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

[Olivier]

2011/5/3 Ernie Dunbar <maillist at lightspeed.ca>

> 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)
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>


Could you add an Answer() statement (just in case) ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110503/e6ef49b3/attachment.htm>