<p class="MsoNormal"><span style="font-size:10.0pt;font-family:Arial;color:navy"><o:p>Sorry my mistake. It is Version 8.9 on Cisco 7960 ( POS3-08-9-00).</o:p></span></p><p class="MsoNormal"><span style="font-size:10.0pt;font-family:Arial;color:navy"><o:p>Gary</o:p></span></p><p class="MsoNormal"><span style="font-size:10.0pt;font-family:Arial;color:navy"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:Tahoma">From:</span></b><span style="font-size:10.0pt;font-family:Tahoma">
freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>On Behalf Of </b>Peder<br>
<b>Sent:</b> Wednesday, January 12, 2011 11:12 AM<br>
<b>To:</b> 'FreeSWITCH Users Help'<br>
<b>Subject:</b> Re: [Freeswitch-users] Possible Freeswitch bug with Cisco 7960</span><o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:Calibri;
color:#1F497D">I’d say it is much more likely a Cisco 7960 bug. Cisco’s
SIP software is total crap. It’s too bad because their phones are
physically very nice and very reliable. I’d verify what release you are
on as a search of Cisco’s site doesn’t show a 7.9. It shows up to
7.5. 8 goes all the way to 8.9. In * land, we never went past 7.4
as anything after that had issues.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:Calibri;
color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:Tahoma">From:</span></b><span style="font-size:10.0pt;font-family:Tahoma"> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>
<a href="mailto:%5bmailto:freeswitch-users-bounces@lists.freeswitch.org%5d">[mailto:freeswitch-users-bounces@lists.freeswitch.org]</a>
<b>On Behalf Of </b>Gary Chen<br>
<b>Sent:</b> Wednesday, January 12, 2011 8:51 AM<br>
<b>To:</b> <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
<b>Subject:</b> [Freeswitch-users] Possible Freeswitch bug with Cisco 7960<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">FreeSWITCH Version 1.0.head (git-9350fb9 2010-12-07 00-20-07
-0600)<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I just install freeswitch and have two cisco 7960 phones
(firmware sip 7.9) registered.<o:p></o:p></p>
<p class="MsoNormal">I can call out to our service provider or make 7960 call
each other. They all work fine except this:<o:p></o:p></p>
<p class="MsoNormal">If I call into freeswitch echo test extension or check
voicemail using one of Cisco 7960, the call will be droped after 30
seconds. By looking at SIP messages, it appears that After FS send out 200 OK
to Cisco 7960, it never received ACK sip messages back and timeout. Here is the
debug info in Cisco 7960:<o:p></o:p></p>
<p class="MsoNormal">[11:56:30:41354610] SIP/2.0 200 OK<o:p></o:p></p>
<p class="MsoNormal">Via: SIP/2.0/UDP xxx.xxx.xxx.144:5060;branch=z9hG4bK2be0821e<o:p></o:p></p>
<p class="MsoNormal">From: "Line1" <<a href="sip:1007@xxx.xxx.xxx.177">sip:1007@xxx.xxx.xxx.177</a>>;tag=00082166efcb015d72e42433-5254ac8f<o:p></o:p></p>
<p class="MsoNormal">To: <<a href="sip:9196@xxx.xxx.xxx.177;user=phone">sip:9196@xxx.xxx.xxx.177;user=phone</a>>;tag=p9H5Qy5r993mB<o:p></o:p></p>
<p class="MsoNormal">Call-ID: <a href="mailto:00082166-efcb0007-04fe7b08-3c126cb0@xxx.xxx.xxx.144">00082166-efcb0007-04fe7b08-3c126cb0@xxx.xxx.xxx.144</a><o:p></o:p></p>
<p class="MsoNormal">CSeq: 102 INVITE<o:p></o:p></p>
<p class="MsoNormal">Contact: <<a href="sip:9196@xxx.xxx.xxx.177:5060;transport=udp">sip:9196@xxx.xxx.xxx.177:5060;transport=udp</a>><o:p></o:p></p>
<p class="MsoNormal">User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9350fb9
2010-12-07 00-20-07 -0600<o:p></o:p></p>
<p class="MsoNormal">Accept: application/sdp<o:p></o:p></p>
<p class="MsoNormal">Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE<o:p></o:p></p>
<p class="MsoNormal">Supported: timer, precondition, path, replaces<o:p></o:p></p>
<p class="MsoNormal">Allow-Events: talk, hold, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo, message-summary,
refer<o:p></o:p></p>
<p class="MsoNormal">Content-Type: application/sdp<o:p></o:p></p>
<p class="MsoNormal">Content-Disposition: session<o:p></o:p></p>
<p class="MsoNormal">Content-Length: 253<o:p></o:p></p>
<p class="MsoNormal">Remote-Party-ID: "9196" <<a href="sip:9196@xxx.xxx.xxx.177">sip:9196@xxx.xxx.xxx.177</a>>;party=calling;privacy=off;screen=no<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">v=0<o:p></o:p></p>
<p class="MsoNormal">o=FreeSWITCH 1294739962 1294739963 IN IP4 xxx.xxx.xxx.177<o:p></o:p></p>
<p class="MsoNormal">s=FreeSWITCH<o:p></o:p></p>
<p class="MsoNormal">c=IN IP4 xxx.xxx.xxx.177<o:p></o:p></p>
<p class="MsoNormal">t=0 0<o:p></o:p></p>
<p class="MsoNormal">m=audio 24992 RTP/AVP 0 101<o:p></o:p></p>
<p class="MsoNormal">a=rtpmap:0 PCMU/8000<o:p></o:p></p>
<p class="MsoNormal">a=rtpmap:101 telephone-event/8000<o:p></o:p></p>
<p class="MsoNormal">a=fmtp:101 0-16<o:p></o:p></p>
<p class="MsoNormal">a=silenceSupp:off - - - -<o:p></o:p></p>
<p class="MsoNormal">a=ptime:20<o:p></o:p></p>
<p class="MsoNormal">[11:56:30:41354614] SIPTaskProcessSIPMessage: Line filter:
Determining destination line...<o:p></o:p></p>
<p class="MsoNormal">[11:56:30:41354615] sip_sm_determine_ccb: Matched to_tag<o:p></o:p></p>
<p class="MsoNormal">[11:56:30:41354616] sip_sm_ccb_match_branch_cseq: Method
index not found<o:p></o:p></p>
<p class="MsoNormal">[11:56:30:41354616] SIPTaskProcessSIPMessage: Error:
sip_sm_determine_ccb(): bad response. Dropping message.<o:p></o:p></p>
<p class="MsoNormal">[11:56:34:41355009] SIPTaskProcessListEvent: cmd = 0x160200<o:p></o:p></p>
<p class="MsoNormal">[11:56:34:41355009] SIPProcessUDPMessage: recv UDP message
from <xxx.xxx.xxx.177>:<50195>, length=<1185>, message=<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Looks like cisco 7960 having trouble to process 200 OK
message from FS.<o:p></o:p></p>
<p class="MsoNormal">Is this a possible FS bug? Does anybody know how to fix it?<o:p></o:p></p>
<p class="MsoNormal">By searching on the WEB, I can see that Asterisk 1.6 also
has the same problem in some revisions and been fixed.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Gary<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>