almost there, but the ../sip_profiles ../public ../default ../dialplan directories have me confused, especially since all these files 'look the same'. (almost)
I must REALLY be dense:

if I just want to bridge (since the provider won't ever send me anything but the did's they are getting paid for):

so: just TWO files?

cd /usr/local/freeswitch/conf

in ../dialplan/public/from_provider.xml I have: (do I need <include> ..... </include> if this in in a file? tried both ways.)

<extension name="from_provider">
   <condition field="network_addr" expression="68.100.226.97"/>
   <condition field="destination_number" expression="^(.*)$">
       <action application="bridge" data="sofia/gateway/sipxecs/$1"/>
   </condition>
</extension>

and in ../sip_profiles/external/sipxecs.xml I have: (I don't need authentication/registration).. im local :-)

<include>
    <gateway name="sipxecs">
      <param name="username" value="username"/>
      <param name="password" value="password"/>
      <param name="from-user" value="fromuser"/>
      <param name="from-domain" value="192.168.0.2"/>
      <param name="proxy" value="192.168.0.2"/>
      <param name="expire-seconds" value="600"/>
      <param name="register" value="false"/>
      <param name="retry-seconds" value="30"/>
      <param name="context" value="public"/>
    </gateway>
</include>

I must really be an idiot, because now, I don't even get anything in the logs.  tcpdump shows INVITES from the provider.
(I added <include /> to the from_provider.xml just to see)

../log/freeswitch.log doesn't show anything after the restart stuff.  and when I had it wrong, at least I logged the trys and 480 'try later' error.

taking a break for breakfast.. its 6am. back in 2 hours.

I want a sample setup.
I dial the 301 number, come in freeswitch, is gatewayed to my sipxecs.  I get audio!  I get DTMF, I can internally transfer,
(oh, and when I finally get this to work. how do I send this to port 5080 on sipx? so I can transfer from AA and/ or CFNA?)  I don't need the outbound gateway (yet)

(to complicate matters, this is the freeswitch package on pfsense.  no, not the 'new one', but the one I can install via the pfsense gui.  and restarts are slow)


On 8/26/10 1:49 AM, jesse wrote: 
I feel Michael's idea should also work:  If call is from X.X.X.X, then
bridge the call with the same
destination_number ($1) to sipxecs gateway.

<extension name="from_provider">
    <condition field="network_addr" expression="X.X.X.X"/>
    <condition field="destination_number" expression="^(.*)$">
      <action application="bridge" data="sofia/gateway/sipxecs/$1"/>
    </condition>
  </extension>


On Wed, Aug 25, 2010 at 4:33 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
  
It is right.
You just transfer to another dialplan (context) in order to make it more
readable.
You'll be able to add all your rules in the provider context, because only
calls coming from provider X.X.X.X will go into this dialplan.
Remember: extension name has absolutely NO importance in FS. It's just a
label.
So to sum it all:
if call from X.X.X.X, transfer to context provider, with the same
destination_number ($1).
In context provider, if destination_number is XXX, then bridge the call to
sipxecs.
David Ponzone  Direction Technique
email: david.ponzone@ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34
Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

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