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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]

If I have a global variable in my dialplan and I change it, does that change immediately take affect for all calls that are active? Here is my situation. The company I work for has two office groups that share a building. The two offices are separate companies but support one another and want to be able to transfer calls as if they were all on the same phone system. Each company has 4

I am trying to setup an interactive menu where a caller hits the main menu and can then dial an extension. As far as I can tell the "Waitexten" app is failing to read 3 digits and just reading the first and then announcing that it is invalid since all extensions are 3 digits. How do I make Waitexten wait for 3 digits? I have setup the extension "100" for users to reach the

Hi Guys. I've had a lay off from Asterisk for 12 months but I am starting to look into it again. I am not very Linux savvy and found it hard going the last time. I've started playing with it in the last 3 weeks and I have to admit to making more head way this time. The first problem I'm stuck on and I cant find a solution to is that sound files that I have recorded (be it by

Hi I can't for the life of me work out why this is not working. When in the campon contect if you hit a DTMF key 2 you get moved to the exten => 2 defined in the mainmenu context not the exten => 2 defined in the campon context. What is wrong? The same happens if you hit key 1. [campon] exten => _*1XXX,1,Answer exten => _*1XXX,2,SetCallerID(${CALLERIDNUM}) exten =>

I am trying to get my xterm window under gnome to open with large fonts, with light green foreground and dark green background. I have the following .Xdefaults file contents: $ cat .Xdefaults ! This is a comment ;-) #ifdef COLOR *customization: -color #endif !! Let's cast a wide net, for any app supporting these ! Blink instead of beeping *visualBell: True *scrollTtyOutput: False

Hi list I encountered problem in using Background command. Below is my extensions.conf. [mainmenu] exten => 4,1,Wait(1) exten => 4,2,Background(thank-you-for-calling) exten => 4,3,Goto(n01|s|1) [n01] exten => s,1,NoOp(${CONTEXT}) exten => s,2,Background(thank-you-cooperation|m) exten => s,3,WaitExten() exten => s,4,Playback(digits/pound) exten => 1,1,Playback(digits/1)

Just another bit of info which might help solve this: Looking at the Asterisk log messages - I notice when I start up Asterisk, I see the error: pbx_config.c: Can't use 'next' priority on the first entry! Could I be right that its something got to do with priorities? I changed the incomingpstn context to the following eliminating the 'n' field and still the same errors were

Hi, At my house, I have two POTS lines. Both are connected to my * server on a TDM400P card. As an example, say the phone numbers are (919)555-1212 and (919)555-1213. I also have four SIP extensions, an ATA with a fax machine, and a DID coming in from an ITSP. I have an autoattendant configured that talks and allows users to forward to the extension they choose, but my family doesn't like

Hi, Yes InternalExtension is the context and 2093 the extension. Just to explain something odd that?s happening (and I?m very stumped with this) .I think my contexts are definately the reason that I can?t interrupt the menu for incoming pstn calls to choose a submenu: My users register with my sip proxy (SER). Therefore when I create an entry for them in sip.conf I set only one context. Also to

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

Has anyone used Gastman/Astman successfully? I have it up and running (Gastman win32), but have a problem with the creation of end stations on the map. I'm not sure of the format of the extension to use when creating a end station icon. Services like Conference bridge and Musichonhold seem to work ok (I use 555@mainmenu and 666@mainmenu) for the Icon extensions. IAX softphone seems to work

I have a situation where gnome console does not handle vt102 escape sequences properly and therefor need to employ xterm instead. When I run xterm from a gnome terminal window I am presented with an extremely small terminal window employing an almost unreadably small font. I have attempted to set the font size using xrdb and a custom .Xresources file. I can change the colour scheme. I can

When a person calls into the Asterisk voicemail or auto attendant, the first second or two are cut-off. This happens with custom prompts I have created (with or without 1 or 2 second delays) and with the default prompts that come with Asterisk. Does anyone have a solution to this problem? I'm running the current CVS. My default menu config is: [mainmenu] ; ; We start with what to do when a

Hello all, I have one Asterisk 1.4.21 system connected to a North American POTS line. Normally hangup detection works fine, and Asterisk hangs up properly if you are talking to a caller and they hang up; but occasionally a call comes in (typically from a US telemarketer) where the caller hangs up just as voicemail recording is starting, and you get a voicemail recording of dialtone (then

I have set up an auto attendant and all is working but I am bothered by a long pause when switching between menus. This pause is between 5 and 7 seconds and is quite annoying. Is there anyway to address this. One other thing I find interesting is that when I move from the main menu to the sub menu the delay is there but when I move from the sub menu to the main menu the delay is not there.

Hi all, Running bristuffed 1.2.4 system with solely Cisco 7940/7960 phones with SIP. I'm using also op_panel 0.25 (snapshot). I'm using * queues. I want to properly implement DND via *78 and *79. I'm using op_panel's documentation RECIPE 1 solution with astdb and dnd variables and this is fine for FOP. The DND works in normal cases, since I catch it with my Macro dialsip, HOWEVER

Hello all! I'd like my * to call out to somebody and offer the called party a menu system. Everything should just be as if the called party had initiated the call themselves. This is my try: exten => 100,1,Dial(Modem/g1:0555321) exten => 100,2,Goto(mainmenu,s,1) This doesn't really work, because the Dial cmd blocks further execution until the called party hangs up. Next try:

Hello People, I'm newbie in * 1.0.1, running a Linux 2.6.7 in a Debian Sarge, and this is my situation: +------------+ +-------------+ | Sip Server |-------------|CISCO PSTN GW| +------------+ +-------------+ \ || \ || \ +----------+ || | Asterisk |=========

Here is some dialog from the Console: -- Starting simple switch on 'Zap/13-1' Feb 10 07:22:36 NOTICE[21105]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... -- Executing Goto("Zap/13-1", "mainmenu|s|1") in new stack -- Goto (mainmenu,s,1) -- Executing BackGround("Zap/13-1", "thank-you-for-calling-poker -support") in new stack