[asterisk-users] audio session start delay

[Luca Corti]

On Thu, 2006-07-06 at 23:22 -0300, Fabio wrote:
> are you using SIP reinvite ?

Proably not as I'm using "t" in Dial()s for call transfer.

> post a bit more information (sip.conf)

[general]
context=sip
allowguest=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
domain=mydomain.com
domain=1.2.3.4
allowexternalinvites=no
language=it
relaxdtmf=yes

[authentication]

[as5350] ; My PSTN gateway
type=peer
qualify=200
host=1.2.3.5
fromdomain=1.2.3.5
insecure=very

[ser] ; My SIP proxy
type=peer
qualify=200
host=1.2.3.6
fromdomain=1.2.3.6
insecure=very

[01]; Extension example
callerid=My Name <01>
nat=yes
type=friend
username=01
secret=mypass
host=dynamic
dtmfmode=rfc2833
context=uffici
canreinvite=no
callgroup=1
pickupgroup=1
qualify=no

Thanks

-- 
Luca Corti
PGP Key ID 1F38C091
Adesso dico: "L'usignolo chiuso in gabbia smette di cantare."