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 <[EMAIL PROTECTED]> 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 <[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. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> [email protected] >> 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:[EMAIL PROTECTED] <[EMAIL PROTECTED]> >> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> >> iax:[EMAIL PROTECTED]/888 >> googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> >> pstn:213-799-1400 >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> [email protected] >> 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:[EMAIL PROTECTED] <[EMAIL PROTECTED]> >> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> >> iax:[EMAIL PROTECTED]/888 >> googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> >> pstn:213-799-1400 >> >> > _______________________________________________ > Freeswitch-users mailing list > [email protected] > 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:[EMAIL PROTECTED] <[EMAIL PROTECTED]> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> pstn:213-799-1400
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