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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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@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 <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 <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 <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
-- _____________________________________________________________________ -- 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