<br><br><div class="gmail_quote">2011/5/3 Ernie Dunbar <span dir="ltr"><<a href="mailto:maillist@lightspeed.ca">maillist@lightspeed.ca</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I'm kind of at a loss to diagnose problems like this, yet we get them a lot.<br>
<br>
- The ATA (Thomson 784 in this particular case) is logged into the<br>
Asterisk server. 'sip show peer' shows their IP address, port, and<br>
useragent.<br>
- The ATA is connected directly to the internet (no NAT, but the sip<br>
configuration has nat=always) and logs in to our server, which is also<br>
directly connected to the internet without any firewalling.<br>
- When people call this extension, the console shows that Asterisk accepts<br>
the call from the DAHDI channel, executes the SIP call, then... nothing.<br>
It either waits until the timeout set in the dialplan is up, then goes to<br>
voicemail (next step), or it sends a 'hangup cause 102' to the DAHDI<br>
channel. Conspicuously missing is the console saying "SIP/username is<br>
ringing".<br>
<br>
The following is redacted output from such a call:<br>
<br>
<br>
-- Executing [6045551212@local:1] Dial("DAHDI/6-1", "SIP/sipuser|20")<br>
in new stack<br>
-- Called sipuser<br>
-- Accepting call from '7785550001' to '6045551212' on channel 0/6,<br>
span 1<br>
<br>
-- Channel 0/6, span 1 got hangup, cause 102<br>
== Spawn extension (local, 6045551212, 1) exited non-zero on 'DAHDI/6-1'<br>
-- Hungup 'DAHDI/6-1'<br>
-- No one is available to answer at this time (1:0/0/0)<br>
<br>
<br>
<br>
--<br>
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</blockquote></div><br><br>Could you add an Answer() statement (just in case) ?<br>