similar to: Asterisk might be dropping RTP packets before reaching eth int?

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

Hi List; What configuration needed to let my FXS send and receive FAX? Regards Bilal

Hello all, Anyone expecialized with call center monitoring and reporting solution based on asterisk. A client of us, want to install a call center reporting solution for an asterisk server but I do not know which could be the best tool for that. I need a tool for reporting queue calls, agent calls, and disconnect cause. Any clue will be appreciated. Thanks in advance. VoipCrazy

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

Just an FYI for Digium. I received a mailing today from you guys which was nice. The price of mailing was ~$1.60 and inside was an inflatable beach ball. Cool, but I tried to blow up the beach ball and the the seam where the part opens to inflate the ball was not connected to the ball whatsoever, so it went right in the trash. I wonder if the sick heat had anything to do with it, was mine just

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

So we've got a TE410P configured as E-1. The PRI is showing up as normal, I have green lights, but d channel doesnt seem to come up and i keep getting this error if i do a "pri intense debug" The carrier swears up and down that there are no issues on their end. Any thoughts? localhost*CLI> > Unnumbered frame: > SAPI: 00 C/R: 0 EA: 0 > TEI: 000 EA: 1 > M3:

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.

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

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

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:

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

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

Is there a way to busy out a Zap channel? I have a customer who is having problems with a line connected to a TDM800 card and we would like to busy out that line. Since that line is the head of the hunt group I cannot simply disable that channel, I need to busy the line so calls will come over the other lines. -- ?Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director

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

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.

This is a very good example, it seems that their lines have been down the whole day. I'm assuming because of this: http://www.bloomberg.com/apps/news?pid=20601087&sid=anu.mwV7nxuk&refer=home Now imagine they would have another location in Florida, since the 2006 hurricane season is not yet over: http://en.wikipedia.org/wiki/2006_Atlantic_hurricane_season They could have been out in

I have used WIP300, hitachi 5000 wireless phones on asterisk and have had good success. However, I am looking for a WIFI phone with integrated belt clip. Has anyone found any? I have tried after market clips and holders and those just don't work. THanks for sharing if someone has found something that works with asterisk. Jerry

Hi Folks, Recent installation with a simple TDM11B (one FXO, one FXS) that I've set up (at home). I am receiving callerID fine from the telco, as it shows up in my call detail records, AND on 2 SIP phones. However, I'm not reliably receiving it (that is, very seldom does it come through) on the analog phones. Any ideas on where to check configurations, etc? I haven't encountered this

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