<div dir="ltr">Hi, <div><br></div><div>You can set codec from dialplan. </div><div>See</div><div><a href="http://wiki.freeswitch.org/wiki/Codec_negotiation">http://wiki.freeswitch.org/wiki/Codec_negotiation</a>
</div><div><br></div><div>make sure to set inbound-codec-negotiation from generous to greedy in your sip profile </div><div>&lt;param name=&quot;inbound-codec-negotiation&quot; value=&quot;greedy&quot;/&gt;</div><div><br clear="all">

<div dir="ltr">Regards,<br>Zohair Raza<div><br></div></div>
<br><br><div class="gmail_quote">On Fri, Jan 6, 2012 at 5:18 PM, Bharat Lalcheta <span dir="ltr">&lt;<a href="mailto:bharatlalcheta@gmail.com">bharatlalcheta@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div>Hiii,</div><div> </div><div>I am new to freeswitch. Prior to freeswitch i was using asterisk.</div><div> </div><div>I have 200 extensions working in my office and want to move all to freeswitch from asterisk. </div>
<div>

 </div><div>In asterisk, i can give codec selection and preferance in sip.conf to all extensions. In the same way i created 200 extensions under internal profile in freeswitch.</div><div> </div><div>Follwing is one example....</div>


<div>-----------------------------------------------------------------</div><div>&lt;include&gt;<br>  &lt;user id=&quot;590&quot;&gt;<br>    &lt;params&gt;<br>      &lt;param name=&quot;password&quot; value=&quot;590&quot;/&gt;<br>


      &lt;param name=&quot;vm-password&quot; value=&quot;&quot;/&gt;<br>      &lt;param name=&quot;vm-enabled&quot; value=&quot;true&quot;/&gt;</div><div>      &lt;param name=&quot;inbound_codec_prefs&quot; value=&quot;PCMA,H264&quot;/&gt;<br>


      &lt;param name=&quot;outbound_codec_prefs&quot; value=&quot;PCMA,H264&quot;/&gt;<br></div><div>    &lt;/params&gt;<br>    &lt;variables&gt;<br>      &lt;variable name=&quot;accountcode&quot; value=&quot;&quot;/&gt;<br>


      &lt;variable name=&quot;user_context&quot; value=&quot;default&quot;/&gt;<br>      &lt;variable name=&quot;max-calls&quot; value=&quot;2&quot;/&gt;<br>      &lt;variable name=&quot;bypass_media_after_bridge&quot; value=&quot;no&quot;/&gt;<br>


    &lt;/variables&gt;<br>  &lt;/user&gt;<br>&lt;/include&gt;<br>----------------------------------------------------------</div><div> </div><div>Now when ever i called to 590 freeswitch sends all codecs to 590 sip phone other than defined in 590.xml. It is seding codes which is mentioned in my conf/sip_profiles/internal.xml and codec negotiation done on whatever codec my sip phone having. </div>


<div> </div><div>I want to use different codecs for different extensions.</div><div> </div><div>Is it common behaviour of Freeswitch ? Should i override codec prerfrance in my extension list from my internal profile or not ?</div>


<div> </div><div>If no, then is it that i have to create 200 profiles in freeswitch to solve this problem ?</div><div> </div><div>Please guide me and provide solution for the same</div><div> </div><div> </div><div>Thanks in advance</div>

<span class="HOEnZb"><font color="#888888">
<div> </div><div>Bharat Lalcheta<br>
</div>
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<br></blockquote></div><br></div></div>