search for: autoattendant

...e 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 "V" - The caller browses the autoattendant and as a result is transferred to another number in US PSTN (also via "V") - The autoattendant on Asterisk hangs up. When this scenario is attempted in Europe on the excellen...

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,Bac...

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 time. My setup is 6 PSTN (POTS) lines -> AccessBankII channel bank -> T100P card on the asterisk server...

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 it. I figured out how to make the fax work from any extension. How would I make entries in extensions.conf to forward to certain SIP extensions based on the incoming call's DID? I lo...

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 but everything I find always assumes so much and I'm left lost. Plus I haven't found a thing about setting up Asterisk as an SIP server. I installed the Asterisk@Home p...

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 o...

...o(6800-interceptor) ; This is matched when 8 is dialed during macro-6800-interceptor,s,4 exten => _8,1,Playback(welcome) exten => _8,2,Hangup [macro-6800-interceptor] exten => s,1,DigitTimeout,2 exten => s,2,ResponseTimeout,7 exten => s,3,Answer exten => s,4,Background(autoattendant-ivr/grtg-6) ; Play full after-hours greeting exten => t,1,Goto(s,1) exten => i,1,Goto(s,1) ; However, this is never be matched if 8 is dialed during (s,4) above exten => _8,1,Playback(typhoon) exten => _8,2,Hangup So, why does the DTMF detect jump out to [macro-process-routing]...

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 some other zapata.conf tweaks, but nothing seems to work. I also tinkered with adding some wait statements before the 'answer' but only heard silence & then the attendant.....

...itches in place of the proprietary (and ridiculously expensive) switches that we've used in the past (well, and still do use). This opens the door to all kinds of off-the-shelf equipment that we can interface with. My goal now is to usefully integrate Asterisk primarily to provide voicemail and autoattendant feature and probably to demonstrate various VOIP capabilities. Ideally I would interface via a T1/E1 interface (might as well use E1 for the extra channels) but since I don't want to shell out gobs of money for what is at this point still a personal project, I'm opting for ISDN connectivity...

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,

...ming extension). Does anyone have something like this working? Is it possible to have some lines go to the auto-attendent and some ring through? I was thinking it would be possible with something as simple as (assuming 4-digit DID) [default] ; 555-1212 is the main number exten => 1212,1,Goto(AutoAttendant,s,1) ; 555-5000 is the backdoor to the voicemail system exten => 5000,1,VoiceMailMain() exten => 5000,1,Hangup() ; 555-1800 is Bob's phone exten => 1800,1,Dial(Zap/842) ... Is that in line or am I missing something? Does anyone have that working? The service is expensive, so I'...

...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. Just PBX and voicemail, no fancy sutff like call recording... maybe a simple autoattendant like "thank you for calling, please press one for...." -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ------------------...

...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. Thanks, John Lawler

...preted at a broken DTMF tone and getting regenerated by your T1 or POTS card, or Asterisk itself. We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by using ... relaxdtmf=no ...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting) Problem with that it that our autoattendant wasn't recognizing DTMF tone from callers very well. They would dial 4 digits and in my logs, I'd see one or two, maybe three. The autoattendant would tell them they had dialed an invalid extension. So we had to go back to relaxdtmf=yes on the dahdi channels in question. So problem_solved...

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...