similar to: ivr key press?

Can someone point me to some reasonable example / starting point to implement a basic IVR menu? Looking for something rather simple like the press 1 for sales, 2 for tech support, and probably an option to list the voicemail directory kind of thing. Nothing elaborate needed, just basic menu. (Yes, I did look at the wiki and google searched for "ivr menu".)

Hi all, I am trying to get a Menu system to work, and having probs with the internal extensions from the prompts. Below is the extensions.conf section. [mainmenu] ; ;"main menu" context with submenu ; include => default exten => s,1,Answer exten => s,2,Background(hello) exten => s,3,Background(thank_you) exten => s,4,Background(if_you_know_extension) exten =>

Hi, ? I'm trying to figure out what the issue is splicing Asterisk between our Telewest PRI and a GDK-186 with a PRI card. ? We're using the Digium TE405P ? Our telco provider is Telewest, and Telco directly into switch is fine. ? When I splice Asterisk in, I can make and receive calls from Asterisk extensions, I can make outbound calls from the GDK, but inbound calls do not seem to pass

Hi guys still have the problem to setup the IVR correctly. I am forwarding call from ser : if (method == "INVITE") { if (uri =~ "sip:1[0-9]{10}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("xxx.xxx.xxx.xxx:5061"); t_relay(); break; } } inside sip.conf

When I attempt to start asterisk with my modem setup listed it will not start attached are the error messages i get and also the modem.conf that i am currently using. Any assistance would be greatly appreciated. running CVS ver 12/7/03, modified only to allow the RxFax and TxFax to compile and run with it (from http://www.opencall.org) just e-mail me privately if you need more info Thanks in

Here is the current dial plan section: [custom-michael] exten => _900,1,Playback(custom/extn-xfer) exten => _900,2,SayDigits(${EXTEN}) exten => _900,3,MixMonitor........... exten => _900,4,Dial(SIP/${EXTEN}|${DEFRT}) exten => _900,5,Playback(custom/extn-xfer2) exten => _900,6,Goto(custom-michael,901,4) exten => _901,1,Playback(custom/extn-xfer) exten =>

I would like to add the ability for our users to be able to press "0" whenever reaching someone's voicemail box to re-reroute them to the auto-attendant. Here's a sample extensions.conf: [incoming] include => ciscophones exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten => s,5,BackGround(auto-greeting)

if i go off hook and dial 666 from an internal sipura spa-x000 (at extn 141), it rings straight through to extn 666. using the same dialplan, from a cisco 7960 with 7.1 sip code (at extn 142), i have to go off hook hit NewCall punch 142 (or any valid extn in the dialplan) hit Dial then dial 666 wtf? sip.conf for crisco [fiji] callerid="crisco" <142>

Greetings *`s, I am having what appears to be a small problem, but the frustration is erally getting to me, what am I doing wrong here ? I used AMP to set up a custom menu, so if caller presses 1 it goes to ext200, if caller presses 2 it goes to ext201 etc etc... Now I have created a third option that when the caller presses 3 it must play a sound and hang up. No rocket science yet. When

Yo, Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate call divert-feature. This one validates if the extension a call-forward is to be set to is actually valid for the current context and additionally saves this context into the DB and always uses it to originate the divert from, as you can't expect the

Hi all. I am new to the list and new to asterisk. I have asterisk installed and running. I am using it as a voicemail server only. What I would like to do is send users to a general mailbox that will be addressed as <companyname>@asterisk and give them the option to wait for the tone and leave a message, or press 9 to dial by name. My questions are: 1. What is the best way to do

Hi, First i would like to thank the kind people of the list who have answered my previuos mail, but i am still stuck as asterisk still crashes upon startup, i have read the install article at http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation and i have search the asterisk archives, but i still cant get asterisk to work, i have tried reinstalling asterisk but it still complains and

Hello all, HI i am very new to asterisk and my boss needs me to investigate setting up asterisk for a new client. I have downloaded and installed (make, make install and make progdocs)asterisk on my personal computer and when i try to run it (./asterisk -vvvc) i get the following output below: NOTE: i am running REDHAT 9.0 on a 796MHz cpu machine: I am excited to be able to work with asterisk

Do you see any syntax errors? Positive comments welcomed. The short version of the logic is as follows: create a file based on the NUMBER create a an audio file move to a new extension (label) and play the results exten => 621,1,Answer() exten => 621,n,Read(NUMBER,enteryournumberstartingwithaone,12,,5) ; create a variable from a DTMF entry / 12 characters long exten =>

Due to changes at the office, I'm finally getting around to setting up an AA to deal with incoming calls. One of the big changes is that we're dropping the old alphanumeric pager and will just send pages to our phones. I've got the outbound greeting message working in a test context no problem right now, but I'm kind of stuck on how to capture a DTMF sequence from a user and

The options are not seperated by commas. exten => s,1,Dial(SIP/50,23,r,d) should be exten => s,1,Dial(SIP/50,23,rd) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of John Klimek Sent: Thursday, June 22, 2006 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

Folks, I've just successfully set up Asterisk (as part of the Asterisk Management Portal installation). When I say "successfully", I mean that I have gone through all the steps detailed for the installation of AMP and not hit any snags there. I can connect to my asterisk server via ssh and can also connect via Http to the portal to change settings in AMP. Now I'm trying to

Only a warning, and doesn't seem to do anything bad. But I can't seem to figure out what these warnings mean? -- Requested transfer capability: 0x00 - SPEECH [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: k [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: o [Feb 17

Hi! Maybe I have just been looking on the wrong pages but there is a question that is very important for me. I already studied some Demo-Dialplans and made some basic experiences with Asterisk. But what I need to find out is how I can handle this. I am leaving my office and I want to tell asterisk to forward calls now to my mobile phone by just hitting a key (on my IP-Phone) or by using a