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

[Joe Antkowiak]

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

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users