>>
>> I am looking for a vxml (voicexml) parser in python language. Need to use
>> the parsed vxml tags and interact with freeswitch to run IVR. Can anyone
>> help me with any kind of opensource vxml parsers?
>>
>> Regards,
>> Avni Mahajan
>
>
>
> _______________________________________________
> 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110527/ae827f3e/attachment.html 

From gabe at gundy.org  Fri May 27 21:23:50 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 27 May 2011 11:23:50 -0600
Subject: [Freeswitch-dev] VXML parser
In-Reply-To: <BANLkTimo17oWXftLamK6RuBD4TqOwzh-Kw@mail.gmail.com>
References: <BANLkTikGk5TR_TAW8UiF9WRb+f5-ua73+A@mail.gmail.com>
	<BANLkTimo17oWXftLamK6RuBD4TqOwzh-Kw@mail.gmail.com>
Message-ID: <BANLkTi=HWDU9GpzZZH7uvm+w34=4u9Ew7A@mail.gmail.com>

On Mon, May 23, 2011 at 10:28 PM, Avni Mahajan
<avni.mahajan at eng.knowlarity.com> wrote:
> I am looking for a vxml (voicexml) parser in python language. Need to use
> the parsed vxml tags and interact with freeswitch to run IVR. Can anyone
> help me with any kind of opensource vxml parsers?

This inquiry would be more appropriate on the regular FreeSWITCH users
list.  This list should be used for discussion relating to FreeSWITCH
*development*.

Anyway, check these guys out...
http://www.plivo.org/

They're doing something in Python that parses the Twillio XML and
drives the call with FreeSWITCH (I think).  That might be a good
starting place.  Licensed under the MPL (again, I think).

Best,
Gabe


From jan.berger at video24.no  Sat May 28 00:30:08 2011
From: jan.berger at video24.no (Jan Berger)
Date: Fri, 27 May 2011 22:30:08 +0200
Subject: [Freeswitch-dev] VXML parser
In-Reply-To: <BANLkTimo17oWXftLamK6RuBD4TqOwzh-Kw@mail.gmail.com>
References: <BANLkTikGk5TR_TAW8UiF9WRb+f5-ua73+A@mail.gmail.com>
	<BANLkTimo17oWXftLamK6RuBD4TqOwzh-Kw@mail.gmail.com>
Message-ID: <2085A61D84CA450896495327CE33C92C@dell9400>

In C++ yes, but not in Phython :-)

 

Jan

  _____  

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Avni
Mahajan
Sent: 24. mai 2011 06:29
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] VXML parser

 

Hi,

I am looking for a vxml (voicexml) parser in python language. Need to use
the parsed vxml tags and interact with freeswitch to run IVR. Can anyone
help me with any kind of opensource vxml parsers?

Regards,
Avni Mahajan 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110527/3879bcd4/attachment.html 

From steveayre at gmail.com  Sat May 28 22:44:38 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 28 May 2011 19:44:38 +0100
Subject: [Freeswitch-dev] VXML parser
In-Reply-To: <BANLkTi=7Xd+Ek0TyhXq9M9VOJE8-hAoCqg@mail.gmail.com>
References: <BANLkTikGk5TR_TAW8UiF9WRb+f5-ua73+A@mail.gmail.com>
	<BANLkTimo17oWXftLamK6RuBD4TqOwzh-Kw@mail.gmail.com>
	<BANLkTi=7Xd+Ek0TyhXq9M9VOJE8-hAoCqg@mail.gmail.com>
Message-ID: <BANLkTi=0aqpA-q_TYq9e+68p8O_Bxen4BA@mail.gmail.com>

I believe the mod_unimrcp module can use vxml too.

-Steve


On 27 May 2011 14:57, Owen Williams <williams.owen at gmail.com> wrote:

> There is no VoiceXML parser for freeswitch.  It is however possible to
> write a remote application that controls a freeswitch IVR dialog (e.g. using
> the freeswitch event socket layer).
>
> What is it that you are trying to do? Must you use VoiceXML?
>
> Owen
>
> On 24 May 2011 05:28, Avni Mahajan <avni.mahajan at eng.knowlarity.com>wrote:
>
>> Hi,
>>>
>>> I am looking for a vxml (voicexml) parser in python language. Need to use
>>> the parsed vxml tags and interact with freeswitch to run IVR. Can anyone
>>> help me with any kind of opensource vxml parsers?
>>>
>>> Regards,
>>> Avni Mahajan
>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110528/dee12e6a/attachment-0001.html 

From barisyanar at gmail.com  Mon May 30 18:17:38 2011
From: barisyanar at gmail.com (barisyanar)
Date: Mon, 30 May 2011 17:17:38 +0300
Subject: [Freeswitch-dev] Sipx TLS connections make Freeswitch crash
In-Reply-To: <BANLkTin2B8m7X=yQU2HhfCv9QBQpzce_oQ@mail.gmail.com>
References: <BANLkTinSVup=JqM7zPoBh8tadYkzxJcTBA@mail.gmail.com>
	<BANLkTin2B8m7X=yQU2HhfCv9QBQpzce_oQ@mail.gmail.com>
Message-ID: <BANLkTiks6UGgj131t5rLjxzUn3b=Dqyakg@mail.gmail.com>

http://www.ceng.metu.edu.tr/~e1348093/paylasim/core.18007.zip is the new
core got from fs compiled after devel-bootstrap.sh.
Version is: FreeSWITCH version: 1.0.7 (git-e19096c 2011-03-25 17-14-07
-0400)
bt from new core is similar:

#0  __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131
#1  0x00007fbe0eb43913 in sofia_glue_get_url_from_contact (buf=0x0,
    to_dup=0 '\000') at sofia_glue.c:4825
#2  0x00007fbe0eb4536b in sofia_overcome_sip_uri_weakness (
    session=0x7fbdf8016628, uri=<value optimized out>,
    transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE, params=0x0)
    at sofia_glue.c:1293
#3  0x00007fbe0eb24e10 in sofia_handle_sip_i_invite (nua=0x7fbdf80100c0,
    profile=0xbdc8c0, nh=0x7fbdfc009d40, sofia_private=<value optimized
out>,
    sip=<value optimized out>, tags=<value optimized out>) at sofia.c:6732
#4  0x00007fbe0eb4002a in sofia_event_callback (event=nua_i_invite,
    status=100, phrase=0x7fbdfc00a9c0 "Trying", nua=0x7fbdf80100c0,
    profile=0xbdc8c0, nh=0x7fbdfc009d40, sofia_private=0x0,
    sip=0x7fbdfc0064d8, tags=0x7fbdfc00a9b0) at sofia.c:936
#5  0x00007fbe0ebb1de3 in nua_application_event (dummy=<value optimized
out>,
    sumsg=<value optimized out>, ee=0x7fbdfc00a988) at nua_stack.c:393
#6  0x00007fbe0ec0f804 in su_base_port_execute_msgs (queue=0x7fbdfc00d380)
    at su_base_port.c:280
#7  0x00007fbe0ec0fd51 in su_base_port_step (self=0x7fbdf80010f0, tout=0)
    at su_base_port.c:473
#8  0x00007fbe0eb39b8a in sofia_profile_thread_run (
    thread=<value optimized out>, obj=0xbdc8c0) at sofia.c:1617
#9  0x0000003efd2068e0 in start_thread (arg=0x7fbe0f96b710)

