[asterisk-users] audio session start delay
Luca Corti
luca at leenoox.net
Sun Jul 9 18:24:55 MST 2006
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."
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