Hi ,

I am strugling to make two phones to make calls. I 'll appreciate information about the configs that I am listing below.

1s phone - Is an internat phone(behind NAT). This one was working before. It was able to make calls to PSTN and to another  extension.

The config is like

Sip server address: 192.168.1.117
Sip user id : 1000

2nd phone - It is external the of the Network.


The config is like :
Username : 1001
Domain 24.67.78.200:5060


Both are registering in FS. I cannot call between them.

When I try to make PSTN  calls using external phone  I get .


2008-07-15 10:45:34 [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel sofia/internal2/1001@24.67.78.200:5060 [58e39d4a-e78f-4c96-8f43-3e969704d9e1]
2008-07-15 10:45:34 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing FreeSwitch->12508606894@public
2008-07-15 10:45:34 [ERR] mod_sofia.c:1713 sofia_outgoing_channel() Invalid Gateway
2008-07-15 10:45:34 [NOTICE] mod_sofia.c:1904 sofia_outgoing_channel() Close Channel N/A [CS_NEW]
2008-07-15 10:45:34 [ERR] switch_ivr_originate.c:823 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2008-07-15 10:45:34 [INFO] mod_dptools.c:1797 audio_bridge_function() Originate Failed.  Cause: INVALID_NUMBER_FORMAT
2008-07-15 10:45:34 [NOTICE] mod_dptools.c:1824 audio_bridge_function() Hangup sofia/internal2/1001@24.67.78.200:5060 [CS_EXECUTE] [INVALID_NUMBER_FORMAT]
2008-07-15 10:45:34 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 17 (sofia/internal2/1001@24.67.78.200:5060) Ended
2008-07-15 10:45:34 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/internal2/1001@24.67.78.200:5060 [CS_HANGUP]


Please check the settings that I have.

On /conf/dialplan/extensions
teste.xml

<extension name="teste">
   <condition field="destination_number" expression="^(\d+)$">
       <!--- The % behind the username tells FS to lookup the user in it's local sip_registration database -->
       <action application="bridge" data="sofia/gateway/inphonex/$0"/>
   </condition>
 </extension>


On conf/sip_profiles

internal2.xml

<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="internal2">
  <!-- This profile is only for outbound registrations to providers -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

  <aliases>
    <alias name="internal2"/>
  </aliases>

  <domains>
    <domain name="$${domain}" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="5060"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="100"/>
    <param name="codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="use-rtp-timer" value="true"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="manage-presence" value="false"/>
    <param name="aggressive-nat-detection" value="true"/>
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <param name="rtp-timeout-sec" value="1800"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="24.67.78.200"/>
    <param name="ext-sip-ip" value="24.67.78.200"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
  </settings>
</profile>