similar to: Call getting disconnected in queue

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

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

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

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 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, 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

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'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]

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, 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

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'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:

Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using

Hi, I am setting up a test call center using *. I am using one Zap channel (Wildcard TDM400P REV E/F -- 4 FXO modules) for incoming call and sip phones (SjPhone) for call agents. I have setup queues and agents. While testing I found that if the agent presses * key in soft phone while attending calls Zap channel gets hung up, and another customer can call. But if the caller hangs up (for example

-This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow

Asterisk 1.4 r181990 given the dialplan snippet below, can anyone tell me why the h exten is not being run ? ============================================================================ console output: [Mar 29 10:33:49] -- Executing [s at questionnaire-menu:1] Set("Zap/1-1", "TIMEOUT(digit)=3") in new stack [Mar 29 10:33:49] -- Digit timeout set to 3 [Mar 29

Hi I have a situation where I want to be able to count how many times a caller goes round a loop of "Please hold...", "please continue to hold". I have found an example on voip-info but I can't get it to work. Not sure if I've got some syntax wrong somewhere? All that happens at the moment, is I hit is the playback of "som-debug" at 9999. Any ideas would

My SPA-3000 finally arrived and I'm trying to get the FXO port on it to work as if it was a X100P card as far as Asterisk is concerned. I have Asterisk dialing out over the SPA-3000 FXO port no problem. The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk "mainmenu" context (or ext I guess). Currently the

Hello mailinglist, I have a strange problem, which occurs sometimes on some WinXP clients. It is not 100% reproducable for me, but it returns regular. The problem is, that somehow a user can't delete a directory. All files in the directory will be removed successfully, but the directory itself not. Deleting with the Explorer or over the cmd-console returns in the same result. First the