Re: [asterisk-users] Play audio file for both Caller and Callee in a call
Dear Danny: How can you use Playback in the middle of 2 channels engaged in a conversation?ThanksC. Savinovich Original Message Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in a call From: "Danny Nicholas" da...@debsinc.com Date: Thu, December 15, 2011 9:31 am To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.com Playback? What flavor of Asterisk are you using?From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ISABEL ORDAS ARNALSent: Thursday, December 15, 2011 10:29 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Play audio file for both Caller and Callee in a callDear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee?A(x) of Dial application only plays audio for callee. I don’t want to use MeetMe because I want to use Monitor and MixMonitor. Thank you!Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo.This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at.http://www.tid.es/ES/PAGINAS/disclaimer.aspx-- _ -- 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
Re: [asterisk-users] AMI: anything to glue originate to events?
Excuse me if I am off the mark here, I don't have the chance to read too well into your post. But if it is what I think it is, I remember I had a similar situation a few years ago, and I ended up having to create an internal table in my code, so that I could keep track of the channel ids + action ids .Please never mind if it is something elseCS
Original Message
Subject: Re: [asterisk-users] AMI: anything to glue originate to
events?
From: "giovanni.v" i...@keybits.org
Date: Thu, November 17, 2011 11:22 am
To: asterisk-users@lists.digium.com
On 17/11/2011 13.11, Yaroslav Panych wrote:
exten = 384087,1,UserEvent(LinkOriginate,CHANNEL:${CHANNEL(name),ACTIONID:${ActionID}}
UserEvent application will generate event into AMI in form
Event: LinkOriginate
CHANNEL: channle-name (channel id created by asterisk)
ACTIONID: FFA02C6A03 (action id you set in originate)
Thanks Yaroslav,
unfortunately this doesn't add much because only generate another event
but nothing will propagate to Originate child events.
Also the event is fired in asynchronous mode... usually reported later
in the middle of other events when almost of the call flow was already
processed.
Thanks anyway.
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Re: [asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk?
I am writing to you privately because I am an asterisk consultant and if you need any help I can help you for a fee. I have worked with dialogic cards for several years, until I kicked them out my life when Intel bought Dialogic J Having said that however, these are my thoughts: You have to switch your thought frame when you go from dialogic to asterisk. Although asterisk supports dialogic drivers, the entire frame is moot, and the support is in reality just marketing plot, since IVR programming options from asterisk are just different. Beginning with the fact that dialogic's call handling is hardware based and asterisk's is software based. In short, the dialogic card will only end up being an interface card, and all the programming logic will have to be rewritten. In any case, anything I can do for you guys, just ask C. Savinovich From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of AMARDEEP SINGH Sent: Monday, July 05, 2010 7:29 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk? Hello all Asterisk Users, This is my first post here. We are in a process of moving Dialogic 240/JCT-T1 from old voicemail server to Asterisk box. Which card drivers do we need? Please share experience if anyone have successfully configured Dialogic JCT-T1 card with asterisk? Only source proves that this card work with * http://lists.digium.com/pipermail/asterisk-dev/2003-April/000244.html There's not enough info on Dialogic and Asterisk based forums/mailing-list. and Dialogic boards documentation is based on Windows. Thanks: Amardeep -- _ -- 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
Re: [asterisk-users] sip server
Yes CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mohamed daif Sent: Monday, June 28, 2010 2:49 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip server Hi, Can i use asterisk as sip server for manage call Transmission between gateways Best Regards -- _ -- 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
