[Asterisk-Users] ATA186: SIP/2.0 503 Service Unavailable

[Brian Capouch]

I have done my homework on this, I hope.

I have a customer with an ATA186 who uses Nufone as his IAX provider. 
His network operations center in the Bahamas was destroyed by the 
hurricanes, and I'm helping him rebuild.

We have a nagging problem getting his ATAs (located in public IP space) 
to talk through his IAX provider (Nufone) to the outside world.  As far 
as we know, things worked OK before, but when we reconstructed things 
something must be awry.

He dials on the ATA, we see the call being picked up normally by 
Asterisk, the provider accepts the call and shows the codec being used, 
and then immediately hangs up.

The (sanitized) sip debug output at that point looks like this:

     -- Call accepted by W.X.Y.Z (format ULAW)
     -- Format for call is ULAW
     -- Hungup 'IAX2/Carrier/7'
   == No one is available to answer at this time
     -- Executing Congestion("SIP/blah-blah-76d6", "") in new stack
Transmitting (no NAT):
SIP/2.0 503 Service Unavailable

I have confs and dumps but they're laden with his IPs and passwords, 
etc.  I can sanitize and post them if it's helpful, but perhaps someone 
there would know some canonical circumstance that leads to this error.

I set up a context with parameters as nearly identical as I could to 
point his server at mine here in my office, and the call zipped right 
through, same phone, same parameters in the config, same codecs.

Thanks.

B.