Nice answer!<br><br><div class="gmail_quote">On Tue, Jun 14, 2011 at 6:44 PM, Steven Ayre <span dir="ltr">&lt;<a href="mailto:steveayre@gmail.com">steveayre@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;">

At a guess, it&#39;s this line:<div class="im"><br>
&lt;action application=&quot;set&quot; data=&quot;bypass_media=true&quot;/&gt;<br></div>Unless the
 media is going through FS it can&#39;t trancode. The SDP (codec list) will 
be forwarded from the aleg to the bleg so absolute_codec_string would be
 ignored.<br><br>Also this has no effect, unless have proxy_media set to
 true on the sip profile. You don&#39;t need to set them to false, they&#39;ll 
already be false. You&#39;re correct to have this disabled though, as that 
would also prevent transcoding.<div class="im"><br>
&lt;action application=&quot;set&quot; data=&quot;proxy_media=false&quot;/&gt;<br><br></div>This won&#39;t do as you think:<div class="im"><br>&lt;action application=&quot;set&quot; data=&quot;absolute_codec_string=PCMU&quot;/&gt;<br>

</div>It
 chooses the codec before you get to the dialplan. absolute_codec_string
 will only apply to the bleg. The incoming codec will either be picked 
from the inbound-codec-prefs param on the sip profile that receives the 
a-leg, or if you&#39;re using late-negotiation it&#39;ll pick that after the 
bridge so that it&#39;ll try to use the same codec as the bleg if possible 
and pick one of the other choices and transcode if that&#39;s not possible. 
absolute_codec_string overrides the outbound-codec-prefs sip profile 
param on the bleg.<br><br>
Try these on your sip profile:<br>
 &lt;param name=&quot;inbound-codec-prefs&quot; value=&quot;PCMU&quot; /&gt;<br>
 &lt;param name=&quot;outbound-codec-prefs&quot; value=&quot;PCMA&quot; /&gt;<br>
  <br>
Then this will transcode:<div class="im"><br>
&lt;extension name=&quot;external&quot;&gt;<br>    
&lt;condition field=&quot;destination_number&quot; expression=&quot;^666654351428063&quot;&gt;<br>    
    
&lt;action application=&quot;set&quot; data=&quot;hangup_after_bridge=true&quot;/&gt;<br>    
    
&lt;action application=&quot;set&quot; data=&quot;progress_timeout=20&quot;/&gt;<br>    
    
&lt;action application=&quot;set&quot; 
data=&quot;export_vars=et_routing_number,et_routing_host,et_routed_number,et_routed_host,et_random_key&quot;/&gt;<br>    
    
&lt;action application=&quot;set&quot; data=&quot;et_routing_number=666654351428063&quot;/&gt;<br>    
    

&lt;action application=&quot;set&quot; data=&quot;et_routing_host=10.0.0.12&quot;/&gt;<br>    
    

&lt;action application=&quot;set&quot; data=&quot;et_random_key=029645322327599477&quot;/&gt;<br></div><div class="im">    
    

&lt;action application=&quot;set&quot; data=&quot;et_routed_number=555554351428063&quot;/&gt;<br>    

    
&lt;action application=&quot;set&quot; data=&quot;et_routed_host=10.0.0.14&quot;/&gt;<br>    
    

&lt;action application=&quot;bridge&quot; data=&quot;sofia/external/<a href="mailto:555554351428063@10.0.0.14" target="_blank">555554351428063@10.0.0.14</a>&quot;/&gt;<br>    

&lt;/condition&gt;<br>
&lt;/extension&gt;<br>
<br></div>
If you want to offer other codecs you can add them to the codec-pref 
params. They&#39;ll be listed in the preference order. FS will handle the 
transcoding automatically.<br><br>
You might also find it useful to read the Codec Negotiation page on the Wiki, if you haven&#39;t already done so:<br>
<a href="http://wiki.freeswitch.org/wiki/Codec_Negotiation" target="_blank">http://wiki.freeswitch.org/wiki/Codec_Negotiation</a><br>
<br>-Steve<div><div></div><div class="h5"><br><br><br><br><br><div class="gmail_quote">On 14 June 2011 13:35, Gustavo Espeche <span dir="ltr">&lt;<a href="mailto:gustavo.espeche@easyipcall.com" target="_blank">gustavo.espeche@easyipcall.com</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

