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[matteo.campana at tech...]

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

[EWieling at nyigc.com]

How many g729 Licenses do you have? 

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@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 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

[matteo.campana at tech...]

Original Message
Sender: Eric Wieling<EWieling@nyigc.com>
Recipient: Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users@lists.digium.com>
Date: mercoledì, giu 4, 2014 17:17
Subject: Re: [asterisk-users] Renegotiate SIP audio codec after call is up


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