From:
freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Peder
Sent: Wednesday, January 12, 2011 11:12 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Possible Freeswitch bug with Cisco 7960
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