[asterisk-users] ATA refuses to answer a call?

[Ernie Dunbar]

I'm kind of at a loss to diagnose problems like this, yet we get them a lot.

- The ATA (Thomson 784 in this particular case) is logged into the
Asterisk server. 'sip show peer' shows their IP address, port, and
useragent.
- The ATA is connected directly to the internet (no NAT, but the sip
configuration has nat=always) and logs in to our server, which is also
directly connected to the internet without any firewalling.
- When people call this extension, the console shows that Asterisk accepts
the call from the DAHDI channel, executes the SIP call, then... nothing.
It either waits until the timeout set in the dialplan is up, then goes to
voicemail (next step), or it sends a 'hangup cause 102' to the DAHDI
channel. Conspicuously missing is the console saying "SIP/username is
ringing".

The following is redacted output from such a call:


    -- Executing [6045551212 at local:1] Dial("DAHDI/6-1", "SIP/sipuser|20")
in new stack
    -- Called sipuser
    -- Accepting call from '7785550001' to '6045551212' on channel 0/6,
span 1

    -- Channel 0/6, span 1 got hangup, cause 102
  == Spawn extension (local, 6045551212, 1) exited non-zero on 'DAHDI/6-1'
    -- Hungup 'DAHDI/6-1'
    -- No one is available to answer at this time (1:0/0/0)