[Freeswitch-users] Possible Freeswitch bug with Cisco 7960

[Gary Chen]

FreeSWITCH Version 1.0.head (git-9350fb9 2010-12-07 00-20-07 -0600)

I just install freeswitch and have two cisco 7960 phones (firmware sip 7.9) registered.
I can call out to our service provider or make 7960 call each other. They all work fine except this:
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:
[11:56:30:41354610] SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.144:5060;branch=z9hG4bK2be0821e
From: "Line1" <sip:1007 at xxx.xxx.xxx.177>;tag=00082166efcb015d72e42433-5254ac8f
To: <sip:9196 at xxx.xxx.xxx.177;user=phone>;tag=p9H5Qy5r993mB
Call-ID: 00082166-efcb0007-04fe7b08-3c126cb0 at xxx.xxx.xxx.144
CSeq: 102 INVITE
Contact: <sip:9196 at xxx.xxx.xxx.177:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9350fb9 2010-12-07 00-20-07 -0600
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 253
Remote-Party-ID: "9196" <sip:9196 at xxx.xxx.xxx.177>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1294739962 1294739963 IN IP4 xxx.xxx.xxx.177
s=FreeSWITCH
c=IN IP4 xxx.xxx.xxx.177
t=0 0
m=audio 24992 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
[11:56:30:41354614] SIPTaskProcessSIPMessage: Line filter: Determining destination line...
[11:56:30:41354615] sip_sm_determine_ccb: Matched to_tag
[11:56:30:41354616] sip_sm_ccb_match_branch_cseq: Method index not found
[11:56:30:41354616] SIPTaskProcessSIPMessage: Error: sip_sm_determine_ccb(): bad response. Dropping message.
[11:56:34:41355009] SIPTaskProcessListEvent: cmd = 0x160200
[11:56:34:41355009] SIPProcessUDPMessage: recv UDP message from <xxx.xxx.xxx.177>:<50195>, length=<1185>, message=


Looks like cisco 7960 having trouble to process 200 OK message from FS.
Is this a possible FS bug? Does anybody know how to fix it?
By searching on the WEB, I can see that Asterisk 1.6 also has the same problem in some revisions and been fixed.

Gary


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