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

[Vladimir Klejch]

Hi

in dialplan i see:

<condition field="destination_number" expression="^(2[0-9])$">   -> check
 	on variable destination_number

and later

<action application="bridge" data="user/${dialed_extension}@${domain_name}"></action> ->
 	bridge to variable dialed_extension , other then checked destination_number
 	or $1 from regexp


try:

<action application="bridge" 
data="user/${destination_number}%${sip_profile}"></action>

or

<action application="bridge"
data="user/$1%${sip_profile}"></action>





 		By
 					Kleo


On Tue, 22 Sep 2009, Filip Lyncker 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
> [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
>
>
>

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   	~~	~~	~~	~~	~~	~~	~~
Vladimir `KLEO' Klejch     	Kleo'at'netbox.cz
 		...	...	...	...