[Asterisk-Users] SIP CANCEL problem

[justiceguy at pobox.com]

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