You can not do this. You can not have an INVITE that Asterisk originated enter back into Asterisk. Technically this is not a loop, but this is an INVITE glare and the way Asterisk is reacting is correct. <br><br>You'll need to change the Call-Id of the INVITE that goes into Asterisk (a proxy can not do that so you'll need a B2BUA), or else you can do something like what Olle suggested.
<br><br>Thanks,<br>Raj<br><br><br><div class="gmail_quote">On Dec 22, 2007 9:51 AM, Tomasz Zieleniewski <<a href="mailto:tzieleniewski@gmail.com">tzieleniewski@gmail.com</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi,<br><br>The message that asterisk receives is not a retransmission but this is the same message but it enters asterisk from other sip proxy which is not a loop.<br>The flow is the following<br><br>Asterisk SIP Proxy (Location Service)
<br>INVITE (to registrar)<br>---------------------><br>INVITE (to voicemail when not registered) <br><--------------------<br><br>when message enters asterisk for the second time it ofcorse has some extra SIP <br>specific header like Record-Route and Via and the Request-URI is changed.
<br>And this causes 491 response.<br>Can I do something about this?<br>Can this behaviour be controlled, what do I have to change in the message so that asterisk won't treat it with 491 response?<br><br>Thanks<br>Tomasz
<br><br><div class="gmail_quote">On Dec 21, 2007 7:28 PM, Terry Wilson <<a href="mailto:twilson@digium.com" target="_blank">twilson@digium.com</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div><div><div><blockquote type="cite">What is the reason for such response?</blockquote><div><br></div></div><div><div><blockquote type="cite">SIP/2.0 491 Request Pending<br>Via: SIP/2.0/UDP
<a href="http://192.168.129.74:5160" target="_blank">192.168.129.74:5160</a>;branch=z9hG4bK17c3.17db29e7.0;received=<a href="http://192.168.129.74" target="_blank">192.168.129.74</a><br>Via: SIP/2.0/UDP <a href="http://192.168.129.74" target="_blank">
192.168.129.74</a> ;branch=z9hG4bK17c3.23083974.0<br>Via: SIP/2.0/UDP <a href="http://192.168.129.74:5070" target="_blank">192.168.129.74:5070</a>;branch=z9hG4bK5b33ae78;rport=5070<br>From: "IPFon" <<a>sip:0225761853@192.168.129.74:5070
</a>>;tag=as7217acbc <br>To: <<a href="mailto:sip:tzl@voip.touk.pl" target="_blank">sip:tzl@voip.touk.pl</a>>;tag=as7217acbc<br>Call-ID: <a href="mailto:307fda656066a7e264e85cea0742e601@192.168.129.74" target="_blank">
307fda656066a7e264e85cea0742e601@192.168.129.74</a> <br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Content-Length: 0<br>X-Asterisk-HangupCause: Normal Clearing
<br>X-Asterisk-HangupCauseCode: 16 <br></blockquote></div><div><br></div></div><div>Asterisk will send a 491 Request Pending when it is currently processing an INIVTE on a particular call and it gets another INVITE that isn't a retransmission.
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