[asterisk-users] broadcast
virendra bhati
virbhati at gmail.com
Tue Sep 13 02:54:53 CDT 2011
Hi Sam,
I am doing the same things.
into your suggested script you join into context Konference and then .call
file start IVRs .
the same logic I have pasted in which I make .call file and then join into
the Konference and then .call file start it's work.
But As i know they are on different -2 channels and not joined into same
conference. That's why no audio is able to broadcast into conference.
[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(${p}/LQA/12/Biology/Que7)
exten => s,n,Wait(10)
exten => s,n,Hangup()
Where you have mention in which conf. it will be start ?
miss comunication in between .call and rest users.
On Tue, Sep 13, 2011 at 12:34 PM, Sam Govind <govoiper at gmail.com> wrote:
> Virendra,
> you need to change your logic just a bit. in call file a Channel is one
> which needs to be dialled fires (See link<http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out>).
> this will be an extension where your Konference is Hosted for all the other
> callers to join. i.e *Channel: local/s at Konference*
>
> [Konference]
> exten => s,1,ANSWER()
> exten => s,n,if(conference is already started//do nothing else: trigger the
> system command to make a call file...don't forget to move it to outgoing
> directory)
> exten => s,n,SET(some thing else you need to set for each incoming call i.e
> save CallerID etc)
> exten => s,n(message),Konference(43689956,ADMRSTV)
> exten => s,n,Hangup()
>
> Note that the call file should be triggered only for the first caller and
> not every time a participant joins in. That'll case overlap message
> broadcasts.
>
> Next thing in call file is the destination which will be playing broadcast
> message once Konference is called.
>
> *Context:*broadcast-message
> *Extension: *s
> *Priority: *1
> *
> *
> [broadcast-message]
> exten => s,1,Answer()
> exten => s,n,Set(p="/var/spool/asterisk/monitor/")
> exten => s,n,playback(${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
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>>>>>>> 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
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>>>>>>> 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
>>>>>>
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>>>>>> 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
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>>>> 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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>>>
>>
>>
>>
>> --
>>
>>
>>
>> -----
>> 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
>>
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>> 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
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