klejch+freeswitch at n...
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 Posted: Tue Sep 22, 2009 10:58 am Post subject: [Freeswitch-users] Unable to set internal call to registered |
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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:
| Quote: | 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@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@Bigfish> 2009-09-22 13:50:29.367114 [NOTICE]
switch_channel.c:602 New Channel sofia/internal/24@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@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@192.168.1.34) Ended
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close
Channel sofia/internal/24@192.168.1.34 [CS_DESTROY]
thanks again for your help ...
regards,
Filip
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Vladimir `KLEO' Klejch Kleo'at'netbox.cz
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