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

2009-09-22 Thread 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/${dialed_extensi...@${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/5...@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:

freeswi...@bigfish 2009-09-22 13:50:29.367114 [NOTICE] 
switch_channel.c:602 New Channel sofia/internal/2...@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/2...@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/2...@192.168.1.34) Ended
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close 
Channel sofia/internal/2...@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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Re: [Freeswitch-users] Unable to set internal call to registered sip user

2009-09-22 Thread Tihomir Culjaga
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/5...@x.x.x.x/
--

T.


On Tue, Sep 22, 2009 at 1:52 PM, Filip Lyncker lync...@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/${dialed_extensi...@${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/5...@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:

 freeswi...@bigfish 2009-09-22 13:50:29.367114 [NOTICE]
 switch_channel.c:602 New Channel sofia/internal/2...@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/2...@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/2...@192.168.1.34) Ended
 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close
 Channel sofia/internal/2...@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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 FreeSWITCH-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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Re: [Freeswitch-users] Unable to set internal call to registered sip user

2009-09-22 Thread Filip Lyncker

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/${dialed_extensi...@${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/5...@x.x.x.x/ --

 T.


 On Tue, Sep 22, 2009 at 1:52 PM, Filip Lyncker lync...@lyth.de 
 mailto:lync...@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/${dialed_extensi...@${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/5...@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:

 freeswi...@bigfish 2009-09-22 13:50:29.367114 [NOTICE]
 switch_channel.c:602 New Channel sofia/internal/2...@192.168.1.34
 mailto:2...@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/2...@192.168.1.34 mailto:2...@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/2...@192.168.1.34 mailto:2...@192.168.1.34) Ended
 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close
 Channel sofia/internal/2...@192.168.1.34 mailto:2...@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



 ___
 FreeSWITCH-users mailing list
 

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

2009-09-22 Thread 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/${dialed_extensi...@${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/${dialed_extensi...@${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/5...@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:

 freeswi...@bigfish 2009-09-22 13:50:29.367114 [NOTICE]
 switch_channel.c:602 New Channel sofia/internal/2...@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/2...@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/2...@192.168.1.34) Ended
 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close
 Channel sofia/internal/2...@192.168.1.34 [CS_DESTROY]

 thanks again for your help ...


 regards,

 Filip




-- 
_
|  You have moved the mouse.  #
| Windows must be restarted for the changes to take effect.   #
|  OK   #
##/

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



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