Re: [Freeswitch-users] Noob question about xml dialplan

16 May 2017

      Just for the shake of simplicity ... on the Asterisk world would be
something like:

exten => _1NXXNXXXXXX,1,Noop("T38 Gateway Call")
exten => _1NXXNXXXXXX,n,Set(FAXOPT(gateway)=yes)
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@proxy.server.tld)

I just want to make the same with FS, but not allowing T38 passthrought at
all ...  just T38 (a leg) to Ulaw (b leg)

2017-05-16 17:48 GMT+01:00 Raúl Alexis Betancor Santana <rbetancor@gmail.com
...
:
...
Same result ... No Route, Aborting
Lets see if I found some step-by-step tutorial that explains who the
dialplan works. The doc on the Wiki it's not clear enought for me.
2017-05-16 17:40 GMT+01:00 David Villasmil <david.villasmil.work@gmail.com
...
:
...
That was my test extension, you can replace it for whatever you need :)
ᐧ
Regards,
David Villasmil
email: david.villasmil.work@gmail.com
phone: +34669448337 <+34%20669%2044%2083%2037>
On Tue, May 16, 2017 at 6:38 PM, Raúl Alexis Betancor Santana <
rbetancor@gmail.com> wrote:
...
I don't get the point of calling a user/$1 ... as there is no users
registered against this FS ... I just want to use it as a T38->Ulaw gateway.
2017-05-16 17:34 GMT+01:00 David Villasmil <
david.villasmil.work@gmail.com>:
...
that context line is not necessary, btw. point is you need the section:
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="dialplan" description="RE Dial Plan For FreeSwitch">
  <extension name="local_call">
    <condition field="destination_number" expression="^(10..)$">
      <action application="bridge" data="user/$1"/>
      <action application="hangup"/>
    </condition>
  </extension>
<section/>
ᐧ
Regards,
David Villasmil
email: david.villasmil.work@gmail.com
phone: +34669448337 <+34%20669%2044%2083%2037>
On Tue, May 16, 2017 at 6:33 PM, David Villasmil <
david.villasmil.work@gmail.com> wrote:
...
Ok i've looked at the source code for mod_dialplan and in your xml,
you need something like:
<?xml version="1.0" encoding="utf-8"?>
<document type="freeswitch/xml">
  <section name="dialplan" description="RE Dial Plan For FreeSwitch">
<context name="default">
<extension name="local_call">
    <condition field="destination_number" expression="^(10..)$">
      <action application="bridge" data="user/$1"/>
      <action application="hangup"/>
    </condition>
  </extension>
<section/this works for me
ᐧ
Regards,
David Villasmil
email: david.villasmil.work@gmail.com
phone: +34669448337 <+34%20669%2044%2083%2037>
On Tue, May 16, 2017 at 5:26 PM, Raúl Alexis Betancor Santana <
rbetancor@gmail.com> wrote:
...
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:$${homer_ip}:$${hom
er_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/dia
> lplans/t38_transcode.xml"/>
>
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work@gmail.com
> phone: +34669448337 <+34%20669%2044%2083%2037>
>
> 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/dia
>>>> lplans/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/dia
>>>> lplans/t38_transcode.xml"/>
>>>>
>>>> Did I miss something? ... or maybe missundestood who the xml
>>>> dialplan works ?
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting@freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
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>>>>
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>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
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>>> http://www.freeswitchsolutions.com
>>>
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>>
>>
>> ____________________________________________________________
>> _____________
>> Professional FreeSWITCH Consulting Services:
>> consulting@freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
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>>
>
> ᐧ
>
> ____________________________________________________________
> _____________
> Professional FreeSWITCH Consulting Services:
> consulting@freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
> switch-users
> http://www.freeswitch.org
>
____________________________________________________________
_____________
Professional FreeSWITCH Consulting Services:
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Official FreeSWITCH Sites
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____________________________________________________________
_____________
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http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
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_____________
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Official FreeSWITCH Sites
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_________________________________________________________________________
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Official FreeSWITCH Sites
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