<div dir="ltr"><div class="gmail_default" style="font-family:tahoma,sans-serif;color:rgb(11,83,148)">Ok, I also tried to hangup directly through dialplan, it doesn't work.</div><div class="gmail_default" style="font-family:tahoma,sans-serif;color:rgb(11,83,148)"><br></div><div class="gmail_default" style="font-family:tahoma,sans-serif;color:rgb(11,83,148)"><div class="gmail_default"> == Using SIP RTP CoS mark 5</div><div class="gmail_default"> -- Executing [12023300643@default:1] Hangup("SIP/66.226.76.70-0000d0b0", "41") in new stack</div><div class="gmail_default"> == Spawn extension (default, 12023300643, 1) exited non-zero on 'SIP/66.226.76.70-0000d0b0'</div><div class="gmail_default"> == Using SIP RTP CoS mark 5</div><div class="gmail_default"> -- Executing [12023300643@default:1] Hangup("SIP/66.226.76.70-0000d0b1", "41") in new stack</div><div class="gmail_default"> == Spawn extension (default, 12023300643, 1) exited non-zero on 'SIP/66.226.76.70-0000d0b1'</div><div class="gmail_default"> == Using SIP RTP CoS mark 5</div><div class="gmail_default"> -- Executing [12023300643@default:1] Hangup("SIP/66.226.76.70-0000d0b2", "41") in new stack</div><div class="gmail_default"> == Spawn extension (default, 12023300643, 1) exited non-zero on 'SIP/66.226.76.70-0000d0b2'</div><div class="gmail_default"> == Using SIP RTP CoS mark 5</div><div class="gmail_default"> -- Executing [12023300643@default:1] Hangup("SIP/66.226.76.70-0000d0b3", "41") in new stack</div><div class="gmail_default"> == Spawn extension (default, 12023300643, 1) exited non-zero on 'SIP/66.226.76.70-0000d0b3'</div><div class="gmail_default"> == Using SIP RTP CoS mark 5</div><div class="gmail_default"> -- Executing [12023300643@default:1] Hangup("SIP/66.226.76.70-0000d0b4", "41") in new stack</div><div class="gmail_default"> == Spawn extension (default, 12023300643, 1) exited non-zero on 'SIP/66.226.76.70-0000d0b4'</div><div class="gmail_default"> == Using SIP RTP CoS mark 5</div><div class="gmail_default"> -- Executing [12023300643@default:1] Hangup("SIP/66.226.76.70-0000d0b5", "41") in new stack</div><div class="gmail_default"> == Spawn extension (default, 12023300643, 1) exited non-zero on 'SIP/66.226.76.70-0000d0b5'</div><div class="gmail_default"> == Using SIP RTP CoS mark 5</div><div class="gmail_default"> -- Executing [12023300643@default:1] Hangup("SIP/66.226.76.70-0000d0b6", "41") in new stack</div><div class="gmail_default"> == Spawn extension (default, 12023300643, 1) exited non-zero on 'SIP/66.226.76.70-0000d0b6'</div><div class="gmail_default"> == Using SIP RTP CoS mark 5</div><div class="gmail_default"> -- Executing [12023300643@default:1] Hangup("SIP/66.226.76.70-0000d0b7", "41") in new stack</div><div class="gmail_default"> == Spawn extension (default, 12023300643, 1) exited non-zero on 'SIP/66.226.76.70-0000d0b7'</div><div class="gmail_default"> == Using SIP RTP CoS mark 5</div><div class="gmail_default"> -- Executing [12023300643@default:1] Hangup("SIP/66.226.76.70-0000d0b8", "41") in new stack</div><div class="gmail_default"> == Spawn extension (default, 12023300643, 1) exited non-zero on 'SIP/66.226.76.70-0000d0b8'</div><div class="gmail_default"> == Using SIP RTP CoS mark 5</div><div class="gmail_default"> -- Executing [12023300643@default:1] Hangup("SIP/66.226.76.70-0000d0b9", "41") in new stack</div><div class="gmail_default"> == Spawn extension (default, 12023300643, 1) exited non-zero on 'SIP/66.226.76.70-0000d0b9'</div><div class="gmail_default"> == Using SIP RTP CoS mark 5</div><div class="gmail_default"> -- Executing [12023300643@default:1] Hangup("SIP/66.226.76.70-0000d0ba", "41") in new stack</div><div class="gmail_default"> == Spawn extension (default, 12023300643, 1) exited non-zero on 'SIP/66.226.76.70-0000d0ba'</div><div><br></div></div><div class="gmail_extra"><div><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div><br></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div>
<br><div class="gmail_quote">On Tue, Feb 14, 2017 at 6:03 AM, Joshua Colp <span dir="ltr"><<a href="mailto:jcolp@digium.com" target="_blank">jcolp@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="HOEnZb"><div class="h5">On Mon, Feb 13, 2017, at 08:58 PM, Anas Moiz wrote:<br>
> Yes Joshua, Its SIP and but the problem is I have tried everything but it<br>
> doesn't seem to work.<br>
><br>
> In the SIP Trace I can see that I am sending 503 Service Unavailable as a<br>
> response.<br>
><br>
> You can check the SIP trace attached below:<br>
><br>
> <a href="http://162.243.107.173:5060" rel="noreferrer" target="_blank">162.243.107.173:5060</a> -> <a href="http://66.226.76.70:5060" rel="noreferrer" target="_blank">66.226.76.70:5060</a><br>
> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP<br>
> 66.226.76.70:5060;branch=<br>
> z9hG4bK643.e44ea565.0;<wbr>received=66.226.76.70;rport=<wbr>5060 Via: SIP/2.0/UDP<br>
> 74.117.36.136;received=74.117.<wbr>36.136;rport=5060;branch=<wbr>z9hG4bKHBe9cmy3QX2Se<br>
> From: <<a href="http://sip:2126555763@66.226.76.70:5060" rel="noreferrer" target="_blank">sip:2126555763@66.226.76.70:<wbr>5060</a>>;tag=5H54caUKre8gc To: <<br>
> <a href="http://sip:12023300643@162.243.107.173:5060" rel="noreferrer" target="_blank">sip:12023300643@162.243.107.<wbr>173:5060</a>>;tag=as61c328a0 Call-ID:<br>
> 15-8824754a-f58560c9-335bcd48-<wbr>45558f71 CSeq: 103180201 INVITE Server:<br>
> user_Anas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,<br>
> NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length:<br>
> 0<br>
<br>
</div></div>You would need to determine what will stop the remote server from<br>
sending you the call again. Once you do that and can provide what it is<br>
then we can figure out how to get Asterisk to do that. As it is the<br>
problem isn't Asterisk, it is what is sending you the call.<br>
<div class="HOEnZb"><div class="h5"><br>
--<br>
Joshua Colp<br>
Digium, Inc. | Senior Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br>
Check us out at: <a href="http://www.digium.com" rel="noreferrer" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" rel="noreferrer" target="_blank">www.asterisk.org</a><br>
<br>
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