[Freeswitch-users] Hangup problem/SIP BYE lacking authentication
Anthony Minessale
anthony.minessale at gmail.com
Mon Nov 3 09:39:51 PST 2008
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/
>> 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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>>
>>
>>
>> --
>> 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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> http://www.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 <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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