<p>FreePBX questions should be asked at FreePBX forums.</p>
<p>As for the asterisk part, where are you defining the context to receive incoming calls? Probably in the trunk settings (Peer Details) you need to add "context=from-trunk" if FreePBX still uses it as the default context for incoming calls.<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 2010-06-10 11:24 AM, "bruce bruce" <<a href="mailto:bruceb444@gmail.com">bruceb444@gmail.com</a>> wrote:<br><br>Hi Guys,<div><br></div><div>I have Spikko setup as provider of DID and outbound routes and I can make calls out but no inbound calls via DID can be made. I did a sip debug which is reported below. I never receive the call though, I have a catch all in my inbound routes and it doesn't hit my context at all or not sip invite comes in:</div>
<div><br></div><div><br></div><div>FreePBX:</div><div><br></div><div>Trunk Name:</div><div><b>Spikko</b></div><div><br></div><div>Peer Detail</div><div><div><b>username=MyUsername</b></div><div><b>type=friend</b></div><div>
<b>secret=MyPassword</b></div><div><b>host=<a href="http://sip.spikko.com" target="_blank">sip.spikko.com</a></b></div><div><b>nat=no</b></div><div><b>port=5090</b></div><div><b>fromuser=MyUsername</b></div><div><b>disallow=all</b></div>
<div><b>allow=g729&gsm&ulaw&alaw</b></div><div><br></div></div><div>Register String:</div><div><b><a href="http://MyUsername:MyPassword@sip.spikko.com:5090/MyUsername" target="_blank">MyUsername:MyPassword@sip.spikko.com:5090/MyUsername</a></b></div>
<div><br></div><div><br></div><div>Inbound Router:</div><div><b>Send Any DID and ANY CID to Music on Hold</b></div><div><br></div><div><br></div><div>Sip debug:</div><div><br></div><div><div><div><b>Really destroying SIP dialog '<a href="mailto:417b3c8f3a97a82d4629343a53b2feb6@177.177.177.177" target="_blank">417b3c8f3a97a82d4629343a53b2feb6@177.177.177.177</a>' Method: REGISTER</b></div>
<div><b>tel*CLI></b></div><div><b><--- SIP read from UDP:<a href="http://82.80.252.29:5090" target="_blank">82.80.252.29:5090</a> ---></b></div><div><b>INVITE <a href="mailto:sip%3AMyUsername@177.177.177.177" target="_blank">sip:MyUsername@177.177.177.177</a> SIP/2.0</b></div>
<div><b>Via: SIP/2.0/UDP 82.80.252.234:5090;branch=z9hG4bK07b38a0c;rport</b></div><div><b>From: "Unknown" <<a href="http://sip:Unknown@82.80.252.234:5090" target="_blank">sip:Unknown@82.80.252.234:5090</a>>;tag=as24089849</b></div>
<div><b>To: <<a href="mailto:sip%3AMyUsername@177.177.177.177" target="_blank">sip:MyUsername@177.177.177.177</a>></b></div><div><b>Contact: <<a href="http://sip:Unknown@82.80.252.234:5090" target="_blank">sip:Unknown@82.80.252.234:5090</a>></b></div>
<div><b>Call-ID: <a href="mailto:55a4cf1f4e5575e97f8b3b23495f0a43@82.80.252.234" target="_blank">55a4cf1f4e5575e97f8b3b23495f0a43@82.80.252.234</a></b></div><div><b>CSeq: 102 INVITE</b></div><div><b>User-Agent: AG1</b></div>
<div><b>Max-Forwards: 70</b></div>
<div><b>Date: Thu, 10 Jun 2010 14:58:09 GMT</b></div><div><b>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</b></div><div><b>Supported: replaces</b></div><div><b>Content-Type: application/sdp</b></div>
<div><b>Content-Length: 331</b></div><div><b><br></b></div><div><b>v=0</b></div><div><b>o=root 6129 6129 IN IP4 82.80.252.234</b></div><div><b>s=session</b></div><div><b>c=IN IP4 82.80.252.234</b></div><div><b>t=0 0</b></div>
<div><b>m=audio 10172 RTP/AVP 18 3 97 101</b></div><div><b>a=rtpmap:18 G729/8000</b></div><div><b>a=fmtp:18 annexb=no</b></div><div><b>a=rtpmap:3 GSM/8000</b></div><div><b>a=rtpmap:97 iLBC/8000</b></div><div><b>a=fmtp:97 mode=30</b></div>
<div><b>a=rtpmap:101 telephone-event/8000</b></div><div><b>a=fmtp:101 0-16</b></div><div><b>a=silenceSupp:off - - - -</b></div><div><b>a=ptime:20</b></div><div><b>a=sendrecv</b></div><div><b><br></b></div><div><b><-------------></b></div>
<div><b>--- (14 headers 16 lines) ---</b></div><div><b>Using INVITE request as basis request - <a href="mailto:55a4cf1f4e5575e97f8b3b23495f0a43@82.80.252.234" target="_blank">55a4cf1f4e5575e97f8b3b23495f0a43@82.80.252.234</a></b></div>
<div>
<b>Found peer 'Spikko' for 'Unknown' from <a href="http://82.80.252.29:5090" target="_blank">82.80.252.29:5090</a></b></div></div></div><div><br></div><div><br></div><div>I also sometimes get this even though trunk shows registered and can make calls out:</div>
<div><div><b><--- Transmitting (no NAT) to <a href="http://82.80.252.29:5090" target="_blank">82.80.252.29:5090</a> ---></b></div><div><b>SIP/2.0 489 Bad event</b></div><div><b>Via: SIP/2.0/UDP 82.80.252.234:5090;branch=z9hG4bK463b703d;received=82.80.252.29;rport=5090</b></div>
<div><b>From: "asterisk" <<a href="http://sip:asterisk@82.80.252.234:5090" target="_blank">sip:asterisk@82.80.252.234:5090</a>>;tag=as4af8cf81</b></div><div><b>To: <<a href="mailto:sip%3Asaarshalom@173.203.29.102" target="_blank">sip:saarshalom@173.203.29.102</a>>;tag=as64c0ba34</b></div>
<div><b>Call-ID: <a href="mailto:497197a679122f5d448d324f571f3c3e@82.80.252.234" target="_blank">497197a679122f5d448d324f571f3c3e@82.80.252.234</a></b></div><div><b>CSeq: 102 NOTIFY</b></div><div><b>Server: Asterisk PBX 1.6.2.7</b></div>
<div><b>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</b></div><div><b>Supported: replaces, timer</b></div><div><b>Content-Length: 0</b></div></div><div><br></div><div>Thanks,</div><div>Bruce</div>
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