[Freeswitch-users] Some Newbie questions about dialplan and local Sip registration

18 Sep 2009

      Hi List,

for the first experiments with freeswitch I downloaded the Windows 
installation.
Now Im trying to get my 2 Sipphones get connected to. Later I want 
connect the freeswitch to my asterisk gateway.

I find the examples pretty complex therfore Im trying to build up a 
simple solution to understand the functions from the scratch ..

my current problem is , that I cant route my local sips to each other ( 
registration seems to work now).
the next is , that freeshwitch is not able to connect to asterisk. but I 
will describe this later.

I installed in the Directory a xml file ( called 22.xml) with the 
following content :

<include>
<domain name="$${domain}">
  <user id="22" mailbox="22">
    <params>
      <param name="password" value="Xk21%"></param>
      <param name="vm-password" value="22"></param>     
      <param name="sip-port" value="5060"></param>

    </params>
    <variables>
      <variable name="accountcode" value="22"></variable>
      <variable name="user_context" value="default"></variable>
      <variable name="effective_caller_id_name" value="Extension 
22"></variable>
      <variable name="effective_caller_id_number" value="22"></variable>
    </variables>
  </user>
  <user id="24" mailbox="24">
    <params>
      <param name="password" value="dudeldum"></param>
      <param name="vm-password" value="24"></param>     
      <param name="sip-port" value="5060"></param>

    </params>
    <variables>
      <variable name="accountcode" value="24"></variable>
      <variable name="user_context" value="default"></variable>
      <variable name="effective_caller_id_name" value="Extension 
24"></variable>
      <variable name="effective_caller_id_number" value="24"></variable>
    </variables>
  </user>  
  </domain>
</include>

This seems to be ok now. Now I want to dial from 22 to 24 , wherefore I 
configured this dialplan :

<include>
  <context name="any">   
   <condition field="destination_number" expression="^(2[0-9])$">

      <action application="bridge" data="user/${dialed_extension}"/>

   </condition>
</include>

wich doesnt work , mybe b/c the user/${dialed_extension} I dont know...
Freeswitch says:
[INFO] switch_core_state_machine.c:136 No Route, Aborting
[NOTICE] switch_core_state_machine.c:137 Hangup 
sofia/internal/24@192.168.1.34 [CS_ROUTING] [NO_ROUTE_DESTINATION]
[NOTICE] switch_core_session.c:1086 Session 17 
(sofia/internal/24@192.168.1.34) Ended
[NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/24@192.168.1.34 [CS_DESTROY]

Im sure , for you guys this cant be a big deal;)

Next Point is my Asterisk registration , mybe you can help me out here 
to .. :

In the sip-profiles/external I installed the my_asterisk.xml with that 
content :

<include>
  <gateway name="asterisk">
    <param name="username" value="28"></param>
    <param name="password" value="test"></param>
    <param name="realm" value="28"></param>
    <param name="proxy" value="192.168.1.119"></param>
    <param name="register" value="true"></param>
  </gateway>
</include>

Freeswitch allways complains a timeout error :
 [ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request 
Timeout [408]. failure #17
 [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry 
to 540 seconds.

it seems that It cant connect ( I also tried out to explicit set the 
port to 5060 b/c I read something about 5080 .. : <param name="sip-port" 
value="5060"></param> but this didnt help)
In my Asterisk I set in the sip.conf the entry 28 with the pw test ....

If someone could help me with my first steps I would be verrry thankful ;))

cheers

Filip

-- 
_________________________________
Filip Lyncker, Dipl.-Inform. (FH)

Lyncker & Theis GmbH
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