similar to: pull a call from a queue

When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more

I was asked today if we could somehow have a trainee on the phone with a supervisor conferenced in, but somehow have it so anything the supervisor says is only heard by the trainee and not the customer. Is there a feature like that?

I'm asking about this scenario: Asterisk(public IP) <--> Internet <--> Router (public IP) <--> SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia

someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. Thanks in advance.! -- Esteban L. Cacavelos de Amoriza Cel: 0981 220 429 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Folks, Is there a way to find out the mobile/landline carrier name based on the phone number? For example, who is the mobile carrier for (415)2345678 I had heard about some query but just don't remember how/what? Thanks in advance. Ritesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a customer that has a campground. Wants to see who's at the gate, remotely, via camera, and talk to that person through a "traditional squawk box" and be able to open the gate remotely from that phone. He doesn't want to have a separate camera feed, etc, he wants to do it all on one phone. Does such a way to do this exist by using Asterisk and some kind of relay

I was looking at the ILECs' web sites to determine how their users access voicemail. I looked at AT&T, Verizon, Qwest, and Embarq. They supported one or a combination of the following for calling from your phone: *98 #55 Toll free number Your number A varying phone number, based on your number's location. Calling from anywhere else, they supported: Hitting star when

In the simplest terms I can think of, I'm going to describe what I want to do and I want to know if it's possible in the current version of asterisk. Can I take a T38 call from an ATA, convert that back to analog and have asterisk screech that out on a POTS line to a remote fax machine. Would it work? And could I receive an incoming fax the same way? Please don't talk to me

Hello everybody. I want to make a menu with the incoming calls, I want that when someone calls the Asterisk play a record (in gsm) and them the caller must choose a option (1,2 or 3). if he choose 1 it will redirect to 101 extension if he choose 2 it will redirect to 102 extension if he choose 3 it will redirect to 103 extension my extensions.conf is this one: [default] exten =>

SIP channel hang ups are progressively getting worse and I'm really grasping at straws here trying to find out what the cause is. The problem start, once a week or so the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. It's now doing it multiple times a day and I fear having to go back to our

Hi, I wish to use Asterisk as a SIP server. How do I use Asterisk in a redundent network? So, if one Asterisk server fails, how does failover work? James

I'm in the Northeast US and looking for any recommendations on Level3 resellers. I don't do enough volume to go to Level3 directly. If there's anybody you'd definitely avoid I'd love to hear about that too. Thanks, Adam

What is the deal with "CSR's"? TWTelecom is telling me that I can't port a number to their service without a Customer Service Record. Apparently this is easy with Verizon, and not so easy with some other companies. Basically I'm at a brick wall with a couple of ports because TWTelecom is telling me I HAVE to get a CSR and certain other providers (Time Warner Cable for

I read the thread about what IP phone is best for business deployment with great interest. Our need is slightly different however. We are deploying VoiP as a value-add with our high speed internet service and are having trouble finding the right SIP analog terminal adapter. In order to support people's existing phones and wiring we need to use an ATA. 1) The first priority is we want

This is an email to all New York based Asterisk users. For some time it's been bugging me that we don't have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I'm going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many

Hi, I'm looking for adressbook that could easily integrate into Asterisk, so it should: - have agi script to search for caller id name - have fields for notes on previous contacts (for CRM spawning of FOP) - have web interface to edit entries easily ... Any advice, pointers ? What is your favourite address book to use with Asterisk ? Regards, Rob.

Is there a memory leak in asterisk 1.4? The other day with asterisk 1.4.0 I noticed that top was reporting a RES of 106 meg for the asterisk process. Restarting the process brought it down to more like 4 meg, but it grew over time to be 20+. So yesterday morning I upgraded to 1.4.4 in case this is something that had been addressed. Again I started with a RES of like 4meg or so, but this

Hello all, I had a quick look at the horde site and noticed that horde is being advertised as, let's say, "smartphone friendly". Does anyone know if the newest horde version can "talk" to? smart phones in regards to e-mails ? In other words, can a user owning a smartphone get his/her e-mails on it apart from the webpage ? Thank you, spyros ? ---- "I merely

Hi all, I have been playing with the AMI quite a bit lately - mostly debugging WombatDialer in production, but that's a different story - and I have been frustrated by the lack of a simple way to interact CLI-like with the AMI itself. So I have decided to write something myself to make my life easier, or at least a bit less miserable. The result is a little webapp that you can use as a sort

Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a "channel" format, like SIp/1234 or Local/1234 at ext) and "park" it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the