[Asterisk-Users] Codec negotation with re-invites..

Billy Huddleston billy at nxs.net
Fri Mar 12 12:02:51 MST 2004


I'm about over this.. okay,, here is what I got..

[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = inbound                       ; Default for incoming calls
tos=lowdelay
tos=184
disallow=all                    ; Disallow all codecs
allow=ulaw

[gateway]
type=friend
host=1.1.6.9
canreinvite=yes
qualify=yes
dtmfmode=rfc2833
context=default
disallow=all
allow=ulaw
allow=g729

[sipphoneg729]
type=friend
secret=password
nat=yes
host=dynamic
canreinvite=yes
qualify=200
context=longdistance-g729
dtmfmode=rfc2833
mailbox=2199
disallow=all
allow=g729

[sipphoneulaw]
type=friend
secret=password
nat=yes
host=dynamic
canreinvite=yes
qualify=200
context=longdistance
dtmfmode=rfc2833
mailbox=2199
disallow=all
allow=ulaw


okay, when I place a call from sipphoneulaw to the outside world via
gateway, everything works fine..
If I place a call from sipphoneg729, it doesn't work..  One leg to the
gateway will be ulaw, the leg to the phone will be g729, and, I have 1 way
audio.. The sip phone can hear anything from the gateway, but, the gateway
can't hear the phone.

I've even went as far as to setup a seperate context for the g729 phone and
do this..
,SetVar,SIP_CODEC=g729  which, says it sets it to g729, but it's still a
ulaw call..  Guys, this is a real problem... We're going be doing mixed
configs.. and if a gateway says it can do both, and phone says it can only
do one... then we should be using the compatable codec...  PLEASE help..
This is going to cause problems in our rollout.

Thanks, Billy


             +--------------------------------------------------+
             | Billy Huddleston   Senior Systems Administrator  |
             | Net-Express                  http://www.nxs.net  |
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             | billy at nxs.net                                    |
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