[asterisk-users] Problem with Cisco Phones

Scott Griepentrog sgriepentrog at digium.com
Tue Jan 20 11:26:38 CST 2015


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
>
>
> <------------>
>
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Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
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