Hi we are trying to do transcoding with freeswitch g711a--&gt;g711u or<br>
g711--&gt;speex but we can&#39;t do that FS do a trascoding, follow is a<br>
dialplan that we use with mod_curl<br>
<br>
&lt;?xml version=&quot;1.0&quot; encoding=&quot;UTF-8&quot; standalone=&quot;no&quot;?&gt;<br>
&lt;document type=&quot;freeswitch/xml&quot;&gt;<br>
&lt;section name=&quot;dialplan&quot; description=&quot;Regex/XML Dialplan&quot;&gt;<br>
&lt;context name=&quot;public&quot;&gt;<br>
&lt;extension name=&quot;external&quot;&gt;<br>
&lt;condition field=&quot;destination_number&quot; expression=&quot;^666654351428063&quot;&gt;<br>
&lt;action application=&quot;set&quot; data=&quot;hangup_after_bridge=true&quot;/&gt;<br>
&lt;action application=&quot;set&quot; data=&quot;progress_timeout=20&quot;/&gt;<br>
&lt;action application=&quot;set&quot;<br>
data=&quot;export_vars=et_routing_number,et_routing_host,et_routed_number,et_routed_host,et_random_key&quot;/&gt;<br>
&lt;action application=&quot;set&quot; data=&quot;et_routing_number=666654351428063&quot;/&gt;<br>
&lt;action application=&quot;set&quot; data=&quot;et_routing_host=10.0.0.12&quot;/&gt;<br>
&lt;action application=&quot;set&quot; data=&quot;et_random_key=029645322327599477&quot;/&gt;<br>
&lt;action application=&quot;set&quot; data=&quot;proxy_media=false&quot;/&gt;<br>
&lt;action application=&quot;set&quot; data=&quot;bypass_media=true&quot;/&gt;<br>
&lt;action application=&quot;set&quot; data=&quot;absolute_codec_string=PCMU&quot;/&gt;<br>
&lt;action application=&quot;export&quot; data=&quot;nolocal:absolute_codec_string=PCMA&quot;/&gt;<br>
&lt;action application=&quot;set&quot; data=&quot;et_routed_number=555554351428063&quot;/&gt;<br>
&lt;action application=&quot;set&quot; data=&quot;et_routed_host=10.0.0.14&quot;/&gt;<br>
&lt;action application=&quot;bridge&quot;<br>
data=&quot;sofia/external/<a href="mailto:555554351428063@10.0.0.14" target="_blank">555554351428063@10.0.0.14</a>&quot;/&gt;<br>
&lt;/condition&gt;<br>
&lt;/extension&gt;<br>
&lt;/context&gt;<br>
&lt;/section&gt;<br>
&lt;/document&gt;<br>
<br>
 we relay can&#39;t understand what are we doing wrong.<br>
are we need configure someone special in FS? we are using<br>
sip_external_profile with port 5060.<br>
<br>
<br>
Regards.<br>
<br>
--<br>
<br>
Gustavo Espeche<br>
EasyIpCall S.R.L.<br>
<a href="http://www.easyipcall.com" target="_blank">www.easyipcall.com</a><br>
Bv Mitre 517 24° E<br>
Cordoba - Argentina<br>
Te: +54 - 351 - 4280633<br>
<br>
<br>
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</blockquote></div><br><div style="padding:0px;margin-left:0px;margin-top:0px;overflow:hidden;word-wrap:break-word;color:black;font-size:10px;text-align:left;line-height:130%">

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<br></blockquote></div><br><br clear="all"><br>-- <br>Sidharth Kshatriya<br><a href="http://www.sidk.info">www.sidk.info</a><br><br>