search for: autoattendants

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

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)

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

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

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

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

Are IVR and "Auto Attendant" interchangeable terms? They both do the "Press 1 for" thing. Sales is asking me how to word it and I've always used both terms interchangeably.

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

I'm having trouble finding documentation for the auto attendant does anyone have an idea where there might be some???

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

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

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

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

Is there any way that you could trigger events based upon actual pickup of calls and hangups of lines in ALL cases(parked calls, queued calls, calls triggerd by .call queue files)? It seems like Asterisk needs something a little lower level to allow for this, is it even possible? MATT--- -----Original Message----- From: asterisk@analitica.md [mailto:asterisk@analitica.md] Sent: Thursday,

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?

I have this server I need to put to work. The option I have is to make it work as a small office PBX with SIP users and a Digium E1 Card for PSTN service. 24 SIP users and one E1 card in an Intel 945board (533 Front side bus) with 1GB DDR 533mhz of ram, one Pentium Dual Core 2.66 ghz (FSB 533MHZ) and two 80GB SATA disks. Can the box sustain the load? I can add another 1gb of ram if necessary.

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.

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 currently working on a perl script convert csv logs to a http log equiv: LogFormat "%h %l %u %t \"%r\" %>s %b \"%{Referer}i\" \"%{User-agent}i\"" Right now I have output something similar to this: 111 - - [02/Aug/2003:16:39:15 -0500] "GET /300 HTTP/1.0" 200 6144 "sipext" "ANSWERED" INCOMING - -

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