[Freeswitch-users] SIP dialout problems

[Chris Danielson]

Nicolas,
    I had a similar problem months ago.  It turns out that you really 
want to add the gateway information into the "directory" and not the 
"sip_profiles".
Remove the information you have added into the outbound directory and 
try adding your gateway into the following directory path. 
/usr/local/freeswitch/conf/directory/default/
For an example config for sipphone have a look in the directory from you 
compiled freeswitch.
~/projects/freeswitch/conf/directory/default/  <-- inside here you will 
find a file named:   sipphone.com.noload

Just edit the information accordingly and copy this into the directory 
mentioned previously.  Stop and restart your FreeSWITCH process.

Let's see how that works for you.
Regards,
Chris


Nicolas Brenner wrote:
> 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,
>
>