November 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP/2.0 500 No session set by user
by Brian Wiese 15 Nov '11

15 Nov '11

Hello again everyone! I'm getting a strange result when calling from a Polycom endpoint on FS server "A" to FS server "B": About one out of ten calls won't complete with the error "SIP/2.0 500 No session set by user". If I immediately redial the number again it works. Here's my log: http://pastebin.freeswitch.org/17622 Note the response to the endpoint from server "A" on line 452. Important IP addresses: Polycom endpoint: 10.15.100.250 Server "A": 10.15.100.20 Server "B": 10.15.100.10 I'm on the latest revision (git-2f786a0 2011-10-26 12-57-59 -0500) as well. I appreciate your help. ~Brian

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UUID logging - help test patch?
by Avi Marcus 14 Nov '11

14 Nov '11

If you use uuid prepending on your logging to help you grep out a call, Math wrote a patch to apply that to the multi-line SDP, too. (See http://wiki.freeswitch.org/wiki/Mod_logfile#Logging_UUIDs for how to turn on) If you can test it out and comment, the patch is here: http://jira.freeswitch.org/browse/FS-3572 Also, there's still at least 4 log types that don't get the UUID, I unfortunately still don't know enough of C & FS to check for myself why: 2011-11-03 08:41:12.696695 [INFO] switch_nat.c:589 NAT port mapping disabled 2011-11-03 08:41:12.876687 [DEBUG] switch_rtp.c:3181 Correct ip/port confirmed. 2011-11-03 08:41:12.736698 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz 2011-09-16 15:44:06.776690 [DEBUG] switch_core_file.c:180 File .....v3f.wav sample rate 44100 doesn't match requested rate 8000 -Avi Marcus

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FreeSWITCH
by Craig Smith 14 Nov '11

14 Nov '11

Some recent frustrations with Asterisk have me looking at FreeSWITCH, but it’s lack? of adoption leaves me wondering. Here are some questions I have as a long time Asterisk user, dCAP, and Astricon attendee. Why hasn’t FreeSWITCH reached a critical mass? Is the XML configuration that scary? What does Asterisk have that FreeSWITCH doesn’t? Does FreeSWITCH need better documentation and/or tutorials? Is it git? Would a FreeSWITCH certification help? I want to help the FreeSWITCH project, but I’m not developer or philanthropist , what can I do?

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Applying a patch to freeswitch
by Gilad Abada 14 Nov '11

14 Nov '11

Hi All Can anyone tell me how to apply a patch to freeswitch? Thanks Gill

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http://latest.freeswitch.org/ not found
by Thomas Hoellriegel 14 Nov '11

14 Nov '11

Hi all, i read the install-instuction from: http://wiki.freeswitch.org/wiki/Download_FreeSWITCH#Source_Tarball Why is this link http://latest.freeswitch.org/ not working? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

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Expires ignored
by Hynek Cihlar 14 Nov '11

14 Nov '11

Freeswitch ignores the Expires header, instead registers SIP clients with expiration timeout of 90 seconds. I know there are few variables/parameters which control the Expiration. I use pretty default SIP profile configuration but the directory is built with with a lua script. If the lua script defines sip-force-expires, the registrations is ok forced to this value. If the variable sip-force-expires is missing, 90 seconds is used regardless what the client sends. Any ideas? Thanks for any help. Hynek

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Anser confirmation via script?
by Thomas Hoellriegel 14 Nov '11

14 Nov '11

Hi all, I am a newbi, what lua perl and python are concerned. Has someone already managed to write a answerconfirmation? My approach: I like to setup a greate callscreening. For example: press 1 to accept the caller. press 2 to forward to voicemail. Press 3 to transfer the call to my cellphone. Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

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Re: [Freeswitch-users] FreeTDM PRI tapping issue
by kiruthika sri 13 Nov '11

13 Nov '11

Resolved SEGFAULT by changing the following line, crv = tap_pri_get_crv(pritap->pri, e->ring.call) into crv = tap_pri_get_crv(pritap->pri, e->proceeding.call) under the case PRI_EVENT_PROGRESS from ftmod_pritap.c source. On Sat, Nov 5, 2011 at 9:55 AM, kiruthika sri <kiruthika.bkite(a)gmail.com>wrote: > > > ---------- Forwarded message ---------- > From: kiruthika sri <kiruthika.bkite(a)gmail.com> > Date: Thu, Nov 3, 2011 at 10:11 AM > Subject: Re: [Freeswitch-users] FreeTDM PRI tapping issue > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Getting segfault after getting RINGING event in the first call itself. > > > On Wed, Nov 2, 2011 at 9:01 PM, Moises Silva <moises.silva(a)gmail.com>wrote: > >> getting segfault right away? after a few calls? days? >> >> On Wed, Nov 2, 2011 at 12:35 AM, kiruthika sri <kiruthika.bkite(a)gmail.com >> > wrote: >> >>> >>> Hi, >>> >>> I am trying to use the ftmod_pritap module for passive call recording. >>> >>> I did the followings. >>> * Installed wanpipe 3.5.23 for sangoma card AFT A102. >>> * Installed tap 1.4 (customized version of librpi for passive tapping) >>> * Installed freeswitch and freetdm with --with-pritap option >>> * Configured wanpipe, freetdm and freeswitch. >>> >>> Facing the following problem : >>> * FreeSWITCH is getting killed >>> * Getting SIGSEGV >>> >>> # wanrouter messages >>> >>> freeswitch[19720]: segfault at 8 ip 00ae00d1 sp b6ead208 error 4 in >>> libpri.so.1.4[ac5000+4c000] >>> >>> # gdb freeswitch core >>> >>> Program terminated with signal 11, Segmentation fault. >>> #0 0x00ae00d1 in q931_call_getcrv (ctrl=0x99fd9c0, call=0x0, >>> callmode=0xb6ead24c) at q931.c:5335 >>> 5335 *callmode = call->cr & 0x7; >>> >>> Kindly do the help on this. >>> >>> Thanks in advance. >>> >>> Regards >>> Kiruthika.U >>> >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > >

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Adjust nat-options-ping interval
by Spencer Thomason 13 Nov '11

13 Nov '11

Hello all, Is there any way to adjust the interval of the nat-options-ping parameter? All of our endpoints are behind NAT using TCP and 30 seconds is a little short as pretty much every router I can think of has a TCP timeout greater than that. I realize that this is a very specific change so if someone could point me to the right place to look I can patch this for our use. Thanks! Spencer

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Conference and room numbers in a multi tenant environment
by Peter P GMX 12 Nov '11

12 Nov '11

Hello, I created 2 conferences * 1000@domain1 * 1000@doamin2 When I called * conference 1000@domain1 from user 100@domain1 and * conference 1000@domain2 from user 100@domain2 both users are in the same conference in my case. Question: Is 1000@domain1 distinct from 1000@domain2 or do I have to use seperate room numbers? Best regards Peter

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