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</o:shapelayout></xml><![endif]--></head><body bgcolor=white lang=EN-GB link="#0563C1" vlink="#954F72"><div class=WordSection1><p class=MsoNormal><span style='color:windowtext'>I think I get it now thank you Bogdan.<o:p></o:p></span></p><p class=MsoNormal><span style='color:windowtext'>So I would forward the traffic using the opensips proxy, using the if (is_method(“refer”)) to the opensips box that would be the B2BUA? To bridge the call ?.<o:p></o:p></span></p><p class=MsoNormal><span style='color:windowtext'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:windowtext'>Also look forward to Opensips summit in may </span><span style='font-family:"Segoe UI Emoji",sans-serif;color:windowtext'>😊</span><span style='color:windowtext'> ill see you all there got it booked Saturday </span><span style='font-family:"Segoe UI Emoji",sans-serif;color:windowtext'>😊</span><span style='color:windowtext'><o:p></o:p></span></p><p class=MsoNormal><span style='color:windowtext'><o:p> </o:p></span></p><div><p class=MsoNormal><span style='font-size:12.0pt;mso-fareast-language:EN-GB'>Regards,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt;mso-fareast-language:EN-GB'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:16.0pt;mso-fareast-language:EN-GB'>Brian Southworth<o:p></o:p></span></p></div><p class=MsoNormal><span style='color:windowtext'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span lang=EN-US style='color:windowtext;mso-fareast-language:EN-GB'>From:</span></b><span lang=EN-US style='color:windowtext;mso-fareast-language:EN-GB'> Bogdan-Andrei Iancu [mailto:bogdan@opensips.org] <br><b>Sent:</b> 05 February 2018 15:47<br><b>To:</b> Brian Southworth <brian.southworth@clocom.uk>; OpenSIPS users mailling list <users@lists.opensips.org><br><b>Subject:</b> Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><tt><span style='font-size:10.0pt'>Hi Brian,</span></tt><span style='font-size:10.0pt;font-family:"Courier New"'><br><br><tt>Keep in mind that you cannot make opensips act in the same time as proxy (as required by the load balancer) and as a end-point (as required by the B2BUA). Ideally is to run the two services (LB and B2B) on two opensips instances in a chain.</tt><br><br><tt>Best regards,</tt><br><br></span><span style='mso-fareast-language:EN-GB'><o:p></o:p></span></p><pre>Bogdan-Andrei Iancu<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>OpenSIPS Founder and Developer<o:p></o:p></pre><pre> <a href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a><o:p></o:p></pre><pre>OpenSIPS Summit 2018<o:p></o:p></pre><pre> <a href="http://www.opensips.org/events/Summit-2018Amsterdam">http://www.opensips.org/events/Summit-2018Amsterdam</a><o:p></o:p></pre><div><p class=MsoNormal>On 02/02/2018 07:03 PM, Brian Southworth wrote:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal><span style='color:windowtext'>Sorry my apologies.</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>So from the beginning opensips acts as an authorization proxy which passes the call on to an asterisk box based on load (using load balancer).</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>I am trying to get the opensips proxy to handle call transfers and I thought the b2bua would be the best way. Initially the refer was sent to the asterisk box.</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>On inbound calls </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>The call comes in from the carrier goes to asterisk, asterisk then passes the sip invite to the proxy which then rings the sip phone.</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>What I wish to achieve is a way to transfer an inbound call to an internal extension or external number.</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>Example: </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>Caller A receives call </span><span style='font-family:Wingdings;color:windowtext'>à</span><span style='color:windowtext'> caller A places call on hold and dials caller B </span><span style='font-family:Wingdings;color:windowtext'>à</span><span style='color:windowtext'> caller B picks up </span><span style='font-family:Wingdings;color:windowtext'>à</span><span style='color:windowtext'> caller A presses cisco xfer and call is passed to caller B</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>I was hoping to achieve this using the proxy or asterisk box if possible.</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>I hope this helps.</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='font-size:12.0pt;mso-fareast-language:EN-GB'>Regards,</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:16.0pt;mso-fareast-language:EN-GB'> </span><o:p></o:p></p><p class=MsoNormal><span style='font-size:16.0pt;mso-fareast-language:EN-GB'>Brian Southworth</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><div><div style='border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span lang=EN-US style='color:windowtext;mso-fareast-language:EN-GB'>From:</span></b><span lang=EN-US style='color:windowtext;mso-fareast-language:EN-GB'> Bogdan-Andrei Iancu [<a href="mailto:bogdan@opensips.org">mailto:bogdan@opensips.org</a>] <br><b>Sent:</b> 02 February 2018 16:50<br><b>To:</b> Brian Southworth <a href="mailto:brian.southworth@clocom.uk"><brian.southworth@clocom.uk></a>; OpenSIPS users mailling list <a href="mailto:users@lists.opensips.org"><users@lists.opensips.org></a><br><b>Subject:</b> Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]</span><o:p></o:p></p></div></div><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal><tt><span style='font-size:10.0pt'>I'm a bit confused. The original report was on a record_route() / loose_route() matter. But you say you have opensips as B2B, so the RR mechanism must not be used in such a case - you act either as a end-point, either as a proxy - you cannot be both for the same call.