Re: [Freeswitch-users] Skypiax Parameters Informations Request

2009-07-10 Thread Meftah Tayeb

hello Muhammad ,
thank you
what about hig cality audio codec to use?
speex is good?
thanks

Muhammad Shahzad wrote:
Destination parameter actually specifies the extension on which this 
Skype user is reachable within FreeSWITCH dialplan for incoming calls.


If this parameter is specified in per_interface_settings xml tag then 
it will override the value of this parameter in global_settings xml 
tag, otherwise value of this parameter from global_settings xml tag 
will be used.


Here is an example (see below), the user test.01 is reachable on 
dialplan extension 2000 (since it has its own destination defined in 
per_interface_settings xml tag), whereas test.02 is reachable on 
dialplan extension 5000 (since it does not have destination parameter 
defined and thus it will use value for this parameter in 
global_settings xml tag).


  global_settings
param name=debug value=8/
param name=codec-master value=us/
param name=dialplan value=XML/
param name=context value=default/
param name=codec-prefs value=gsm,ulaw/
param name=codec-rates value=8000,16000/
param name=hold-music value=$${moh_uri}/
param name=destination value=5000/
  /global_settings

  per_interface_settings
interface id=1 name=test.01
param name=hold-music value=$${moh_uri}/
param name=dialplan value=XML/
param name=context value=default/
param name=X11-display value=:101/
param name=tcp_cli_port value=15556/
param name=tcp_srv_port value=15557/
param name=skype_user value=test.01/
param name=destination value=2000/
/interface
interface id=2 name=test.02
param name=hold-music value=$${moh_uri}/
param name=dialplan value=XML/
param name=context value=default/
param name=X11-display value=:102/
param name=tcp_cli_port value=15558/
param name=tcp_srv_port value=15559/
param name=skype_user value=test.02/
/interface
  /per_interface_settings


Now the codec, Skype has its own proprietory code for Skype to Skype 
calls. The codec we specify in Skypiax configuration file is actually 
used for Skype to/from non-Skype calls. Consider following dial plan 
example (with skypiax configuration given above),


extension name=skype_incoming-01
  condition field=destination_number expression=^2000$
action application=bridge data=sofia/internal/1000/
  /condition
/extension

If a remote Skype user dials test.01 from his/her Skype client, then 
FreeSWITCH will route this call to SIP user 1000 and codecs specified 
in Skypiax configuration will be offered to destination SIP endpoint 
(SIP user 1000 in this case).


Thank you.


On Fri, Jul 10, 2009 at 5:58 AM, Meftah Tayeb tayeb.mef...@gmail.com 
mailto:tayeb.mef...@gmail.com wrote:


hello,
i have the folowing parameter in Skypiax.conf.xml:
configuration name=skypiax.conf description=Skypiax
Configuration
 global_settings
   param name=destination value=1000/
each call that will to by routed to this destination??

 per_interface_settings
   interface id=1 name=skypiax1
   param name=destination value=1000/

Each Call will to by routed to this destination?

each codecs that is pocible to use it with Skypiax? all? speex?
this codecs is used beetwan skypiax and the remote peer?
thanks


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--
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_78...@hotmail.com mailto:shari_78...@hotmail.com
Email: shaherya...@googlemail.com mailto:shaherya...@googlemail.com


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database 4231 (20090710) __

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

  





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database 4231 (20090710) __

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


Re: [Freeswitch-users] Skypiax Parameters Informations Request

2009-07-10 Thread Muhammad Shahzad
I think you can use it has long as remote end-point supports it.

Thank you.


On Fri, Jul 10, 2009 at 3:48 PM, Meftah Tayeb tayeb.mef...@gmail.comwrote:

  hello Muhammad ,
 thank you
 what about hig cality audio codec to use?
 speex is good?
 thanks

 Muhammad Shahzad wrote:

 Destination parameter actually specifies the extension on which this Skype
 user is reachable within FreeSWITCH dialplan for incoming calls.

 If this parameter is specified in per_interface_settings xml tag then it
 will override the value of this parameter in global_settings xml tag,
 otherwise value of this parameter from global_settings xml tag will be used.

 Here is an example (see below), the user test.01 is reachable on dialplan
 extension 2000 (since it has its own destination defined in
 per_interface_settings xml tag), whereas test.02 is reachable on dialplan
 extension 5000 (since it does not have destination parameter defined and
 thus it will use value for this parameter in global_settings xml tag).

   global_settings
 param name=debug value=8/
 param name=codec-master value=us/
 param name=dialplan value=XML/
 param name=context value=default/
 param name=codec-prefs value=gsm,ulaw/
 param name=codec-rates value=8000,16000/
 param name=hold-music value=$${moh_uri}/
 param name=destination value=5000/
   /global_settings

   per_interface_settings
 interface id=1 name=test.01
 param name=hold-music value=$${moh_uri}/
 param name=dialplan value=XML/
 param name=context value=default/
 param name=X11-display value=:101/
 param name=tcp_cli_port value=15556/
 param name=tcp_srv_port value=15557/
 param name=skype_user value=test.01/
 param name=destination value=2000/
 /interface
 interface id=2 name=test.02
 param name=hold-music value=$${moh_uri}/
 param name=dialplan value=XML/
 param name=context value=default/
 param name=X11-display value=:102/
 param name=tcp_cli_port value=15558/
 param name=tcp_srv_port value=15559/
 param name=skype_user value=test.02/
 /interface
   /per_interface_settings


 Now the codec, Skype has its own proprietory code for Skype to Skype calls.
 The codec we specify in Skypiax configuration file is actually used for
 Skype to/from non-Skype calls. Consider following dial plan example (with
 skypiax configuration given above),

 extension name=skype_incoming-01
   condition field=destination_number expression=^2000$
 action application=bridge data=sofia/internal/1000/
   /condition
 /extension

 If a remote Skype user dials test.01 from his/her Skype client, then
 FreeSWITCH will route this call to SIP user 1000 and codecs specified in
 Skypiax configuration will be offered to destination SIP endpoint (SIP user
 1000 in this case).

