Hrm, setting either directmedia=yes and directrtpsetup=yes in sip.conf does not seem to fix the issue.
I wonder if this is a network issue, everything is on routable address's endpoint wise, and the gateway in between routes between my RFC1918 address network (which the asterisk server sits on), i've done this sort of setup in prod before tho and it works well. On 13 April 2011 10:39, Joel Wiramu Pauling <j...@aenertia.net> wrote: > Cheers will give that a go, thanks for your input Gunnar. > > > wrt @amit: Codec is supported, it's the SDP/ATV combination ( I assume > that's the resolution ) that it is saying is unsupported - h264 ( the > codec for video ) is working fine, I think you are seeing the Siren > (audio) mismatches thats fine it falls back to ulaw. > > > Kind regards > > -JoelW > > > On 13 April 2011 08:35, Gunnar Schaller <li...@nowin.de> wrote: >> Hello, >> Try a Dial without "tr" parameters and with "directmedia=yes" in >> sip.conf. >> http://www.voip-info.org/wiki/view/Asterisk+SIP+media+path >> >> Regards, >> Gunnar >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-video mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-video >> > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video
