On Wed, Jun 4, 2014 at 4:51 AM, Matteo Campana <[email protected]> wrote: > Hi Devs, > > 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? >
This is not a development question. The asterisk-dev mailing list is used to discuss changes to the Asterisk source code, not to solicit help for deployment situations. Please use the asterisk-users mailing list for questions of this type - you will be far more likely to receive useful responses. (And, 99% of the developers are subscribed to and read that mailing list as well) Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