On Sun, May 22, 2011 at 12:06 PM, Anton VG <anton.vazir at gmail.com> wrote:

> for valuable backtrace, you should compile FS with no optimization
> (-O0), running devel-bootstrap.sh will do it for you.
>
> 2011/5/16 barisyanar <barisyanar at gmail.com>:
> > Hi all,
> > I am using Freeswitch 1.0.7 with sipx release-4.4.
> > Whenever I try to make an IVR call with TLS, Freeswitch crashes dumping
> the
> > attached core file. The output bt command is:
> > #0  __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131
> > #1  0x00007f9c0b4373c3 in sofia_glue_get_url_from_contact (buf=0x0,
> >     to_dup=0 '\000') at sofia_glue.c:4943
> > #2  0x00007f9c0b438e1b in sofia_overcome_sip_uri_weakness (
> >     session=0x7f9c04017108, uri=<value optimized out>,
> >     transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE, params=0x0)
> >     at sofia_glue.c:1296
> > #3  0x00007f9c0b4182b0 in sofia_handle_sip_i_invite (nua=0x7f9c04008890,
> >     profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=<value optimized
> > out>,
> >     sip=<value optimized out>, tags=<value optimized out>) at
> sofia.c:6793
> > #4  0x00007f9c0b433aca in sofia_event_callback (event=nua_i_invite,
> >     status=100, phrase=0x7f9c0000ab40 "Trying", nua=0x7f9c04008890,
> >     profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=0x0,
> >     sip=0x7f9c00006658, tags=0x7f9c0000ab30) at sofia.c:942
> > #5  0x00007f9c0b4a5f53 in nua_application_event (dummy=<value optimized
> > out>,
> >     sumsg=<value optimized out>, ee=0x7f9c0000ab08) at nua_stack.c:393
> > #6  0x00007f9c0b503974 in su_base_port_execute_msgs (queue=0x0)
> >     at su_base_port.c:280
> > #7  0x00007f9c0b503ec1 in su_base_port_step (self=0x7f9c040010f0, tout=0)
> >     at su_base_port.c:473
> > #8  0x00007f9c0b42d4ea in sofia_profile_thread_run (
> >     thread=<value optimized out>, obj=0x148dad0) at sofia.c:1623
> > #9  0x0000003efd2068e0 in start_thread (arg=0x7f9c18116710)
> > ---Type <return> to continue, or q <return> to quit---
> >     at pthread_create.c:297
> > #10 0x0000003efcee0c9d in clone ()
> > And sipx's freeswitch log prints the line:
> > 2011-05-16 15:04:51.530426 [WARNING] switch_core.c:1120 Cannot locate
> domain
> > deneme.deneme.karelarge.com,
> > where deneme.deneme.karelarge.com is my fqdn.
> > No problem occurs during TCP connection.
> > Problem is possibly caused by sipx, but i'll be glad to hear FS
> community's
> > ideas.
> > _______________________________________________
> > 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
> >
> >
>
> _______________________________________________
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110530/ecaaca34/attachment.html 

From steveayre at gmail.com  Mon May 30 18:34:30 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 30 May 2011 15:34:30 +0100
Subject: [Freeswitch-dev] Sipx TLS connections make Freeswitch crash
In-Reply-To: <BANLkTiks6UGgj131t5rLjxzUn3b=Dqyakg@mail.gmail.com>
References: <BANLkTinSVup=JqM7zPoBh8tadYkzxJcTBA@mail.gmail.com>
	<BANLkTin2B8m7X=yQU2HhfCv9QBQpzce_oQ@mail.gmail.com>
	<BANLkTiks6UGgj131t5rLjxzUn3b=Dqyakg@mail.gmail.com>
Message-ID: <BANLkTin4Y8MqJ7yB+70SqsyhxsvbWmXpcg@mail.gmail.com>

It might also be useful to see the debug logs prior to the crash, with
siptrace enabled (sofia global siptrace on).

Since this is a bug, you should submit it as a bug report at
http://jira.freeswitch.org/.

Include a description of the problem, and attach the debug log including the
siptrace and how to reproduce it.

Unfortunately the coredump you attached is not much use to anyone without
the local directory you compiled FS from. Please load the coredump in gdb
(the gnu debugger) and collect the backtrace and also attach that. There are
instructions of how to do so at
http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29and
the support-d/fscore_pb script in the git checkout will automate that
process by posting it to our pastebin.

-Steve



On 30 May 2011 15:17, barisyanar <barisyanar at gmail.com> wrote:

> http://www.ceng.metu.edu.tr/~e1348093/paylasim/core.18007.zip is the new
> core got from fs compiled after devel-bootstrap.sh.
> Version is: FreeSWITCH version: 1.0.7 (git-e19096c 2011-03-25 17-14-07
> -0400)
> bt from new core is similar:
>
> #0  __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131
> #1  0x00007fbe0eb43913 in sofia_glue_get_url_from_contact (buf=0x0,
>      to_dup=0 '\000') at sofia_glue.c:4825
> #2  0x00007fbe0eb4536b in sofia_overcome_sip_uri_weakness (
>     session=0x7fbdf8016628, uri=<value optimized out>,
>     transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE, params=0x0)
>     at sofia_glue.c:1293
> #3  0x00007fbe0eb24e10 in sofia_handle_sip_i_invite (nua=0x7fbdf80100c0,
>     profile=0xbdc8c0, nh=0x7fbdfc009d40, sofia_private=<value optimized
> out>,
>     sip=<value optimized out>, tags=<value optimized out>) at sofia.c:6732
> #4  0x00007fbe0eb4002a in sofia_event_callback (event=nua_i_invite,
>     status=100, phrase=0x7fbdfc00a9c0 "Trying", nua=0x7fbdf80100c0,
>     profile=0xbdc8c0, nh=0x7fbdfc009d40, sofia_private=0x0,
>     sip=0x7fbdfc0064d8, tags=0x7fbdfc00a9b0) at sofia.c:936
> #5  0x00007fbe0ebb1de3 in nua_application_event (dummy=<value optimized
> out>,
>     sumsg=<value optimized out>, ee=0x7fbdfc00a988) at nua_stack.c:393
> #6  0x00007fbe0ec0f804 in su_base_port_execute_msgs (queue=0x7fbdfc00d380)
>     at su_base_port.c:280
> #7  0x00007fbe0ec0fd51 in su_base_port_step (self=0x7fbdf80010f0, tout=0)
>     at su_base_port.c:473
> #8  0x00007fbe0eb39b8a in sofia_profile_thread_run (
>     thread=<value optimized out>, obj=0xbdc8c0) at sofia.c:1617
> #9  0x0000003efd2068e0 in start_thread (arg=0x7fbe0f96b710)
>
> On Sun, May 22, 2011 at 12:06 PM, Anton VG <anton.vazir at gmail.com> wrote:
>
>> for valuable backtrace, you should compile FS with no optimization
>> (-O0), running devel-bootstrap.sh will do it for you.
>>
>> 2011/5/16 barisyanar <barisyanar at gmail.com>:
>> > Hi all,
>> > I am using Freeswitch 1.0.7 with sipx release-4.4.
>> > Whenever I try to make an IVR call with TLS, Freeswitch crashes dumping
>> the
>> > attached core file. The output bt command is:
>> > #0  __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131
>> > #1  0x00007f9c0b4373c3 in sofia_glue_get_url_from_contact (buf=0x0,
>> >     to_dup=0 '\000') at sofia_glue.c:4943
>> > #2  0x00007f9c0b438e1b in sofia_overcome_sip_uri_weakness (
>> >     session=0x7f9c04017108, uri=<value optimized out>,
>> >     transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE, params=0x0)
>> >     at sofia_glue.c:1296
>> > #3  0x00007f9c0b4182b0 in sofia_handle_sip_i_invite (nua=0x7f9c04008890,
>> >     profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=<value optimized
>> > out>,
>> >     sip=<value optimized out>, tags=<value optimized out>) at
>> sofia.c:6793
>> > #4  0x00007f9c0b433aca in sofia_event_callback (event=nua_i_invite,
>> >     status=100, phrase=0x7f9c0000ab40 "Trying", nua=0x7f9c04008890,
>> >     profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=0x0,
>> >     sip=0x7f9c00006658, tags=0x7f9c0000ab30) at sofia.c:942
>> > #5  0x00007f9c0b4a5f53 in nua_application_event (dummy=<value optimized
>> > out>,
>> >     sumsg=<value optimized out>, ee=0x7f9c0000ab08) at nua_stack.c:393
>> > #6  0x00007f9c0b503974 in su_base_port_execute_msgs (queue=0x0)
>> >     at su_base_port.c:280
>> > #7  0x00007f9c0b503ec1 in su_base_port_step (self=0x7f9c040010f0,
>> tout=0)
>> >     at su_base_port.c:473
>> > #8  0x00007f9c0b42d4ea in sofia_profile_thread_run (
>> >     thread=<value optimized out>, obj=0x148dad0) at sofia.c:1623
>> > #9  0x0000003efd2068e0 in start_thread (arg=0x7f9c18116710)
>> > ---Type <return> to continue, or q <return> to quit---
>> >     at pthread_create.c:297
>> > #10 0x0000003efcee0c9d in clone ()
>> > And sipx's freeswitch log prints the line:
>> > 2011-05-16 15:04:51.530426 [WARNING] switch_core.c:1120 Cannot locate
>> domain
>> > deneme.deneme.karelarge.com,
>> > where deneme.deneme.karelarge.com is my fqdn.
>> > No problem occurs during TCP connection.
>> > Problem is possibly caused by sipx, but i'll be glad to hear FS
>> community's
>> > ideas.
>> > _______________________________________________
>> > 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
>> >
>> >
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110530/334e1a14/attachment-0001.html 

From barisyanar at gmail.com  Tue May 31 14:51:56 2011
From: barisyanar at gmail.com (barisyanar)
Date: Tue, 31 May 2011 13:51:56 +0300
Subject: [Freeswitch-dev] Sipx TLS connections make Freeswitch crash
In-Reply-To: <BANLkTin4Y8MqJ7yB+70SqsyhxsvbWmXpcg@mail.gmail.com>
References: <BANLkTinSVup=JqM7zPoBh8tadYkzxJcTBA@mail.gmail.com>
	<BANLkTin2B8m7X=yQU2HhfCv9QBQpzce_oQ@mail.gmail.com>
	<BANLkTiks6UGgj131t5rLjxzUn3b=Dqyakg@mail.gmail.com>
	<BANLkTin4Y8MqJ7yB+70SqsyhxsvbWmXpcg@mail.gmail.com>
Message-ID: <BANLkTimLyRJ=33HGttiUXsp5EtPiG3qhAA@mail.gmail.com>

Here's the pastebin link: http://pastebin.freeswitch.org/16413
After getting sofia siptrace I'll report the bug.

On Mon, May 30, 2011 at 5:34 PM, Steven Ayre <steveayre at gmail.com> wrote:

