similar to: Autodialer - call simultaneously to both ends

Has anyone here implemented "Ring back when free" in Asterisk? The way it works in the UK is as follows: 1. A calls B. B is engaged (busy). 2. A hears "The number you called is busy. To use ringback, press 5" 3. A presses 5, and hears "Your ringback request has been accepted". 4. A hangs up. 5. Later, B hangs up. The system then calls A (if A is now busy, it

Good afternoon list. I am experiencing a problem with the CDR and callfiles. What is happening is this: When generating a call with a callfile, everything works perfectly, but the CDR is recorded in the table when they answer the call destination. The field disposition is being recorded correctly, but the duration field is marked with the ring time and billsec is marked with 0. This just happens

Hi there, I?ve been looking to do an outbound dialer for systems alerting, etc. and have in large part followed the recipe here: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out That and the associated pages at voip-info give a basic set of recipes for callfiles, but nowhere there or in my copy of the O?Reilly book by Meggelen, Madsen, & Smith can I find a detailed

Hi Everybody I've a requirement for one of my operators for an autodialler for which i plan to deploy asterisk (I already have 3 asterisk servers on PRI running very well ! ). The scene is like : Asterisk will call a customer and play a prompt that prompts him to press 1 if he wishes to talk to an agent , If the customer presses 1 then the call gets connected to one of my proffessional agents

I'm looking for an easy way to make asterisk perform as a basic (broadcast)autodialer. Basically all I want to do is give it a list of phone #'s and a pre-recorded message and have it call each one and play the message or leave it on the person's answering machine. The people I'll be calling are all our customers, etc. so I don't need to do any do-not-call checking. Just

Hi Group, I have following case scenario. Through call file, Asterisk makes a call to SIP extension. When Extension answers the call, Asterisk reads customer numbers (set in callfile) and calls them one by one untill one of the customers answeres the call. Here customer and SIP extension gets patched and talk to each other. Now if outgoing call is answered by Answering machine,I don't want

Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is

I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/5551212 at provider Variable: destination=SIP/8675309 at provider Callerid: 5551212 Context: default ActionID: 849120

On Tuesday 22 September 2020 at 13:27:27, Joshua C. Colp wrote: > On Tue, Sep 22, 2020 at 7:37 AM Antony Stone wrote: > > Hi. > > > > (Asterisk 16.2.1) > > > > I'm using AMI Originate to initiate calls, and I'm passing some > > additional data in to the dialplan context using the Variable: > > parameter. Works fine. > > > >

Hi, I copied a setup from an older 1.8.5 installation to an 11.15 installation, and I'm having problems getting call files to work. Here is the extension setup I'm using: [outbound-swift] exten => _[a-zA-Z].,1,Answer exten => _[a-zA-Z].,n,Playback(AAA/check_ip_failure) ;exten => _[a-zA-Z].,1,Swift("${EXTEN}") exten => _[a-zA-Z].,n,Goto(1) [mis-phone] exten =>

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), . . . The softphones are mostly on machines without proper sound hardware (no mics, no speakers/headsets); This is partly because the workforce is quite conservative in what they want to use :) meaning handsets are important; As the handsets have

Good afternoon, I'm trying to write a simple callback context, but i need to hangup an incoming call and then call the origin number back, the problem is that asterisk stops processing the call after Hangup() application then it is not able to dial the origin number back. Sorry for the grammatical erros. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone know of any plans to add an intercom/all-page feature in *? The few SIP phones that have auto-answer capability would be better if Asterisk could broadcast one leg of a channel to many legs at one time. Thank you, Steve Maroney

Hi all- I have asterisk running on 2 systems, with four E1 spans each. Each system is connected to a (big) NT DMS-100 switch. For load testing an IVR system running on one of the asterisk systems, I'd like to use the other system to generate a lot of outbound calls under program control - on most or all of its channels simultaneously. All of the asterisk dialplan and agi programming

I am looking to see if anyone has any info on auto dialer software that connects directly to a voip provider without using any third party boards or digium cards? I've been building dialers for the past 5 years and I want to get out of using add on cards and just make calls from the software directly using voip. The software would need reporting features, answering machine detection, hangup

Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system & restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository:

Hello, I am using Asterisk 1.4 version. How to dial multiple numbers per second through asterisk manager???? Thanks and regards

Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890

Hello, I'm currently developing a simple asterisk application using SFS (Skype For SIP) which tries to call to an outbound number, play a message and read DMTF digits. My first approach used the Manager to originate calls and then called an agi script to deal with the rest. Anyway, this ended up being not so clear because the call did not start on the Originate extension that it was supposed

Not sure if anybody noticed, but it seems like Skype For Asterisk is out. $66 per channels, pretty pricey http://store.digium.com/productview.php?product_code=1SFA0001 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090817/cd8c6546/attachment.htm