<p>In theory he could use those variable names but it would involve some extra work ;)</p>
<div class="gmail_quote">On 2012-01-06 7:24 AM, "Kristian Kielhofner" <<a href="mailto:kris@kriskinc.com">kris@kriskinc.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
inbound_codec_prefs<br>
outbound_codec_prefs<br>
<br>
These are Sofia profile configuration options. They are NOT valid<br>
options for the directory UNLESS you're setting them as variable to<br>
use in your dialplan later.<br>
<br>
If you want to control codes on a per-user basis you have to set late<br>
negotiation and use the dialplan.<br>
<br>
On Fri, Jan 6, 2012 at 8:18 AM, Bharat Lalcheta<br>
<<a href="mailto:bharatlalcheta@gmail.com">bharatlalcheta@gmail.com</a>> wrote:<br>
> Hiii,<br>
><br>
> I am new to freeswitch. Prior to freeswitch i was using asterisk.<br>
><br>
> I have 200 extensions working in my office and want to move all to<br>
> freeswitch from asterisk.<br>
><br>
> In asterisk, i can give codec selection and preferance in sip.conf to all<br>
> extensions. In the same way i created 200 extensions under internal profile<br>
> in freeswitch.<br>
><br>
> Follwing is one example....<br>
> -----------------------------------------------------------------<br>
> <include><br>
> <user id="590"><br>
> <params><br>
> <param name="password" value="590"/><br>
> <param name="vm-password" value=""/><br>
> <param name="vm-enabled" value="true"/><br>
> <param name="inbound_codec_prefs" value="PCMA,H264"/><br>
> <param name="outbound_codec_prefs" value="PCMA,H264"/><br>
> </params><br>
> <variables><br>
> <variable name="accountcode" value=""/><br>
> <variable name="user_context" value="default"/><br>
> <variable name="max-calls" value="2"/><br>
> <variable name="bypass_media_after_bridge" value="no"/><br>
> </variables><br>
> </user><br>
> </include><br>
> ----------------------------------------------------------<br>
><br>
> Now when ever i called to 590 freeswitch sends all codecs to 590 sip phone<br>
> other than defined in 590.xml. It is seding codes which is mentioned in my<br>
> conf/sip_profiles/internal.xml and codec negotiation done on whatever codec<br>
> my sip phone having.<br>
><br>
> I want to use different codecs for different extensions.<br>
><br>
> Is it common behaviour of Freeswitch ? Should i override codec prerfrance in<br>
> my extension list from my internal profile or not ?<br>
><br>
> If no, then is it that i have to create 200 profiles in freeswitch to solve<br>
> this problem ?<br>
><br>
> Please guide me and provide solution for the same<br>
><br>
><br>
> Thanks in advance<br>
><br>
> Bharat Lalcheta<br>
><br>
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<br>
<br>
<br>
--<br>
Kristian Kielhofner<br>
<br>
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</blockquote></div>