[asterisk-users] Question about codec re-negotiation in asterisk 1.4.X

[Eric Wieling]

Asterisk supports reinvites (if reinvites are enabled in sip.conf), just not changing codecs in the middle of the call.      If anyone has managed to get it to work, I'd love to hear about it.

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matteo Campana
Sent: Friday, July 22, 2011 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X


On Fri, Jul 22, 2011 at 4:46 PM, Eric Wieling <EWieling at nyigc.com<mailto:EWieling at nyigc.com>> wrote:

Asterisk does not support changing codecs on the fly.


And why asterisk sends 200 OK to the provider, if does not support its re-invite?

M.




From: asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com> [mailto:asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>] On Behalf Of Matteo Campana
Sent: Friday, July 22, 2011 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X

Hi all,
I have a major issue with a codec renegotiation in an asterisk 1.4.33.1 setup, which leads me to ask a general question about asterisk 1.4.X codec negotiation: asterisk can support a re-negotiation of a codec "on the fly" through a re-Invite? If my SIP provider sends me a re-invite changing codec from g729 to g711, asterisk properly handle the matter?
I see in the trace that asterisk responds 200 OK to the provider, but does not send the re-invite to the UAC, and stops to send rtp to the UAC.



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