August 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Please Help
by Chris Williams 21 Aug '08

21 Aug '08

How do I get off this mailing list? I am not a programmer and have no idea why I'm signed up for this mailing list. If someone can help, please email me directly and let me know what to do. Thank you. chris(a)guidepointsystems.com

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Re: [Freeswitch-users] mod_perl buid errors on Solaris
by Andy Spitzer 21 Aug '08

21 Aug '08

Woof! On Mon, 18 Aug 2008 14:25:45 -0400, Bruce McAlister <bruce.mcalister(a)blueface.ie> wrote: > Does anyone have any idea's on the error listed below? > /opt/SUNWspro/bin/cc -w -DMULTIPLICITY -D_LARGEFILE_SOURCE > -D_FILE_OFFSET_BITS=64 -D_TS_ERRNO > -I/usr/perl5/5.8.4/lib/i86pc-solaris-64int/CORE -DEMBED_PERL > -I/export/home/vzhsxn/packages/BUILD/freeswitch-1.0.1/src/include > -I/export/home/vzhsxn/packages/BUILD/freeswitch-1.0.1/libs/libteletone/src > -KPIC -DPIC -DPATH_MAX=2048 -g -v -Xc -xc99=all -D_GNU_SOURCE > -DHAVE_CONFIG_H -c -o perlxsi.o perlxsi.c > > "/usr/perl5/5.8.4/lib/i86pc-solaris-64int/CORE/scope.h", line 221: syntax > error before or at: sigjmp_buf Just a guess (from previous solaris experience), that somewhere in the perlxsi.c file, there is an include that is re-defining a symbol used in scope.h. I don't have access to scope.h here to see what it is complaining about, but that is where I'd look. Another thing to try would be use gcc instead of Sun's compiler...see if that makes a difference or changes the error message to something a bit more useful. --Woof!

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recording module
by Ilan Perez 21 Aug '08

21 Aug '08

Where can I find the code in FS tha deals with the actual recording. Not the xml code to record a session but the pro-compiled record function? Ilan Perez

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Re: [Freeswitch-users] Voice not work in bgapi
by msp 21 Aug '08

21 Aug '08

Hi all, It was the problem of softphone. FS is ok. My version of Zoiper was having problem with sound/voice while using bgapi. I changed it and its working fine. Thsanks msp

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Problem in geting call duration
by Anand Kumar 21 Aug '08

21 Aug '08

Hi All, Please help me, I am trying to find out call answer time duration. i have tested in 2 way.. 1. Make one call using FS and call is terminate on pstn and i hangup from the softphone that time i got answer time using .. var callduration=session.getVariable("duration"); console_log() Your call Answer time :4 2. when i disconnect the call from pstn site that time i am not get answer time duration... Your call Answer time :false same code is return ""*false*" why ? please help me.. Thanks Anand Dayal

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Fwd: FreeSwitch Branch
by Gayatri Kulkarni 21 Aug '08

21 Aug '08

Where exactly do I login to execute svn checkout ?? ---------- Forwarded message ---------- From: Gayatri Kulkarni <xtpl.gayatri(a)gmail.com> Date: Wed, Aug 20, 2008 at 1:29 AM Subject: Re: FreeSwitch Branch To: "freeswitch. org" <dev(a)freeswitch.org> On Wed, Aug 20, 2008 at 12:46 AM, freeswitch.org <dev(a)freeswitch.org> wrote: > Congratulations, your FreeSwitch branch has been created and you now have > read and write access to it. It is accessible using SVN via: > > http://svn.freeswitch.org/svn/freeswitch/branches/gakul > > Using your regular username and password. > > Username: gakul > Password: kulkarn1 > > Using your SVN Branch: > > cd /usr/src > svn co http://svn.freeswitch.org/svn/freeswitch/branches/gakul freeswitch > cd freeswitch > (merge trunk to your branch) > svn merge http://svn.freeswitch.org/svn/freeswitch/branches/gakul > http://svn.freeswitch.org/svn/freeswitch/trunk > svn commit > > Please note that since you are a developer you have been subscribed > automatically to freeswitch-dev(a)lists.freeswitch.org (please allow up to 2 > hours). If you wish not to participate in this you may unsubscribe at > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev. > > Thanks, > FreeSwitch >

