I am trying to configure Asterisk to receive an inbound call on a
Zap channel T1 and Dial a SIP UA registered to Asterisk. SIP
Debug and pcap output shows that asterisk is sending an INVITE,
followed by an immediate SIP Cancel message. I hear one ring on
the called party and then an immediate disconnect - don't know
why. The phone actually sends back a 100 Trying, 180 Ringing, 200
Ok, then a 487 Request Cancelled. I've searched all over the Wiki
and can't find out why I am missing this. When I do a
playback(demo-congrats) instead of the Dial command, demo audio
plays back just fine. What am I missing and where can I look more
to find the problem?
Extensions.conf:
[Provider_T14]
exten => 2145551212,1,Dial(SIP/User1,30,r)
CLI debug:
*CLI>
-- Starting simple switch on 'Zap/73-1'
-- Executing Dial("Zap/73-1", "SIP/User1|30|r") in new
stack
-- Called User1
== Spawn extension (Provider_T14, 2145551212, 1) exited non-zero
on 'Zap/73-1'
-- Hungup 'Zap/73-1'
*CLI>
sip.conf:
[User1]
type=friend
host=dynamic
username=User1
secret=User1
qualify=200
nat=no
allow=ulaw
allow=alaw
context=intern
