[Freeswitch-users] Hangup problem/SIP BYE lacking authentication
wchao at yahoo.com
wchao at yahoo.com
Sun Nov 9 13:11:39 PST 2008
OK, I retrieved the latest trunk and compiled and now I am having
different problems. I noticed the configuration files changed somewhat,
but I tried to carry over my configuration from version 1.0.1. It wasn't
too hard, but I may have messed some things up.
In any case, here is what is happening now:
* I can make outbound calls from my snom 320 (configured to register on
extension 1001 on Freeswitch). I can hang up my snom 320 phone and the
other side (an external POTS line) will also get the signal to hang up and
does so properly. However, if I hang up the other side first (the external
POTS line), my snom 320 phone waits on the line forever -- it seems it's
not receiving a hangup. This is the reverse of the problem I had before!
* I can receive inbound calls to the IVR, but not to my extension (1001,
the snom 320 phone). Here is what the Freeswitch log says:
2008-11-09 15:59:58 [INFO] mod_dialplan_xml.c:233 dialplan_hunt()
Processing 9172388084->2675379325 in context public
2008-11-09 15:59:58 [NOTICE] switch_ivr.c:1116
switch_ivr_session_transfer() Transfer
sofia/internal/9172388084 at 64.115.128.6:5060 to XML[2675379325 at default]
2008-11-09 15:59:58 [INFO] mod_dialplan_xml.c:233 dialplan_hunt()
Processing 9172388084->2675379325 in context default
2008-11-09 15:59:59 [NOTICE] switch_ivr.c:1116
switch_ivr_session_transfer() Transfer
sofia/internal/9172388084 at 64.115.128.6:5060 to XML[1001 at default]
2008-11-09 15:59:59 [INFO] mod_dialplan_xml.c:233 dialplan_hunt()
Processing 9172388084->1001 in context default
2008-11-09 15:59:59 [INFO] switch_ivr_async.c:1536
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx
XML features
2008-11-09 15:59:59 [INFO] switch_ivr_async.c:1536
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2
record_session::/usr/local/freeswitch/recordings/9172388084.2008-11-09-15-59-59.wav
2008-11-09 15:59:59 [INFO] switch_ivr_async.c:1536
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf
XML features
2008-11-09 16:00:00 [NOTICE] switch_channel.c:551
switch_channel_set_name() New Channel
sofia/internal/1001 at 192.168.216.104:2051;line=4lp4nzfu
[ebe41989-0c8c-4cc6-8e2d-20c56907385f]
2008-11-09 16:00:00 [NOTICE] sofia.c:2784 sofia_handle_sip_i_state()
Hangup sofia/internal/1001 at 192.168.216.104:2051;line=4lp4nzfu
[CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2008-11-09 16:00:00 [ERR] switch_ivr_originate.c:1064
switch_ivr_originate() Cannot create outgoing channel of type [user]
cause: [NORMAL_TEMPORARY_FAILURE]
2008-11-09 16:00:00 [INFO] mod_dptools.c:1848 audio_bridge_function()
Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE
2008-11-09 16:00:00 [NOTICE] switch_core_session.c:927
switch_core_session_thread() Session 4
(sofia/internal/1001 at 192.168.216.104:2051;line=4lp4nzfu) Ended
2008-11-09 16:00:00 [NOTICE] switch_core_session.c:929
switch_core_session_thread() Close Channel
sofia/internal/1001 at 192.168.216.104:2051;line=4lp4nzfu [CS_HANGUP]
Do you think I migrated the configuration settings incorrectly, or do you
think this might be a bug in the trunk version of Freeswitch?
Wellie
On Mon, 3 Nov 2008, Anthony Minessale wrote:
> Date: Mon, 3 Nov 2008 11:39:51 -0600
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Reply-To: freeswitch-users at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication
>
> can you try latest trunk. I added a way to save the string into sofia_private so even after it's too late to get session you
> can get the name from there instead.
>
>
> On Sun, Nov 2, 2008 at 11:40 PM, Wellie Chao <wchao at yahoo.com> wrote:
> I added some debug code and determined that session is null in sofia_reg.c in the sofia_reg_handle_sip_r_challenge
> function, which is called by sofia_event_callback in sofia.c. I added further debug code and found that
> sofia_event_callback only sets session if sofia_private->uuid exists. The strange thing is that during the call
> setup for a call from Metaswitch to Freeswitch (which is unauthenticated, remember), sofia_private->uuid exists and
> is a valid call ID, and session is also set to a valid value, but when I hang up from the Freeswitch side,
> sofia_private->uuid is null in that particular call to sofia_event_callback (and thus session is obviously left
> null). On the call setup, there are two legs (Metaswitch to Freeswitch, then Freeswitch to the extension). The call
> hangup is being performed by the extension. The session initiated by Metaswitch is unauthenticated, as I mentioned.