> It might also be useful to see the debug logs prior to the crash, with
> siptrace enabled (sofia global siptrace on).
>
> Since this is a bug, you should submit it as a bug report at
> http://jira.freeswitch.org/.
>
> Include a description of the problem, and attach the debug log including
> the siptrace and how to reproduce it.
>
> Unfortunately the coredump you attached is not much use to anyone without
> the local directory you compiled FS from. Please load the coredump in gdb
> (the gnu debugger) and collect the backtrace and also attach that. There are
> instructions of how to do so at
> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29and the support-d/fscore_pb script in the git checkout will automate that
> process by posting it to our pastebin.
>
> -Steve
>
>
>
>
> On 30 May 2011 15:17, barisyanar <barisyanar at gmail.com> wrote:
>
>> http://www.ceng.metu.edu.tr/~e1348093/paylasim/core.18007.zip is the new
>> core got from fs compiled after devel-bootstrap.sh.
>> Version is: FreeSWITCH version: 1.0.7 (git-e19096c 2011-03-25 17-14-07
>> -0400)
>> bt from new core is similar:
>>
>> #0  __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131
>> #1  0x00007fbe0eb43913 in sofia_glue_get_url_from_contact (buf=0x0,
>>      to_dup=0 '\000') at sofia_glue.c:4825
>> #2  0x00007fbe0eb4536b in sofia_overcome_sip_uri_weakness (
>>     session=0x7fbdf8016628, uri=<value optimized out>,
>>     transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE, params=0x0)
>>     at sofia_glue.c:1293
>> #3  0x00007fbe0eb24e10 in sofia_handle_sip_i_invite (nua=0x7fbdf80100c0,
>>     profile=0xbdc8c0, nh=0x7fbdfc009d40, sofia_private=<value optimized
>> out>,
>>     sip=<value optimized out>, tags=<value optimized out>) at sofia.c:6732
>> #4  0x00007fbe0eb4002a in sofia_event_callback (event=nua_i_invite,
>>     status=100, phrase=0x7fbdfc00a9c0 "Trying", nua=0x7fbdf80100c0,
>>     profile=0xbdc8c0, nh=0x7fbdfc009d40, sofia_private=0x0,
>>     sip=0x7fbdfc0064d8, tags=0x7fbdfc00a9b0) at sofia.c:936
>> #5  0x00007fbe0ebb1de3 in nua_application_event (dummy=<value optimized
>> out>,
>>     sumsg=<value optimized out>, ee=0x7fbdfc00a988) at nua_stack.c:393
>> #6  0x00007fbe0ec0f804 in su_base_port_execute_msgs (queue=0x7fbdfc00d380)
>>     at su_base_port.c:280
>> #7  0x00007fbe0ec0fd51 in su_base_port_step (self=0x7fbdf80010f0, tout=0)
>>     at su_base_port.c:473
>> #8  0x00007fbe0eb39b8a in sofia_profile_thread_run (
>>     thread=<value optimized out>, obj=0xbdc8c0) at sofia.c:1617
>> #9  0x0000003efd2068e0 in start_thread (arg=0x7fbe0f96b710)
>>
>> On Sun, May 22, 2011 at 12:06 PM, Anton VG <anton.vazir at gmail.com> wrote:
>>
>>> for valuable backtrace, you should compile FS with no optimization
>>> (-O0), running devel-bootstrap.sh will do it for you.
>>>
>>> 2011/5/16 barisyanar <barisyanar at gmail.com>:
>>> > Hi all,
>>> > I am using Freeswitch 1.0.7 with sipx release-4.4.
>>> > Whenever I try to make an IVR call with TLS, Freeswitch crashes dumping
>>> the
>>> > attached core file. The output bt command is:
>>> > #0  __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131
>>> > #1  0x00007f9c0b4373c3 in sofia_glue_get_url_from_contact (buf=0x0,
>>> >     to_dup=0 '\000') at sofia_glue.c:4943
>>> > #2  0x00007f9c0b438e1b in sofia_overcome_sip_uri_weakness (
>>> >     session=0x7f9c04017108, uri=<value optimized out>,
>>> >     transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE,
>>> params=0x0)
>>> >     at sofia_glue.c:1296
>>> > #3  0x00007f9c0b4182b0 in sofia_handle_sip_i_invite
>>> (nua=0x7f9c04008890,
>>> >     profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=<value
>>> optimized
>>> > out>,
>>> >     sip=<value optimized out>, tags=<value optimized out>) at
>>> sofia.c:6793
>>> > #4  0x00007f9c0b433aca in sofia_event_callback (event=nua_i_invite,
>>> >     status=100, phrase=0x7f9c0000ab40 "Trying", nua=0x7f9c04008890,
>>> >     profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=0x0,
>>> >     sip=0x7f9c00006658, tags=0x7f9c0000ab30) at sofia.c:942
>>> > #5  0x00007f9c0b4a5f53 in nua_application_event (dummy=<value optimized
>>> > out>,
>>> >     sumsg=<value optimized out>, ee=0x7f9c0000ab08) at nua_stack.c:393
>>> > #6  0x00007f9c0b503974 in su_base_port_execute_msgs (queue=0x0)
>>> >     at su_base_port.c:280
>>> > #7  0x00007f9c0b503ec1 in su_base_port_step (self=0x7f9c040010f0,
>>> tout=0)
>>> >     at su_base_port.c:473
>>> > #8  0x00007f9c0b42d4ea in sofia_profile_thread_run (
>>> >     thread=<value optimized out>, obj=0x148dad0) at sofia.c:1623
>>> > #9  0x0000003efd2068e0 in start_thread (arg=0x7f9c18116710)
>>> > ---Type <return> to continue, or q <return> to quit---
>>> >     at pthread_create.c:297
>>> > #10 0x0000003efcee0c9d in clone ()
>>> > And sipx's freeswitch log prints the line:
>>> > 2011-05-16 15:04:51.530426 [WARNING] switch_core.c:1120 Cannot locate
>>> domain
>>> > deneme.deneme.karelarge.com,
>>> > where deneme.deneme.karelarge.com is my fqdn.
>>> > No problem occurs during TCP connection.
>>> > Problem is possibly caused by sipx, but i'll be glad to hear FS
>>> community's
>>> > ideas.
>>> > _______________________________________________
>>> > 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
>>> >
>>> >
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110531/24302b0d/attachment.html 

From steveayre at gmail.com  Tue May 31 15:42:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 31 May 2011 12:42:10 +0100
Subject: [Freeswitch-dev] Sipx TLS connections make Freeswitch crash
In-Reply-To: <BANLkTimLyRJ=33HGttiUXsp5EtPiG3qhAA@mail.gmail.com>
References: <BANLkTinSVup=JqM7zPoBh8tadYkzxJcTBA@mail.gmail.com>
	<BANLkTin2B8m7X=yQU2HhfCv9QBQpzce_oQ@mail.gmail.com>
	<BANLkTiks6UGgj131t5rLjxzUn3b=Dqyakg@mail.gmail.com>
	<BANLkTin4Y8MqJ7yB+70SqsyhxsvbWmXpcg@mail.gmail.com>
	<BANLkTimLyRJ=33HGttiUXsp5EtPiG3qhAA@mail.gmail.com>
Message-ID: <BANLkTinb97C9mQh0aw4PyP1rCf3iby6gtg@mail.gmail.com>

Unfortunately that coredump doesn't show anything useful:

#0  0x0000003efcf1fc98 in ?? ()
#1  0x00007fbe0eb43913 in ?? ()
#2  0x00007fbdf8016628 in ?? ()
#3  0x0000000000000001 in ?? ()
#4  0x0000000000000003 in ?? ()
#5  0x00007fbe0eb4536b in ?? ()
#6  0x00007fbdf801b160 in ?? ()
#7  0x0000000000bdc8c0 in ?? ()
#8  0x0000000000bdc8c0 in ?? ()
#9  0x00007fbdfc0064d8 in ?? ()
#10 0x0000000000000000 in ?? ()

Usually you'd expect to see the name of the function call instead of ??,
that shows it can't find those. Did you keep the FS git checkout after
installing it, and did you compile it with the debugging symbols?

