[Freeswitch-users] Noob question about xml dialplan

[Raúl Alexis Betancor Santana]

Hi all, till now I'm been working with pre-made setup files for FS, and now
I'm trying to get a deep knowleadge of how the dialplan works.

So I modifed my autoload_configs/sofia.conf.xml file and changed my
dialplan param to something like this:

<param name="dialplan" value="XML:/etc/freeswitch/
dialplans/t38_transcode.xml,inline:socket:127.0.0.1:8022 async full"/>

The Idea is that it loads another .xml file especific task and also use a
dialplan throught a socket to a daemon that handle the rest.

On my t38_transcode.xml file ... very simple:

<extension name="t38_transcode">
 <condition field="sip_h_X-T38-Transcode" expresion="^True"/>
 <condition field="destination_number" expresion="^(\+1|1)?([2-9]\d\
d[2-9]\d{6})$">
   <action application="set" data="fax_enable_t38=true"/>
   <action application="set" data="t38_trace=true"/>
   <action application="set" data="sip_execute_on_image=t38_gateway peer
nocng"/>
   <action application="bridge" data="sofia/external/$1$2 at proxy.server.tld"/>

 </condition>
</extension>

But If I fire a call to FS like [number]@[FS_IP] from a sip testing client
I get

[INFO] switch_core_state_machine.c:311 No Route, Aborting

What I'm doing wrong here?

My target its just I want to ANY call that came in with a SIP header of
X-T39-Transcode=True (market by a sip proxy elsewhere on the net) just do a
T38->Ulaw transcoding saving the T38 trace, so I could inspect it later.
The rest of calls coming in ... as they don't have the sip-header should
end on other app.

For doing the testing I disabled the socket_inline part of the dialplan
string, so it have only this:

<param name="dialplan"
value="XML:/etc/freeswitch/dialplans/t38_transcode.xml"/>

Did I miss something? ... or maybe missundestood who the xml dialplan works
?
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