[Asterisk-Users] Codec Negotiation Does not seem to work as expected ?? Help Please !!

[SW]

Try using canreinvite=yes in all three contexts. Would that screw-up ATA, I
do not know, cause I have no Cisco's ?

SW

Message: 5
Date: Mon, 05 Jan 2004 02:29:49 -0500
From: SamW <swc at svtinc.com>
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Codec Negotiation Does not seem to work as
expected ?? Help  Please !!
Reply-To: asterisk-users at lists.digium.com

Hello,

I have been trying to get my coders to work without a conversion. I have
read all the available asterisk documentation and support groups without
any luck. Here is my issue. (Please feel free to ask questions if you do
not understand what I am talking about.)

I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if
sip-server request g711)

I have 2 SIP-services to which I have to deliver the call in 2 coder
formats. Lets call 2 sip-providers, SIP-A and SIP-B. SIP-A accept g729 and
g711, SIP-B only accept g711.

I do not have any g729 licence, but I believe the * should negotiate to
have the correct passthrough coders as ATA is capable of both coders. (I
think even if you have the licenses, * should try avoid codec-conversions
when ever it can)


Here is my settings in sip.conf. I will only list the required codec
related lines, for easy understanding,

[general]
disallow=all
allow=g729
allow=ulaw
allow=alaw

register => sip-a at foo.com
register => sip-b at bar.com

[sip-a]
....
disallow=all
allow=ulaw


[sip-b]
...
disallow=all
allow=g729

[ATA]
.....
canreinvite=no