[asterisk-users] Call not going through and failing because "never answered"

[Zeeshan Zakaria]

You are getting congestion error message, which in your case only means
failed sip communication, or no sip communication at all. Settings on your
end are just fine.

Can you post the Dial command from your extensions.conf?

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-07-20 12:16 PM, "Andy Beak" <andrewb at cellsmart.co.za> wrote:

Hi,

Thanks, I added that.  I'll ask my network provider if they received these
message tomorrow morning.  That will narrow things down to either an
Asterisk configuration or a network routing issue.

There is not really a caller, I'm trying to use Asterisk as an Automated
Voice Message server to dial phone numbers and play an mp3.

I'm using my mobile phone to test on and it doesn't ring.  Asterisk gives
the following message immediately after reading the .call file from the
spool directory:

-- Attempting call on SIP/MTN-NEW/mynumber for application
MP3Player(/myfile) (Retry 1)


 == Using SIP RTP CoS mark 5
> Channel SIP/MTN-NEW-00000001 was never answered.
[Jul 20 18:07:37] NOTICE[22259]: pbx_spool.c:339 attempt_thread: Call failed
to go through, reason (8) Congestion (circuits busy)

Because the phone doesn't ring and the error message appears immediately I
don't think it's a timeout issue.

Will reading the source for pbx_spool.c at line 339 give any clues as to
what's happening or will that be a waste of time?

Cheers,
 Andy




On 20/07/2010 05:42 PM, Gareth Blades wrote:
>
> If you add qualify=yes to the setting in sip.con...

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