[Freeswitch-users] Newbie question about Cisco phones

[Steven Schoch]

Here are the important lines from the trace:  (I don't think I need to
paste the whole thing.)

recv 1521 bytes from tcp/[192.168.4.254]:4025 at 01:28:45.696774:
   ------------------------------------------------------------------------
   REGISTER sip:192.168.4.1:5060;transport=tcp SIP/2.0
   Via: SIP/2.0/TCP 192.168.4.254:4025
;rport;branch=z9hG4bKPjZiQxS2mheTBbTXalnSTcX-9-vSvvYydR
   Max-Forwards: 70
   From: "110" <sip:1001 at 192.168.4.1>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl
   To: "110" <sip:1001 at 192.168.4.1>
   Call-ID: 6Y7zlZiZJekTAYupYa4SU.v2.8vpto97
   CSeq: 4019 REGISTER
   User-Agent: Cisco-CP3905/9.2.1
[other stuff...]

   ------------------------------------------------------------------------
send 684 bytes to tcp/[192.168.4.254]:4025 at 01:28:45.856104:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/TCP 192.168.4.254:4025
;rport=4025;branch=z9hG4bKPjZiQxS2mheTBbTXalnSTcX-9-vSvvYydR
   From: "110" <sip:1001 at 192.168.4.1>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl
   To: "110" <sip:1001 at 192.168.4.1>;tag=UrQB9NZ278Hpc
   Call-ID: 6Y7zlZiZJekTAYupYa4SU.v2.8vpto97
[I don't think the other stuff is pertinent.]

And later:
   ------------------------------------------------------------------------
send 562 bytes to tcp/[192.168.4.254]:4025 at 01:28:46.032084:
   ------------------------------------------------------------------------
   SIP/2.0 403 Forbidden
   Via: SIP/2.0/TCP 192.168.4.254:4025
;rport=4025;branch=z9hG4bKPjbdyYmV-sH2X-.Cs-LZUG9P36AHF5dn4E
   From: "110" <sip:1001 at 192.168.4.1>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl
   To: "110" <sip:1001 at 192.168.4.1>;tag=v1g4aHg64H88Q
   Call-ID: 6Y7zlZiZJekTAYupYa4SU.v2.8vpto97
   CSeq: 4020 REGISTER


I have put a name/password in the Cisco config file, and the same
name/password in conf/directory/1001.xml.  Should this go in the
conf/sip_profiles/internal section instead?

-- 
Steve

On Thu, Dec 27, 2012 at 4:53 PM, Michael Collins <msc at freeswitch.org> wrote:

> Can you capture the SIP traffic between the phone and the server and drop
> it on pastebin.freeswitch.org? It might be good to review the SIP dialog,
> just in case there are some clues there. You can use the fs_cli if you wish:
> sofia profile internal siptrace on
> (if you're using the example configs that come with FreeSWITCH. Use the
> correct profile name if you have one other than "internal" that you're
> using.)
>
>
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