Hi All,
Asterisk from 11.X branch is able to renegotiate an audio codec after a SIP call session has been established (INVITE and 200 OK)?
I have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone: after the call is established in g729, after a while I have the reinvite sent by the SIP provider with g711 in the SDP; Asterisk (v 1.4.33.1) says "Oooh, we need to change our audio formats since our peer supports only g729" and sends back 200 OK to the provider; at this point I have one no audio.
So it seems that Asterisk responds 200 OK to the reinvite but really can not change the codec.
Is that correct?
Best regards,
Matteo
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