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Alexander Chemeris alexander.chemeris at gmail.comOmar, Just curious - is there any reason you're running RTP through the osmo-sip-connector instead of directly to FreeSWITCH? Please excuse typos. Written with a touchscreen keyboard. -- Regards, Alexander Chemeris CEO Fairwaves, Inc. https://fairwaves.co On Jan 26, 2017 02:31, "OMAR RAMADAN" <omar.ramadan at berkeley.edu> wrote: > I've seen it a few times in production already and it filled the disk. You > should be able to reproduce it by killing an active RTP stream. I have been > using freeswitch, but I don't imagine it is limited to this SIP server. It > looks like sofia-sip is driven to continue to receiving media and gets > nothing back while the call should be terminated. > > On Wed, Jan 25, 2017 at 12:06 PM, Holger Freyther <holger at freyther.de> > wrote: > >> >> > On 25 Jan 2017, at 18:06, OMAR RAMADAN <omar.ramadan at berkeley.edu> >> wrote: >> > >> > If the SIP server dies in the middle of a call, osmo-sip-connector is >> in a bad state and generates a never ending stream of error messages: >> >> >> Can you reliable reproduce it? It seems sofia-sip is struggling with some >> input to it and goes crazy after that. I lack a stable way to reproduce it. >> The lack of \n in that message is annoying too. :( >> >> holger > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.osmocom.org/pipermail/openbsc/attachments/20170126/0ff2a6c6/attachment.htm>