-Steve



On 31 May 2011 11:51, barisyanar <barisyanar at gmail.com> wrote:

> Here's the pastebin link: http://pastebin.freeswitch.org/16413
> After getting sofia siptrace I'll report the bug.
>
> On Mon, May 30, 2011 at 5:34 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> It might also be useful to see the debug logs prior to the crash, with
>> siptrace enabled (sofia global siptrace on).
>>
>> Since this is a bug, you should submit it as a bug report at
>> http://jira.freeswitch.org/.
>>
>> Include a description of the problem, and attach the debug log including
>> the siptrace and how to reproduce it.
>>
>> Unfortunately the coredump you attached is not much use to anyone without
>> the local directory you compiled FS from. Please load the coredump in gdb
>> (the gnu debugger) and collect the backtrace and also attach that. There are
>> instructions of how to do so at
>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29and the support-d/fscore_pb script in the git checkout will automate that
>> process by posting it to our pastebin.
>>
>> -Steve
>>
>>
>>
>>
>> On 30 May 2011 15:17, barisyanar <barisyanar at gmail.com> wrote:
>>
>>> http://www.ceng.metu.edu.tr/~e1348093/paylasim/core.18007.zip is the new
>>> core got from fs compiled after devel-bootstrap.sh.
>>> Version is: FreeSWITCH version: 1.0.7 (git-e19096c 2011-03-25 17-14-07
>>> -0400)
>>> bt from new core is similar:
>>>
>>> #0  __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131
>>> #1  0x00007fbe0eb43913 in sofia_glue_get_url_from_contact (buf=0x0,
>>>      to_dup=0 '\000') at sofia_glue.c:4825
>>> #2  0x00007fbe0eb4536b in sofia_overcome_sip_uri_weakness (
>>>     session=0x7fbdf8016628, uri=<value optimized out>,
>>>     transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE, params=0x0)
>>>     at sofia_glue.c:1293
>>> #3  0x00007fbe0eb24e10 in sofia_handle_sip_i_invite (nua=0x7fbdf80100c0,
>>>     profile=0xbdc8c0, nh=0x7fbdfc009d40, sofia_private=<value optimized
>>> out>,
>>>     sip=<value optimized out>, tags=<value optimized out>) at
>>> sofia.c:6732
>>> #4  0x00007fbe0eb4002a in sofia_event_callback (event=nua_i_invite,
>>>     status=100, phrase=0x7fbdfc00a9c0 "Trying", nua=0x7fbdf80100c0,
>>>     profile=0xbdc8c0, nh=0x7fbdfc009d40, sofia_private=0x0,
>>>     sip=0x7fbdfc0064d8, tags=0x7fbdfc00a9b0) at sofia.c:936
>>> #5  0x00007fbe0ebb1de3 in nua_application_event (dummy=<value optimized
>>> out>,
>>>     sumsg=<value optimized out>, ee=0x7fbdfc00a988) at nua_stack.c:393
>>> #6  0x00007fbe0ec0f804 in su_base_port_execute_msgs
>>> (queue=0x7fbdfc00d380)
>>>     at su_base_port.c:280
>>> #7  0x00007fbe0ec0fd51 in su_base_port_step (self=0x7fbdf80010f0, tout=0)
>>>     at su_base_port.c:473
>>> #8  0x00007fbe0eb39b8a in sofia_profile_thread_run (
>>>     thread=<value optimized out>, obj=0xbdc8c0) at sofia.c:1617
>>> #9  0x0000003efd2068e0 in start_thread (arg=0x7fbe0f96b710)
>>>
>>> On Sun, May 22, 2011 at 12:06 PM, Anton VG <anton.vazir at gmail.com>wrote:
>>>
>>>> for valuable backtrace, you should compile FS with no optimization
>>>> (-O0), running devel-bootstrap.sh will do it for you.
>>>>
>>>> 2011/5/16 barisyanar <barisyanar at gmail.com>:
>>>> > Hi all,
>>>> > I am using Freeswitch 1.0.7 with sipx release-4.4.
>>>> > Whenever I try to make an IVR call with TLS, Freeswitch crashes
>>>> dumping the
>>>> > attached core file. The output bt command is:
>>>> > #0  __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131
>>>> > #1  0x00007f9c0b4373c3 in sofia_glue_get_url_from_contact (buf=0x0,
>>>> >     to_dup=0 '\000') at sofia_glue.c:4943
>>>> > #2  0x00007f9c0b438e1b in sofia_overcome_sip_uri_weakness (
>>>> >     session=0x7f9c04017108, uri=<value optimized out>,
>>>> >     transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE,
>>>> params=0x0)
>>>> >     at sofia_glue.c:1296
>>>> > #3  0x00007f9c0b4182b0 in sofia_handle_sip_i_invite
>>>> (nua=0x7f9c04008890,
>>>> >     profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=<value
>>>> optimized
>>>> > out>,
>>>> >     sip=<value optimized out>, tags=<value optimized out>) at
>>>> sofia.c:6793
>>>> > #4  0x00007f9c0b433aca in sofia_event_callback (event=nua_i_invite,
>>>> >     status=100, phrase=0x7f9c0000ab40 "Trying", nua=0x7f9c04008890,
>>>> >     profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=0x0,
>>>> >     sip=0x7f9c00006658, tags=0x7f9c0000ab30) at sofia.c:942
>>>> > #5  0x00007f9c0b4a5f53 in nua_application_event (dummy=<value
>>>> optimized
>>>> > out>,
>>>> >     sumsg=<value optimized out>, ee=0x7f9c0000ab08) at nua_stack.c:393
>>>> > #6  0x00007f9c0b503974 in su_base_port_execute_msgs (queue=0x0)
>>>> >     at su_base_port.c:280
>>>> > #7  0x00007f9c0b503ec1 in su_base_port_step (self=0x7f9c040010f0,
>>>> tout=0)
>>>> >     at su_base_port.c:473
>>>> > #8  0x00007f9c0b42d4ea in sofia_profile_thread_run (
>>>> >     thread=<value optimized out>, obj=0x148dad0) at sofia.c:1623
>>>> > #9  0x0000003efd2068e0 in start_thread (arg=0x7f9c18116710)
>>>> > ---Type <return> to continue, or q <return> to quit---
>>>> >     at pthread_create.c:297
>>>> > #10 0x0000003efcee0c9d in clone ()
>>>> > And sipx's freeswitch log prints the line:
>>>> > 2011-05-16 15:04:51.530426 [WARNING] switch_core.c:1120 Cannot locate
>>>> domain
>>>> > deneme.deneme.karelarge.com,
>>>> > where deneme.deneme.karelarge.com is my fqdn.
>>>> > No problem occurs during TCP connection.
>>>> > Problem is possibly caused by sipx, but i'll be glad to hear FS
>>>> community's
>>>> > ideas.
>>>> > _______________________________________________
>>>> > 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
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110531/a00c2b38/attachment-0001.html 

