similar to: Attendant configured AutoAttendant

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)

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

Help. Why is it that I can call out from my GSBudgetone SIP phone but the audio is "one-way'? Why is it that when I call my asterisk phone number, I get a fast busy?

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'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, 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'd like to set up a "silent attendant". By this I mean when someone calls me I'd like for them to hear the comfort ringing tones, but for the first 5 seconds I'd like to give them the option of pressing 9 to send the call to an alternate extension; if they don't press 9, the call goes to a default extension. For most callers I just want standard PSTN behaviour, only a

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

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

Hello! I?m trying to add fax functionality to my asterisk installation. Right now I?m focusing on receiving faxes. This is not explained in a book, but I assume that I can use same context, add ?fax? extension and if someone calls to send fax - it will autodetect. Right? Per book, I made following setup additions: 1. In sip.conf [general] I added: ;FAX stuff faxdetect=yes t38pt_udptl=yes 2.

Ah I see, in theory it's possible then. We don't have any IVRs or anything which requires key presses, there isn't even voicemail on this particular phone system so I don't think it will be too much of a problem. I've also updated the firmware on the Cisco phones that have had the issue, just to see if that solves the issue but as it's been going on for a while, I'm

Hello all, I'm planning to create a simple Auto-Attendant (IVR Menu) for my home PBX yet all callers from incoming (trunk) calls must only press the extension numbers from the [analog-ext] else will play the "pbx-invalid". How do you do that using the GotoIf() (or probably using the other applications) but will check if the numbers entered belongs to a specific context? Also, how

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

Dear list, I need to buy a phone which could monitor the state of the maximun number of sip extensions about 200. It is for an attendant. I just saw Snom 370 with keypad and Linksys 962 but they do not let me to monitor 200 extensions states adding keypads. Do you know any kind of phone that let me do that? Which is the maximun number of extensions your phones can monitor and which models phones

We've been playing with Asterisk with an eye towards possibly replacing or augmenting our existing PBX serving about over 600 phones (and needing to expand). The one missing bit that I can't find any mention of is an Attendant Console. Are there any good solutions out there? I've considered that maybe one of the better softphones might suffice, but the ones I've looked at so far

One of our clients is using a Grandstream GXP2000 with an attendant console. We have used the same phone with past clients successfully however this particular operator processes around 200 calls a hours and the GXP2000 for sure does not like the quick line shuffling and call volume. We get the following problems randomly: 1. menu stops working 2. transfer key stops working 3. Line 1 LED gets

Hi: I am trying to setup a simple auto attendant with Asterisk using SIP extensions. I have an IP trunk to voicepulse and my outgoing calls go over that. I can also receive calls on that voicepulse trunk and want it to an auto attendant. Everything works except on the following: - one of the options is to allow the caller to press the extension that they would like to be connected to. I have

Someone will probably tell you with more certainty, but (you don't say but I assume you are talking about FXO) the Caller ID normally comes in between the first and seconds rings, I think you can tell asterisk not to get the CID but if you don't, it waits for it. Also, I remember reading in a modem manual something about the number of rings you had to let your modem ring before doing an

Hi Folks I have had a look through the Features list, and I see that the system does support an auto attendant, however is it possible to have say 5 telephone numbers that a person would dial and have 5 different messages I.e Dial 555-1121 and you get a message for companyA call centre Dial 555-1131 and you get a message for companyB call centre Dial 555-1141 and you get a message for companyC