[Freeswitch-users] Hangup problem/SIP BYE lacking authentication

[Anthony Minessale]

See what they said in the challenge?

WWW-Authenticate: Digest
realm="SipLocal",nonce="3e952db60fb8",stale=false,algorithm=MD5,qop="auth"

Since this is a spontaneous challenge (which i think is somewhat silly since
it lets you talk on the phone for 40
minutes then makes you authenticate to hangup but *shrug*) FS does not know
which gateway to use for credentials.

The realm they sent was SipLocal so FS is looking in its configuration for a
gateway with that name.
The 2nd thing it tries is the host from the To: header (64.115.128.6).  if
there was a gateway with either of those names,
it would find it.

So try naming your gateway SipLocal or 64.115.128.6
or you can try setting the variable sip_use_gateway=<whatever> on the
channel which can give it a hint which
gateway to use.



On Thu, Oct 30, 2008 at 8:42 PM, Brian West <brian at freeswitch.org> wrote:

> They really shouldn't be challenging the bye like that because not every
> device can answer the challenge properly.  Either case it doesn't matter if
> you have a proper domain 206.57.23.143 in your user directory with the
> proper user and realm set to meet this challenge.   We have a setting in
> FreeSWITCH to auth all packets and when we did that we found out that some
> devices just do not work properly when doing that.  I need to see your
> original gateway XML.  You can email it off list if you like and let me try
> to solve this.
> /b
>
> On Oct 30, 2008, at 7:52 PM, Wellie Chao wrote:
>
> Here is the BYE from Freeswitch to Metaswitch:
>
> send 683 bytes to udp/[64.115.128.6]:5060 at 00:44:46.607025:
>
> ------------------------------------------------------------------------
>    BYE sip:9172388084 at 64.115.128.6:5060;transport=udp SIP/2.0
>    Via: SIP/2.0/UDP 216.57.23.143;rport;branch=z9hG4bKNa693jZ8SD54D
>    Max-Forwards: 70
>    From: <sip:2675379325 at 216.57.23.143>;tag=r4yBmtX3U0Hrr
>    To:
> <sip:9172388084 at 64.115.128.6:5060;transport=udp
> >;tag=Broadview1+1+25f76f+cc3ba534
>    Call-ID: CEB9027F at Broadview1
>    CSeq: 106588607 BYE
>    Contact: <sip:mod_sofia at 216.57.23.143:5060;transport=udp>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.1-9171
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: 100rel, timer, precondition, path, replaces
>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>    Content-Length: 0
>
>
> ------------------------------------------------------------------------
>
> Metaswitch is not so happy with the BYE message:
>
> recv 491 bytes from udp/[64.115.128.6]:5060 at 00:44:46.630445:
>
> ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP
> 216.57.23.143;received=216.57.23.143
> ;branch=z9hG4bKNa693jZ8SD54D;rport=5060
>    From: <sip:2675379325 at 216.57.23.143>;tag=r4yBmtX3U0Hrr
>    To:
> <sip:9172388084 at 64.115.128.6:5060;transport=udp
> >;tag=Broadview1+1+25f76f+cc3ba534
>    Call-ID: CEB9027F at Broadview1
>    CSeq: 106588607 BYE
>    WWW-Authenticate: Digest
> realm="SipLocal",nonce="3e952db60fb8",stale=false,algorithm=MD5,qop="auth"
>    Server: DC-SIP/2.0
>    Organization:
>    Supported: 100rel
>    Content-Length: 0
>
>
> ------------------------------------------------------------------------
>
> Right after receiving the 401 Unauthorized message from Metaswitch,
> Freeswitch emits the following error on the console:
>
> 2008-10-30 20:44:46 [ERR] sofia_reg.c:1089
> sofia_reg_handle_sip_r_challenge() No Matching gateway found
>
> At this point, the caller (the endpoint connected to Metaswitch) just
> remains on the line, never having received the BYE.
>
> Did you take a look at the packet traces I captured? The carrier
> gave me a packet trace for an Aastra PBX/softswitch, and it had the same
> interaction, but immediately upon receiving the 401 Unauthorized from
> Metaswitch, the Aastra machine then sent a second BYE, this time with
> authentication. Is there some way I can tell Freeswitch to do the same?
>
> Regards,
> Wellie
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20081031/55ce2cce/attachment-0002.html