Re: [asterisk-users] 3-way calling for IAX channels

2008-07-23 Thread Chento Arohuanca
We are developing an softphone based on IAX client version 1.2 (my current
SIP softphone has many eoors), but it doesn´t have a specific function for
Conferencing (3-way calling) or to place the other party on HOLD.

I´m trying to do it through the PBX because our softphone´s lack of
functions. I´ll be gratefull for further comments.

Thanks again,

Daniel

On Tue, Jul 22, 2008 at 11:49 PM, Noah Miller [EMAIL PROTECTED]
wrote:

 Hi Daniel -

  There is no way to enable it at the softphone itself? As is the case for
  hardphones like my Polycom.

 A phone can definitely do conference mixing.  As you asked about IAX
 channels on the asterisk-users list, I assumed you were asking about
 how to do this in asterisk.

 My experience with IAX softphones is somewhat limited, but maybe if
 you indicate which phone you're using, somebody could provide you with
 assistance.


 - Noah



  Daniel
  On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller [EMAIL PROTECTED]
  wrote:
 
  Hi Daniel -
 
   How can I made a 3-way conference betwwen IAX channels?
   My current version is: 1.4.21.1
 
  Anytime you need a call with more than 2 parties, you need to use some
  kind of conferencing application.  The default conference
  application for asterisk is meetme. You can use meetme with any kind
  of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc).  Just use the meetme()
  application in extensions.conf, and create your conference rooms in
  meetme.conf
 
 
  - Noah
 
  ___
  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 
  AstriCon 2008 - September 22 - 25 Phoenix, Arizona
  Register Now: http://www.astricon.net
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
  ___
  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 
  AstriCon 2008 - September 22 - 25 Phoenix, Arizona
  Register Now: http://www.astricon.net
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2008 - September 22 - 25 Phoenix, Arizona
 Register Now: http://www.astricon.net

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 3-way calling for IAX channels

2008-07-23 Thread MFH
Asterisk supports conferencing without using meetme.  In this case you 
don't have a central dial in number but a single extension can initiate 
the conference call.  Generally this is done the same way as with 
traditional PSTN service which is that while on a call between two 
parties, flash the line, dial out to the third party then flash again 
and all the parties should be connected.

Noah Miller wrote:
 Hi Daniel -

   
 How can I made a 3-way conference betwwen IAX channels?
 My current version is: 1.4.21.1
 

 Anytime you need a call with more than 2 parties, you need to use some
 kind of conferencing application.  The default conference
 application for asterisk is meetme. You can use meetme with any kind
 of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc).  Just use the meetme()
 application in extensions.conf, and create your conference rooms in
 meetme.conf


 - Noah

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2008 - September 22 - 25 Phoenix, Arizona
 Register Now: http://www.astricon.net

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
   

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] 3-way calling for IAX channels

2008-07-23 Thread Steve Davies
2008/7/23 MFH [EMAIL PROTECTED]:
 Noah Miller wrote:
 Hi Daniel -


 How can I made a 3-way conference betwwen IAX channels?
 My current version is: 1.4.21.1


 Anytime you need a call with more than 2 parties, you need to use some
 kind of conferencing application.  The default conference
 application for asterisk is meetme. You can use meetme with any kind
 of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc).  Just use the meetme()
 application in extensions.conf, and create your conference rooms in
 meetme.conf

 Asterisk supports conferencing without using meetme.  In this case you
 don't have a central dial in number but a single extension can initiate
 the conference call.  Generally this is done the same way as with
 traditional PSTN service which is that while on a call between two
 parties, flash the line, dial out to the third party then flash again
 and all the parties should be connected.

I believe that response is slightly misleading - Asterisk does not
support conferencing without using meetme, but Zaptel/DAHDI will
emulate the PSTN flash/recall facility which looks a bit like a
conference. In SIP, IAX, and I believe all other non Zaptel/DAHDI
channel types, the endpoint must manage the equivalent of a PSTN
flash/recall conference.

Anything cross-channel or otherwise more complex does indeed require
app_meetme. Given that the OP was referring to IAX, I believe they
will need app_meetme.

