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

Eric Wieling EWieling at nyigc.com
Wed Jun 4 10:17:09 CDT 2014

How many g729 Licenses do you have?

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matteo Campana
Sent: Wednesday, June 04, 2014 10:48 AM
To: asterisk-users
Subject: [asterisk-users] Renegotiate SIP audio codec after call is up

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 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,

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140604/5131b5f1/attachment.html>

More information about the asterisk-users mailing list