May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Assertion when Transcoding
by Guillermo Ruiz Camauer 24 May '13

24 May '13

I am getting the following assertion when attempting to record a G.729 call: freeswitch: src/switch_channel.c:832: switch_channel_get_variable_dup: Assertion `channel != ((void *)0)' failed. The system has a Sangoma D100 card installed. The call is established using the G.729 codec, and after playing some prompts, I am attempting to record an answer: freeswitch@192.168.9.151:5000@internal> show codecs type,name,ikey codec,ADPCM (IMA),mod_spandsp codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_spandsp codec,G.726 16k,mod_spandsp codec,G.726 16k (AAL2),mod_spandsp codec,G.726 24k,mod_spandsp codec,G.726 24k (AAL2),mod_spandsp codec,G.726 32k,mod_spandsp codec,G.726 32k (AAL2),mod_spandsp codec,G.726 40k,mod_spandsp codec,G.726 40k (AAL2),mod_spandsp codec,GSM,mod_spandsp codec,LPC-10,mod_spandsp codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Sangoma G729,mod_sangoma_codec 18 total. . . . 2013-05-24 12:33:31.071446 [ALERT] switch_core_session.c:2739 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC_COMPLETE] 2013-05-24 12:33:31.071446 [ALERT] switch_ivr.c:650 sofia/crossfonetrunk/52764439 receive message [AUDIO_SYNC] 2013-05-24 12:33:31.111446 [ALERT] switch_core_session.c:2724 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC] 2013-05-24 12:33:31.111446 [INFO] mod_native_file.c:94 Opening File [/mnt/TEMP/beep-7.G729] 8000hz 2013-05-24 12:33:31.231447 [ALERT] switch_core_session.c:2739 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC_COMPLETE] 2013-05-24 12:33:31.231447 [ALERT] switch_ivr.c:650 sofia/crossfonetrunk/52764439 receive message [AUDIO_SYNC] 2013-05-24 12:33:31.271451 [ALERT] switch_core_session.c:2724 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC] 2013-05-24 12:33:31.291450 [ALERT] switch_core_io.c:446 sofia/crossfonetrunk/52764439 receive message [TRANSCODING_NECESSARY] freeswitch@192.168.9.151:5000@internal> freeswitch: src/switch_channel.c:832: switch_channel_get_variable_dup: Assertion `channel != ((void *)0)' failed. FreeSwitch dies after this. Note that this does not happen if I restrict everything to G711 (ULAW). Any thoughts? -- Guillermo Ruiz Camauer

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Freeswitch versioning / tagging /releasemanagement
by mbo 24 May '13

24 May '13

On the wiki I can see that the latest stable freeswitch release is 1.2.9. If I look into git (git tag -l -n1), I can see that there are many tags with a higher version, but older tag date. v1.3.0 1.3.0 release v1.3.1 Tagging version 1.3.1 v1.3.10 tag v1.3.10 v1.3.11 tag v1.3.11 v1.3.12 Retag v1.3.12 v1.3.13 tag v1.3.13 v1.3.14 tag v1.3.14 v1.3.15 tag v1.3.15 v1.3.16 tag v1.3.16 v1.3.17-final tag v1.3.17-final v1.3.2 Tagging version 1.3.2 v1.3.3 Tagging 1.3.3 v1.3.4 release v1.3.4 v1.3.5 release v1.3.5 v1.3.6 tag v1.3.6 v1.3.7 tag v1.3.7 v1.3.8 tag v1.3.8 v1.3.9 tag v1.3.9 v1.5.0 tag v1.5.0 What about those tags? How is the release management organized? Thanks Markus

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Re: [Freeswitch-users] mod_cdr_csv is not working!
by Shahzad Bhatti 24 May '13

24 May '13

i open a jira and here is a link of that http://jira.freeswitch.org/browse/FS-5468 Shahzad Bhatti On Wed, May 22, 2013 at 8:50 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: mod_cdr_csv is not working! (Michael Collins) > 2. Re: One Way Audio (Michael Collins) > 3. Re: spam>spam>Re: One Way Audio (Moishe Grunstein) > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 22 May 2013 08:39:26 -0700 > Subject: Re: [Freeswitch-users] mod_cdr_csv is not working! > Can you reproduce this on latest HEAD? If so, gather details and open a > Jira. > > -MC > > > On Wed, May 22, 2013 at 8:00 AM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com>wrote: > >> Hi, >> >> after loading *mod_cdr_csv, *when i originate test calls the cdr is not >> appended in Master.csv file most of the time but sometime cdr is appended >> in the Master.csv that show that mod_cdr_csv is configured. Also when i try >> to use *cdr_csv rotate* command on fs_cli it also not work. even i also >> try to do using the perl script example available on following url >> >> >> https://wiki.freeswitch.org/wiki/Mod_cdr_csv#Example_Perl_Script_for_CDR_in… >> >> i want to know why this is happening on my server and how i can fix these >> issues >> >> 1. cdrs are not updated in the csv file; >> 2. rotate command is not working; >> >> >> any reply is highly appreciated; >> >> Regards >> >> Shahzad Bhatti >> >>