Of course I could be wrong :)
Steve

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] 3-way calling for IAX channels

2008-07-23 Thread Tilghman Lesher
On Wednesday 23 July 2008 12:17:26 Steve Davies wrote:
 2008/7/23 MFH [EMAIL PROTECTED]:
  Noah Miller wrote:
  Hi Daniel -
 
  How can I made a 3-way conference betwwen IAX channels?
  My current version is: 1.4.21.1
 
  Anytime you need a call with more than 2 parties, you need to use some
  kind of conferencing application.  The default conference
  application for asterisk is meetme. You can use meetme with any kind
  of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc).  Just use the meetme()
  application in extensions.conf, and create your conference rooms in
  meetme.conf
 
  Asterisk supports conferencing without using meetme.  In this case you
  don't have a central dial in number but a single extension can initiate
  the conference call.  Generally this is done the same way as with
  traditional PSTN service which is that while on a call between two
  parties, flash the line, dial out to the third party then flash again
  and all the parties should be connected.

 I believe that response is slightly misleading - Asterisk does not
 support conferencing without using meetme, but Zaptel/DAHDI will
 emulate the PSTN flash/recall facility which looks a bit like a
 conference. In SIP, IAX, and I believe all other non Zaptel/DAHDI
 channel types, the endpoint must manage the equivalent of a PSTN
 flash/recall conference.

 Anything cross-channel or otherwise more complex does indeed require
 app_meetme. Given that the OP was referring to IAX, I believe they
 will need app_meetme.

The interesting thing is that Zaptel/DAHDI is using exactly the same
conferencing/audio mixing engine as app_meetme.  Or more correctly,
app_meetme is using the Zaptel/DAHDI engine for audio mixing.

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] 3-way calling for IAX channels

2008-07-22 Thread Chento Arohuanca
How can I made a 3-way conference betwwen IAX channels?
My current version is: 1.4.21.1

Thanx,

Daniel Arohuanca Lagos
+51 1 3594122
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 3-way calling for IAX channels

2008-07-22 Thread Noah Miller
Hi Daniel -

 How can I made a 3-way conference betwwen IAX channels?
 My current version is: 1.4.21.1

Anytime you need a call with more than 2 parties, you need to use some
kind of conferencing application.  The default conference
application for asterisk is meetme. You can use meetme with any kind
of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc).  Just use the meetme()
application in extensions.conf, and create your conference rooms in
meetme.conf


- Noah

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] 3-way calling for IAX channels

2008-07-22 Thread Chento Arohuanca
Thanks for answering Noah,

There is no way to enable it at the softphone itself? As is the case for
hardphones like my Polycom.

Daniel
On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller [EMAIL PROTECTED]
wrote:

 Hi Daniel -

  How can I made a 3-way conference betwwen IAX channels?
  My current version is: 1.4.21.1

 Anytime you need a call with more than 2 parties, you need to use some
 kind of conferencing application.  The default conference
 application for asterisk is meetme. You can use meetme with any kind
 of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc).  Just use the meetme()
 application in extensions.conf, and create your conference rooms in
 meetme.conf


 - Noah

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2008 - September 22 - 25 Phoenix, Arizona
 Register Now: http://www.astricon.net

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 3-way calling for IAX channels

2008-07-22 Thread Noah Miller
Hi Daniel -

 There is no way to enable it at the softphone itself? As is the case for
 hardphones like my Polycom.

A phone can definitely do conference mixing.  As you asked about IAX
channels on the asterisk-users list, I assumed you were asking about
how to do this in asterisk.

My experience with IAX softphones is somewhat limited, but maybe if
you indicate which phone you're using, somebody could provide you with
assistance.


- Noah



 Daniel
 On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller [EMAIL PROTECTED]
 wrote:

 Hi Daniel -

  How can I made a 3-way conference betwwen IAX channels?
  My current version is: 1.4.21.1

 Anytime you need a call with more than 2 parties, you need to use some
 kind of conferencing application.  The default conference
 application for asterisk is meetme. You can use meetme with any kind
 of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc).  Just use the meetme()
 application in extensions.conf, and create your conference rooms in
 meetme.conf


 - Noah

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2008 - September 22 - 25 Phoenix, Arizona
 Register Now: http://www.astricon.net

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2008 - September 22 - 25 Phoenix, Arizona
 Register Now: http://www.astricon.net

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users