similar to: ivr testing script

Hello, I'm having trouble working out how to send DTMF tones to an external IVR. My system has an analog phone connected to a TDM400P card, a SIP software phone (Zultys LIPZ4) and is connected to a BRI in Australia with a NETjet-S card. I'm using ISDN4Linux and a 2.4.25 kernel patched with the ISDN audio patch from Traverse (which allows the card to do voice). DTMF works fine between

New to asterisk so please be gentle. I'm guessing I'm among a number of recent additions to the list after the article in Linux Mag. I gotta say I'm *very* intersted in the project and will be doing lots of reading shortly. A couble quick questions first... How suitable is Asterisk for use as an IVR providing callers with textual data out of a database? Can it be combined easily

This was submitted to the Dev list last week, but there was no response, and perhaps it wasn't the right group. I am developing an application in which I need asterisk to pass on an incoming call to a separate IVR server. The problem is that asterisk appears to hang up while the IVR is playing back a sequence of recorded voice and systhesized voice prompts. My setup is: Analog line

I connected two plain old telephones to FXS lines of a TDM400P (defined as fxoks in zaptel.conf), one of them dials the other with Dial(Zap/2), I talk to myself for a while, hang up either of the phones, and the phone that remains off-hook gets the congestion tone until it goes on-hook (at least as long as I've cared to wait). I don't have a voltmeter on the line, but if I'm hearing

I thought I had posted this, but I didn't see it in the archives, so I guess I hadn't. I've got FXS lines going to a legacy IVR. When I Dial into one of these lines and then hang up, FXS plays the Congestion tone until the IVR drops voltage. I would like the IVR to hang up sooner. I could do this by either making the IVR recognize the standard Congestion tone, or changing the

On Sun, 31 May 2015, Luca Bertoncello wrote: > Now, it would be nice, if I can signaling on the phone which number will > be called, so that, for example, if I receive a call for +493511111111 I > get a message on the display or the phone ring with a particular tone, > and if I receive a call for +493512222222 the phone write something > other on the display or ring with

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

Hi, Someone knows an Open Source solution that can handle Outbound IVR for asterisk?. What I'm looking it a pre-preprogrammed a telephone call that reach a Person and start making an Interview over the telephone. Specifically I want to call all my customers exactly one hour after the service has been performed and ask some questions in an IVR, also the results of the Interview I will need

We are running 1.2.14 with an IVR in the dialplan. If I connect to the IVR with a SIP phone (Polycom or Xlite) and press a couple of digits very rapidly (I found this with 33 on a sticky keypad) which are an invalid response, Allison will go into a loop saying 'I'm sorry, that is an invalid response, please try again.' over and over. This does not happen with a commercial

To insert long pause during dialing and submitting multiple DTMF tones, is there better solution then below: exten => _51,1,Dial(SIP/18778794590 at pstn-5665,300,D(wwwwwwwwwww1www),D(wwwwwwww005893884053811#)) I think submitting multiple DTMF tones is not allow from one command line. The first part D(wwwwwwwwwww1www) worked, but not the second one D(wwwwwwww005893884053811#) I'm trying

hello, i try to build a IVR for our company my problem is that the dtmf tones are not recognized by the phones i tried several phones. BUT when i call the voicemail i can navigate with all phones through the menu. I use * 1.2 here is the context: [ivr] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 ;SAI menu -

I'm trying to setup a VERY simple mainmenu ivr but can't seem to get the tones to be recognized during the background( ) the playback and background files play, but asterisk doesn't do anything when I start pushing keys - I've tried it from softphones and pstn line phones Can anyone tell me what I'm doing wrong? Required contexts Exentions.conf below [from-broadvoice]

Hi, How can I stress test an asterisk IVR? I am looking for some kind of sip phone which can be "programmed" to send out digits after specified time to simulate users pressing menu items. If it can originate large number of calls simultaneously then it's great! Does any one have any recommendations ? Any other method to stress test an IVR call flow? with regards, raj

I have an IVR application that works fine from multiple DID sources, unless the call to that DID was from a Vonage service user. In this case about half the DTMF tones never get recognized by Asterisk. Has anyone else seen this? Suggestions? I'm running 1.0.9.

Steve Edwards <asterisk.org at sedwards.com> schrieb: > You can fiddle with the ring tone by phone specific configuration and > phone specific SIP headers (sipaddheader(Alert-Info: ...)). > > These seem relevant: > > http://www.voip-info.org/wiki/view/RTTTL+melodies+for+ST2030 (the > discussion looks relevant as well). > >

Though I've had implementations of Asterisk, I havent encountered this one yet, so i'd like to seek your advise if this possible. I would want asterisk to be stand between the dialer the destination. The dialer will now dial asterisk access number. Asterisk will acknowledge user by using CallerID and check against its database for authentication and then sends out a DTMF A tone for ?

We have a couple of parallel ring settings (and this has worked well for eons). Either in the form of same => n,Dial(SIP/1001 & SIP/1002 & SIP/1003 …..) Or via a subroutine (below) that has a bit of extra logic: FOO = 1010 & 1019 & 1017 & 1033 ... same => n,gosub(sub-callout,s,1,(${FOO},”Ringing all class FOO telefons")) Now I have two types of phones

Dear All, Situation (1) is SIP agent -> Asterisk TE410P -> Telcom Mobile System PRI debug Message type: (just hear about Mobile System VoiceMail IVR introduction, hangup before beep tone) Message type: SETUP (5) Message type: CALL PROCEEDING (2) Message type: ALERTING (1) Message type: DISCONNECT (69) Message type: RELEASE (77) Message type: RELEASE COMPLETE (90) GSM Mobile still

Does anyone know why? ThePBX*CLI> -- Executing [310-456-7890 at from-trunk:1] Set("SIP/202.101.202.101-b763ce60", "__FROM_DID=310-456-7890") in new stack -- Executing [310-456-7890 at from-trunk:2] ExecIf("SIP/202.101.202.101-b763ce60", "1 |Set|CALLERID(name)=310-456-0987") in new stack -- Executing [310-456-7890 at from-trunk:3]

May be I have fiddled too much with my sipura settings but I can't get it to give the stutter tone when there is a new voice mail waiting on the asterisk box. I can either get a stutter tone all the time or not at all. Anyone got this working. Thanks Chris