----- "Steve Underwood" <ste...@coppice.org> escreveu: > On 02/03/2010 12:45 AM, Vinícius Fontes wrote: > > ----- "Kevin P. Fleming"<kpflem...@digium.com> escreveu: > > > > > >> Vinícius Fontes wrote: > >> > >> > >>> I couldn't agree more Steve. > >>> > >>> Is there any other info I could provide in order to help you find > >>> > >> out what's wrong? I could even open an issue on Mantis if the > Digium > >> staff think it's worth it. > >> > >> Post a 'sip set debug' capture of the failing call in this thread; > >> that > >> will make it much more obvious what is happening. > >> > >> -- > >> Kevin P. Fleming > >> Digium, Inc. | Director of Software Technologies > >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > >> skype: kpfleming | jabber: kpflem...@digium.com > >> Check us out at www.digium.com& www.asterisk.org > >> > >> -- > >> > _____________________________________________________________________ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com > -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > I've put it on pastebin because is was a lot of text. Here's the > link: http://pastebin.com/m7467cea1. That's all the information on the > CLI with verbose=3 and "sip set debug peer voxip". > > > > > I wonder why Asterisk would say: > > X-asterisk-Info: SIP re-invite (External RTP bridge) > Content-Type: application/sdp > Content-Length: 344 > > v=0 > o=root 44350963 44350964 IN IP4 10.153.66.146 > s=Asterisk PBX 1.6.1.13 > c=IN IP4 10.153.66.146 > t=0 0 > m=image 4819 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxTranscodingMMR > a=T38FaxTranscodingJBIG > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxDatagram:1400 > a=T38FaxUdpEC:t38UDPRedundancy > > I'm pretty sure it doesn't support T38FaxTranscodingMMR or > T38FaxTranscodingJBIG, so they should not be there. Perhaps more > relevant to you, though, is why is * saying "(External RTP bridge)". > Does it really mean it? > > Steve
I'm not really sure. What I know is that this telco has separate boxes for SIP signalling and RTP media. Not even sure if that's related to your question which, to be honest, I didn't fully understand. :) -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users