>
> I can look into this further, but I wanted to see if you had any quick pointers before delving in more deeply.
>
> On Fri, 31 Oct 2008, Anthony Minessale wrote:
>
> Date: Fri, 31 Oct 2008 10:16:25 -0500
>
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Reply-To: freeswitch-users at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication
>
> Yes i mean add it to the dial string inside the {}
> it only will work if the channel with the variable set is tied to the FS session obj.
>
> sofia_reg.c 1122 is where it all happens
> so if session is null there the var code won't work.
>
> you can add some debug code there and try to figure out what's wrong.
>
>
>
> On Fri, Oct 31, 2008 at 10:06 AM, Wellie Chao <wchao at yahoo.com> wrote:
> I tried the following in conf/dialplan/extensions/7_inbound.xml:
>
> <extension name="broadview_inbound_9325">
> <condition field="destination_number" expression="^12675379325|2675379325$">
> <action application="export" data="sip_use_gateway=broadview"/>
> <action application="transfer" data="1001"/>
> </condition>
> </extension>
>
> Also tried the following in conf/dialplan/public.xml:
>
> <extension name="public_did_broadview">
> <condition field="destination_number" expression="^(12675379324|2675379324|12675379325|2675379325)$">
> <action application="export" data="sip_use_gateway=broadview"/>
> <action application="transfer" data="$1 XML default"/>
> </condition>
> </extension>
>
> Neither helped. When you say add it to the dial string directly that calls it, I'm not sure what you mean (I know
> the
> general format of {var_name=var_value}, so that's not my question). Do you mean add it in front of the 1001 as the
> target
> of the transfer?
>
> By the way, hangup DOES work properly if I create another gateway and name it 64.115.128.6. However, I'd love to get
> it
> working without having to create a duplicate gateway with a non-intuitive name. It's definitely a lot better than
> nothing
> to do it that way, but I'd prefer to have it work with the sip_use_gateway scheme you mention. I'm assuming I'm just
> doing
> something wrong with how sip_use_gateway should be specified in the XML configuration files. Can you tell what I am
> doing
> wrong?
>
> On Fri, 31 Oct 2008, Anthony Minessale wrote:
>
> Date: Fri, 31 Oct 2008 09:49:18 -0500
>
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Reply-To: freeswitch-users at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication
>
> try using "export" instead of "set" or add it to the dial string directly that calls it
>
> {sip_use_gateway=broadview}sofia/.......
>
>
> On Fri, Oct 31, 2008 at 9:42 AM, Wellie Chao <wchao at yahoo.com> wrote:
> Where do you recommend I put the sip_use_gateway=broadview action?
>
> I have tried in the conf/dialplan/public.xml like so:
>
> <extension name="public_did_broadview">
> <condition field="destination_number" expression="^(12675379324|2675379324|12675379325|2675379325)$">
> <action application="set" data="sip_use_gateway=broadview"/>
> <action application="transfer" data="$1 XML default"/>
> </condition>
> </extension>
>
> I've also tried in conf/dialplan/extensions/7_inbound.xml (a file I created that is pulled in via an include
> pre-processor directive):
>
> <extension name="broadview_inbound_9325">
> <condition field="destination_number" expression="^12675379325|2675379325$">
> <action application="set" data="sip_use_gateway=broadview"/>
> <action application="transfer" data="1001"/>
> </condition>
> </extension>
>
> I have a gateway named broadview in conf/sip_profiles/external. In both cases, I still get the following error
> on
> the Freeswitch console:
>
> 2008-10-31 10:37:28 [ERR] sofia_reg.c:1089 sofia_reg_handle_sip_r_challenge() No Matching gateway found
>
> On Fri, 31 Oct 2008, Anthony Minessale wrote:
>
> Date: Fri, 31 Oct 2008 08:04:23 -0500
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Reply-To: freeswitch-users at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication
>
> 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.
>
>
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>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
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>
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> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
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>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
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