similar to: Equipment for small office setup

Hi, I'm looking for a reliable provider to use mainly for outgoing calls in the UK, incoming isn't so much of a worry as I think I'm going to accept them over ISDN. Cheers! Steve -- Steve Foy steve@narnian.org

Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for

We are planning our office communications system using the asterisk PBX and will be using an VOIP origination and termination service such as voicepulse to do so. I know the possibilities are endless, but I'm just looking for a typical asterisk setup for a small/medium business which will use a VOIP origination and termination service. For instance, let's say we have one receptionist,

Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue

I am looking at placing a system in an office with a central receptionist, and phones for each individual employee thereafter. Could I use a Snom 220 with additional keypads to view if the lines are in use by the other employees? Fred is in sales... A call comes into the receptionist and they transfer the call to Fred. The receptionist can tell Fred is still on the phone by viewing the assigned

Hi, We are looking towards a 8+ CO line setup (20 extensions) in our office but we do not want an IVR(auto-attendant) feature. All incoming will be answered by a receptionist. I have read the multi-line configuration for cisco 7960 thread in this list but that way I believe we could only display 6 incoming lines. What will happen to the rest? Does the expansion module for the cisco 7960 work

Hi, I just joined the list, anyways i am trying to setup an @home box with a x100p card and so far i can't even get the box to pickup the incoming call and in the amp management under the section "send calls from PSTN too" page all the radio buttons are blank and i want to use the digital receptionist, also when i try to setup digital receptionist via uploading wav file and save, it

hy all actually i want to have a setup of five offices having round about 200 extensions ( each office having 35 to 45 ) which will be connected through asterisk. now either i should go for voip phones( hard phones ). or use any interface card to asterisk server to which the analogue phones will be connected. ----- if i use analogue phones in the above case ( we have analogue phones already )

I have a customer who wants their receptionist to input the users' long distance PINs for the because they use each others pins. I am having trouble coming up with a way to do this because of creating a channel between the user and receptionist, dropping the channel and its variables and creating a new one for the actual long distance call. Any advice is really needed. 1. User Dials Long

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

I have a Polycom IP600 serving as a receptionist phone. We developed a call manager via c/gtk that runs on a touchpad. It allows them to transfer calls, transfer to voicemail, page, etc. The problem is this: When paging another phone from the touchpad, I have to open a channel to the receptionist phone. This rings the receptionist phone. When she picks up, it then pages the desired person. This is

Hi, Quick question. With an old phone system a receptionist receiving a call has 1 button to push to transfer calls to a specific extension, with Asterisk, a receptionist would actually put the caller on hold, pick up another line, call the extension, ask if the person is available, hang up pick up the caller again and transfer. To me it's seems a long way to simply do a receptionist

Hi! I've got a number of extensions (about 50) on a working Asterisk setup. For each user, I have two extensions configured (for example 11021 for a Cisco 79XX phone and 11022 for X-Lite), and a ring group that ties the two extensions together (for example, 1102). Reason being that if the user is away from his/her desk or working offsite, they can answer the soft phone on the PC. From

Hey all... Scenario (INTERNAL) 1 Call comes in to receptionist and gets transferred to someone 2 No one picks up that transfer 3 Call goes back to receptionist Now when the call goes back to the receptionist, how can I change either the ringer, the callerID or both? * TIA

Hello- I have a low traffic AAH setup, a few hardphones, a few softphones, 50 calls per day max. I used the AMP Digital Receptionist to make a simple voice menu: "Thank you for calling xxxx". I did this for both Normal times and After Hours times. It worked fine. I then went to the AMP Maintenance window, Config Edit, got the "phpconfig for Asterisk PBX" page, and selected

Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies We have an installation of 35 SIP phones (Polycom 501) and one receptionist phone (Polycom 601). I have 15 of the 501s set up to accept a "Page". From what I understand, the "Page" is done using the asterisk page application that throws the extensions into a conference room and then set the originating caller

Here is the situation that I would like to create: Call comes in Receptionist sees that the caller ID is Jenny <8675309> Receptionist picks up phone and transfers call to Batman Batman looks at his phone and sees that the caller ID is Jenny <8675309> I can't seem to figure out how to forward the caller ID. Is this possible with Asterisk?

Hi Running bristuffed 0.3.0-PRE-1f which is 1.2.1. Using *2 in features.conf for attended transfer. Works well if someone answers. But the following sequence causes issue: 1. Receptionist takes call. 2. *2 then 123 to transfer to extension 123. 3. 123 is busy or does not answer so receptionist hears 123 voicemail 4. How can receptionist reconnect to calling user - could wait for voicemail to

Hi, I want to start recording the caller automatically when the receptionist transfers a new sales lead to 567. I don't want the receptionist to have to press *1 manually for automon. Can someone recommend how best to accomplish this? exten => 567,1,Set(CALLERID(name)=SALES CALL) exten => 567,n,Playback(recorded-for-training) exten =>

If extension 101 calls 102 and user 102 hits # and then 103, the caller ID of 103's phone says 102. I've been looking for a way to have 103's Caller ID show the person that is being transferred not the person transferring. So if my receptionist answers the phone and transfers it to one of my techs, I want my techs phone to display the caller ID of the person who called the