From barisyanar at gmail.com  Tue May 31 16:09:48 2011
From: barisyanar at gmail.com (barisyanar)
Date: Tue, 31 May 2011 15:09:48 +0300
Subject: [Freeswitch-dev] Sipx TLS connections make Freeswitch crash
In-Reply-To: <BANLkTinb97C9mQh0aw4PyP1rCf3iby6gtg@mail.gmail.com>
References: <BANLkTinSVup=JqM7zPoBh8tadYkzxJcTBA@mail.gmail.com>
	<BANLkTin2B8m7X=yQU2HhfCv9QBQpzce_oQ@mail.gmail.com>
	<BANLkTiks6UGgj131t5rLjxzUn3b=Dqyakg@mail.gmail.com>
	<BANLkTin4Y8MqJ7yB+70SqsyhxsvbWmXpcg@mail.gmail.com>
	<BANLkTimLyRJ=33HGttiUXsp5EtPiG3qhAA@mail.gmail.com>
	<BANLkTinb97C9mQh0aw4PyP1rCf3iby6gtg@mail.gmail.com>
Message-ID: <BANLkTinL+Xtg+TknK+ygMObGpF5P5KORGQ@mail.gmail.com>

sorry, i ran script at wrong place. here is the new link:
http://pastebin.freeswitch.org/16418

and the siptrace is here - after printing fs_cli closes:

http://pastebin.com/dCCEvUEL

On Tue, May 31, 2011 at 2:42 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Unfortunately that coredump doesn't show anything useful:
>
> #0  0x0000003efcf1fc98 in ?? ()
> #1  0x00007fbe0eb43913 in ?? ()
> #2  0x00007fbdf8016628 in ?? ()
> #3  0x0000000000000001 in ?? ()
> #4  0x0000000000000003 in ?? ()
> #5  0x00007fbe0eb4536b in ?? ()
> #6  0x00007fbdf801b160 in ?? ()
> #7  0x0000000000bdc8c0 in ?? ()
> #8  0x0000000000bdc8c0 in ?? ()
> #9  0x00007fbdfc0064d8 in ?? ()
> #10 0x0000000000000000 in ?? ()
>
> Usually you'd expect to see the name of the function call instead of ??,
> that shows it can't find those. Did you keep the FS git checkout after
> installing it, and did you compile it with the debugging symbols?
>
> -Steve
>
>
>
>
> On 31 May 2011 11:51, barisyanar <barisyanar at gmail.com> wrote:
>
>> Here's the pastebin link: http://pastebin.freeswitch.org/16413
>> After getting sofia siptrace I'll report the bug.
>>
>> On Mon, May 30, 2011 at 5:34 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> It might also be useful to see the debug logs prior to the crash, with
>>> siptrace enabled (sofia global siptrace on).
>>>
>>> Since this is a bug, you should submit it as a bug report at
>>> http://jira.freeswitch.org/.
>>>
>>> Include a description of the problem, and attach the debug log including
>>> the siptrace and how to reproduce it.
>>>
>>> Unfortunately the coredump you attached is not much use to anyone without
>>> the local directory you compiled FS from. Please load the coredump in gdb
>>> (the gnu debugger) and collect the backtrace and also attach that. There are
>>> instructions of how to do so at
>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29and the support-d/fscore_pb script in the git checkout will automate that
>>> process by posting it to our pastebin.
>>>
>>> -Steve
>>>
>>>
>>>
>>>
>>> On 30 May 2011 15:17, barisyanar <barisyanar at gmail.com> wrote:
>>>
>>>> http://www.ceng.metu.edu.tr/~e1348093/paylasim/core.18007.zip is the
>>>> new core got from fs compiled after devel-bootstrap.sh.
>>>> Version is: FreeSWITCH version: 1.0.7 (git-e19096c 2011-03-25 17-14-07
>>>> -0400)
>>>> bt from new core is similar:
>>>>
>>>> #0  __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131
>>>> #1  0x00007fbe0eb43913 in sofia_glue_get_url_from_contact (buf=0x0,
>>>>      to_dup=0 '\000') at sofia_glue.c:4825
>>>> #2  0x00007fbe0eb4536b in sofia_overcome_sip_uri_weakness (
>>>>     session=0x7fbdf8016628, uri=<value optimized out>,
>>>>     transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE, params=0x0)
>>>>     at sofia_glue.c:1293
>>>> #3  0x00007fbe0eb24e10 in sofia_handle_sip_i_invite
>>>> (nua=0x7fbdf80100c0,
>>>>     profile=0xbdc8c0, nh=0x7fbdfc009d40, sofia_private=<value optimized
>>>> out>,
>>>>     sip=<value optimized out>, tags=<value optimized out>) at
>>>> sofia.c:6732
>>>> #4  0x00007fbe0eb4002a in sofia_event_callback (event=nua_i_invite,
>>>>     status=100, phrase=0x7fbdfc00a9c0 "Trying", nua=0x7fbdf80100c0,
>>>>     profile=0xbdc8c0, nh=0x7fbdfc009d40, sofia_private=0x0,
>>>>     sip=0x7fbdfc0064d8, tags=0x7fbdfc00a9b0) at sofia.c:936
>>>> #5  0x00007fbe0ebb1de3 in nua_application_event (dummy=<value optimized
>>>> out>,
>>>>     sumsg=<value optimized out>, ee=0x7fbdfc00a988) at nua_stack.c:393
>>>> #6  0x00007fbe0ec0f804 in su_base_port_execute_msgs
>>>> (queue=0x7fbdfc00d380)
>>>>     at su_base_port.c:280
>>>> #7  0x00007fbe0ec0fd51 in su_base_port_step (self=0x7fbdf80010f0,
>>>> tout=0)
>>>>     at su_base_port.c:473
>>>> #8  0x00007fbe0eb39b8a in sofia_profile_thread_run (
>>>>     thread=<value optimized out>, obj=0xbdc8c0) at sofia.c:1617
>>>> #9  0x0000003efd2068e0 in start_thread (arg=0x7fbe0f96b710)
>>>>
>>>> On Sun, May 22, 2011 at 12:06 PM, Anton VG <anton.vazir at gmail.com>wrote:
>>>>
>>>>> for valuable backtrace, you should compile FS with no optimization
>>>>> (-O0), running devel-bootstrap.sh will do it for you.
>>>>>
>>>>> 2011/5/16 barisyanar <barisyanar at gmail.com>:
>>>>> > Hi all,
>>>>> > I am using Freeswitch 1.0.7 with sipx release-4.4.
>>>>> > Whenever I try to make an IVR call with TLS, Freeswitch crashes
>>>>> dumping the
>>>>> > attached core file. The output bt command is:
>>>>> > #0  __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131
>>>>> > #1  0x00007f9c0b4373c3 in sofia_glue_get_url_from_contact (buf=0x0,
>>>>> >     to_dup=0 '\000') at sofia_glue.c:4943
>>>>> > #2  0x00007f9c0b438e1b in sofia_overcome_sip_uri_weakness (
>>>>> >     session=0x7f9c04017108, uri=<value optimized out>,
>>>>> >     transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE,
>>>>> params=0x0)
>>>>> >     at sofia_glue.c:1296
>>>>> > #3  0x00007f9c0b4182b0 in sofia_handle_sip_i_invite
>>>>> (nua=0x7f9c04008890,
>>>>> >     profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=<value
>>>>> optimized
>>>>> > out>,
>>>>> >     sip=<value optimized out>, tags=<value optimized out>) at
>>>>> sofia.c:6793
>>>>> > #4  0x00007f9c0b433aca in sofia_event_callback (event=nua_i_invite,
>>>>> >     status=100, phrase=0x7f9c0000ab40 "Trying", nua=0x7f9c04008890,
>>>>> >     profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=0x0,
>>>>> >     sip=0x7f9c00006658, tags=0x7f9c0000ab30) at sofia.c:942
>>>>> > #5  0x00007f9c0b4a5f53 in nua_application_event (dummy=<value
>>>>> optimized
>>>>> > out>,
>>>>> >     sumsg=<value optimized out>, ee=0x7f9c0000ab08) at
>>>>> nua_stack.c:393
>>>>> > #6  0x00007f9c0b503974 in su_base_port_execute_msgs (queue=0x0)
>>>>> >     at su_base_port.c:280
>>>>> > #7  0x00007f9c0b503ec1 in su_base_port_step (self=0x7f9c040010f0,
>>>>> tout=0)
>>>>> >     at su_base_port.c:473
>>>>> > #8  0x00007f9c0b42d4ea in sofia_profile_thread_run (
>>>>> >     thread=<value optimized out>, obj=0x148dad0) at sofia.c:1623
>>>>> > #9  0x0000003efd2068e0 in start_thread (arg=0x7f9c18116710)
>>>>> > ---Type <return> to continue, or q <return> to quit---
>>>>> >     at pthread_create.c:297
>>>>> > #10 0x0000003efcee0c9d in clone ()
>>>>> > And sipx's freeswitch log prints the line:
>>>>> > 2011-05-16 15:04:51.530426 [WARNING] switch_core.c:1120 Cannot locate
>>>>> domain
>>>>> > deneme.deneme.karelarge.com,
>>>>> > where deneme.deneme.karelarge.com is my fqdn.
>>>>> > No problem occurs during TCP connection.
>>>>> > Problem is possibly caused by sipx, but i'll be glad to hear FS
>>>>> community's
>>>>> > ideas.
>>>>> > _______________________________________________
>>>>> > 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
>>>>> >
>>>>> >
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110531/def62d2a/attachment.html 

