similar to: Autoattendant Configuration

So I have whose autoattendant is colliding with their extensions... Quick fix anyone? Second someone presses say a person's extension (101) ... Autoattendant sends them to the first context... [companyx-main-aa] exten => s,1,Background(companyx/companyx-main) exten => s,2,Background(silence/10) exten => s,3,Background(companyx/companyx-main) exten => s,4,Background(silence/10)

Just to make sure this isn't a typo in your original email... Is this example from your zapata.conf? Also, the extension you have shown are in extensions.conf not zapata.conf correct? Here is an example of a good zapata.conf.... [channels] language=en busydetect=yes faxdetect=both busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes useincomingcalleridonzaptransfer=yes

I'm trying to get into the world of Asterisk in order to use the voicemail and autoattendat features (and more stuff later) with a Redcom switch. But, I've only started and haven't gotten to that yet. At this point my solitary goal is to talk to the autoattendant via an SIP phone (SJPhone). I've spent countless hours trying to find the documentation I need to accomplish my goals

Hi, I setup asterisk as an autoattendant. When I call using IAX I get the autoattendent okay, but when I dial one of the extensions, there is no ringing sound passed back to the caller. It happens when I use my DID number, but I also configured a context so I can get it to happen with Firefly (iax client) as the caller. It seems that once the Answer command is executed in the dialplan, status

All, Has anyone ever seen a problem where the autoattend detects the first digit twice? What I am seeing is this: My extensions are 421-468. When a caller calls in and dials exten 433 from the autoattendant, they get exten 443. This is happen for any extension that is valid in the 44x range (i.e. 42x -> 442, 43x -> 443, 44x -> 444, etc.). I am seeing this problem about 1/3 of the

Anyone have a user configured auto attendant setup? Something that can be used without the * admin helping to make changes. Something where the operator can record the message like 'press 1 for john, 2 for bill, 3 for jean' and then the operator can enter the extension that gets dialed when the caller presses 1 or 2 or 3? This would be useful if Bill leaves the company, the operator can

Consider this dialplan fragment, where the call is being dialed into [macro-process-routing] over an iax2 channel from another (same build) Asterisk server: [macro-process-routing] ; This is the entrypoint of the debug call but is also refered to by Macro(process-routing) elsewhere in the dialplan ; XXX-NNN-6800 exten => _6800,1,Macro(6800-interceptor) ; This is matched when 8 is

I've spent the past several weeks reading up and playing around with Asterisk while I've been waiting for an ISDN card I got on ebay to arrive so I can really get to business. I'd just like to run my project ideaa by some of you to hopefully get a little feedback. I aplogize if this ends up being a somewhat long message. In the Marine Corps we've somewhat recently started using

Hi all, been searching & not finding an answer to this, although I'm guessing it's absurdly simple... I just hooked up a T1 to our * box (1.2.0), which had been using POTS lines via a channel bank.. Now when I call the new T1 circuit, there are no rings, the Autoattendant just picks up right away.. Any clue on how to make it ring twice before getting picked up? I tried immedate=no and

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've been asked to create a graphical "call-queue" manager. That is, use the existing call queues application but allow a way to view what's coming and attach information to it. As far as the "attaching information" that's in the realm of my application, but I'm trying to figure out if the internals of queues are exposed through any interface. Any help there?

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

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

I know that Asterisk supports DID, but does anyone have documentation on how to write the configuration for it? I'll be trying to setup a hybrid system where some incoming numbers will be DID enabled and others won't, so I'll need to be able to sort between the two, i.e. directly connect the DID dialed numbers and route the others to an autoattendant for extension dialing.

Could anyone help me set up Asterisk in such a way that it makes SIP-SIP transfers using the REFER / NOTIFY method according to RFC-3515 ? SCANARIO: - Asterisk registers with PSTN<->SIP VoIP provider "V" (Vonage) as a friend - Asterisk is located in Europe, Vonage in located US. - Asterisk acts as an autoattendant located in Europe. - Asterisk answers and incoming call from

hi, i am trying to do autoattendant but failing. as in the manual i inserted the background(welcome-mainmenu) file so that after the sound the caller can dial the extension he wants to call. i figured that the background sound wasn't coming in the asterisk. how do we do this without first loading the welcome message? for example after certain rings the caller can dial the extension no to

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 =>

Hi Tom, Thank you for your informative and helpful reply. I had considered using the relaxdtmf setting but held off this due to not using any physical connection hardware -Asterik uses both SIP in and out from an upstream provider (Gradwell.com). Is it still possible to set this when using SIP trunks only and not physical hardware? The box does have a Digium ISDN card but the ISDN is no longer

I'm pretty much a newbie to this but still think I've been around the various help pages, voip-info.org etc to be fairly sure I'm not missing something here so your help is appreciated! I have a box running RedHat9 at home with the latest CVS of Asterisk and all works fine. At the office, we installed Gentoo linux on a machine, downloaded the latest CVS of Asterisk, set it up. All

hi, i am trying to do to autoattendant. here is my extension.conf part [tumpak] exten=>s,1,Dial,Zap/4|10 exten=>s,2,Voicemail,u9999 exten=>s,102,Voicemail,b9999 exten=>t,1,hangup so when a caller dials the extension 2 suppose, it enters to the above context.. everything is fine. the problem is when the caller hangs up the asterisk does not. after caller hangs up and tries again he