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@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@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@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@mydomain:5060) Ended 2008-03-27 12:44:02 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/1000@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