similar to: Interact with IVR

Hi! I'm setting up my IVR system, how can I register in a mysql database the IVR menus accessed by the clients ? Thanks a lot, Szasz Szabolcs

On Wed, 2006-12-06 at 23:51 -0700, asterisk-users-request@lists.digium.com wrote: > Date: Wed, 06 Dec 2006 22:37:01 -0500 > From: Steve Prior <sprior@geekster.com> > Subject: Re: [asterisk-users] Is there any Asterisk controllable > thermostat? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >

With one IVR payment system, I noticed quite a difference in DTMF transmission between these two cards. The IVR missed nearly all digits from X100P, while receiving digits from TDM fine. Since neither card process or synthesize audio, what can the difference be? (This particular IVR has problem with some regular phone devices, too.) Yuan Liu

I'm fairly new to Asterisk, but I've been searching the archives extensively and haven't seen much information on using IVR for more than menus. I'd like to prompt a caller to enter his ID (employee, customer, account, etc). The business use I have in mind requires a five-digit ID. Then I need to be able to capture the ID entered, validate it (probably by playing it back with

I have asterisk 1.0.5 installed via apt on a debian system. It's a custom distrobution called Voyage Linux that runs from a flash card and I have a hard drive installed with mysql installed on it as well as apache. I have been though the AMP install guide (asterisk management portal) and in the interface it has a place for me to record new IVR menus. I have to dial *77 to begin recording

Hello Team, IVR selection of QUEUEMETRICS As we know queuemetrics had an IVR selection functionality where it can get the IVR keypress of a caller. We saw this link http://forum.queuemetrics.com/index.php?action=printpage;topic=503.0 and upon checking, its only determined the Queue, I want to get is the per IVR of a caller. Can you help me guys regarding this? I want to implement this with

I have really tried to do my best googling and wiki-reading before asking this question. I couldn't find the answers there so I throw myself at the mercy of the list... I get excellent quality for SIP -> PSTN and PSTN -> SIP calls, however when I or anyone else calls from PSTN -> * the voice menus are oftentimes very choppy. Sometimes they are absolutely perfect and I cannot tell

Does anyone know of an (E)AGI or program to develop a IVR dial-plan which will take a list of words and then do something when a unique branch has been found. i.e. Say there's 3 words demon deacon bishop On a phone they'd be represented as 33666 332266 247467 So if the user enters "2" we know they want bishop if they enter "336" they want demon and "332"

The SM56 MODEM manual says it does. But when used with zaptel 1.2.12, nothing shows up. Yuan Liu

I've written myself a easy to use telephone directory which I use at home and thought it may be of interrest to others. The purpose of this agi script is to provide an online telephone directory that can be easily accessed using the numbers on the phone dial pad. You select entries by spelling out the name of the person you want to contact using the phone dial pad. Now this is normally

Hi, One of our customers has an issue with the callee not being able to hear them. It seems to happen very frequently on one number in particular where there are about 3 IVR menus to dial through before getting to a live person. However, this does not happen on every call. Running tcpdump on the RTP packets, I can see that RTP is setting sent, but the values in the packet are all very close to

I upgraded from 1.0.9 to 1.2.1. In 1.0.9 everything worked perfect. Now, I call in my IVR, and after navigating in menus when I get dialtone for dialing extension, Sound is choppy and I get bunch of messagess: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence

I'm working on an IVR that gives the users the option (number 5 in the main menu) to dial by extension: exten => 5,1,Set(TIMEOUT(digit)=5) ; Dial Extension exten => 5,2,Set(TIMEOUT(response)=10) exten => 5,3,Background(LCL/prompt-60) exten => 5,4,WaitExten(15) When going option 5 you can dial some extensions such as 2802, it goes to the extension (all extens start with 28 on the

Hi all, Can someone point me to some information on the type of hardware that might / should be used for a high load on an asterisk machine ? I know that this is dependant on what services you plan to have running, and it's relevant to what you plan to do. We are likely to be running 4 E1's, Voicemail, IVR menus, Music on Hold, Pay-Over-The-Phone, lots of interdepartmental

I've added the ability for a user to record a custom message associated with a special IVR menu for occasions when business will be closed for some non-standard amount of time (Maybe 4 days at Christmas...) They just dial 800, record the message then hang up and dial 801 to enable it. Presumably, when they return after the holiday, they should dial 802 to disable it and return to the

Hi all, I am trying to understand how I can get a simple IVR scenario to work properly (having already removed most of my hair...). The basic requirement is as follows: * Caller arrives at our main number * Caller is greeted and then told they can enter an extension number, if known, or wait and their call will be connected to an available rep. * The IVR then dials a group of extensions (if

Hello all. I'm having a problem debugging an IVR I'm building. I can't see any reason this shouldn't be working. Firstly the asterisk version is: Asterisk SVN-trunk-r7230 built by root @ localhost.localdomain on a i686 running Linux on 2006-02-17 22:44:48 UTC Basically the problem is this. While the playbacks are happening you can push any one of the options and to happily

I don't have a particular reason to upgrade, but I'm installing a new box, so I have the opportunity to go 1.4. On the other hand, I'm not familiar with 1.4, and relatively new to Asterisk. So instead of trying to keep up with two different versions, I want to tie my handful of boxes to one, before any of them grow too complex. Is there a document about the main motivations to

Hi All, I have setup 2 trunks using 2 different voip providers using sip. the first one i have no problem calling inbound then redirected to an IVR, i can hear the IVR. the second one has issues, inbound works going to IVR as i can see it on the CLI, but i don't hear anything. i tried redirecting it to an extension not an IVR just to see if inbound really works, and it rings the

hi all, i have gradwell DID i am using it for inbound dialing with IVR when ever customer call my DID some times other IVR is cumming on my IVR that IVR is not even related with my server .can u please help me on this -------------- next part -------------- An HTML attachment was scrubbed... URL: