[Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem.

[julien]

I'm currently trying to connect FreeSwitch to a PBX (Alcatel-Lucent), 
thanks to a SIP trunk.
SIP trunks are available and working on the PBX thanks to a recent update.

My problem is that I can't call phones linked to the PBX.
When I try to call 300, I've got this message in freeswitch console :


2009-07-27 17:38:48.514105 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/[EMAIL PROTECTED] [93d5a10e-7ac3-11de-b456-e5e56113066d]
2009-07-27 17:38:48.516907 [INFO] mod_dialplan_xml.c:252 Processing 
jgonzalez jgonzalez->300 in context default
2009-07-27 17:38:48.521084 [NOTICE] switch_channel.c:602 New Channel 
sofia/external/[EMAIL PROTECTED] [93d69816-7ac3-11de-b456-e5e56113066d]
2009-07-27 17:38:48.636073 [NOTICE] sofia.c:3775 Hangup 
sofia/external/[EMAIL PROTECTED] [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
2009-07-27 17:38:48.636073 [INFO] mod_dptools.c:2091 Originate Failed.  
Cause: NO_ROUTE_DESTINATION
2009-07-27 17:38:48.637788 [NOTICE] mod_dptools.c:633 Hangup 
sofia/internal/[EMAIL PROTECTED] [CS_EXECUTE] [NORMAL_CLEARING]
2009-07-27 17:38:48.639047 [NOTICE] switch_core_session.c:1085 Session 
15 (sofia/internal/[EMAIL PROTECTED]) Ended
2009-07-27 17:38:48.639047 [NOTICE] switch_core_session.c:1087 Close 
Channel sofia/internal/[EMAIL PROTECTED] [CS_DESTROY]
2009-07-27 17:38:48.643042 [NOTICE] switch_core_session.c:1085 Session 
16 (sofia/external/[EMAIL PROTECTED]) Ended
2009-07-27 17:38:48.643042 [NOTICE] switch_core_session.c:1087 Close 
Channel sofia/external/[EMAIL PROTECTED] [CS_DESTROY]


I've defined, in sip_profiles/external, a gateway to the PBX this way :


<include>
  <gateway name="pbxlyon">
  <param name="username" value="pbxlyon"/>
  <param name="realm" value="[PBX IP address]"/>
  <param name="password" value="pbxlyon"/>
  <param name="register" value="false"/>
  <param name="register-transport" value="udp"/>
  <param name="retry_seconds" value="30"/>
  </gateway>
</include>


And in the dialplan default.xml :


    <extension name="pbxlyon">
      <condition field="destination_number" expression="300">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="sofia/gateway/pbxlyon/300"/>
        <action application="hangup"/>
      </condition>
    </extension>


(for the moment, I'm trying only with the number 300 which is a correct 
number of the phone system).
As you can see, I'm far from being an expert of FreeSwitch, SIP or even 
VoIP in general. I'm learning.
I hope you can help me.

Regards,
Julien Gonzalez.