We experience the same thing. The solution we use is to not change codecs in the middle of a call. I assumed it was an issue with our upstream.
> -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > Larry Moore > Sent: Thursday, June 16, 2011 10:32 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] No audio after a reinvite changing codec > > On 15/06/2011 8:15 PM, Matteo Campana wrote: > > HI list, > no idea?? :) > > > > There not much substance in the information provided for an > assessment to be made. > > I would suggest you capture the network traffic between UAC > (g711) & Asterisk UAS ensuring the snap length is large > enough to capture the whole packet and do the same with > traffic between Asterisk UAC & Provider then use Wireshark > and its telephony feature to analyse VoIP calls, check the > flows, you may discover the problem this way! > > Larry. > > > > M. > > > On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana > <matteo.camp...@gmail.com> wrote: > > > Hi all, > we have a problem with a reinvite sent by our > SIP provider to change audio codec due to the recognition of > a fax tone. > After that the SIP call session has been > established (INVITE and 200 OK) we have the following codec > situation: > > UAC > ASTERISK UAS | ASTERISK UAC PROVIDER > g711 <----------------------> > g711 | g729 <---------------------------> g729 > rtp > rtp > > After a while, we have the reinvite sent by the > SIP provider with g711 in the SDP. > So asterisk need to change audio codec from > g729 to g711 and correctly we see on debug the following line: > "Oooh, we need to change our audio formats > since our peer supports only g729" and asterisk send back 200 > OK to the provider. > At this point we have one way rtp audio: > > UAC > ASTERISK UAS | ASTERISK UAC PROVIDER > g711 ----------------------> > g711 | g711 ---------------------------> g711 > rtp > rtp > > So the problem is that UAC does not hear audio at all. > Any idea? > > (Asterisk version: 1.4.33.1). > > Thanks in advance, > Matteo > > > > > -- > > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory > webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
