[Asterisk-Users] G723 - Has anyone gotten SIP_CODEC= to work?

[Dan Fernandez]

Joe,

Thanks. I know * doesn´t support g723 (actually you can still have a SIP to
SIP call with g723) and that´s exactly why I need to change codecs on the
fly.

Dan

----- Original Message -----
From: "Joe Antkowiak" <joe at jsci.net>
To: <asterisk-users at lists.digium.com>; "Dan Fernandez"
<danfernandez00 at hotmail.com>
Sent: Monday, May 05, 2003 7:19 PM
Subject: RE: [Asterisk-Users] G723 - Has anyone gotten SIP_CODEC= to work?


> FYI, asterisk DOES now support g723, but you have to pay for it:
>
> http://store.yahoo.com/asteriskpbx/asteriskg729.html
>
> -----Original Message-----
> From: Dan Fernandez <danfernandez00 at hotmail.com>
> Date: Mon, 5 May 2003 17:33:05 -0300
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC=   to work?
>
> Basically, since I´d like to use g723 for sip communication between
> endpoints and * does not support it, I need to change codecs when a user
> wants to check voicemail, use a zap channel, etc.
>
> I have configured sip.conf and extensions.conf as below but when I try it
I
> keep getting the following:
>
> chan_sip.c ...(sip_answer):Changing codec to GSM for this call because of
> ${SIP_CODEC} variable
> channel.c ..(ast_set_write_format): Unable to find a path from 2 to 1.
>
> Any ideas?
>
>
>
> sip.conf
> disallow=all
> allow=g723.1
> allow=gsm
>
>
> extensions.conf
>
> exten => 1000,1,SetVar,SIP_CODEC=gsm
> exten => 1000,2,VoiceMailMain
>
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