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[tayeb.meftah at gmail.com]

hello,
i have the folowing parameter in Skypiax.conf.xml:
<configuration name="skypiax.conf" description="Skypiax Configuration">
<global_settings>
<param name="destination" value="1000"/>
each call that will to by routed to this destination??

<per_interface_settings>
<interface id="1" name="skypiax1">
<param name="destination" value="1000"/>

Each Call will to by routed to this destination?

each codecs that is pocible to use it with Skypiax? all? speex?
this codecs is used beetwan skypiax and the remote peer?
thanks


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[shaheryarkh at googlem...]

Destination parameter actually specifies the extension on which this Skype user is reachable within FreeSWITCH dialplan for incoming calls.

If this parameter is specified in per_interface_settings xml tag then it will override the value of this parameter in global_settings xml tag, otherwise value of this parameter from global_settings xml tag will be used.

Here is an example (see below), the user test.01 is reachable on dialplan extension 2000 (since it has its own destination defined in per_interface_settings xml tag), whereas test.02 is reachable on dialplan extension 5000 (since it does not have destination parameter defined and thus it will use value for this parameter in global_settings xml tag).

  <global_settings>
    <param name="debug" value="8"/>
    <param name="codec-master" value="us"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="default"/>
    <param name="codec-prefs" value="gsm,ulaw"/>
    <param name="codec-rates" value="8000,16000"/>
    <param name="hold-music" value="$${moh_uri}"/>
    <param name="destination" value="5000"/>
  </global_settings>

  <per_interface_settings>
    <interface id="1" name="test.01">
        <param name="hold-music" value="$${moh_uri}"/>
        <param name="dialplan" value="XML"/>
        <param name="context" value="default"/>
        <param name="X11-display" value=":101"/>
        <param name="tcp_cli_port" value="15556"/>
        <param name="tcp_srv_port" value="15557"/>
        <param name="skype_user" value="test.01"/>
        <param name="destination" value="2000"/>
    </interface>
    <interface id="2" name="test.02">
        <param name="hold-music" value="$${moh_uri}"/>
        <param name="dialplan" value="XML"/>
        <param name="context" value="default"/>
        <param name="X11-display" value=":102"/>
        <param name="tcp_cli_port" value="15558"/>
        <param name="tcp_srv_port" value="15559"/>
        <param name="skype_user" value="test.02"/>
    </interface>
  </per_interface_settings>


Now the codec, Skype has its own proprietory code for Skype to Skype calls. The codec we specify in Skypiax configuration file is actually used for Skype to/from non-Skype calls. Consider following dial plan example (with skypiax configuration given above),

    <extension name="skype_incoming-01">
      <condition field="destination_number" expression="^2000$">
        <action application="bridge" data="sofia/internal/1000"/>
      </condition>
    </extension>

If a remote Skype user dials test.01 from his/her Skype client, then FreeSWITCH will route this call to SIP user 1000 and codecs specified in Skypiax configuration will be offered to destination SIP endpoint (SIP user 1000 in this case).

Thank you.


On Fri, Jul 10, 2009 at 5:58 AM, Meftah Tayeb <tayeb.meftah@gmail.com (tayeb.meftah@gmail.com)> wrote:

[tayeb.meftah at gmail.com]

hello Muhammad ,
thank you
what about hig cality audio codec to use?
speex is good?
thanks

Muhammad Shahzad wrote:

[shaheryarkh at googlem...]

I think you can use it has long as remote end-point supports it.

Thank you.


On Fri, Jul 10, 2009 at 3:48 PM, Meftah Tayeb <tayeb.meftah@gmail.com (tayeb.meftah@gmail.com)> wrote: