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

Filip Lyncker lyncker at lyth.de
Tue Sep 22 06:55:09 PDT 2009


ok , i tried several things :
<action application="bridge" 
data="user/${dialed_extension}%${domain_name}"></action>
<action application="bridge" 
data="user/${dialed_extension}%192.168.1.34"></action>
<action application="bridge" data="user/${dialed_extension}"></action>


but all this doesnt work sorry mybe I dont see something apparent , but 
I dont have a clue...




Tihomir Culjaga schrieb:
> and this is not enough for you?
>
>       <!--- 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}"/>
>       <!--- 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"/> -->
>
> T.
>
>
> On Tue, Sep 22, 2009 at 1:52 PM, Filip Lyncker <lyncker at lyth.de 
> <mailto:lyncker at lyth.de>> wrote:
>
>     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
>     <mailto: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 <http://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 <mailto: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 <mailto: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 <mailto:24 at 192.168.1.34>
>     [CS_DESTROY]
>
>     thanks again for your help ...
>
>
>     regards,
>
>     Filip
>
>
>     --
>     _________________________________
>     Filip Lyncker, Dipl.-Inform. (FH)
>
>
>     Lyncker & Theis GmbH
>     Wilhelmstr. 16
>     65185 Wiesbaden
>     Germany
>
>     Fon +49 611/9006951
>     Fax +49 611/9406125
>
>
>     Handelsregister: HRB 23156 Amtsgericht Wiesbaden
>     Steuernummer: 4023897051
>     USt-IdNr.: DE255806399
>
>     Geschäftsführer:
>     Filip Lyncker,
>     Armin Theis
>
>
>
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>
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-- 
_________________________________
Filip Lyncker, Dipl.-Inform. (FH)


Lyncker & Theis GmbH
Wilhelmstr. 16
65185 Wiesbaden
Germany

Fon +49 611/9006951
Fax +49 611/9406125


Handelsregister: HRB 23156 Amtsgericht Wiesbaden
Steuernummer: 4023897051
USt-IdNr.: DE255806399

Geschäftsführer:
Filip Lyncker,
Armin Theis 






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