I think you can use it has long as remote end-point supports it.<br><br>Thank you.<br><br><br><div class="gmail_quote">On Fri, Jul 10, 2009 at 3:48 PM, Meftah Tayeb <span dir="ltr"><<a href="mailto:tayeb.meftah@gmail.com">tayeb.meftah@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000">
hello Muhammad ,<br>
thank you<br>
what about hig cality audio codec to use?<br>
speex is good?<br>
thanks<br>
<br>
Muhammad Shahzad wrote:
<blockquote type="cite"><div><div></div><div class="h5">Destination parameter actually specifies the extension on
which this Skype user is reachable within FreeSWITCH dialplan for
incoming calls.<br>
<br>
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.<br>
<br>
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).<br>
<br>
<global_settings><br>
<param name="debug" value="8"/><br>
<param name="codec-master" value="us"/><br>
<param name="dialplan" value="XML"/><br>
<param name="context" value="default"/><br>
<param name="codec-prefs" value="gsm,ulaw"/><br>
<param name="codec-rates" value="8000,16000"/><br>
<param name="hold-music" value="$${moh_uri}"/><br>
<param name="destination" value="5000"/><br>
</global_settings><br>
<br>
<per_interface_settings><br>
<interface id="1" name="test.01"><br>
<param name="hold-music" value="$${moh_uri}"/><br>
<param name="dialplan" value="XML"/><br>
<param name="context" value="default"/><br>
<param name="X11-display" value=":101"/><br>
<param name="tcp_cli_port" value="15556"/><br>
<param name="tcp_srv_port" value="15557"/><br>
<param name="skype_user" value="test.01"/><br>
<param name="destination" value="2000"/><br>
</interface><br>
<interface id="2" name="test.02"><br>
<param name="hold-music" value="$${moh_uri}"/><br>
<param name="dialplan" value="XML"/><br>
<param name="context" value="default"/><br>
<param name="X11-display" value=":102"/><br>
<param name="tcp_cli_port" value="15558"/><br>
<param name="tcp_srv_port" value="15559"/><br>
<param name="skype_user" value="test.02"/><br>
</interface><br>
</per_interface_settings><br>
<br>
<br>
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),<br>
<br>
<extension name="skype_incoming-01"><br>
<condition field="destination_number" expression="^2000$"><br>
<action application="bridge" data="sofia/internal/1000"/><br>
</condition><br>
</extension><br>
<br>
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).<br>
<br>
Thank you.<br>
<br>
<br>
<div class="gmail_quote">On Fri, Jul 10, 2009 at 5:58 AM, Meftah
Tayeb <span dir="ltr"><<a href="mailto:tayeb.meftah@gmail.com" target="_blank">tayeb.meftah@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">hello,<br>
i have the folowing parameter in Skypiax.conf.xml:<br>
<configuration name="skypiax.conf" description="Skypiax
Configuration"><br>
<global_settings><br>
<param name="destination" value="1000"/><br>
each call that will to by routed to this destination??<br>
<br>
<per_interface_settings><br>
<interface id="1" name="skypiax1"><br>
<param name="destination" value="1000"/><br>
<br>
Each Call will to by routed to this destination?<br>
<br>
each codecs that is pocible to use it with Skypiax? all? speex?<br>
this codecs is used beetwan skypiax and the remote peer?<br>
thanks<br>
<br>
<br>
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-- <br>
Muhammad Shahzad<br>
-----------------------------------<br>
CISCO Rich Media Communication Specialist (CRMCS)<br>
CISCO Certified Network Associate (CCNA)<br>
Cell: +92 334 422 40 88<br>
MSN: <a href="mailto:shari_786pk@hotmail.com" target="_blank">shari_786pk@hotmail.com</a><br>
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<br>
<br>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Muhammad Shahzad<br>-----------------------------------<br>CISCO Rich Media Communication Specialist (CRMCS)<br>CISCO Certified Network Associate (CCNA)<br>Cell: +92 334 422 40 88<br>
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