Ok, I think the whole point is that I don't undestand how the dialplan works.

I started from 0 ... now on the sofia.conf.xml I have this:
<configuration name="sofia.conf" description="sofia Endpoint">

 <global_settings>
   <param name="log-level" value="0"/>
   <!-- <param name="auto-restart" value="false"/> -->
   <param name="debug-presence" value="0"/>
 </global_settings>

 <profiles>
       <profile name="fax">
         <gateways>
         <X-PRE-PROCESS cmd="include" data="../gateways/*.xml"/>
         </gateways>

         <domains>
           <domain name="all" alias="false" parse="false"/>
         </domains>

         <settings>
           <param name="debug" value="0"/>
           <param name="shutdown-on-fail" value="true"/>
           <param name="sip-trace" value="no"/>
           <param name="sip-capture" value="yes"/>
           <param name="capture-server" value="udp:$${homer_ip}:$${homer_port};hep=3;capture_id=2001"/>
           <param name="rfc2833-pt" value="101"/>
           <!--<param name="enable-rfc-5626" value="true"/> -->
            
           <param name="t38-trace" value="true"/>
           <param name="dialplan" value="XML"/>  

           <param name="context" value="incoming"/>

           <param name="dtmf-duration" value="2000"/>
           <param name="codec-prefs" value="PCMU"/>
           <param name="disable-hold" value="true"/>

           <param name="rtp-timer-name" value="soft"/>
           <param name="local-network-acl" value="localnet.auto"/>
           <param name="manage-presence" value="false"/>

           <param name="inbound-codec-negotiation" value="generous"/>
           <param name="nonce-ttl" value="60"/>
           <param name="auth-calls" value="false"/>
           <param name="disable-register" value="true"/>

           <param name="inbound-late-negotiation" value="false"/>
           <param name="inbound-zrtp-passthru" value="false"/>

           <param name="sip-port" value="5060"/>
           <param name="rtp-ip" value="$${sofia_ip}"/>
           <param name="sip-ip" value="$${sofia_ip}"/>
           <param name="ext-rtp-ip" value="$${sofia_ip}"/>
           <param name="ext-sip-ip" value="$${sofia_ip}"/>

           <param name="rtp-timeout-sec" value="300"/>
           <param name="rtp-hold-timeout-sec" value="1800"/>

           <param name="tls" value="false"/>
           <param name="tls-only" value="false"/>
           <param name="tls-bind-params" value="transport=tls"/>
           <param name="tls-sip-port" value="5061"/>
           <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
           <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/>
           <param name="tls-passphrase" value=""/>
           <param name="tls-verify-date" value="true"/>
           <!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
           <param name="tls-verify-policy" value="none"/>
           <param name="tls-verify-depth" value="2"/>
           <!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
           <param name="tls-verify-in-subjects" value=""/>
           <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
           <param name="tls-version" value="tlsv1"/>
         </settings>
       </profile>
 </profiles>

</configuration>


Then on the t38_transconde.xml file I have this.

<include>
<context name="incoming">
<extension name="t38_transcode">
 <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="sip_execute_on_image=t38_gateway peer nocng"/>     
   <action application="bridge" data="sofia/external/$1$2@$${external_sip_proxy}"/>
 </condition>
</extension>
</context>
</include>


I could see on the /var/log/freeswitch/freeswitch.xml.fsxml file that the t38_transcode.xml file its included ... but still get the same error on on the console, so something terrible wrong I'm missing here ... :-(

Also ... it's possible to directly send the calls sofia/$1$2@$${external} ? ... I mean, sending the call directly to a URI, with username/password and so on ... instead of having to define a profile for it.


2017-05-16 16:13 GMT+01:00 David Villasmil <david.villasmil.work@gmail.com>:
No, you may be sending it to the right ports, but the profile attached to that port must have the context set to the correct dialplan.

I just did exactly what you're doing and i also got "No Route, Aborting"

I've never seen this type of "dialplan" value, tbh

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


Regards,

David Villasmil

On Tue, May 16, 2017 at 3:30 PM, Raúl Alexis Betancor Santana <rbetancor@gmail.com> wrote:
Only one profile defined on the sofia.conf.xml and I'm sending the traffict to the wright ports, if not I whould get the logs on the console, as that are the only ports enabled.

2017-05-16 14:22 GMT+01:00 David Villasmil <david.villasmil.work@gmail.com>:
Take a look at your profile, it should be listening on the port you're sending to, and must have the context parameter set to your dialplan name.
On Tue, May 16, 2017 at 3:15 PM Raúl Alexis Betancor Santana <rbetancor@gmail.com> wrote:

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@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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_________________________________________________________________________
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ᐧ

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