[Freeswitch-users] Hangup problem/SIP BYE lacking authentication
anthony.minessale at gmail.com
Thu Oct 30 14:10:58 PDT 2008
just setup a gateway in fs that has reg=false and the proper credentials to
pass the challenge.
On Thu, Oct 30, 2008 at 2:49 PM, Wellie Chao <wchao at yahoo.com> wrote:
> Hangups do not work for me under certain circumstances. Here is the
> background information:
> * Our carrier uses a Metaswitch server with Acme Packet in front as a
> proxy/SBC. Only the Acme Packet machine is publicly visible
> * Our Freeswitch server is at 220.127.116.11.
> * For calls originating from Freeswitch and terminating on Metaswitch
> (18.104.22.168 -> 22.214.171.124), Freeswitch authenticates with
> Metaswitch and everything works hunky-dorey. Either side can hang up and
> the other side will automatically hang up without requiring a manual
> Now the problem:
> For calls originating from Metaswitch to Freeswitch (126.96.36.199 ->
> 188.8.131.52), Metaswitch does not authenticate with Freeswitch.
> Metaswitch also does not use the existing authenticated registration that
> our Freeswitch server initiates with Metaswitch upon startup of
> Freeswitch. Metaswitch just begins a new (unauthenticated) session and we
> have configured Freeswitch to allow any inbound calls from 184.108.40.206
> without requiring authentication.
> We receive inbound calls (Metaswitch to Freeswitch, 220.127.116.11 ->
> 18.104.22.168) just fine. The phone rings and we can have a normal
> conversation. If the caller (the endpoint attached to Metaswitch) hangs
> up, both sides hang up. If I hang up (remember, I'm at the endpoint
> attached to Freeswitch), the caller's line remains attached forever.
> I have recorded a packet trace. Look at freeswitch_2.cap in the ZIP file,
> and you want to graph the first call starting at 21.202 and ending 53.798.
> If you go to time 53.087, you can see that my Freeswitch server sends a
> BYE to Metaswitch. This is a result of me hanging up my phone. At time
> 53.089, you see Metaswitch responding with 401 Unauthorized. Later at time
> 53.777, you see a BYE from Metaswitch to Freeswitch, but you should ignore
> this because that was a result of the caller (the guy hooked up to
> Metaswitch) manually hanging up. If he had not hung up his phone, the BYE
> from Metaswitch to Freeswitch would not have been issued and his phone
> would just stay on the line forever. Also, when I hang up my phone, I see
> the following at the Freeswitch console:
> 2008-10-29 23:03:28 [ERR] sofia_reg.c:1089
> sofia_reg_handle_sip_r_challenge() No Matching gateway found
> I presume that Freeswitch emits this error because it got the 401
> Unauthorized from Metaswitch.
> I also asked our carrier for a packet trace of a successful hangup on the
> Aastra platform (the engineer at the carrier says it is an Asterisk
> derivative -- I'm not sure about that). Look at
> Aastra_authentication_test.cap in the ZIP file. Graph the first call
> starting at 43.633 and ending 93.156. If you go to 93.118, you'll see that
> the Aastra server sends a BYE. Just like our Freeswitch scenario,
> Metaswitch sends back a 401 Unauthorized, but in response to the 401
> Unauthorized, Aastra then sends back another BYE with the difference that
> the second BYE is authenticated. Metaswitch gets the second BYE and
> responds with 200 OK.
> I am pretty sure that if Freeswitch were to send back a second BYE (but
> with authentication), it would work fine. Now my question is how can I do
> this? I am not sure if this divergence of behavior is caused by: (a) my
> own error in configuring Freeswitch, (b) Metaswitch lacking standard SIP
> support (maybe it's not supposed to send the 401 Unauthorized), or (c)
> Freeswitch lacking standard SIP support (maybe it's supposed to send back
> a second BYE with authentication automatically). I don't know the SIP
> standards (or Freeswitch) well enough to know whether this problem is
> caused by me or by a deficiency in one of the two products (Metaswitch or
> Can you provide some pointers?
> The ZIP file with the packet traces can be downloaded here:
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
Anthony Minessale II
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>
-------------- next part --------------
An HTML attachment was scrubbed...
More information about the FreeSWITCH-users