[Freeswitch-users] Two Major Problems

8 Apr 2010

      I'm using the external sip profile to route calls from a voice gateway
router to an internal PBX.

In separate xml files in the sofia/external directory, I have the following:

<include>

    <gateway name="Server-1">

      <param name="username" value="gateway"/>

      <param name="password" value="gateway"/>

      <param name="realm" value=" X.X.X.X "/>

      <param name="proxy" value=" X.X.X.X "/>

      <param name="register" value="false"/>

      <param name="context" value="public"/>

    </gateway>

</include>

<include>

    <gateway name="Server-2">

      <param name="username" value="gateway"/>

      <param name="password" value="gateway"/>

      <param name="realm" value="X.X.X.X"/>

      <param name="proxy" value="X.X.X.X"/>

      <param name="register" value="false"/>

      <param name="context" value="public"/>

    </gateway>

</include>

<include>

    <gateway name="Voice Gateway">

      <param name="username" value="gateway"/>

      <param name="password" value="gateway"/>

      <param name="realm" value="X.X.X.X"/>

      <param name="proxy" value="X.X.X.X"/>

      <param name="register" value="false"/>

      <param name="context" value="public"/>

    </gateway>

</include>

In the dial plan/public directory, I have the following (different xml
files):

<extension name="PBX">

   <condition field="destination_number" expression="^(Directory Numbers)$">

       <action application="bridge" data="sofia/gateway/Server-1/$1"/>

       <action application="bridge" data="sofia/gateway/Server-2/$1"/>

   </condition>

</extension>

<extension name="PSTN">

   <condition field="destination_number" expression="^(\d{10})$">

       <action application="bridge" data="sofia/gateway/Voice Gateway/$1"/>

   </condition>

</extension>

In the sip profiles/eternal.xml file, I have the following major settings:

  <settings>

    <param name="inbound-proxy-media" value="false"/>

    <param name="disable-transcoding" value="false"/>

    <param name="inbound-late-negotiation" value="true"/>

    <param name="user-agent-string" value="My Company Name"/>

    <param name="disable-register" value="true"/>

    <param name="inbound-codec-negotiation" value="generous"/>

    <param name="inbound-codec-prefs" value="G729,PCMU"

    <param name="outbound-codec-prefs" value="G729,PCMU"/>

    <param name="media-option" value="resume-media-on-hold"/>

    <param name="stun-enabled" value="false"/>

    <param name="debug" value="0"/>

    <param name="sip-trace" value="no"/>

    <param name="rfc2833-pt" value="101"/>

    <param name="sip-port" value="5080"/>

    <param name="rtp-ip" value="X.X.X.X"/>

    <param name="sip-ip" value="X.X.X.X"/>

    <param name="dialplan" value="XML"/>

    <param name="context" value="public"/>

    <param name="dtmf-duration" value="2000"/>

    <param name="rtp-timer-name" value="soft"/>

    <param name="rtp-timeout-sec" value="300"/>

    <param name="rtp-hold-timeout-sec" value="1800"/>

  </settings>

</profile>

Currently, I'm experiencing two issues.

1.       Inbound calls work as long as I'm using "inbound-proxy-media". When
I disable that setting and force transcoding, the phone rings but
immediately hangs up when I pick up the receiver. I see an "Incompatible
Destination" message on the console.

2.       Outbound calls don't work at all. On the console, I see "407 Proxy
Authentication Required". If I'm using the external profile, how could it
require authentication? 

Any ideas? 

Thanks,

Ken Fulmer