try pressing f8 and try again you will see much more info.<br>I believe your user is not properly registered and you may have the domain wrong in your config.<br><br>You can try installing the default config and test using your box's ip as the domain.<br>
and id 1000 - 1016 with pass 1234 <br><br><br><br><div class="gmail_quote">On Sun, Nov 9, 2008 at 3:11 PM, <span dir="ltr"><<a href="mailto:wchao@yahoo.com">wchao@yahoo.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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 <a href="http://1.0.1." target="_blank">1.0.1.</a> It wasn't too hard, but I may have messed some things up.<br>
<br>
In any case, here is what is happening now:<br>
<br>
* 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!<br>
<br>
* 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:<br>
<br>
2008-11-09 15:59:58 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 9172388084->2675379325 in context public<br>
2008-11-09 15:59:58 [NOTICE] switch_ivr.c:1116 switch_ivr_session_transfer() Transfer sofia/internal/<a href="http://9172388084@64.115.128.6:5060" target="_blank">9172388084@64.115.128.6:5060</a> to XML[2675379325@default]<br>
2008-11-09 15:59:58 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 9172388084->2675379325 in context default<br>
2008-11-09 15:59:59 [NOTICE] switch_ivr.c:1116 switch_ivr_session_transfer() Transfer sofia/internal/<a href="http://9172388084@64.115.128.6:5060" target="_blank">9172388084@64.115.128.6:5060</a> to XML[1001@default]<br>
2008-11-09 15:59:59 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 9172388084->1001 in context default<br>
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<br>
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<br>
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<br>
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]<br>
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]<br>
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]<br>
2008-11-09 16:00:00 [INFO] mod_dptools.c:1848 audio_bridge_function() Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE<br>
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<br>
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]<br>
<br>
Do you think I migrated the configuration settings incorrectly, or do you think this might be a bug in the trunk version of Freeswitch?<br>
<br>
Wellie<br>
<br>
On Mon, 3 Nov 2008, Anthony Minessale wrote:<br>
<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Date: Mon, 3 Nov 2008 11:39:51 -0600<div><div></div><div class="Wj3C7c"><br>
From: Anthony Minessale <<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>><br>
Reply-To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication<br>
<br>
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<br>
can get the name from there instead.<br>
<br>
<br>
On Sun, Nov 2, 2008 at 11:40 PM, Wellie Chao <<a href="mailto:wchao@yahoo.com" target="_blank">wchao@yahoo.com</a>> wrote:<br>
I added some debug code and determined that session is null in sofia_reg.c in the sofia_reg_handle_sip_r_challenge<br>
function, which is called by sofia_event_callback in sofia.c. I added further debug code and found that<br>
sofia_event_callback only sets session if sofia_private->uuid exists. The strange thing is that during the call<br>
setup for a call from Metaswitch to Freeswitch (which is unauthenticated, remember), sofia_private->uuid exists and<br>
is a valid call ID, and session is also set to a valid value, but when I hang up from the Freeswitch side,<br>
sofia_private->uuid is null in that particular call to sofia_event_callback (and thus session is obviously left<br>
null). On the call setup, there are two legs (Metaswitch to Freeswitch, then Freeswitch to the extension). The call<br>
hangup is being performed by the extension. The session initiated by Metaswitch is unauthenticated, as I mentioned.<br>
<br>
I can look into this further, but I wanted to see if you had any quick pointers before delving in more deeply.<br>
<br>
On Fri, 31 Oct 2008, Anthony Minessale wrote:<br>
<br>
Date: Fri, 31 Oct 2008 10:16:25 -0500<br>
<br>
From: Anthony Minessale <<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>><br>
Reply-To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication<br>
<br>
Yes i mean add it to the dial string inside the {}<br>
it only will work if the channel with the variable set is tied to the FS session obj.<br>
<br>
sofia_reg.c 1122 is where it all happens<br>
so if session is null there the var code won't work.<br>
<br>
you can add some debug code there and try to figure out what's wrong.<br>
<br>
<br>
<br>
On Fri, Oct 31, 2008 at 10:06 AM, Wellie Chao <<a href="mailto:wchao@yahoo.com" target="_blank">wchao@yahoo.com</a>> wrote:<br>
I tried the following in conf/dialplan/extensions/7_inbound.xml:<br>
<br>
<extension name="broadview_inbound_9325"><br>
<condition field="destination_number" expression="^12675379325|2675379325$"><br>
<action application="export" data="sip_use_gateway=broadview"/><br>
<action application="transfer" data="1001"/><br>
</condition><br>
