[Asterisk-Users] DID problem

[Sam Tam]

Hello All,

I don't know whether this has been talked about before or not but it seems
that from time to time I always bump into problem with the DID.

What happening at the moment is that I have a incoming number pointing to a
calling card software.  I have setup a inbound route in AMP and have it
point to the calling card software which works completely fine.

But half of the time the DID will reach the server and terminate without
passing it to calling card and then a message of "The service cannot be
connected" will come out.


    -- Executing AbsoluteTimeout("SIP/195.8.117.11-b7815b90", "15") in new
stack
    -- Set Absolute Timeout to 15
    -- Executing Congestion("SIP/195.8.117.11-b7815b90", "") in new stack
  == Spawn extension (from-sip-external, 02080359600, 2) exited non-zero on
'SIP/195.8.117.11-b7815b90'
    -- Executing AbsoluteTimeout("SIP/195.8.117.11-b7815b90", "15") in new
stack
    -- Set Absolute Timeout to 15
    -- Executing Congestion("SIP/195.8.117.11-b7815b90", "") in new stack
  == Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/195.8.117.11-b7815b90'
asterisk1*CLI>

And the other half of the time it will pass it on to the calling card.

I am quite confused on why that is happening and would love to hear if
anybody has experienced such before.

Sam