[Freeswitch-users] Phone registration error

15 Jul 2008

      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>

doublenat.xml

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

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

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

  <settings>
    <param name="debug" value="0"/>
    <param name="force-register-domain" value="$${domain}"/>
    <param name="apply-nat-acl" value="rfc1918"/>
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="5090"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="100"/>
    <param name="codec-prefs" value="$${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="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
  </settings>
</profile>

On /conf/directory/default

inphonex.xml

<include>
   <user id="inphonex">
     <gateways>
        <gateway name="inphonex">
                <param name="username" value="3462101"/>
                <param name="realm" value="sip.varphonex.com"/>
                <param name="password" value="606545"/>
                <param name="proxy" value="sip.varphonex.com"/>
                <param name="register" value="true"/>
                <param name="expire-seconds" value="3600"/>
         </gateway>
</gateways>
<params>
       <param name="password" value="606545"/>
     </params>
   </user>
</include>

Thank you 

Jair santos

[Freeswitch-users] Phone registration error

Jair Santos