</span></tt><span style='font-size:10.0pt;font-family:"Courier New ,serif",serif'><br><br></span><tt><span style='font-size:10.0pt'>Now you have this b2b error, during a call transfer scenario. and you mentioned LB also :)...so I'm a bit confused - could please try to put all these pieces together, so I can understand what you are doing ?</span></tt><span style='font-size:10.0pt;font-family:"Courier New ,serif",serif'><br><br></span><tt><span style='font-size:10.0pt'>Regards,</span></tt><span style='font-size:10.0pt;font-family:"Courier New ,serif",serif'><br><br><br></span><o:p></o:p></p><pre>Bogdan-Andrei Iancu<o:p></o:p></pre><pre> <o:p></o:p></pre><pre>OpenSIPS Founder and Developer<o:p></o:p></pre><pre> <a href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a><o:p></o:p></pre><pre>OpenSIPS Summit 2018<o:p></o:p></pre><pre> <a href="http://www.opensips.org/events/Summit-2018Amsterdam">http://www.opensips.org/events/Summit-2018Amsterdam</a><o:p></o:p></pre><div><p class=MsoNormal>On 02/02/2018 04:27 PM, Brian Southworth wrote:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal><span style='color:windowtext'>Maybe I am doing this wrong but I wanted the B2BUA module to handle the refer and bridge the calls. </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>I have the B2bUA working now. However my issue is that its not able to send the replies.</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>incoming reply</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>b2b_reply (B2B.222.7591351.1517580641)</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>Feb 2 14:10:47 [22664] ERROR:tm:_reply_light: failed to generate 408 reply when a final 200 was sent out</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>Feb 2 14:10:47 [22664] ERROR:b2b_entities:b2b_send_reply: failed to send reply with tm</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>Feb 2 14:10:47 [22664] ERROR:b2b_logic:b2b_logic_notify_reply: Sending reply failed - 408, [B2B.452.342.1517580641]</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>Do you need anything else to help me debug this ? I am not sure why its failing to pass the reply with tm, I have enabled the param:</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:10.0pt'>modparam("tm", "pass_provisional_replies", 1)</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>I should also note that I am using the load balancer module also. This normally deals with all call distribution. In and out.</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><div><p class=MsoNormal><span style='font-size:12.0pt;mso-fareast-language:EN-GB'>Regards,</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:16.0pt;mso-fareast-language:EN-GB'> </span><o:p></o:p></p><p class=MsoNormal><span style='font-size:16.0pt;mso-fareast-language:EN-GB'>Brian Southworth</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:16.0pt;mso-fareast-language:EN-GB'> </span><o:p></o:p></p></div><div><div style='border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span lang=EN-US style='color:windowtext;mso-fareast-language:EN-GB'>From:</span></b><span lang=EN-US style='color:windowtext;mso-fareast-language:EN-GB'> Bogdan-Andrei Iancu [<a href="mailto:bogdan@opensips.org">mailto:bogdan@opensips.org</a>] <br><b>Sent:</b> 02 February 2018 14:20<br><b>To:</b> Brian Southworth <a href="mailto:brian.southworth@clocom.uk"><brian.southworth@clocom.uk></a>; OpenSIPS users mailling list <a href="mailto:users@lists.opensips.org"><users@lists.opensips.org></a><br><b>Subject:</b> Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]</span><o:p></o:p></p></div></div><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal><tt><span style='font-size:10.0pt'>Hi Brian,</span></tt><span style='font-size:10.0pt;font-family:"Courier New ,serif",serif'><br><br></span><tt><span style='font-size:10.0pt'>Maybe that warning points to a routing error that prevents the REFER to be route to carrier - make a sip capture to be sure the REFER from A is properly routed and accepted by the carrier.</span></tt><span style='font-size:10.0pt;font-family:"Courier New ,serif",serif'><br><br></span><tt><span style='font-size:10.0pt'>Regards,</span></tt><span style='font-size:10.0pt;font-family:"Courier New ,serif",serif'><br><br><br><br></span><o:p></o:p></p><pre>Bogdan-Andrei Iancu<o:p></o:p></pre><pre> <o:p></o:p></pre><pre>OpenSIPS Founder and Developer<o:p></o:p></pre><pre> <a href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a><o:p></o:p></pre><pre>OpenSIPS Summit 2018<o:p></o:p></pre><pre> <a href="http://www.opensips.org/events/Summit-2018Amsterdam">http://www.opensips.org/events/Summit-2018Amsterdam</a><o:p></o:p></pre><div><p class=MsoNormal>On 02/02/2018 01:38 PM, Brian Southworth wrote:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal><span style='color:windowtext'>Hi Bogdan,</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>Thank you very much, so this doesn’t have any impact on why the call being transferred are dropped ?