RE: [Asterisk-Users] Inbound IAX to SIP

2004-02-17 Thread 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:[EMAIL PROTECTED] 
Sent: Tuesday, February 17, 2004 6:46 PM
To: [EMAIL PROTECTED]
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 => _NXXNXX,1,Playback(demo-congrats)
exten => _NXXNXX,1,Dial(SIP/248379)
exten => h,1,Hangup
exten => i,1,Hangup
exten => t,1,Hangup
; busy condition N+101...
exten => _NXXNXX,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/usernameHost Mask Port Status

248379   (Unspecified)   (D)  255.255.255.255  0
Unmonitored


Clues gratefully accepted.



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[Asterisk-Users] Inbound IAX to SIP

2004-02-17 Thread Bill Michaelson
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 => _NXXNXX,1,Playback(demo-congrats)
exten => _NXXNXX,1,Dial(SIP/248379)
exten => h,1,Hangup
exten => i,1,Hangup
exten => t,1,Hangup
; busy condition N+101...
exten => _NXXNXX,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/usernameHost Mask Port Status   
248379   (Unspecified)   (D)  255.255.255.255  0Unmonitored

Clues gratefully accepted.



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