Ok so I found these threads in which some people have had similar issues
but no solutions yet...
http://lists.digium.com/pipermail/asterisk-users/2003-April/009595.html
http://lists.digium.com/pipermail/asterisk-users/2005-March/095725.html
I've tried using iaxtel and BroadVoice to route toll free calls and the
call appears to connect ok (see log snippet below) but it just rings and
rings and eventually it times out and I get
"The person you are calling is unavailable...."
The really dumb thing is all the numbers I am trying always pick up on
the first ring or without a ring when dialing from the PSTN.
Any ideas?
Log snippet below:
-- Executing Dial("SIP/116-3e81",
"SIP/[EMAIL PROTECTED]|45") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/sip.broadvoice.com-ace8 is making progress passing it to
SIP/116-3e81
-- Nobody picked up in 45000 ms
-- Executing Congestion("SIP/116-3e81", "") in new stack
== Spawn extension (inside, 18887467426, 2) exited non-zero on
'SIP/116-3e81'
_____________________________________________________________________
Shadow Roldan
IT Manager
Zero G Software, Inc.
mailto:[EMAIL PROTECTED]
www.ZeroG.com
The leading provider of multiplatform software deployment solutions.
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