<p>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.</p>
<p>Can you post the Dial command from your extensions.conf?<br></p>
<p>Zeeshan A Zakaria</p>
<p>--<br>
<a href="http://www.ilovetovoip.com">www.ilovetovoip.com</a></p>
<p><blockquote type="cite">On 2010-07-20 12:16 PM, "Andy Beak" <<a href="mailto:andrewb@cellsmart.co.za">andrewb@cellsmart.co.za</a>> wrote:<br><br>Hi,<br>
<br>
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.<br>
<br>
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.<br>
<br>
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:<br>
<br>
-- Attempting call on SIP/MTN-NEW/mynumber for application MP3Player(/myfile) (Retry 1)<p><font color="#500050"><br> == Using SIP RTP CoS mark 5</font></p>
> Channel SIP/MTN-NEW-00000001 was never answered.<br>
[Jul 20 18:07:37] NOTICE[22259]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy)<br>
<br>
Because the phone doesn't ring and the error message appears immediately I don't think it's a timeout issue.<br>
<br>
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?<br>
<br>
Cheers,<br><font color="#888888">
Andy</font><p><font color="#500050"><br><br><br>On 20/07/2010 05:42 PM, Gareth Blades wrote:<br>><br>> If you add qualify=yes to the setting in sip.con...</font></p><br>--<br>
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