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 <[EMAIL PROTECTED]>
Reply-To: [email protected]
To: [email protected]
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 <[EMAIL PROTECTED]> 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="$1 XML default"/>
</condition>
</extension>
Neither helped. When you say add it to the dial string directly that calls it,
I'm not sure what you mean (I know the
general format of {var_name=var_value}, so that's not my question). Do you mean
add it in front of the 1001 as the target
of the transfer?
By the way, hangup DOES work properly if I create another gateway and name it
64.115.128.6. However, I'd love to get it
working without having to create a duplicate gateway with a non-intuitive name.
It's definitely a lot better than nothing
to do it that way, but I'd prefer to have it work with the sip_use_gateway
scheme you mention. I'm assuming I'm just doing
something wrong with how sip_use_gateway should be specified in the XML
configuration files. Can you tell what I am doing
wrong?
On Fri, 31 Oct 2008, Anthony Minessale wrote:
Date: Fri, 31 Oct 2008 09:49:18 -0500
From: Anthony Minessale <[EMAIL PROTECTED]>
Reply-To: [email protected]
To: [email protected]
Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication
try using "export" instead of "set" or add it to the dial string directly that
calls it
{sip_use_gateway=broadview}sofia/.......
On Fri, Oct 31, 2008 at 9:42 AM, Wellie Chao <[EMAIL PROTECTED]> wrote:
Where do you recommend I put the sip_use_gateway=broadview action?
I have tried in the conf/dialplan/public.xml like so:
<extension name="public_did_broadview">
<condition field="destination_number"
expression="^(12675379324|2675379324|12675379325|2675379325)$">
<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 <[EMAIL PROTECTED]>
Reply-To: [email protected]
To: [email protected]
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:[EMAIL PROTECTED]
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED]
iax:[EMAIL PROTECTED]/888
googletalk:[EMAIL PROTECTED]
pstn:213-799-1400
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