</extension><br>
<br>
Also tried the following in conf/dialplan/public.xml:<br>
<br>
<extension name="public_did_broadview"><br>
<condition field="destination_number" expression="^(12675379324|2675379324|12675379325|2675379325)$"><br>
<action application="export" data="sip_use_gateway=broadview"/><br>
<action application="transfer" data="$1 XML default"/><br>
</condition><br>
</extension><br>
<br>
Neither helped. When you say add it to the dial string directly that calls it, I'm not sure what you mean (I know<br>
the<br>
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<br>
target<br>
of the transfer?<br>
<br>
By the way, hangup DOES work properly if I create another gateway and name it <a href="http://64.115.128.6" target="_blank">64.115.128.6</a>. However, I'd love to get<br>
it<br>
working without having to create a duplicate gateway with a non-intuitive name. It's definitely a lot better than<br>
nothing<br>
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<br>
doing<br>
something wrong with how sip_use_gateway should be specified in the XML configuration files. Can you tell what I am<br>
doing<br>
wrong?<br>
<br>
On Fri, 31 Oct 2008, Anthony Minessale wrote:<br>
<br>
Date: Fri, 31 Oct 2008 09:49:18 -0500<br>
<br>
From: Anthony Minessale <<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>><br>
Reply-To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication<br>
<br>
try using "export" instead of "set" or add it to the dial string directly that calls it<br>
<br>
{sip_use_gateway=broadview}sofia/.......<br>
<br>
<br>
On Fri, Oct 31, 2008 at 9:42 AM, Wellie Chao <<a href="mailto:wchao@yahoo.com" target="_blank">wchao@yahoo.com</a>> wrote:<br>
Where do you recommend I put the sip_use_gateway=broadview action?<br>
<br>
I have tried in the conf/dialplan/public.xml like so:<br>
<br>
<extension name="public_did_broadview"><br>
<condition field="destination_number" expression="^(12675379324|2675379324|12675379325|2675379325)$"><br>
<action application="set" data="sip_use_gateway=broadview"/><br>
<action application="transfer" data="$1 XML default"/><br>
</condition><br>
</extension><br>
<br>
I've also tried in conf/dialplan/extensions/7_inbound.xml (a file I created that is pulled in via an include<br>
pre-processor directive):<br>
<br>
<extension name="broadview_inbound_9325"><br>
<condition field="destination_number" expression="^12675379325|2675379325$"><br>
<action application="set" data="sip_use_gateway=broadview"/><br>
<action application="transfer" data="1001"/><br>
</condition><br>
</extension><br>
<br>
I have a gateway named broadview in conf/sip_profiles/external. In both cases, I still get the following error<br>
on<br>
the Freeswitch console:<br>
<br>
2008-10-31 10:37:28 [ERR] sofia_reg.c:1089 sofia_reg_handle_sip_r_challenge() No Matching gateway found<br>
<br>
On Fri, 31 Oct 2008, Anthony Minessale wrote:<br>
<br>
Date: Fri, 31 Oct 2008 08:04:23 -0500<br>
From: Anthony Minessale <<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>><br>
Reply-To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication<br>
<br>
See what they said in the challenge?<br>
<br>
WWW-Authenticate: Digest <br>
realm="SipLocal",nonce="3e952db60fb8",stale=false,algorithm=MD5,qop="auth"<br>
<br>
Since this is a spontaneous challenge (which i think is somewhat silly since it lets you talk on the phone for 40<br>
minutes then makes you authenticate to hangup but *shrug*) FS does not know which gateway to use for credentials.<br>
<br>
The realm they sent was SipLocal so FS is looking in its configuration for a gateway with that name.<br>
The 2nd thing it tries is the host from the To: header (<a href="http://64.115.128.6" target="_blank">64.115.128.6</a>). if there was a gateway with either of those<br>
names,<br>
it would find it.<br>
<br>
So try naming your gateway SipLocal or <a href="http://64.115.128.6" target="_blank">64.115.128.6</a><br>
or you can try setting the variable sip_use_gateway=<whatever> on the channel which can give it a hint which<br>
gateway to use.<br>
<br>
<br>
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<br>
<br>
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<br>
FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
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<br>
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<a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a><br>
<a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
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--<br>
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<br>
FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
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<a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a><br>
<a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
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<br>
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FreeSWITCH Developer Conference<br>
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<a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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