[Freeswitch-users] Two Major Problems

Ken Fulmer kenfulmer at icstechnologysolutions.com
Thu Apr 8 15:18:40 PDT 2010


Actually, I did purchase a license and installed it today. One call
establishes at 729. When I hang up the phone and try again, it's 711. 

 

The Proxy Authentication Required is being sent by FreeSwitch to the
internal PBX. I have registration disabled on the FreeSwitch gateway and the
internal server. 

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, April 08, 2010 4:25 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Two Major Problems

 

Did you purchase g.729 licenses for this system? If not then you won't be
able to transcode g.729 and the incompatible destination error will keep
popping up if you're not in proxy media mode.

Where is the 407 coming from? If it's coming from your provider then they
want you to auth calls, which means you need to make sure your gateway has
the correct username and password information.

-MC

On Thu, Apr 8, 2010 at 12:54 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

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

 


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