Hi,
I have one problem, i´ve a trunk sip Asterisk--- Cisco 2600. Call
inbound work very good, but call outbound don´t work. Call progress but no
audio. Canreinvite=no , no Nat, No problem Codec.
Any idea???
Thanks in advance,
D
___
--
Sorry, This is a mistake, sip.conf:
[302]canreinvite=no[301]canreinvite=noAny idea?Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Hi list,
I got a question:
When I try to ChanSpy a
SIP channel I only listen one channel, for example,
I call from 302
extension and I have two active channels:
SIP/r1-voip-1b7b
(None)
Up Bridged Call(SIP/302-f1f1)
SIP/302-f1f1
[EMAIL PROTECTED] Up
Dial(SIP/[EMAIL
Hi list,
I got a question:
When I try to ChanSpy a SIP channel I only listen one
channel, for example,
I call from 302 extension and I have two active channels:
SIP/r1-voip-1b7b
(None)
Up Bridged Call(SIP/302-f1f1)
SIP/302-f1f1
[EMAIL PROTECTED] Up Dial(SIP/[EMAIL PROTECTED]|4
