[Freeswitch-users] Unable to set internal call to registered sip user

[Filip Lyncker]

Dear List,

I read the documentation, but Im still confused about how to dial a 
internal registered sip user.

I configured the both sip phones in the directory in my local.xml file :

<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>

It seems, that they can connect to the freeswitch.

I configured the dialplan like following :

<include>
  <context name="default"> 
   <extension name="diallocal">
   <condition field="destination_number" expression="^(2[0-9])$">
       <!--- The % behind the username tells FS to lookup the user in 
it's local sip_registration database -->
      <action application="bridge" 
data="user/${dialed_extension}@${domain_name}"></action>
       <!--- x.x.x.x in the line above is the IP address to the 
FreeSWITCH server/device -->
       <!--- If you don't want to bridge a call to a local registered 
user, but to a SIP URI, use the @ instead of %:
       <action application="bridge" 
data="sofia/profilename/500 at x.x.x.x"/> -->
   </condition> 
   </extension>
...


If I call from the sip user 24 to 22 , freeswitch logs the following and 
gives an busy tone immediately:

freeswitch at Bigfish> 2009-09-22 13:50:29.367114 [NOTICE] 
switch_channel.c:602 New Channel sofia/internal/24 at 192.168.1.34 
[decc119c-a973-6b4c-bf11-ec251c653cda]
2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing 
24->22 in context default
2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find user 
[@192.168.1.34]
2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot 
create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed.  
Cause: SUBSCRIBER_ABSENT
2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup 
sofia/internal/24 at 192.168.1.34 [CS_EXECUTE] [SUBSCRIBER_ABSENT]
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session 
13 (sofia/internal/24 at 192.168.1.34) Ended
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close 
Channel sofia/internal/24 at 192.168.1.34 [CS_DESTROY]

thanks again for your help ...


regards,

Filip


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


Lyncker & Theis GmbH
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