[asterisk-dev] Renegotiate SIP audio codec after call is up

[Matthew Jordan]

On Wed, Jun 4, 2014 at 4:51 AM, Matteo Campana
<matteo.campana at techlan.it> 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