From peter.olsson at visionutveckling.se  Tue May 31 18:10:28 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 31 May 2011 16:10:28 +0200
Subject: [Freeswitch-dev] Sipx TLS connections make Freeswitch crash
In-Reply-To: <BANLkTinL+Xtg+TknK+ygMObGpF5P5KORGQ@mail.gmail.com>
References: <BANLkTinSVup=JqM7zPoBh8tadYkzxJcTBA@mail.gmail.com>
	<BANLkTin2B8m7X=yQU2HhfCv9QBQpzce_oQ@mail.gmail.com>
	<BANLkTiks6UGgj131t5rLjxzUn3b=Dqyakg@mail.gmail.com>
	<BANLkTin4Y8MqJ7yB+70SqsyhxsvbWmXpcg@mail.gmail.com>
	<BANLkTimLyRJ=33HGttiUXsp5EtPiG3qhAA@mail.gmail.com>
	<BANLkTinb97C9mQh0aw4PyP1rCf3iby6gtg@mail.gmail.com>
	<BANLkTinL+Xtg+TknK+ygMObGpF5P5KORGQ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59DE54929C@cooper>

That's not latest git head. Try latest, if it still occurs, report a bug in Jira.

/Peter


Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r barisyanar
Skickat: den 31 maj 2011 14:10
Till: freeswitch-dev at lists.freeswitch.org
?mne: Re: [Freeswitch-dev] Sipx TLS connections make Freeswitch crash

sorry, i ran script at wrong place. here is the new link:
http://pastebin.freeswitch.org/16418

and the siptrace is here - after printing fs_cli closes:

http://pastebin.com/dCCEvUEL
On Tue, May 31, 2011 at 2:42 PM, Steven Ayre <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
Unfortunately that coredump doesn't show anything useful:
#0  0x0000003efcf1fc98 in ?? ()
#1  0x00007fbe0eb43913 in ?? ()
#2  0x00007fbdf8016628 in ?? ()
#3  0x0000000000000001 in ?? ()
#4  0x0000000000000003 in ?? ()
#5  0x00007fbe0eb4536b in ?? ()
#6  0x00007fbdf801b160 in ?? ()
#7  0x0000000000bdc8c0 in ?? ()
#8  0x0000000000bdc8c0 in ?? ()
#9  0x00007fbdfc0064d8 in ?? ()
#10 0x0000000000000000 in ?? ()

Usually you'd expect to see the name of the function call instead of ??, that shows it can't find those. Did you keep the FS git checkout after installing it, and did you compile it with the debugging symbols?

-Steve




On 31 May 2011 11:51, barisyanar <barisyanar at gmail.com<mailto:barisyanar at gmail.com>> wrote:
Here's the pastebin link: http://pastebin.freeswitch.org/16413
After getting sofia siptrace I'll report the bug.

On Mon, May 30, 2011 at 5:34 PM, Steven Ayre <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
It might also be useful to see the debug logs prior to the crash, with siptrace enabled (sofia global siptrace on).

Since this is a bug, you should submit it as a bug report at http://jira.freeswitch.org/.

Include a description of the problem, and attach the debug log including the siptrace and how to reproduce it.

Unfortunately the coredump you attached is not much use to anyone without the local directory you compiled FS from. Please load the coredump in gdb (the gnu debugger) and collect the backtrace and also attach that. There are instructions of how to do so at http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 and the support-d/fscore_pb script in the git checkout will automate that process by posting it to our pastebin.

-Steve



On 30 May 2011 15:17, barisyanar <barisyanar at gmail.com<mailto:barisyanar at gmail.com>> wrote:
http://www.ceng.metu.edu.tr/~e1348093/paylasim/core.18007.zip<http://www.ceng.metu.edu.tr/%7Ee1348093/paylasim/core.18007.zip> is the new core got from fs compiled after devel-bootstrap.sh.
Version is: FreeSWITCH version: 1.0.7 (git-e19096c 2011-03-25 17-14-07 -0400)
bt from new core is similar:

