[asterisk-users] broadcast
virendra bhati
virbhati at gmail.com
Tue Sep 13 01:25:06 CDT 2011
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
>>>
>>> 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
>
> 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/c78348b7/attachment.htm>
More information about the asterisk-users
mailing list