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Best performing codec on mobile/slow networks
by Daniel Ivanov 24 May '13

24 May '13

I've officially given up on AMR, due to it's licensing issues and sketchy supports for my mobile clients. I am looking for new options now as CsipSimple has a new codec pack supporting G.726.1, opus and codec2. Which do you think would perform best on mobile networks? And which has FS support for transcoding as well?

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Dialplan not executing on continue_on_fail=true
by Ashish gautam 24 May '13

24 May '13

Hi, I have a dialplan as follows: <include> <extension name="public_did"> <condition field="destination_number" expression="^(47673501)$"> <action application="answer"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="api_hangup_hook=perl hook.pl"/> <action application="set" data="session_in_hangup_hook=true"/> <action application="perl" data="perl/ash.pl"/> </condition> </extension> </include> when the called party does not pick up the phone or is busy, the dialplan does not proceed and hook.pl does not get executed. Please help -- Ashish Gautam IVR Developer Nucleus Microsystems (Pvt.) Ltd.

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Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg
by Adam Raszynski 24 May '13

24 May '13

Hi, I have very strange problem with audio after upgrading of FS to the newest GIT version. Previous version was quite old, but worked without problems for a long time: FreeSWITCH version: 1.0.head (git-313b164 2011-11-26 08-53-01 -0600) Current version is: FreeSWITCH version: 1.5.1b () All configuration files are the same (exact copy) System: Debian Lenny 32-bit All calls are made over SIP Symptoms: - IVR prompt playback is OK - FAX sending/receiving is OK - Calling to remote servers via VoIP provider is OK - The only problem is calling to local user and it's very strange. When external user calls local user (SIP call, via IVR, then bridge), audio stream from external user is OK, but audio stream from local user sent to external user is very delayed. Delay increases during a call. At the beginning is about 1 second, all the time delay is increasing. Audio is reaching external user but after about 30 seconds of call stream is delayed for about 10 seconds. At the same time audio stream from external user is heard by local user with no delays. Propably something changed in FS core or config files and I need to adjust to the new version. As I stated before configuration is exact copy and worked perfectly on older version. Any hints what should I check or change to solve this problem? Kind Regards

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problem bridging lots of dtmf in a session
by Michel Brabants 24 May '13

24 May '13

Hello, we have a problem with dtmf not being bridged by freeswitch when there is lots of dtmf in the call. Freeswitch resides between an asterisk gateway (which sends lot of duplicated dtmf) and an ivr. When there is a lot of dtmf being bridged in a single call, we can see in a wireshark-trace that it stops bridging the dtmf at a certain moment. I can still see the dtmf being received in the freeswitch-logs, bit I'm not seeing it in the wireshark-trace towards the ivr. We're using freeswitch 1.2.7. Thanks, Michel

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Google Talk XMPP Support
by Andrew Cassidy 24 May '13

24 May '13

I've heard rumors that XMPP support in Google Talk is being killed. I'm guessing this means no more Google Talk in FreeSWITCH? -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F<info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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how to turn an ongoing call in to a conference
by Vincent Xia 24 May '13

24 May '13

Dear Support, my question for using freeswitch is: how to turn an ongoing call in to a conference so i can have the third party in without breaking up the call (no hangup, dial stuff)? something like call intrusion，A and B are already talking on the phone, then C wants to talk to both A and B. any response will be greatly appriciated. Jacob

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why it hangs up after originate
by Vincent Xia 24 May '13

24 May '13

when im executing console commands like "originate user/1005 @eavesdrop(uuid)" to have 1005 listen to a call party, 1005 rings then i click answer and find out it hangs up immediately, the freeswitch log says "[NOTICE] switch_core_state_machine.c:262 sofia/internal/sip:1005@IPaddr:5088 has executed the last dialplan instruction, hanging up." does that mean i need to add something the to the dialplan? is there a simple solution by extending the originate command like "originate user/1005 @eavesdrop(uuid) &wait_for_hangup"?

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