1) update to lastest trunk (you are at least 1000 revisions behind) 2) disable the presence debug in sofia.conf 3) enable sip trace instead "sofia profile internal siptrace on" 4) reproduce your problem.
Make sure you include more of the log from before the hangup happened. The one you posted here is missing some of the info from the few seconds prior but with the incomplete info it looks like the other side sent a BYE ending the call. On Thu, May 21, 2009 at 10:09 PM, Dale Trub <dalet...@gmail.com> wrote: > Thanks Brian! To answer your questions: > Freeswitch svn revision: 12148 > Centos rev: 2.6.18-92.el5 > > And apologies, actually I guess we're using g711 not 729. > > Jason: I agree it would seem to be on the switch/telco side. And, the > telco says many other people are in the same set-up as us and don't have any > issues, so they're insisting it's on our end. > > On Thu, May 21, 2009 at 7:28 PM, Brian West <br...@freeswitch.org> wrote: > >> >> On May 21, 2009, at 9:15 PM, Dale Trub wrote: >> >> We're running FreeSwitch as part of a teleconferencing service, inside a >> telcom (so no >> internet latency/NAT issues) and using g.729 >> >> >> So you're using g729 with conferences? >> >> We are receiving some complaints of dropped calls, >> including from landlines. This means they join the conference, and x >> minutes in they simply drop. >> >> I know that cellphones tend to drop calls frequently, but landlines >> are pretty reliable, and we're hearing it a lot. From the FreeSwitch side >> of things, it just >> looks like those callers hung up (but then dialed back in just a moment >> later). >> >> I'm attaching two different snippets of the FS log files where these >> issues are occurring. >> >> >> Next time please call them .txt because you cause extra work to have to >> open them otherwise. >> >> Does anyone have any recommendations about how to troubleshoot this? >> >> Any known issues/patches in FS that could be biting us? >> >> >> Depends you failed to include some very valid info such as what version or >> svn rev you're running and what linux distro. >> >> Is there some SIP logging we can do to debug? >> >> >> Yes covered on the wiki. >> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch >> >> Are there any paid contractors avail who would have the expertise to look >> into this? >> >> >> email consult...@freeswitch.org >> >> Any help appreciated ... this is a major issue for us! >> >> Thanks much, >> >> -Dale >> >> >> Brian West >> br...@freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users@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@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com <msn%3aanthony_miness...@hotmail.com> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org <sip%3a...@conference.freeswitch.org> iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org> pstn:213-799-1400
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