Nice answer!<br><br><div class="gmail_quote">On Tue, Jun 14, 2011 at 6:44 PM, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>></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's this line:<div class="im"><br>
<action application="set" data="bypass_media=true"/><br></div>Unless the
media is going through FS it can'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't need to set them to false, they'll
already be false. You're correct to have this disabled though, as that
would also prevent transcoding.<div class="im"><br>
<action application="set" data="proxy_media=false"/><br><br></div>This won't do as you think:<div class="im"><br><action application="set" data="absolute_codec_string=PCMU"/><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're using late-negotiation it'll pick that after the
bridge so that it'll try to use the same codec as the bleg if possible
and pick one of the other choices and transcode if that'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>
<param name="inbound-codec-prefs" value="PCMU" /><br>
<param name="outbound-codec-prefs" value="PCMA" /><br>
<br>
Then this will transcode:<div class="im"><br>
<extension name="external"><br>
<condition field="destination_number" expression="^666654351428063"><br>
<action application="set" data="hangup_after_bridge=true"/><br>
<action application="set" data="progress_timeout=20"/><br>
<action application="set"
data="export_vars=et_routing_number,et_routing_host,et_routed_number,et_routed_host,et_random_key"/><br>
<action application="set" data="et_routing_number=666654351428063"/><br>
<action application="set" data="et_routing_host=10.0.0.12"/><br>
<action application="set" data="et_random_key=029645322327599477"/><br></div><div class="im">
<action application="set" data="et_routed_number=555554351428063"/><br>
<action application="set" data="et_routed_host=10.0.0.14"/><br>
<action application="bridge" data="sofia/external/<a href="mailto:555554351428063@10.0.0.14" target="_blank">555554351428063@10.0.0.14</a>"/><br>
</condition><br>
</extension><br>
<br></div>
If you want to offer other codecs you can add them to the codec-pref
params. They'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'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"><<a href="mailto:gustavo.espeche@easyipcall.com" target="_blank">gustavo.espeche@easyipcall.com</a>></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-->g711u or<br>
g711-->speex but we can't do that FS do a trascoding, follow is a<br>
dialplan that we use with mod_curl<br>
<br>
<?xml version="1.0" encoding="UTF-8" standalone="no"?><br>
<document type="freeswitch/xml"><br>
<section name="dialplan" description="Regex/XML Dialplan"><br>
<context name="public"><br>
<extension name="external"><br>
<condition field="destination_number" expression="^666654351428063"><br>
<action application="set" data="hangup_after_bridge=true"/><br>
<action application="set" data="progress_timeout=20"/><br>
<action application="set"<br>
data="export_vars=et_routing_number,et_routing_host,et_routed_number,et_routed_host,et_random_key"/><br>
<action application="set" data="et_routing_number=666654351428063"/><br>
<action application="set" data="et_routing_host=10.0.0.12"/><br>
<action application="set" data="et_random_key=029645322327599477"/><br>
<action application="set" data="proxy_media=false"/><br>
<action application="set" data="bypass_media=true"/><br>
<action application="set" data="absolute_codec_string=PCMU"/><br>
<action application="export" data="nolocal:absolute_codec_string=PCMA"/><br>
<action application="set" data="et_routed_number=555554351428063"/><br>
<action application="set" data="et_routed_host=10.0.0.14"/><br>
<action application="bridge"<br>
data="sofia/external/<a href="mailto:555554351428063@10.0.0.14" target="_blank">555554351428063@10.0.0.14</a>"/><br>
</condition><br>
</extension><br>
</context><br>
</section><br>
</document><br>
<br>
we relay can'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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