[asterisk-users] broadcast

Gohar Ahmed gohar.ahmed at vopium.com
Tue Sep 13 02:06:44 CDT 2011


Hey there

You are not moving the call file to spool/outgoing directory. Maybe that's
why you aren't getting anything. I don't feel good about the call file also.
Its not doing what you want it to do.

 

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of virendra bhati
Sent: Tuesday, September 13, 2011 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] broadcast

 

Hi List,

I make a script for .call file and then I started playback on local channel
but nothing was hearing at another channles.

exten => 1234,1,Answer()
exten => 1234,n,System(echo -e "Channel: Channel:
local/23 at contest-call\\nContext: contest-call\\nExtension: 23\\nPriority: 1"
> /tmp/${UNIQUEID}.call)
exten => 1234,n,Konference(43689956,ADMRSTVL)

[contest-call]

exten => _X!,1,Answer()
exten => _X!,n,Set(p="/var/spool/asterisk/monitor/")
exten => _X!,n,playback(${p}/LQA/12/Biology/Que3)
exten => _X!,n,playback(${p}/LQA/12/Biology/Que4)
exten => _X!,n,playback(${p}/LQA/12/Biology/Que5)
exten => _X!,n,playback(${p}/LQA/12/Biology/Que6)
exten => _X!,n,playback(${p}/LQA/12/Biology/Que7)
exten => _X!,n,Konference(43689956,ADMRSTV)
exten => _X!,n,Wait(10)
exten => _X!,n,Hangup()

in it I am dialing 1234 from softphone then join to conf in mute mode, after
it .call file start playback at it's own channels but I am not able to hear
anything into conf.

As i know localdial is not joining into the conf. but how I will do it so
that I will be able to hear any played file into conference ?

 

On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind <govoiper at gmail.com> wrote:

Good to know,

 

I think it'll be a feedback score or a poll from members of the conference.
So if you use the R option and collect DTMF from members, and an AMI script
listening to that particular DTMF event collects all. This way your AMI
listener script should be able to tell you at the end of poll what user
inserted with DTMF.

 

So overall insertion of a broadcast message using Ahmed's method of .call
file and later on collecting DTMF events from AMI script should
theoretically work for you. 

 

On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati <virbhati at gmail.com> wrote:

Hi Sam,

You are right. I am looking for the same 

 

On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind <govoiper at gmail.com> wrote:

IMHO, I think Bhaati is trying to get feedback from multiple conference
users. See DTMF options in Konference module. 

 'R' : enable DTMF relay: DTMF tones generate a manager event 
 If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
members in the conference

 

While some file is played and users press any DTMF collect the AMI events
from each user and use them as you require.

 

Ref: http://main.voiptoday.org/index.php?option=com_content
<http://main.voiptoday.org/index.php?option=com_content&view=article&id=566:
asterisk-conferencing-module-appkonference-16-is-now-available&catid=35:gene
ral&Itemid=173>
&view=article&id=566:asterisk-conferencing-module-appkonference-16-is-now-av
ailable&catid=35:general&Itemid=173

 

 

On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati <virbhati at gmail.com> wrote:

Hi Ahmed,

Konference is also an conferencing application.

On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed <gohar.ahmed at vopium.com> wrote:

Hhhmmm..I dunt have any experience with module Konference. Maybe anyone else
can help you on that. 

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of virendra bhati
Sent: Monday, September 12, 2011 1:28 PM


To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] broadcast

 

Hi Ahmed,

I did the same thing earlier to test the load of Digium card. But this time
I want to play file and want to get some DTMF from all the members of
conference.

So in this case I need more control into Konference module. But when I use
.call files then control will not go longer with all events.

Is there any alternate way to do it? 

I appreciate your suggestion and will doing in parallel at higher priority

On Mon, Sep 12, 2011 at 12:33 PM, Gohar Ahmed <gohar.ahmed at vopium.com>
wrote:

Make a .call file..join one leg to local extension which plays the file and
the other leg to conference. The local extension will be like a conference
member.

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of virendra bhati
Sent: Monday, September 12, 2011 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] broadcast

 

Hi List,

Is there any way by which I can broadcast any audio file to all members into
the conference ?
I don't want to play file individual channels.

-- 




-----
Thanks and regards

 Virendra Bhati
+91-9172341457 <tel:%2B91-9172341457> 
Software Engineer

 


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

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




-- 




-----
Thanks and regards

 Virendra Bhati
+91-9172341457 <tel:%2B91-9172341457> 
Software Engineer

 


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

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




-- 




-----
Thanks and regards

 Virendra Bhati
+91-9172341457 <tel:%2B91-9172341457> 
Software Engineer

 


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

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




-- 




-----
Thanks and regards

 Virendra Bhati
+91-9172341457 <tel:%2B91-9172341457> 
Software Engineer

 


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

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




-- 




-----
Thanks and regards

 Virendra Bhati
+91-9172341457
Software Engineer

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110913/0e53c678/attachment.htm>


More information about the asterisk-users mailing list