similar to: OT - best policy for logs

There was a flurry of "Vonage is going to unlock SIP" activity last year; did anything productive ever come of it? Are *you* using your Vonage lines directly into Asterisk? In lieu of that, for a 4 line small business that doesn't need to pay Vonage $150 a month, who? Broadvoice? Someone else? I'm a touch unimpressed with the fact that BV's website *won't quote you

Hi, I am requesting for a E1 connection from my telco. They are asking if I want DSS1 or SS7, and I am stuck here. Could someone tell me the difference between the two? How should I decide which one to use? Thanks in advance for your help. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

So this is just a general question, Is Asterisk really good? Reliability? Functionality? Customization's? I am coming from a Nortel world, were you pay for everything, and you can't delve into the software. But it seems that customization would be a great thing. Like, setting up a war-dialer to customer lists, incoming/outgoing faxes (that's possible with Asterisk, right?) and

Any know what Digium hasn't released the DS3 card? It was supposed to be out a while ago. -Matt

On Thu, Sep 13, 2007 at 04:32:27PM -0400, Jay R. Ashworth wrote: > I'm about to (finally) do my first Asterisk install; SMB, 4 FXO, 4-6 > stations, mostly IP (I'm looking at the Grandstream 201, to start), and > maybe X-lite on a couple of laptops via VPN. > > We've got a 4xFXO box we bought off eBay, which unfortunately I can't > find to quote a model number off

Does anyone know enough about the implementation of AstDB to know whether the data structure is a Hash function, a Balanced-Tree, a b-Tree, or a Linked List? I'm trying to estimate the lookup 'cost' of a AstDB with around 160,000 keys? Obviously I already know that it WILL WORK, but the question is whether the data structure is optimal in the Berkeley DB AS IMPLEMENTED in Asterisk.

Been working with Samba back to 1.9 or so, but this is the first time I've tried to set it up as a domain member server. I followed (I think) the instructions on how to do this in TOSHARG (which are about as disjointed as the book, which I took back), but I continue to be unable to get the box, running winbind, to reliably authenticate users off the PDC, which is running NT4 (and I'm not

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Hi Folks, I'm trying to send CallerID Name information out to the PSTN via a PRI with Bell Canada with no success. With inbound calls (originating from the PSTN) CNAM is received successfully, and we've not had any similar problems with other Telco PRIs, so I'm stumped.

Greetings everyone, I've been working on a (yet another) "all-in-one" Asterisk based project. It is aimed at embedded / low power systems (but scales fine on more capable hardware) and is based on Asterisk 1.4.x and FreeBSD 6.2. Because of this, I've mostly been hanging out on the asterisk-bsd list as bugs rolled in and the system's features were improved. We're

My wife really likes the fit and feel of my SPA-942. Anyone know of a POTS telephone with similar rugged construction?

Hello everyone, As I am new to the asterisk community (although have been on the list reading for about 6 months) I wanted to see what users would recommend for security to protect several asterisk/ vicidial servers over a fiber connection. Currently I have a managed switch (Tellabs 8813-310) from time warner but I am having intrusion issues on my linux server which I think are contributing

As I am working my way to understand Asterisk, I have a couple of questions that hopefully someone will answer. I know there are differences between a PBX switch and a CO switch. Can Asterisk completely replace and act as a CO switch? Are there any telecoms out there using Asterisk as a CO switch? If so, how well does it work? If not, why not?

Hi, all. I see that the Polycom SoundPoint IP 330 supports VLAN... but I don't quite see how that works. Do you point a non-VLAN'd segment at it (akin to when you uplink a VLAN_enabled switch), and have the phone implement the VLAN? Or...? *puzzled* Thanks much, -Ken

http://www.asterisk.org/node/48327 I mean, really... you're kidding me, right? Lee.

On Fri, 2007-08-17 at 18:22 +0200, Trixter aka Bret McDanel wrote: > On 8/17/07, Aleks Clark <aleks.clark at gmail.com> wrote: > > Actually, the crazy p2p connections actually reinforce their algorithm > > story. If their p2p algorithms have flaked out, it could cause all sorts of > > trouble. OTOH, I don't think they'd run logins over p2p.... > > >

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "user1" calls user "user2" 2. "user1" then presses the feature code "*0" to redirect "user2" to conference room 300 3. "user1" then dials the user "user3" 4.

When I downloaded the sip and bootrom from Polycom website, I noticed a file called SoundPointIPWelcome.wav. However, I have no idea where and when it was used. I played the wav file but I have never heard the phone using this wav file before. Does anyone know what it is used for?

This seems like a piece members of this list would find interesting... === There is growing concern over the interaction of VoIP systems with the legacy PSTN, and the transmission of caller identity data--most notably, Caller ID on the PSTN. It is not always possible, or obvious how, to handle Caller ID data when moving to or from VoIP and the PSTN networks. There are even business models

Hi guys, I've been experiencing a very strange issue with my Digium Card TDM400 as of this week. It has two FXS and two FXO. The FXO modules (both of them) never goes on-hook after hanging up in Asterisk. It had worked perfectly well for over four years. I put an ammeter in series with the line and the card, and immediately after plugging the connector to the card, I got 26mA in the circuit