I just checked a bit better and noticed this error while transfering: U 172.16.1.10:5060 -> 172.16.1.14:5060 NOTIFY sip:1...@172.16.0.24 SIP/2.0. Via: SIP/2.0/UDP 172.16.1.10:5060;branch=z9hG4bK23a1000e;rport. Route: <sip:172.16.1.14;lr=on>. From: "0624469780" <sip:0624469...@172.16.1.10>;tag=as47c203e8. To: <sip:0031851119...@172.16.1.14>;tag=2AE312D6-A13BBC6D. Contact: <sip:0624469...@172.16.1.10>. Call-ID: 05aedaab03eadeca3b42d0b84d880...@172.16.1.10. CSeq: 103 NOTIFY. User-Agent: Asterisk PBX. Max-Forwards: 70. Event: refer;id=2. Subscription-state: terminated;reason=noresource. Content-Type: message/sipfrag;version=2.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 49. . SIP/2.0 481 Call leg/transaction does not exist.
U 172.16.1.14:5060 -> 172.16.0.24:5060 NOTIFY sip:1...@172.16.0.24 SIP/2.0. Via: SIP/2.0/UDP 172.16.1.14;branch=z9hG4bK1df2.c4df24a7.0. Via: SIP/2.0/UDP 172.16.1.10:5060;received=172.16.1.10;branch=z9hG4bK23a1000e;rport=5060. From: "0624469780" <sip:0624469...@172.16.1.10>;tag=as47c203e8. To: <sip:0031851119...@172.16.1.14>;tag=2AE312D6-A13BBC6D. Contact: <sip:0624469...@172.16.1.10>. Call-ID: 05aedaab03eadeca3b42d0b84d880...@172.16.1.10. CSeq: 103 NOTIFY. User-Agent: Asterisk PBX. Max-Forwards: 69. Event: refer;id=2. Subscription-state: terminated;reason=noresource. Content-Type: message/sipfrag;version=2.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 49. . SIP/2.0 481 Call leg/transaction does not exist. That is the message that apears when pressing the transfer button. Iñaki Baz Castillo wrote: > > El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió: >> Well yes, it does work for the internal calls, but >> when a call comes in true asterisk to an opensips extention i CAN'T >> transfer it :-), i get transfer failed in my screen of my phone, and the >> call stays on the original called extention. This is only for announced >> transfers, unannounced works fine. >> >> Flavio post stated something about routing your REFER's back to asterisk, >> so it should work.. but i don't know how to route these calls back to >> the >> asterisk. > > Please, you *already* have the answer. When a phone is speaking with > Asterisk > (through OpenSIPS) you must route REFER to Asterisk as *any* other > in-dialog > request, this is, the *same* as when a phone is speaking with other phone > directly (through OpenSIPS). > > If the REFER fails this is because Asterisk is rejecting it !!! > > I already suggested you to do a SIP capture (using ngrep) to inspect which > error replies Asterisk when the REFER arrives to it. Please do it and > paste it > here (I expect a 403 or 404, so it means a wrong configuration in you > Asterisk, no more). > > And please, forget anything about exotic routing of the REFER. > > > -- > Iñaki Baz Castillo <i...@aliax.net> > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- View this message in context: http://n2.nabble.com/Transfer-issue-tp3877950p3898287.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users