similar to: Video Door Phone

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

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:

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 =>

I'm using the M() option to Dial() and having problems. In the following dialplan example ANY digit exits the macro. When the callee presses 1 the Noop(Reset AbsoluteTimeout(0)) does not get run. Does anyone have any ideas as to what I'm doing wrong? Asterisk 1.2.x [extensions] exten => 2998,1,Dial(Zap/1/5551212,,wM(answer-confirmation^20)) [macro-answer-confirmation] 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

Campers -- In a few weeks there will be a bunch of people in Portland, OR, for a conference. A long time ago some campers mentioned that might be a good time for us to get together and hold a mini CampingConf for a few hours. Is anyone interested? Does anyone want to organize it? I''m thinking Wednesday night, May 16. We could possibly get a spot at FreeGeek (not likely since they are

Is anyone successfully passing Outbound CallerID to Teliax? If so can you please tell me how. Thanks in advance! Dan

We have a queue monitoring application running so we can see the caller ID of callers in a queue. If we see a VIP in the queue, is there any method to force that call to be first in line? If there's a softphone, or queue managing application already written that does this, I'd love to know.

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

> Why is the LKG build so important? Because in software development in > general, you want to reduce the "granularity" of the process as much as > possible, approximating a kurve rather than a series of steps. If people > adopt LLVM v3.1 and then have to wait one year for another update (v3.2) in > which LOTS and LOTS of things have changed, they are in for a tough ride

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

| Do you have any information about how people are setting up their | projects? I ask because there are different ways to set up a project | to depend on LLVM. No, I don't have that kind of information. All I know is that it is quite difficult to find a working version on the trunk. I think this especially hit those that are new to LLVM and yet want to play around with the bleeding edge

Some new stuff in the directories horn and strings. I am indebted to Nathan Sharfi for totally reorganizing the web page of pan.zipcon. It was a heroic effort, as there was considerable mess. Included were important help for Windows and Mac users, -- how to get free music players and how to use wget for download. This contribution is a milestone in the popularization of ogg-vorbis. See

Hi, I know this is not 100% on-topic, but I'll try anyway... I'm looking for a way to strip attachments from incoming mails on the server. The attachments should be stored on a file server (where users have access via SAMBA) and only a link to the file (UNC path) should remain in the mail body. This is to prevent the mail storage from filling up with a large amount of big mails where

On Jul 29, 2015, at 6:19 PM, Nathan Duehr <denverpilot at me.com> wrote: > >> On Jul 28, 2015, at 6:32 PM, Warren Young <wyml at etr-usa.com> wrote: >> >> Now we have entrenched commercial interests that get paid more when you get DDoS?d. I?ll give you one guess what happens in such a world. > > What happens? Folks have to think harder about connecting

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 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

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:

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