And I've got a phone number which does DID in via IAX from Voicepulse. I want it to ring the GS phone for now.
I have this in extensions.conf:
[voicepulse-incoming] ; This context tells Asterisk what to do with ; incoming calls from VoicePulse (if you have signed ; up for DIDs ; ; We should now hear a "congratulations" recording ; on incoming calls to our VoicePulse phone number. ; Once we know that's working, we'll change this to a ; "Dial" statement (or something else depending on our ; needs). ;exten => _NXXNXXXXXX,1,Playback(demo-congrats) exten => _NXXNXXXXXX,1,Dial(SIP/248379) exten => h,1,Hangup exten => i,1,Hangup exten => t,1,Hangup ; busy condition N+101... exten => _NXXNXXXXXX,102,Playback(demo-congrats)
And sip.conf:
[248379] type=friend host=dynamic canreinvite=no mailbox=1234 context=demo disallow=gsm dtmfmode=inband
But the phone won't ring... it acts busy and I don't understand why. Here is some console info...
-- Accepting AUTHENTICATED call from 66.234.228.132, requested format = 4, actual format = 4
-- Executing Dial("[EMAIL PROTECTED]/2", "Sip/248379") in new stack
Feb 17 18:17:56 NOTICE[1209214528]: app_dial.c:506 dial_exec: Unable to create channel of type 'Sip'
== Everyone is busy at this time
-- Executing Playback("[EMAIL PROTECTED]/2", "demo-congrats") in new stack
-- Playing 'demo-congrats' (language 'en')
== Spawn extension (voicepulse-incoming, 6094556707, 102) exited non-zero on '[EMAIL PROTECTED]/2'
-- Executing Hangup("[EMAIL PROTECTED]/2", "") in new stack
== Spawn extension (voicepulse-incoming, h, 1) exited non-zero on '[EMAIL PROTECTED]/2'
-- Hungup '[EMAIL PROTECTED]/2'
There is also:
*CLI> sip show peers
Name/username Host Mask Port Status 248379 (Unspecified) (D) 255.255.255.255 0 Unmonitored
Clues gratefully accepted.
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