[asterisk-users] broadcast

Sam Govind govoiper at gmail.com
Tue Sep 13 02:04:39 CDT 2011 

Virendra,
you need to change your logic just a bit. in call file a Channel is one
which needs to be dialled fires (See
link<http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out>).
this will be an extension where your Konference is Hosted for all the other
callers to join. i.e *Channel: local/s at Konference*

[Konference]
exten => s,1,ANSWER()
exten => s,n,if(conference is already started//do nothing else: trigger the
system command to make a call file...don't forget to move it to outgoing
directory)
exten => s,n,SET(some thing else you need to set for each incoming call i.e
save CallerID etc)
exten => s,n(message),Konference(43689956,ADMRSTV)
exten => s,n,Hangup()

Note that the call file should be triggered only for the first caller and
not every time a participant joins in. That'll case overlap message
broadcasts.

Next thing in call file is the destination which will be playing broadcast
message once Konference is called.

*Context:*broadcast-message
*Extension: *s
*Priority: *1
*
*
[broadcast-message]
exten => s,1,Answer()
exten => s,n,Set(p="/var/spool/asterisk/monitor/")
exten => s,n,playback(${p}/LQA/12/Biology/Que3)
exten => s,n,playback(${p}/LQA/12/Biology/Que4)
exten => s,n,playback(${p}/LQA/12/Biology/Que5)
exten => s,n,playback(${p}/LQA/12/Biology/Que6)
exten => s,n,playback(${p}/LQA/12/Biology/Que7)
exten => s,n,Wait(10)
exten => s,n,Hangup()

This should work and konference should listen to the playbacks.

Regards,
Sammy.

On Tue, Sep 13, 2011 at 11:25 AM, virendra bhati <virbhati at gmail.com> wrote:

> 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&view=article&id=566:asterisk-conferencing-module-appkonference-16-is-now-available&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
>>>>>> 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
>>>>>> 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
>>>>> 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
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>>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>>
>>>
>>> --
>>>
>>>
>>>
>>> -----
>>> Thanks and regards
>>>
>>>  Virendra Bhati
>>> +91-9172341457
>>> 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
>
>
> --
> _____________________________________________________________________
> -- 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
>
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