similar to: Setting auto-attendant to answer immediately

I'm looking to buy a channel bank. I'll be using this to basically dual a Toshiba PBX and Asterisk so that we can deploy Asterisk in small steps and have the Toshiba as a fallback in case something goes wrong. What I need is a channel bank that will allow me to take in 12 lines using the FXO ports in the CB and then output those lines to the Digium T100P. Has anyone had experience

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

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

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

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.

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

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

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

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

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

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

All, I'm trying to hear the asterisk's auto attendant in its default configuration. According to VoIP Hacks in Chapter 4, I found the following excerpt after successfully configuring my SIP IP Phone (Cisco 7960G): In its default configuration, Asterisk has an auto-attendant that can route calls. To try it out, take the IP phone off the hook and dial 2. Then dial the BudgeTone's Send

Using asterisk@home 1.1 and 1.3 ip phones are on the same network as asterisk. I can call another extension and talk/listen w/ no problem. If i dial my own extension (or do anything that makes asterisk play sounds back to me ) I cannot hear anything. this is what shows up in the CLI, when i dial my own extension. -- Goto (macro-vm,s-BUSY,1) -- Executing

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

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

Folks, I have a NEC 2400 pbx(non-voip) behind a Cisco 3725, connected via standard wic-t1 card. The NEC needs to call two different asterisk servers with 4 digits. I have two way calling working with the one * box, but the other is perplexing me. Here's the layout * <--> Cisco 2811(192.168.13.1) <--> 1.54 point to point <- Cisco 3725(192.168.8.1)<-> NEC 2400. The

I never heard anything on the AMP list, so figured maybe someone here might be able to help me sort this one out.. I was making some updates to my attendant config, which is really very basic, and now incoming call processing stopped. Not sure exactly what the heck happened, but figured maybe someone could help me with a clue as to what broke. Now incoming calls are not being answered at

Hi I was given the task to try and build a VOIP solution to an office building with multiple tenants in all sizes and shapes. Some of them will require auto attendants and some will simply want direct lines to their phones. The question I have is: Can asterisk be configured to handle multiple auto attendants? TIA Paolo

Hello dear list. I am currently working on a Automatic attendant, and the core things work, but I think the loop function isn't working as expected. I am testing this environment: a sip internal call from 301 to 501. The setup here is when 301 calls 501, and 301 doesn't enter an extension, it will go in loop, 3 times, and then hangup...Can't get that working. Could someone please

If you really want to do this the asterisk list is based off of mailman. You can learn all about mailman here: http://list.org/ But really, what are the odds that newbs will know to go there first? Are you going to moderate it? Someone has to actually answer the questions you know, if a newb only list is going to exist. Look, don't answer lame questions if you don't want to. Flaming a