similar to: Silent Attendant

Somebody know if they are down? Let me know, Julius C. Barber ventas@gringotel.com www.GringoTel.com Tel. USA: 1-408-705-1189 GringoTel - ahorre en sus llamadas internacionales. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060509/924605b6/attachment.htm

I have gotten intercom working on my office phones (Linksys SPA-942s), but I have noticed that if someone is in a call, it places the call on hold and sends the intercom audio to the person holding the phone that is being paged. I'd like to add logic to my dialplan that doesn't send a page to a phone that is currently in a call. But to do this I need a function that will tell me if a

In the following macro, a call is dialed and control branches according to DIALSTATUS, much like the default std-exten macro. What I'm trying to figure out is how to regain control when the call is answered. ; Standard extension logic [macro-stdexten] ; ${ARG1}=Extension ${ARG2}=Device(s) to ring exten => s,1,NoOp(stdexten ${EXTEN}) exten =>

Well this is disappointing. Time to find somebody else... -- Wes -----Original Message----- From: NuFone Operations [mailto:support@nufone.net] Sent: Tuesday, April 18, 2006 3:44 PM To: wbaehr@totalmac.net Subject: NuFone Update: DIDs Effective 3pm EST Today, April 18th, 2006 Telesthetic, the carrier supporting the Toll-Free and Michigan DID operations of NuFone, has threatened to terminate

I have a problem with my Asterisk system. When I use my phone to call my office mailbox I have to end my password with #. (The office do not use Asterisk) " # " is also used as a transfer button on my asterisk, so when I press it I hear my Asterisk trying to transfer the call. Is there any way to change the transfer button or remove it ? Fredrik

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 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

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

Currently when I call in to my * box it answers after two rings. I'd like for it to answer without ringing. Is this an option somewhere in the dialplan that I'm missing? Thanks, Andrew

Anyone know how many simultaneous calls you can receive on a NuFone DID? -Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050310/e20d2eb8/attachment.htm

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

(Sometimes) When I?m monitoring calls, I hear a very bad jitter ? usually only on one of the bridged channels. So at first I thought it was just the one end of the conversation actually causing the jitter ? but it?s not. So I called in from another device to spy at the same time ? and the other chanspy sounds perfectly normal. (And neither party is complaining of bad sound) So, periodically,

Well I know from personal experience that NuFone is working on a solution for its customers as fast as it can. I know they found an alternate termination provider and are working to have a solution for the TF and Local DID's he currently has on his platform. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

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