similar to: How can i test a modem with Asterisk?

Has anyone gotten CID from Bell Canada to work properly with *? We have our * box down at our datacentre in St Louis, and whenever we call it from a Bell Canada Telephone line, all we see is '' for the CID. I did some digging on google and the mailing lists and couldn't find anything pertaining directly to Bell-Canada and * CID, but didn't find much. I did however find :

Hi, I have implemented a custom application module based in some esqueletone code I will provide below. I have tested it with asterisk 1.2.23 and it works fine. But when I tested the same application with a newest version of asterisk like 1.4.* it always returns an error trying to load the module or more specific, trying to register the application. here is the code: int load_module(void){

Hi, To make a long story short, I can't install any TAPI driver on my XP platform. A. Within Config Panel|Modems and Telephony options|Advanced parameters, I've got a list of 7 TAPI drivers. Among them is Omniis TAPI driver for Asterisk. B. I can properly configure this driver (line, context, ...). C. When I open Outlook 2002 Contacts panel, I can select "Call this contact"

Hi, This is a bit off-topic, but still related to telephony. Is there a barebones TAPI driver that exists that would allow me to call up a command line with, as parameter, the number to dial. For exemple, Outlook integrates with TAPI, so that TAPI driver would allow me to call my own app with the phone number as argument. ex when clicking on 555-555-5555: the TAPI driver would call

Gentlemen We are attempting to use * in a simple switching application: +-----> office lines | V LEC <--PRI--> * <--PRI--> modem bank (56k dialup modems) The problem is that (even with no office lines active) the modems have difficulty establishing a connection, the connection is slow (way too slow for 56k modems) and the connections are

I have a friend with a PRI coming into a modem bank that is receiving 56K modem calls and some ISDN data calls. He wants to dump his analog office phone lines and use some of the capacity on the PRI. I have been digging through the mail archives and Wiki site on this subject but the information I found doesn't give me a high enough confidence to go buy a few T1 cards and try it out.

I have a modem in my system that comes up with this in lspci: 01:07.0 Communication controller: Agere Systems 56k WinModem (rev 01) What programs under CentOS, if any, can use this modem and how? Thanks. mhr -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20070807/fbeb461b/attachment-0001.html>

Hi I am in the UK and am looking for a device that will allow me to connect two sim cards (read wireless lines) to either the port on the back of my fritz card or any other connection direct to the PC that provides a usable telephony interface. I will even plug two devices into a windows box and have that do ISDN to ISDN if required. All I want is two GSM lines that look like voice modems to

Modem connect speeds on calls through * seem to be lower than calls made through the telephone company lines or our old Rolm PBX. All data calls have 2 wire analog modems on both ends. For my set up I have channels of a Zhone channel bank tied to 2 modems. The Zhone channel bank interfaces my * server with a T400P card. modem --- Zhone Channel bank - * via T400P card - Zhone channel bank -

Hi everybody. I know maybe this question has been posted some time ago, but I need your updated opinion on the subject. I'm replacing our old pbx with asterisk. I have two TE207 dual pri (e1) cards on a clustered system (one on each node). I absolutely need to connect 4/5 analog extensions with modems, they're being used for remote assistance on very old systems which cannot be upgraded

Depending on what you are broadcasting, for 28.8 I would go 16kbps and be really safe... Lithium ----- Original Message ----- From: "John Griffiths" <john@capmon.com> To: <icecast@xiph.org> Cc: <icecast@xiph.org> Sent: Thursday, April 05, 2001 3:12 PM Subject: [icecast] bitrate for slow modems > ok so 24kbps for 56k modems... > > can i go any lower and get

Hi, Thest are two methods:- 1) redirect_to :action => ''list'' 2) render :action => ''list'' what is difference between these two methods?????? Thanks. Prash -- Posted via http://www.ruby-forum.com/.

Hi Not entirely centos related but that what the boxes are running so here i go ;) 2 of our centos boxes run nagios for network monitoring and send email and paging alerts and are connected to standard phone lines using modems to send the pages. I'm looking for a way in the event of provider failure to login to these machines and so am looking at the ability to connect to the console over

I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI->T400P->Asterisk->T400P->Adtran 750(L36 Firmware)->RAS Server. I have 4 Zap channels signalled FXO_KS to the 750

Hi all, I've tried researching this, and so far, have struggled to find any contemporary information on the issue, so I do apologise if asking this irritates people who have answered this before. I have managed to set up Asterisk 1.8 on the web server here. I have two softphones (Ekiga) able to communicate with it. So far so good. I'm now curious to see if I can link it with the PSTN

Since the goal is to start llvm-tapi more or less from scratch, I feel the best approach initially is to focus on the structure as a key point of feedback in initial reviews. Once the foundations are set, integrating Mach-O TAPI in parallel with the ELF implementation should be relatively straightforward. The features outside of stubbing aren't as appealing for ELF, so I probably won't be

Hello all, LLVM-TAPI seeks to decouple the necessary link-time information for a dynamic shared object from the implementation of the runtime object. This process will be referred to as dynamic shared object (DSO) stubbing throughout this proposal. A number of projects have implemented their own versions of shared object stubbing for a variety of reasons related to improving the overall linking

Oof, I didn't think about Clang not being in the same place. Perhaps we could put this in clang-tools-extra to solve that? As for the unification of the code bases. I was assuming we didn't want to just throw a ton of code over the wall anyway so the merge was going to need to be reviewed chunk by chunk anyhow. Support for the two formats should be possible to add in parallel (although, I

I've been testing the debug version of AstTAPI, which worked for a few calls, then a bit later in the day (and ever since), when the call is hung up, the TAPI client doesn't get notified. Looking at the server logs, The TAPI message that is sent upon hangup, isn't being sent. exten => h,1,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE) This is in the same context as

I am trying to use a the SIP tapi from www.enum.at <http://www.enum.at/> . This works fine from all kinds of applications which support TAPI, like outlook and Dialer Pro. However when making tapi controlled calls, the signaling to and from PSTN seems to fail. I have used the digium hardware ISDN PRI boards, but also a SIP gateway. Both result in a audio message from asterisk