[Asterisk-Users] Inbound IAX to SIP

[Sean Cheesman]

It looks like your phone is not registering correctly.  try doing a sip
show users and see if it's registered.  also, I've found that many of
the sip.conf entries require a username=248379 in your case, matching
the sip entry name.  but as I look again, it could be the context.  make
sure that your voicepulse-incoming and your demo contexts are linked
somehow.

Sean

-----Original Message-----
From: Bill Michaelson [mailto:bill at cosi.com] 
Sent: Tuesday, February 17, 2004 6:46 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Inbound IAX to SIP


I've a SIP phone (GS 100) which dials out fine through a Voicepulse 
Connect account via *.

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("IAX2[voicepulse at voicepulse]/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("IAX2[voicepulse at voicepulse]/2", 
"demo-congrats") in new stack
    -- Playing 'demo-congrats' (language 'en')
  == Spawn extension (voicepulse-incoming, 6094556707, 102) exited 
non-zero on 'IAX2[voicepulse at voicepulse]/2'
    -- Executing Hangup("IAX2[voicepulse at voicepulse]/2", "") in new
stack
  == Spawn extension (voicepulse-incoming, h, 1) exited non-zero on 
'IAX2[voicepulse at voicepulse]/2'
    -- Hungup 'IAX2[voicepulse at voicepulse]/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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