[Freeswitch-users] Possible Freeswitch bug with Cisco 7960

[Peder]

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 at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gary
Chen
Sent: Wednesday, January 12, 2011 8:51 AM
To: freeswitch-users at 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 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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