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
Hello everyone,
I've set up an asterisk box with basic PBX features (DiD, MoH, MoT,
Blind and Attended Call Transfer, PickUp, ecc.) for 10 Cisco 79xx (7912
and 7960) with SIP image (8.0). PSTN gateway is done using a Cisco
AS5350 with two ISDN PRIs connected to Asterisk via SIP. Between the
Asunto: [asterisk-users] audio session start delay
Hello everyone,
I've set up an asterisk box with basic PBX features (DiD, MoH, MoT,
Blind and Attended Call Transfer, PickUp, ecc.) for 10 Cisco 79xx (7912
and 7960) with SIP image (8.0). PSTN gateway is done using a Cisco
AS5350 with two ISDN PRIs
