[Freeswitch-users] Hangup problem/SIP BYE lacking authentication
Anthony Minessale
anthony.minessale at gmail.com
Mon Nov 10 07:34:08 PST 2008
try pressing f8 and try again you will see much more info.
I believe your user is not properly registered and you may have the domain
wrong in your config.
You can try installing the default config and test using your box's ip as
the domain.
and id 1000 - 1016 with pass 1234
On Sun, Nov 9, 2008 at 3:11 PM, <wchao at yahoo.com> wrote:
> 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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>> 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
>>
>>
>> _______________________________________________
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>> 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
>>
>>
>> _______________________________________________
>> 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
>>
>>
> _______________________________________________
> 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
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