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. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Gary Chen Sent: Wednesday, January 12, 2011 8:51 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Possible Freeswitch bug with Cisco 7960 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@xxx.xxx.xxx.177>;tag=00082166efcb015d72e42433-5254ac8f To: <sip:9196@xxx.xxx.xxx.177;user=phone>;tag=p9H5Qy5r993mB Call-ID: 00082166-efcb0007-04fe7b08-3c126cb0@xxx.xxx.xxx.144 CSeq: 102 INVITE Contact: <sip:9196@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@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