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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal>I asked the question on asterisk–users but did not receive a response, so I am sending the question here.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I am running Asterisk 13.5.0.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>A call comes in, Asterisk answers it. After some actions, the call needs to be Transferred (SIP REFER) to another number. The other switch is responsible for accepting the Transfer and tromboning the lines internally. It will also send a BYE so Asterisk no longer has the call.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>The behavior works when I have the endpoint configured at chan_sip. It does not work when the endpoint is configured as PJSIP. I worked with the other switch vendor and he determined chan_sip includes the Referred-By header. PJSIP does not include the Referred-By header. The other switch requires the Referred-By header to be present.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I tried setting the channel’s SIPREFERREDBYHDR variable before the Transfer command and that still did not force the Referred-By header to be part of the REFER packet.<o:p></o:p></p><p class=MsoNormal>I tried the PJSIP_HEADER add and it still did not add the Referred-By header to the REFER packet.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Is there a PJSIP setting to force the Referred-By to be part of the REFER packet?<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>chan_sip (succeeds)<o:p></o:p></p><p class=MsoNormal>19:27:32.512123 IP (tos 0x0, ttl 64, id 11492, offset 0, flags [none], proto UDP (17), length 630)<o:p></o:p></p><p class=MsoNormal> 192.168.xxx.xxx.sip > 192.168.yyy.yyy.sip: SIP, length: 602<o:p></o:p></p><p class=MsoNormal> REFER <a href="sip:3400@192.168.yyy.yyy:5060">sip:3400@192.168.yyy.yyy:5060</a> SIP/2.0<o:p></o:p></p><p class=MsoNormal> Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;branch=z9hG4bK58f4bd1d<o:p></o:p></p><p class=MsoNormal> Max-Forwards: 70<o:p></o:p></p><p class=MsoNormal> From: <<a href="sip:3344@192.168.xxx.xxx">sip:3344@192.168.xxx.xxx</a>>;tag=as44000cf4<o:p></o:p></p><p class=MsoNormal> To: <<a href="sip:3400@192.168.yyy.yyy">sip:3400@192.168.yyy.yyy</a>>;tag=7Iy0JkwDC<o:p></o:p></p><p class=MsoNormal> Contact: <<a href="sip:3344@192.168.xxx.xxx:5060">sip:3344@192.168.xxx.xxx:5060</a>><o:p></o:p></p><p class=MsoNormal> Call-ID: <a href="mailto:jdEuqpAK-0002-@192.168.yyy.yyy">jdEuqpAK-0002-@192.168.yyy.yyy</a><o:p></o:p></p><p class=MsoNormal> CSeq: 102 REFER<o:p></o:p></p><p class=MsoNormal> User-Agent: Asterisk PBX 13.5.0<o:p></o:p></p><p class=MsoNormal> Date: Thu, 20 Aug 2015 19:27:32 GMT<o:p></o:p></p><p class=MsoNormal> Refer-To: <<a href="sip:370@192.168.yyy.yyy">sip:370@192.168.yyy.yyy</a>><o:p></o:p></p><p class=MsoNormal> Referred-By: <<a href="sip:3344@192.168.xxx.xxx:5060">sip:3344@192.168.xxx.xxx:5060</a>><o:p></o:p></p><p class=MsoNormal> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE<o:p></o:p></p><p class=MsoNormal> Supported: replaces, timer<o:p></o:p></p><p class=MsoNormal> Content-Length: 0<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Pjsip<o:p></o:p></p><p class=MsoNormal>18:46:58.386372 IP (tos 0x0, ttl 64, id 38690, offset 0, flags [DF], proto UDP (17), length 654)<o:p></o:p></p><p class=MsoNormal> 192.168.xxx.xxx.sip > 192.168.yyy.yyy.sip: SIP, length: 626<o:p></o:p></p><p class=MsoNormal> REFER <a href="sip:3400@192.168.yyy.yyy:5060">sip:3400@192.168.yyy.yyy:5060</a> SIP/2.0<o:p></o:p></p><p class=MsoNormal> Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPjec41c3b9-d734-482d-82c1-2a6f8d9452a3<o:p></o:p></p><p class=MsoNormal> From: <<a href="sip:3344@192.168.xxx.xxx">sip:3344@192.168.xxx.xxx</a>>;tag=3c10f423-e468-42ea-87a1-658ae106581c<o:p></o:p></p><p class=MsoNormal> To: <<a href="sip:3400@192.168.yyy.yyy">sip:3400@192.168.yyy.yyy</a>>;tag=WITKDakt<o:p></o:p></p><p class=MsoNormal> Contact: <<a href="sip:192.168.xxx.xxx:5060">sip:192.168.xxx.xxx:5060</a>><o:p></o:p></p><p class=MsoNormal> Call-ID: <a href="mailto:s6Wk6l6Q-0001-@192.168.yyy.yyy">s6Wk6l6Q-0001-@192.168.yyy.yyy</a><o:p></o:p></p><p class=MsoNormal> CSeq: 981 REFER<o:p></o:p></p><p class=MsoNormal> Event: refer<o:p></o:p></p><p class=MsoNormal> Expires: 600<o:p></o:p></p><p class=MsoNormal> Supported: 100rel, timer, replaces, norefersub<o:p></o:p></p><p class=MsoNormal> Accept: message/sipfrag;version=2.0<o:p></o:p></p><p class=MsoNormal> Allow-Events: message-summary, presence, dialog, refer<o:p></o:p></p><p class=MsoNormal> Refer-To: <<a href="sip:370@192.168.yyy.yyy">sip:370@192.168.yyy.yyy</a>><o:p></o:p></p><p class=MsoNormal> Max-Forwards: 70<o:p></o:p></p><p class=MsoNormal> User-Agent: Asterisk PBX 13.5.0<o:p></o:p></p><p class=MsoNormal> Content-Length: 0<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>One other slight oddity.<o:p></o:p></p><p class=MsoNormal>To get chan_sip to Transfer<o:p></o:p></p><p class=MsoNormal><span style='color:#1F497D'><a href="mailto:370@192.168.yyy.yyy">370@192.168.yyy.yyy</a><o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>To get PJSIP to Transfer with the correct Refer-To header, I had to include the <> and sip:<o:p></o:p></span></p><p class=MsoNormal><span style='color:#4F81BD'><<u><a href="sip:370@192.168.yyy.">sip:370@192.168.yyy.</a></u></span><u><span style='color:#1F497D'>yyy</span></u><span style='color:#4F81BD'>><o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Have a great day!<o:p></o:p></p><p class=MsoNormal>Dan<o:p></o:p></p></div></body></html>