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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><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>I have the Transfer working when using the chan_sip support.<o:p></o:p></p><p class=MsoNormal>However, when I try to perform a Transfer using pjsip, it is failing.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>The one difference I am seeing in the SIP trace is chan_sip automatically sends the Referred-By. PJSIP does not.<o:p></o:p></p><p class=MsoNormal>The switch provider I am working with has never seen a REFER without the “Referred-By” line<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>In both cases, I am performing the Transfer via AMI <o:p></o:p></p><p class=MsoNormal>EXEC Transfer ….<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Does Asterisk 13.5.0 PJSIP support require a flag or something to force the Referred-By line to automatically be passed when a Transfer is performed?<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 sip:3400@192.168.yyy.yyy:5060 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: <sip:3344@192.168.xxx.xxx>;tag=as44000cf4<o:p></o:p></p><p class=MsoNormal> To: <sip:3400@192.168.yyy.yyy>;tag=7Iy0JkwDC<o:p></o:p></p><p class=MsoNormal> Contact: <sip:3344@192.168.xxx.xxx:5060><o:p></o:p></p><p class=MsoNormal> Call-ID: jdEuqpAK-0002-@192.168.yyy.yyy<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: <sip:370@192.168.yyy.yyy><o:p></o:p></p><p class=MsoNormal> Referred-By: <sip:3344@192.168.xxx.xxx:5060><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 sip:3400@192.168.yyy.yyy:5060 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: <sip:3344@192.168.xxx.xxx>;tag=3c10f423-e468-42ea-87a1-658ae106581c<o:p></o:p></p><p class=MsoNormal> To: <sip:3400@192.168.yyy.yyy>;tag=WITKDakt<o:p></o:p></p><p class=MsoNormal> Contact: <sip:192.168.xxx.xxx:5060><o:p></o:p></p><p class=MsoNormal> Call-ID: s6Wk6l6Q-0001-@192.168.yyy.yyy<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: <sip:370@192.168.yyy.yyy><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></div></body></html>