 Thank you.


 On Fri, Jul 10, 2009 at 5:58 AM, Meftah Tayeb tayeb.mef...@gmail.comwrote:

 hello,
 i have the folowing parameter in Skypiax.conf.xml:
 configuration name=skypiax.conf description=Skypiax Configuration
  global_settings
param name=destination value=1000/
 each call that will to by routed to this destination??

  per_interface_settings
interface id=1 name=skypiax1
param name=destination value=1000/

 Each Call will to by routed to this destination?

 each codecs that is pocible to use it with Skypiax? all? speex?
 this codecs is used beetwan skypiax and the remote peer?
 thanks


 __ Information from ESET NOD32 Antivirus, version of virus
 signature database 4229 (20090709) __

 The message was checked by ESET NOD32 Antivirus.

 http://www.eset.com



 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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 http://www.freeswitch.org




 --
 Muhammad Shahzad
 ---
 CISCO Rich Media Communication Specialist (CRMCS)
 CISCO Certified Network Associate (CCNA)
 Cell: +92 334 422 40 88
 MSN: shari_78...@hotmail.com
 Email: shaherya...@googlemail.com

 --

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 __ Information from ESET NOD32 Antivirus, version of virus signature 
 database 4231 (20090710) __

 The message was checked by ESET NOD32 Antivirus.
 http://www.eset.com




 __ Information from ESET NOD32 Antivirus, version of virus
 signature database 4231 (20090710) __

 The message was checked by ESET NOD32 Antivirus.

 http://www.eset.com

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 Freeswitch-users 

[Freeswitch-users] Skypiax Parameters Informations Request

2009-07-09 Thread Meftah Tayeb
hello,
i have the folowing parameter in Skypiax.conf.xml:
configuration name=skypiax.conf description=Skypiax Configuration
  global_settings
param name=destination value=1000/
each call that will to by routed to this destination??

  per_interface_settings
interface id=1 name=skypiax1
param name=destination value=1000/

Each Call will to by routed to this destination?

each codecs that is pocible to use it with Skypiax? all? speex?
this codecs is used beetwan skypiax and the remote peer?
thanks


__ Information from ESET NOD32 Antivirus, version of virus signature 
database 4229 (20090709) __

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com



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Re: [Freeswitch-users] Skypiax Parameters Informations Request

2009-07-09 Thread Muhammad Shahzad
Destination parameter actually specifies the extension on which this Skype
user is reachable within FreeSWITCH dialplan for incoming calls.

If this parameter is specified in per_interface_settings xml tag then it
will override the value of this parameter in global_settings xml tag,
otherwise value of this parameter from global_settings xml tag will be used.

Here is an example (see below), the user test.01 is reachable on dialplan
extension 2000 (since it has its own destination defined in
per_interface_settings xml tag), whereas test.02 is reachable on dialplan
extension 5000 (since it does not have destination parameter defined and
thus it will use value for this parameter in global_settings xml tag).

  global_settings
param name=debug value=8/
param name=codec-master value=us/
param name=dialplan value=XML/
param name=context value=default/
param name=codec-prefs value=gsm,ulaw/
param name=codec-rates value=8000,16000/
param name=hold-music value=$${moh_uri}/
param name=destination value=5000/
  /global_settings

  per_interface_settings
interface id=1 name=test.01
param name=hold-music value=$${moh_uri}/
param name=dialplan value=XML/
param name=context value=default/
param name=X11-display value=:101/
param name=tcp_cli_port value=15556/
param name=tcp_srv_port value=15557/
param name=skype_user value=test.01/
param name=destination value=2000/
/interface
interface id=2 name=test.02
param name=hold-music value=$${moh_uri}/
param name=dialplan value=XML/
param name=context value=default/
param name=X11-display value=:102/
param name=tcp_cli_port value=15558/
param name=tcp_srv_port value=15559/
param name=skype_user value=test.02/
/interface
  /per_interface_settings


Now the codec, Skype has its own proprietory code for Skype to Skype calls.
The codec we specify in Skypiax configuration file is actually used for
Skype to/from non-Skype calls. Consider following dial plan example (with
skypiax configuration given above),

extension name=skype_incoming-01
  condition field=destination_number expression=^2000$
action application=bridge data=sofia/internal/1000/
  /condition
/extension

If a remote Skype user dials test.01 from his/her Skype client, then
FreeSWITCH will route this call to SIP user 1000 and codecs specified in
Skypiax configuration will be offered to destination SIP endpoint (SIP user
1000 in this case).

Thank you.


On Fri, Jul 10, 2009 at 5:58 AM, Meftah Tayeb tayeb.mef...@gmail.comwrote:

 hello,
 i have the folowing parameter in Skypiax.conf.xml:
 configuration name=skypiax.conf description=Skypiax Configuration
  global_settings
param name=destination value=1000/
 each call that will to by routed to this destination??

  per_interface_settings
interface id=1 name=skypiax1
param name=destination value=1000/

 Each Call will to by routed to this destination?

 each codecs that is pocible to use it with Skypiax? all? speex?
 this codecs is used beetwan skypiax and the remote peer?
 thanks


 __ Information from ESET NOD32 Antivirus, version of virus
 signature database 4229 (20090709) __

 The message was checked by ESET NOD32 Antivirus.

 http://www.eset.com



 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org




-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
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