Jordan Cook - Gyron Networks
2015-Jan-20 17:16 UTC
[asterisk-users] Problem with Cisco Phones
> 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:4005 at xxx.xxx.xxx.xxx>;tag=203a07fceb4b00eff1377deb-da93e2ee To: <sip:4004 at xxx.xxx.xxx.xxx> Call-ID: OutOfDialog--001e-67a906f5-5333c2b8 at xxx.xxx.xxx.xxx Max-Forwards: 70 Date: Tue, 20 Jan 2015 17:10:19 GMT CSeq: 101 REFER User-Agent: Cisco-CP7945G/9.4.2 Contact: <sip:4005 at xxx.xxx.xxx.xxx:50604;transport=tcp> Referred-By: "4005" <sip:4005 at xxx.xxx.xxx.xxx> Refer-To: cid:9a2a9191 at xxx.xxx.xxx.xxx Content-Length: 963 Content-Type: application/x-cisco-remotecc-request+xml Content-Disposition: session;handling=required Content-Id: <9a2a9191 at xxx.xxx.xxx.xxx> <?xml version="1.0" encoding="UTF-8"?> <x-cisco-remotecc-request> <softkeyeventmsg> <softkeyevent>Conference</softkeyevent> <dialogid> <callid>203a07fc-eb4b001c-1bf7ad61-614d38c1 at xxx.xxx.xxx.xxx</callid> <localtag>203a07fceb4b00ed3e4e2321-d9cb1581</localtag> <remotetag>as4a087ee2</remotetag> </dialogid> <linenumber>0</linenumber> <participantnum>0</participantnum> <consultdialogid> <callid>203a07fc-eb4b001d-14750420-d3d10a57 at xxx.xxx.xxx.xxx</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 OutOfDialog--001e-67a906f5-5333c2b8 at xxx.xxx.xxx.xxx 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:4005 at xxx.xxx.xxx.xxx>;tag=203a07fceb4b00eff1377deb-da93e2ee To: <sip:4004 at xxx.xxx.xxx.xxx>;tag=as141fffdd Call-ID: OutOfDialog--001e-67a906f5-5333c2b8 at xxx.xxx.xxx.xxx 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.
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 < jordan.cook at gyron.net> 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:4005 at xxx.xxx.xxx.xxx > >;tag=203a07fceb4b00eff1377deb-da93e2ee > To: <sip:4004 at xxx.xxx.xxx.xxx> > Call-ID: OutOfDialog--001e-67a906f5-5333c2b8 at xxx.xxx.xxx.xxx > Max-Forwards: 70 > Date: Tue, 20 Jan 2015 17:10:19 GMT > CSeq: 101 REFER > User-Agent: Cisco-CP7945G/9.4.2 > Contact: <sip:4005 at xxx.xxx.xxx.xxx:50604;transport=tcp> > Referred-By: "4005" <sip:4005 at xxx.xxx.xxx.xxx> > Refer-To: cid:9a2a9191 at xxx.xxx.xxx.xxx > Content-Length: 963 > Content-Type: application/x-cisco-remotecc-request+xml > Content-Disposition: session;handling=required > Content-Id: <9a2a9191 at xxx.xxx.xxx.xxx> > > <?xml version="1.0" encoding="UTF-8"?> > <x-cisco-remotecc-request> <softkeyeventmsg> > <softkeyevent>Conference</softkeyevent> <dialogid> > <callid>203a07fc-eb4b001c-1bf7ad61-614d38c1 at xxx.xxx.xxx.xxx</callid> > <localtag>203a07fceb4b00ed3e4e2321-d9cb1581</localtag> > <remotetag>as4a087ee2</remotetag> </dialogid> <linenumber>0</linenumber> > <participantnum>0</participantnum> <consultdialogid> > <callid>203a07fc-eb4b001d-14750420-d3d10a57 at xxx.xxx.xxx.xxx</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 OutOfDialog--001e-67a906f5-5333c2b8 at xxx.xxx.xxx.xxx 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:4005 at xxx.xxx.xxx.xxx > >;tag=203a07fceb4b00eff1377deb-da93e2ee > To: <sip:4004 at xxx.xxx.xxx.xxx>;tag=as141fffdd > Call-ID: OutOfDialog--001e-67a906f5-5333c2b8 at xxx.xxx.xxx.xxx > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150120/89439baa/attachment.html>
Jordan Cook - Gyron Networks
2015-Jan-22 10:31 UTC
[asterisk-users] Problem with Cisco Phones
> Apparently this is a known problem past v8 firmware: > http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update- > version-9/I've done some more playing about and what I've noticed is that even when using TCP SIP on the 8.x Firmware conferencing doesn?t work - making it use UDP fixes this. So has anyone managed to get the 9.x firmware working with UDP? Possibly worth a try to see if this resolves the issue? 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.