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(${p}/LQA/12/Biology/Que7) exten => s,n,Wait(10) exten => s,n,Hangup() Where you have mention in which conf. it will be start ? miss comunication in between .call and rest users. On Tue, Sep 13, 2011 at 12:34 PM, Sam Govind <govoi...@gmail.com> wrote: > Virendra, > you need to change your logic just a bit. in call file a Channel is one > which needs to be dialled fires (See > link<http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out>). > this will be an extension where your Konference is Hosted for all the other > callers to join. i.e *Channel: local/s@Konference* > > [Konference] > exten => s,1,ANSWER() > exten => s,n,if(conference is already started//do nothing else: trigger the > system command to make a call file...don't forget to move it to outgoing > directory) > exten => s,n,SET(some thing else you need to set for each incoming call i.e > save CallerID etc) > exten => s,n(message),Konference(43689956,ADMRSTV) > exten => s,n,Hangup() > > Note that the call file should be triggered only for the first caller and > not every time a participant joins in. That'll case overlap message > broadcasts. > > Next thing in call file is the destination which will be playing broadcast > message once Konference is called. > > *Context:*broadcast-message > *Extension: *s > *Priority: *1 > * > * > [broadcast-message] > exten => s,1,Answer() > exten => s,n,Set(p="/var/spool/asterisk/monitor/") > exten => s,n,playback(${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 <virbh...@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@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 <govoi...@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 <virbh...@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 <govoi...@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 <virbh...@gmail.com>wrote: >>>>> >>>>>> Hi Ahmed, >>>>>> >>>>>> Konference is also an conferencing application. >>>>>> >>>>>> On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed >>>>>> <gohar.ah...@vopium.com>wrote: >>>>>> >>>>>>> Hhhmmm..I dunt have any experience with module Konference. Maybe >>>>>>> anyone else can help you on that. **** >>>>>>> >>>>>>> ** ** >>>>>>> >>>>>>> *From:* asterisk-users-boun...@lists.digium.com [mailto: >>>>>>> asterisk-users-boun...@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.ah...@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-boun...@lists.digium.com [mailto: >>>>>>> asterisk-users-boun...@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 >> >> >> -- >> _____________________________________________________________________ >> -- 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