Priyaranjan Nayak
2018-Sep-17 13:42 UTC
[asterisk-users] IVR call simulation on Asterisk 15 server
Hi All, I am using Asterisk 15 server and wanted to configure IVR call simulation. My configuration scenario is 1. A subscriber will register to Asterisk server and start a call. 2. The IVR audio will come from the Asterisk sever to sbscriber. 3. Once the subscriber pressed the botton, the call will connect to a number based on DTMF digit pressed by subscriber. Then call will continue for 30 seconds. I observered for normal call pjsip.conf file is used for configuration of a subscribers. Could you please help me on below queries ? 1. Which file we need to configure for the IVR call simulation ? 2. Please suggest a good documentation for IVR simulation. Best regards, Priyaranjan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180917/10980caa/attachment.html>
Antony Stone
2018-Sep-17 13:47 UTC
[asterisk-users] IVR call simulation on Asterisk 15 server
On Monday 17 September 2018 at 15:42:50, Priyaranjan Nayak wrote:> Hi All, > > I am using Asterisk 15 server and wanted to configure IVR call simulation.What do you mean by "simulation"?> My configuration scenario is > 1. A subscriber will register to Asterisk server and start a call. > 2. The IVR audio will come from the Asterisk sever to sbscriber. > 3. Once the subscriber pressed the botton, the call will connect to a > number based on DTMF digit pressed by subscriber. Then call will continue > for 30 seconds. > > I observered for normal call pjsip.conf file is used for configuration of a > subscribers. > > Could you please help me on below queries ? > 1. Which file we need to configure for the IVR call simulation ? > 2. Please suggest a good documentation for IVR simulation.Do you actually want to create an IVR system, or do you just want the caller to be answered by a pre-recorded menu, and it doesn't matter what digit they press, they always get connected to the same number for 30 seconds? Antony. -- BASIC is to computer languages what Roman numerals are to arithmetic. Please reply to the list; please *don't* CC me.
Priyaranjan Nayak
2018-Sep-18 04:36 UTC
[asterisk-users] IVR call simulation on Asterisk 15 server
Hi Antony, Thanks for your quick response. I wanted to simulate actual IVR system for my application testing. Prerecorded english menu is fine for me. Based on DTMF tone pressed by the user, it will connect to different number. The call duration will be 30/60 sec as per the test scenario. We will configure in our local setup for internal application testing. Best regards, On Mon, Sep 17, 2018 at 7:18 PM Antony Stone < Antony.Stone at asterisk.open.source.it> wrote:> On Monday 17 September 2018 at 15:42:50, Priyaranjan Nayak wrote: > > > Hi All, > > > > I am using Asterisk 15 server and wanted to configure IVR call > simulation. > > What do you mean by "simulation"? > > > My configuration scenario is > > 1. A subscriber will register to Asterisk server and start a call. > > 2. The IVR audio will come from the Asterisk sever to sbscriber. > > 3. Once the subscriber pressed the botton, the call will connect to a > > number based on DTMF digit pressed by subscriber. Then call will continue > > for 30 seconds. > > > > I observered for normal call pjsip.conf file is used for configuration > of a > > subscribers. > > > > Could you please help me on below queries ? > > 1. Which file we need to configure for the IVR call simulation ? > > 2. Please suggest a good documentation for IVR simulation. > > Do you actually want to create an IVR system, or do you just want the > caller > to be answered by a pre-recorded menu, and it doesn't matter what digit > they > press, they always get connected to the same number for 30 seconds? > > > Antony. > > -- > BASIC is to computer languages what Roman numerals are to arithmetic. > > Please reply to the > list; > please *don't* CC > me. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Thanks Priyaranjan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180918/e6355821/attachment.html>