Thanks guys. I wasn't able to collect enough SIP debug as the problem was resolved as I was testing different configuration for the trunk. Probably a change on the provider side.<div><br></div><div>John Novack: Unfortunately, it seems that this list has a non-stop list of people who like to stir up things or try to censor people who bring legit questions without the consideration that they are not moderators of the list at any level. They forget to remember that AsteriskNow uses FreePBX as well and that Asterisk IS the underlying technology for all the flavours. Thanks for the feedback.</div>
<div><br></div><div>Most I was able to collect was that:</div><div><br></div><div>- if the trunk configuration even included "context=from-pstn", the CLI would show "Executing .....@from-sip-external".</div>
<div>- if SIP Anonymous was set to YES then the [from-sip-external] context would match the peer to the right trunk defined as that is what is expected of that context from the code. If SIP Anonymous was set to off @from-sip-external is set to go to ss-noservice.</div>
<div>- Later on when the calls resumed and the problem was fixed, calls were coming in with "Executing @from-pstn" which should have always been the case regardless of the SIP Anonymous or not. </div><div><br></div>
<div>I was puzzled because the FISRT line of the CLI was the "Executing ....@from-pstn" or "Executing .....@from-sip-external" and that made a world of difference. The latter one not working.</div><div>
<br></div><div>I just couldn't pinpoint where FreePBX failed to read the "context=from-pstn". If it was something to do with the MySQL database or of parsing the _custom.conf files as the problem was fixed all a sudden. I guess now I have to wait and see if it comes back.</div>
<div><br></div><div>Thanks,</div><div><br></div><div><br><br><div class="gmail_quote">On Tue, Sep 14, 2010 at 10:47 AM, Zeeshan Zakaria <span dir="ltr"><<a href="mailto:zishanov@gmail.com">zishanov@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div>This might help to answer poster's question. It tells how the allow anonymous sip connections work in FreePBX, and shows the code.</div>
<div><br></div><a href="http://www.geekzone.co.nz/sbiddle/7183" target="_blank">http://www.geekzone.co.nz/sbiddle/7183</a><div>
<br></div><div><a href="http://www.geekzone.co.nz/sbiddle/7183" target="_blank"></a>--</div><div>Zeeshan<div><div></div><div class="h5"><br><br><div class="gmail_quote">On Sun, Sep 12, 2010 at 12:11 AM, Paul Belanger <span dir="ltr"><<a href="mailto:paul.belanger@polybeacon.com" target="_blank">paul.belanger@polybeacon.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>On Sat, Sep 11, 2010 at 9:40 PM, Zeeshan Zakaria <<a href="mailto:zishanov@gmail.com" target="_blank">zishanov@gmail.com</a>> wrote:<br>
> Poster is having problem when he disallows anonymous sip peers. Do you know<br>
> at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet<br>
> seen the dialplan for FreePBX.<br>
><br>
</div>It's very simple to find the actually issue, if the OP does the following:<br>
<br>
<a href="http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt" target="_blank">http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt</a><br>
<br>
The attached the debug log to thread.<br>
<br>
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</div>Jabber: <a href="mailto:paul.belanger@polybeacon.com" target="_blank">paul.belanger@polybeacon.com</a> | IRC: pabelanger (Freenode)<br>
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