I don't know. > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of > Matteo Campana > Sent: Friday, June 17, 2011 5:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] No audio after a reinvite changing codec > > > > Inviato da iPhone > > Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling > <[email protected]> ha scritto: > > > > > 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. > > > Hi Eric, > this behavior is an asterisk bug or asterisk can never > change the codec "on the fly"? > > > Thanks, > Matteo > > > > > > > >> -----Original Message----- > >> From: [email protected] > >> [mailto:[email protected]] 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 > >> <[email protected]> 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 > > -- > _____________________________________________________________________ > -- 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 >
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