[asterisk-users] Problem with Cisco Phones

[Jordan Cook - Gyron Networks]

> 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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