Apparently this is a known problem past v8 firmware: http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update-version-9/
On Tue, Jan 20, 2015 at 11:16 AM, Jordan Cook - Gyron Networks < [email protected]> wrote: > > 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 > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org
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