Also you should be putting these details in a jira report. http://jira.freeswitch.org
open an issue report and attach all relevant logs, do not attach tarballs or gzipped files and make sure text files have a .txt extension. On Wed, May 27, 2009 at 6:58 PM, Dale Trub <dalet...@gmail.com> wrote: > Anthony, > > Thank you for your suggestions! We are working on 1), but need to > re-integrate code we've changed, and do regression testing. That's in > progress, and we expect to be able to upgrade by the end of next week. > > We did manage to do 3) and 4), and we now have SIP logs (attached). Are you > able to see anything that's out of the ordinary that we should be paying > attention to? > > Best, > Dale > > On Fri, May 22, 2009 at 5:47 AM, Anthony Minessale < > anthony.miness...@gmail.com> wrote: > >> 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 >> >> _______________________________________________ >> 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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