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@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@64.115.128.6:5060 to XML[2675379325@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@64.115.128.6:5060 to XML[1001@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@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@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@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@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@gmail.com> Reply-To: freeswitch-users@lists.freeswitch.org To: freeswitch-users@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@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@gmail.com> Reply-To: freeswitch-users@lists.freeswitch.org To: freeswitch-users@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@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="$1 XML default"/> </condition> </extension>
Neither helped. When you say add it to the dial string directly that calls it, I'm not sure what you mean (I know the general format of {var_name=var_value}, so that's not my question). Do you mean add it in front of the 1001 as the target of the transfer?
By the way, hangup DOES work properly if I create another gateway and name it 64.115.128.6. However, I'd love to get it working without having to create a duplicate gateway with a non-intuitive name. It's definitely a lot better than nothing to do it that way, but I'd prefer to have it work with the sip_use_gateway scheme you mention. I'm assuming I'm just doing something wrong with how sip_use_gateway should be specified in the XML configuration files. Can you tell what I am doing wrong?
On Fri, 31 Oct 2008, Anthony Minessale wrote:
Date: Fri, 31 Oct 2008 09:49:18 -0500
From: Anthony Minessale <anthony.minessale@gmail.com> Reply-To: freeswitch-users@lists.freeswitch.org To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication
try using "export" instead of "set" or add it to the dial string directly that calls it
{sip_use_gateway=broadview}sofia/.......
On Fri, Oct 31, 2008 at 9:42 AM, Wellie Chao <wchao@yahoo.com> wrote: Where do you recommend I put the sip_use_gateway=broadview action?
I have tried in the conf/dialplan/public.xml like so:
<extension name="public_did_broadview"> <condition field="destination_number" expression="^(12675379324|2675379324|12675379325|2675379325)$"> <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@gmail.com> Reply-To: freeswitch-users@lists.freeswitch.org To: freeswitch-users@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.
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400