[Freeswitch-users] SIP dialout problems

[Nicolas Brenner]

Hi, I'm having trouble dialing out some phone numbers through a SIP gateway.

I configured the gateway on conf/sip_profiles/outbound/gizmo.xml as:

<include>
  <gateway name="gizmo">
          <param name="debug" value="1"/>
          <param name="username" value="1747xxxxxx"/>
          <param name="from-domain" value="proxy01.sipphone.com"/>
          <param name="password" value="MYPASSWORD"/>
          <param name="caller-id-in-from" value="true"/>
          <param name="proxy" value="proxy01.sipphone.com"/>
          <param name="expire-seconds" value="3600"/>
          <param name="register" value="true"/>
          <param name="retry_seconds" value="3600"/>
  </gateway>
</include>

Then I created an extension on conf/dialplan/public.xml like this:

<extension name="chile">
                  <condition field="destination_number" expression="^0056(.*)">
                    <action application="bridge"
data="sofia/gateway/gizmo/0056$1"/>
                </condition>
</extension>

The above catches all calls that begin with the 0056 prefix (calls to Chile).

So what I do is, register with a softphone using extension 1000, then
I dial 0056xxxxx, and I get a recording saying the extension I'm
calling is unavailable. The weird thing is that if I don't use 0056,
the call is made, but then I get a recording from Gizmo (my test
gateway) saying the number is wrong (of course). If instead of an
expression, I dial the fixed number 1-800-466-4411 (Google 411), the
called is successfully bridged. Also if I dial a Gizmo user number
(1747xxxxxx) the call works. What's the problem with 0056? Apparently
the call is never "leaving" freeswitch.

For an unsuccessful call, I get this on the console:

2008-03-27 12:43:59 [NOTICE] switch_channel.c:531
switch_channel_set_name() New Channel
sofia/default/1000 at medularis01.dyndns.org:5060
[dbbaff68-0791-4253-a07c-ab2c4130df18]
2008-03-27 12:43:59 [INFO] mod_dialplan_xml.c:223 dialplan_hunt()
Processing Nico->005624949458 at default
2008-03-27 12:43:59 [NOTICE] switch_channel.c:531
switch_channel_set_name() New Channel sofia/outbound/005624949458
[c0b1791c-eb96-464a-bc30-5d6481ed0c10]
2008-03-27 12:44:02 [NOTICE] sofia.c:1950 sofia_handle_sip_i_state()
Hangup sofia/outbound/005624949458 [CS_HOLD] [NORMAL_UNSPECIFIED]
2008-03-27 12:44:02 [INFO] mod_dptools.c:1514 audio_bridge_function()
Originate Failed.  Cause: NORMAL_UNSPECIFIED
2008-03-27 12:44:02 [NOTICE] switch_core_session.c:748
switch_core_session_thread() Session 16 (sofia/outbound/005624949458)
Ended
2008-03-27 12:44:02 [NOTICE] switch_core_session.c:750
switch_core_session_thread() Close Channel sofia/outbound/005624949458
[CS_HANGUP]
2008-03-27 12:44:02 [NOTICE] mod_dptools.c:1541
audio_bridge_function() Hangup sofia/default/1000 at mydomain:5060
[CS_EXECUTE] [NORMAL_UNSPECIFIED]
2008-03-27 12:44:02 [NOTICE] switch_core_session.c:748
switch_core_session_thread() Session 15
(sofia/default/1000 at mydomain:5060) Ended
2008-03-27 12:44:02 [NOTICE] switch_core_session.c:750
switch_core_session_thread() Close Channel
sofia/default/1000 at mydomain:5060 [CS_HANGUP]

What does NORMAL_UNSPECIFIED mean?! I found this page on the wiki:
http://wiki.freeswitch.org/wiki/Hangup_causes but it's only a list of
the codes. The page says "The codes are documented in
src\switch_channel.c" but there's no documentation there at all, just
the code.

Any help would be greatly appreciated. Thanks in advance for your time and help.

Regards,

-- 
Nicolás Brenner