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/[EMAIL PROTECTED]:5060 to [EMAIL PROTECTED]
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/[EMAIL PROTECTED]:5060 to [EMAIL PROTECTED]
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/[EMAIL PROTECTED]: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/[EMAIL PROTECTED]: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/[EMAIL PROTECTED]:2051;line=4lp4nzfu) Ended
2008-11-09 16:00:00 [NOTICE] switch_core_session.c:929
switch_core_session_thread() Close Channel
sofia/internal/[EMAIL PROTECTED]: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 <[EMAIL PROTECTED]>
Reply-To: [email protected]
To: [email protected]
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 <[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.
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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
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googletalk:[EMAIL PROTECTED]
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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]
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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