Hi, Seeing the firmware version in the User Agent header this phone is probably build on the same platform as the 79XX series phones and having the same sip-stack. From experience I can tell you that sip-stack is really really horrible. I don’t know if you have bought a lot of these phone or just testing one. If you’re just testing one you’re better of with the Cisco SPA500 series phones instead of these ones. Especially if you’re new to ip telephony, these phones will give you bad headaches. Erik [cid:imagedf2f15.PNG@6f1c3921.4486eef7] ABC Westland 555 Tel: +31 174 22 50 80 P.O. Box 90 Fax: +31 174 22 50 81 Mob: +31 624 423 009 2685 ZH Poeldijk erik.dekkers@certhon.com<mailto:erik.dekkers@certhon.com> The Netherlands www.certhon.com<http://www.certhon.com> DISCLAIMER All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing. ISO9001 CERTIFIED Van: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Namens Steven Ayre Verzonden: vrijdag 28 december 2012 11:14 Aan: FreeSWITCH Users Help Onderwerp: Re: [Freeswitch-users] Newbie question about Cisco phones This is incomplete. The normal flow is -> REGISTER <- 401 -> REGISTER <- 200 or 403 If you look at the CSeq in the 403 you'll see its a reply to the register after the initial REGISTER. 401 contains a challenge that's used to generate the auth data to put in the 2nd register - that allows digest authentication which avoids sending the password in plaintext and uses a nonce to prevent replay attacks (an attacker can't capture the register and resend it later to auth themselves). The digest includes the domain, if the user and password match perhaps the problem lies there... Sent from my iPad On 28 Dec 2012, at 01:38, Steven Schoch <schoch+freeswitch.org@xwin32.com<mailto:schoch+freeswitch.org@xwin32.com>> wrote: 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@192.168.4.1<mailto:sip%3A1001@192.168.4.1>>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl To: "110" <sip:1001@192.168.4.1<mailto:sip%3A1001@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@192.168.4.1<mailto:sip%3A1001@192.168.4.1>>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl To: "110" <sip:1001@192.168.4.1<mailto:sip%3A1001@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@192.168.4.1<mailto:sip%3A1001@192.168.4.1>>;tag=SoDYLlxdRfdsHMTYaCd6LprFvMe6ROpl To: "110" <sip:1001@192.168.4.1<mailto:sip%3A1001@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@freeswitch.org<mailto:msc@freeswitch.org>> wrote: Can you capture the SIP traffic between the phone and the server and drop it on pastebin.freeswitch.org<http://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.) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org