#0  __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131
#1  0x00007fbe0eb43913 in sofia_glue_get_url_from_contact (buf=0x0,
    to_dup=0 '\000') at sofia_glue.c:4825
#2  0x00007fbe0eb4536b in sofia_overcome_sip_uri_weakness (
    session=0x7fbdf8016628, uri=<value optimized out>,
    transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE, params=0x0)
    at sofia_glue.c:1293
#3  0x00007fbe0eb24e10 in sofia_handle_sip_i_invite (nua=0x7fbdf80100c0,
    profile=0xbdc8c0, nh=0x7fbdfc009d40, sofia_private=<value optimized out>,
    sip=<value optimized out>, tags=<value optimized out>) at sofia.c:6732
#4  0x00007fbe0eb4002a in sofia_event_callback (event=nua_i_invite,
    status=100, phrase=0x7fbdfc00a9c0 "Trying", nua=0x7fbdf80100c0,
    profile=0xbdc8c0, nh=0x7fbdfc009d40, sofia_private=0x0,
    sip=0x7fbdfc0064d8, tags=0x7fbdfc00a9b0) at sofia.c:936
#5  0x00007fbe0ebb1de3 in nua_application_event (dummy=<value optimized out>,
    sumsg=<value optimized out>, ee=0x7fbdfc00a988) at nua_stack.c:393
#6  0x00007fbe0ec0f804 in su_base_port_execute_msgs (queue=0x7fbdfc00d380)
    at su_base_port.c:280
#7  0x00007fbe0ec0fd51 in su_base_port_step (self=0x7fbdf80010f0, tout=0)
    at su_base_port.c:473
#8  0x00007fbe0eb39b8a in sofia_profile_thread_run (
    thread=<value optimized out>, obj=0xbdc8c0) at sofia.c:1617
#9  0x0000003efd2068e0 in start_thread (arg=0x7fbe0f96b710)

On Sun, May 22, 2011 at 12:06 PM, Anton VG <anton.vazir at gmail.com<mailto:anton.vazir at gmail.com>> wrote:
for valuable backtrace, you should compile FS with no optimization
(-O0), running devel-bootstrap.sh will do it for you.

2011/5/16 barisyanar <barisyanar at gmail.com<mailto:barisyanar at gmail.com>>:
> Hi all,
> I am using Freeswitch 1.0.7 with sipx release-4.4.
> Whenever I try to make an IVR call with TLS, Freeswitch crashes dumping the
> attached core file. The output bt command is:
> #0  __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131
> #1  0x00007f9c0b4373c3 in sofia_glue_get_url_from_contact (buf=0x0,
>     to_dup=0 '\000') at sofia_glue.c:4943
> #2  0x00007f9c0b438e1b in sofia_overcome_sip_uri_weakness (
>     session=0x7f9c04017108, uri=<value optimized out>,
>     transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE, params=0x0)
>     at sofia_glue.c:1296
> #3  0x00007f9c0b4182b0 in sofia_handle_sip_i_invite (nua=0x7f9c04008890,
>     profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=<value optimized
> out>,
>     sip=<value optimized out>, tags=<value optimized out>) at sofia.c:6793
> #4  0x00007f9c0b433aca in sofia_event_callback (event=nua_i_invite,
>     status=100, phrase=0x7f9c0000ab40 "Trying", nua=0x7f9c04008890,
>     profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=0x0,
>     sip=0x7f9c00006658, tags=0x7f9c0000ab30) at sofia.c:942
> #5  0x00007f9c0b4a5f53 in nua_application_event (dummy=<value optimized
> out>,
>     sumsg=<value optimized out>, ee=0x7f9c0000ab08) at nua_stack.c:393
> #6  0x00007f9c0b503974 in su_base_port_execute_msgs (queue=0x0)
>     at su_base_port.c:280
> #7  0x00007f9c0b503ec1 in su_base_port_step (self=0x7f9c040010f0, tout=0)
>     at su_base_port.c:473
> #8  0x00007f9c0b42d4ea in sofia_profile_thread_run (
>     thread=<value optimized out>, obj=0x148dad0) at sofia.c:1623
> #9  0x0000003efd2068e0 in start_thread (arg=0x7f9c18116710)
> ---Type <return> to continue, or q <return> to quit---
>     at pthread_create.c:297
> #10 0x0000003efcee0c9d in clone ()
> And sipx's freeswitch log prints the line:
> 2011-05-16 15:04:51.530426 [WARNING] switch_core.c:1120 Cannot locate domain
> deneme.deneme.karelarge.com<http://deneme.deneme.karelarge.com>,
> where deneme.deneme.karelarge.com<http://deneme.deneme.karelarge.com> is my fqdn.
> No problem occurs during TCP connection.
> Problem is possibly caused by sipx, but i'll be glad to hear FS community's
> ideas.
> _______________________________________________
> 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
>
>

_______________________________________________
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


_______________________________________________
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


_______________________________________________
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


_______________________________________________
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


_______________________________________________
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:4de4db0432761032611453!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110531/d2c6cb34/attachment-0001.html 

From msc at freeswitch.org  Tue May 31 21:35:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 31 May 2011 10:35:54 -0700
Subject: [Freeswitch-dev] Call For Help: FreeSWITCH Cookbook
Message-ID: <BANLkTinuuqCh4duDNsnkt33k1aNgqmAZuw@mail.gmail.com>

Hello FreeSWITCHers!

We have a need for some assistance with a few of our recipes for the
cookbook. If you are able to write technical documentation in English and
have first-hand knowledge of these topics then you might be in a position to
help:

CDRs:
Parsing XML CDRs
Handling A and B leg CDRs
Using a web server to handle XML CDRs

Event Socket:
Inbound ESL connections (basic how-to)
Outbound ESL connections (basic how-to)
Launch an outbound call with inbound event socket & ESL
Handle inbound call with socket app & ESL
Reacting to events (events for "my call" vs. system-wide events, etc.)

Misc:
Presence for BLF and SLA
Presence for FIFO agent status

If you think you've got the chops to write one or more recipes for the
cookbook then email me off list and we'll discuss the specifics. I will get
you all the information you need to get started.

Thanks!

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110531/7f3e6c9f/attachment.html 

From acichocki at supermedia.pl  Fri May 27 12:12:32 2011
From: acichocki at supermedia.pl (Artur Cichocki)
Date: Fri, 27 May 2011 08:12:32 -0000
Subject: [Freeswitch-dev] Removing SDP from 183/180
Message-ID: <4DDF5CEE.5060505@supermedia.pl>

Hi All.

I just wrote this patch. It ads 'sip_remove_183sdp' flag, which removes 
SDP from 183 and returns 180 (it also removes SDP from 180, because FS
changes 180 with SDP to 183).

To turn it on add {sip_remove_183sdp=true} to the dial string.

Why? Because in some situations we don't want to transfer voice between 
clients before a real call establishment (and yes, I know that early 
media in this stage is allowed by RFC).

Real case:
Try to setup two clients, one of them using Sipdroid. Turn on bypass 
mode. Try to call to Sipdroid client, you will hear caller voice 
together with the ring (because Sipdroid sends 180 with SDP, and puts 
voice on the speaker).

Patch attached.

-- 
Artur Cichocki
-------------- next part --------------
A non-text attachment was scrubbed...
Name: sip_remove_183sdp.patch
Type: text/x-diff
Size: 3684 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110527/7db6afe9/attachment.bin