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What does it mean by LOSE_RACE
by Adeel Ansari 21 Aug '08

21 Aug '08

Hi Guys, I have got this error lately, while trying to bridge 2 calls. ==== 2008-08-20 16:27:48 [NOTICE] switch_ivr_originate.c:1311 switch_ivr_originate() Hangup sofia/external/01160163472932 [CS_CONSUME_MEDIA] [LOSE_RACE] 2008-08-20 16:27:48 [DEBUG] switch_channel.c:1362 switch_channel_perform_hangup() Kill sofia/external/01160163472932 [KILL] 2008-08-20 16:27:48 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Kill sofia/external/01160163472932 [BREAK] 2008-08-20 16:27:48 [DEBUG] sofia_glue.c:1756 sofia_glue_activate_rtp() AUDIO RTP [sofia/internal/1001(a)192.168.253.101] 192.168.253.101 port 24120 -> 192.168.253.101 port 8880 codec: 8 ms: 20 2008-08-20 16:27:48 [DEBUG] switch_rtp.c:813 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2008-08-20 16:27:48 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/01160163472932) State CONSUME_MEDIA going to sleep 2008-08-20 16:27:48 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() sofia/external/01160163472932 Running State Change CS_HANGUP 2008-08-20 16:27:48 [DEBUG] switch_core_state_machine.c:393 switch_core_session_run() (sofia/external/01160163472932) State HANGUP 2008-08-20 16:27:48 [DEBUG] mod_sofia.c:264 sofia_on_hangup() Channel sofia/external/01160163472932 hanging up, cause: LOSE_RACE 2008-08-20 16:27:48 [DEBUG] mod_sofia.c:328 sofia_on_hangup() Sending CANCEL to sofia/external/01160163472932 2008-08-20 16:27:48 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() Standard HANGUP sofia/external/01160163472932, cause: LOSE_RACE ==== One phone rang not the other, as we can see the cause from the error log. Below is the way I am bridging calls. ==== <extension name="con-1019" continue="true"> <condition field="destination_number" expression="^1019$"> <action application="set" data="call_timeout=20"/> <action application="bridge" data="sofia/gateway/gizmo1/011xxxxxxxx,sofia/gateway/gizmo2/011yyyyyyyy" /> </condition> </extension> ==== Thanks for your support. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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unusal hangup and continuous calls
by Ilan Perez 21 Aug '08

21 Aug '08

Just an interesting point and question. The system I have set up is fairly simple. There is 1 pstn line coming through a gateway. When FS answers the call from the PSTN line it is re-direct to an ivr What I find is two fold: 1. If the caller hangs up unexpectedly in some cases the call continues, and I have to shutdown and restart (only because I don't know how to end a call manually) 2. Sometimes the call connects for literally 2 seconds and then both parties are hung up. Is this normal for FS or is it because of something I did? Is it the gateway? Is it the PSTN line? Any advise welcome Ilan Perez

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Ending recordFile with a touchtone in Python
by Marc Orenberg 20 Aug '08

20 Aug '08

Hello. In the modLua documentation (pasted below), it suggests in the comment that recordings using recordFile may be ended by pressing the # key. For Python, the documentation for the API points to modLua, but the touch-tone barge-in doesn't seem to be working for me. And I can't figure out a simple way to accomplish this. Any advice? session:recordFile syntax is session:recordFile(file_name, max_len, silence_threshold, silence_secs) Example: session:recordFile("/tmp/blah.wav", 30000, 10, 10); -- pressing # ends the recording session:streamFile("/tmp/blah.wav");

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Asterisk Conferencing vs FreeSWITCH
by Gavin Henry 20 Aug '08

20 Aug '08

Hi all, Is there a comparison of each or anythign obvious? We've got freeswitch install here for testing, just getting our heads round it. Thanks. -- http://www.suretecsystems.com/services/openldap/

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