> Next step is packet capture to see if there is a clue as to the cause of the > failure in the SIP signalling.
Right, I see the following when running SIP Debug. Looks to me like the phones are expecting the server to do the conference mixing, which I guess it would do in CallManager? <--- SIP read from TCP:xxx.xxx.xxx.xxx:50604 ---> REFER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/TCP xxx.xxx.xxx.xxx:50604;branch=z9hG4bK48c7492c From: "4005" <sip:[email protected]>;tag=203a07fceb4b00eff1377deb-da93e2ee To: <sip:[email protected]> Call-ID: [email protected] Max-Forwards: 70 Date: Tue, 20 Jan 2015 17:10:19 GMT CSeq: 101 REFER User-Agent: Cisco-CP7945G/9.4.2 Contact: <sip:[email protected]:50604;transport=tcp> Referred-By: "4005" <sip:[email protected]> Refer-To: cid:[email protected] Content-Length: 963 Content-Type: application/x-cisco-remotecc-request+xml Content-Disposition: session;handling=required Content-Id: <[email protected]> <?xml version="1.0" encoding="UTF-8"?> <x-cisco-remotecc-request> <softkeyeventmsg> <softkeyevent>Conference</softkeyevent> <dialogid> <callid>[email protected]</callid> <localtag>203a07fceb4b00ed3e4e2321-d9cb1581</localtag> <remotetag>as4a087ee2</remotetag> </dialogid> <linenumber>0</linenumber> <participantnum>0</participantnum> <consultdialogid> <callid>[email protected]</callid> <localtag>203a07fceb4b00ee46f74fd6-4ed3acbd</localtag> <remotetag>as18747c6d</remotetag> </consultdialogid> <state>false</state> <joindialogid> <callid></callid> <localtag></localtag> <remotetag></remotetag> </joindialogid> <eventdata> <invocationtype>explicit</invocationtype> </eventdata> <userdata></userdata> <softkeyid>0</softkeyid> <applicationid>0</applicationid> </softkeyeventmsg> </x-cisco-remotecc-request> <-------------> --- (16 headers 3 lines) --- Sending to xxx.xxx.xxx.xxx:50604 (no NAT) Call [email protected] got a SIP call transfer from caller: (REFER)! <--- Transmitting (no NAT) to xxx.xxx.xxx.xxx:50604 ---> SIP/2.0 603 Declined (No dialog) Via: SIP/2.0/TCP xxx.xxx.xxx.xxx:50604;branch=z9hG4bK48c7492c;received=xxx.xxx.xxx.xxx From: "4005" <sip:[email protected]>;tag=203a07fceb4b00eff1377deb-da93e2ee To: <sip:[email protected]>;tag=as141fffdd Call-ID: [email protected] CSeq: 101 REFER Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
