HI list, no idea?? :) 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
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