[asterisk-dev] SIP codec selection. Can I select a codecand reinvite?

[asterisk]

Yes sorry my typo.  

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Jared Smith
Sent: Monday, November 19, 2007 3:54 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] SIP codec selection. Can I select a codecand
reinvite?

On Mon, 2007-11-19 at 15:12 -0500, asterisk wrote:
> I am having an issue with codec selections. I am using only 2 codecs
> ULAW & G729.  On some peers I want the proffered codec to be G729
other
> ULAW.    I have a SIP trunk form a carrier that supports both G729 and
> ULAW. I would like asterisk to use the [proffered codec and not
> transcode.
> 
> My peers are setup like this. 
> 
> [siptrunk] 
> Deny=all
> Allow=ulaw
> Allow=g729

Maybe you meant to have "disallow=all" instead of "deny=all"?  There's a
subtle but important difference between the two statements.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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