FW: [Asterisk-Users] Sip codec preferences

[DUSTIN WILDES]

Did anyone have a way to make codec negotiation work with Asterisk?
This is something I would love to have working as well.

I won't need PSTN -> G729 mixing.  Just SIP -> SIP using G729 for calling remote offices via VPN, but everything else use G711.

-----Original Message-----
From: Brancaleoni Matteo [mailto:mbrancaleoni at espia.it]
Sent: Wednesday, July 16, 2003 11:32 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Sip codec preferences


Hi.
I'm experiencing a issue (not big, but important)
I have an asterisk installation with a buch of sip
phones & analog ones.
I have 2 1 sip phone that's outside in the "world",
and is nat'ed. I'm using g.729 with it.
I wanna use g.729 only for the remote phone, and ulaw
for the local ones, since they're on a lan.
What happens? when I call the remote phone, g.729 is used,
but when the remote calls ulaw is used... beside there's
a disallow=all , allow=g729 is the user definition.

so seems that when we call it, the codec definitions
are taken from the user config itself, but when
it call us, the codec defs are from the global settings.

that's the same if we call remote (or receive) from an analog
or iax phone.

Here's a snippet of my sip.conf:


;
; SIP Configuration for Asterisk
;
[general]
port = 5060
bindaddr = 0.0.0.0
context = local
tos = lowdelay
disallow = all
allow = ulaw

;local phone definition

[200]
accountcode=localphone
mailbox=200
type=friend
secret=secret
username=200
host=dynamic
callgroup=1
pickupgroup=1

; remote phone definition
[250]
accountcode=remotephone
type=friend
secret=XXXXX
nat=yes
username=250
context=local
reinvite=no
disallow=all
allow=g729
canreinvite=no
host=dynamic
qualify=1000
callgroup=1
pickupgroup=1

Any hint?

-- 
Matteo Brancaleoni
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-- 
Matteo Brancaleoni
Espia System Administrator
Email : mbrancaleoni at espia.it
Web   : http://www.espia.it
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