</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>I am trying to transfer a call using the refer method as that is what the cisco phones use.</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>The network is setup like so opensips proxy </span><span style='font-family:Wingdings;color:windowtext'>à</span><span style='color:windowtext'> asterisk gateway </span><span style='font-family:Wingdings;color:windowtext'>à</span><span style='color:windowtext'> carrier</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>Scenario:</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>Inbound call comes into the phone like so: carrier </span><span style='font-family:Wingdings;color:windowtext'>à</span><span style='color:windowtext'> ast </span><span style='font-family:Wingdings;color:windowtext'>à</span><span style='color:windowtext'> proxy </span><span style='font-family:Wingdings;color:windowtext'>à</span><span style='color:windowtext'> phone A</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>Phone A needs to transfer call to phone B: Phone A dials phone B </span><span style='font-family:Wingdings;color:windowtext'>à</span><span style='color:windowtext'> phone B picks up </span><span style='font-family:Wingdings;color:windowtext'>à</span><span style='color:windowtext'> phone A presses xfer button and call is dropped.</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'>Any help would be appreciated.</span><o:p></o:p></p><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><div><p class=MsoNormal><span style='font-size:12.0pt;mso-fareast-language:EN-GB'>Regards,</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:16.0pt;mso-fareast-language:EN-GB'> </span><o:p></o:p></p><p class=MsoNormal><span style='font-size:16.0pt;mso-fareast-language:EN-GB'>Brian Southworth</span><o:p></o:p></p></div><p class=MsoNormal><span style='color:windowtext'> </span><o:p></o:p></p><div><div style='border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span lang=EN-US style='color:windowtext;mso-fareast-language:EN-GB'>From:</span></b><span lang=EN-US style='color:windowtext;mso-fareast-language:EN-GB'> Bogdan-Andrei Iancu [<a href="mailto:bogdan@opensips.org">mailto:bogdan@opensips.org</a>] <br><b>Sent:</b> 02 February 2018 11:29<br><b>To:</b> OpenSIPS users mailling list <a href="mailto:users@lists.opensips.org"><users@lists.opensips.org></a>; Brian Southworth <a href="mailto:brian.southworth@clocom.uk"><brian.southworth@clocom.uk></a><br><b>Subject:</b> Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]</span><o:p></o:p></p></div></div><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal><tt><span style='font-size:10.0pt'>Hi Brian,</span></tt><span style='font-size:10.0pt;font-family:"Courier New ,serif",serif'><br><br></span><tt><span style='font-size:10.0pt'>That warning means OpenSIPS found a Route header (while doing loose_route) that is suppose to be of its own, but the network information from the header does not match any of the OpenSIPS SIP listeners.</span></tt><span style='font-size:10.0pt;font-family:"Courier New ,serif",serif'><br><br></span><tt><span style='font-size:10.0pt'>Best regards,</span></tt><span style='font-size:10.0pt;font-family:"Courier New ,serif",serif'><br><br><br><br><br></span><o:p></o:p></p><pre>Bogdan-Andrei Iancu<o:p></o:p></pre><pre> <o:p></o:p></pre><pre>OpenSIPS Founder and Developer<o:p></o:p></pre><pre> <a href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a><o:p></o:p></pre><pre>OpenSIPS Summit 2018<o:p></o:p></pre><pre> <a href="http://www.opensips.org/events/Summit-2018Amsterdam">http://www.opensips.org/events/Summit-2018Amsterdam</a><o:p></o:p></pre><div><p class=MsoNormal>On 02/02/2018 11:14 AM, Brian Southworth wrote:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal>I get this when trying to transfer calls using the B2BUA:<o:p></o:p></p><p class=MsoNormal>[15066] WARNING:rr:after_strict: no socket found to match RR [1][xxx.xxx.xxx.xxx:5060]<o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>When I try looking on the mailing list there are no other similar posts, could you please shed some light on what maybe causing this please.<o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal><span style='font-size:12.0pt;mso-fareast-language:EN-GB'>Regards,</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:16.0pt;mso-fareast-language:EN-GB'> </span><o:p></o:p></p><p class=MsoNormal><span style='font-size:16.0pt;mso-fareast-language:EN-GB'>Brian Southworth</span><o:p></o:p></p><p class=MsoNormal><span style='mso-fareast-language:EN-GB'><br><br><br><br><br><br></span><o:p></o:p></p><pre>_______________________________________________<o:p></o:p></pre><pre>Users mailing list<o:p></o:p></pre><pre><a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><o:p></o:p></pre><pre><a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><o:p></o:p></pre></blockquote><p class=MsoNormal><span style='mso-fareast-language:EN-GB'> </span><o:p></o:p></p></blockquote><p class=MsoNormal><span style='mso-fareast-language:EN-GB'> </span><o:p></o:p></p></blockquote><p class=MsoNormal><span style='mso-fareast-language:EN-GB'> </span><o:p></o:p></p></blockquote><p class=MsoNormal><span style='mso-fareast-language:EN-GB'><o:p> </o:p></span></p></div></body></html>