similar to: Anyone know about callerid format used by NTL in Cambridge?

I'm looking at using a modem to provide caller ID info on my NTL line following the steps in the article posted by Darren Poulson: http://www.22balmoralroad.net/modules.php?name=Sections&op=viewarticle&artid=1 I was wondering if anybody had experience of using any modems (can be pci/isa/external) with an ntl line (in an ex. Cable & Wireless area if it makes a difference) and

Anyone here that knows some hardware USB "client" (DECT) that can connect to a DECT base station and act as a "provider" in asterisk? Ergo, asterisk is the DECT handset, and connects to the DECT base station. Its also important that it also works with most asterisk-compatible devices, including for example raspberry's. The reason I ask, is that many providers in

Hi all, I'm currently on an NTL PSTN line, using Asterisk 1.2 and a X101P cheapo card. The problem lies with detecting when the far end has hung up. It fails to detect it, and will only cleardown when the silence timeout has been reached. Now, I've seen the thread at http://www.mail-archive.com/asterisk-users@lists.digium.com/msg32337.html, to which nothing has come of it. That was

Hi, I've today received an E100P. Can anyone help me out with the config of this card please. We have an NTL PRI circuit installed with 10 channels active. Also what do the LEDs on the back of the card mean ? Many thanks Jon

Chris you might know the answer to my HUUUUUUGE problem A few weeks ago you posted this message: "I have an ISDN PRI supplied by NTL (ex Diamond Cable, Nottingham) which is currently working happily with an SDX Index phone system. I have to replace this phone system shortly and I've been trying to get a * system working for some weeks now. I have configured the dial plan (which works)

On Sat, 2020-10-03 at 22:25 +0200, Sebastian Nielsen wrote: > many providers in sweden have started disabling SIP account details > and now require usage of their own ”router’s”. That's very irritating and make me angry. Few of my client had the same problem. The solution: write a letter asking the SIP credentials explaining you want configure your own equipment and tell them you

Firstly, let me just say I am new to asterisk and if anything I've said is covered in an FAQ or in previous posts I apologise but I have tried searching and I've attempted a few of the things I found but they didn't help. Has anybody got any experience using an X100P on an NTL phone line in the UK (I'm in an ex Cable & Wireless area if that makes any difference). The

Hi, I've installed a TDM400 card in an aging Dell Optiplex GXa (see my post a few weeks ago). The machine powers the card okay, it shows up in an 'lspci', and asterisk runs fine with it. I've tried both 1.0.4 and 1.0.5. The box in question is running a 2.4 kernel. However... I'm having trouble getting the card to work with a variaty of handsets attached to the FXS ports.

Hi, I'm away at a conference in Amsterdam. My home is in Cambridge in the UK. On a whim, I tossed an ATA186 and a phone into my bags before leaving home. I was able to plug my ATA186 into a LAN here at the conference and was connected to my home Asterisk in a few seconds. Total time from unzipping my bag to talking to home no more than 15 seconds. OK, so the kit could be more portable,

Hi all, I am close to purchasing some new DECT phones for our home office here in the UK. We use Asterisk and I am sorely tempted by the Siemens C475IP or the "soon-to-become-available-in-the-uk" S685IP. Both systems have great feature sets and, on-paper at least, look to be the bee's knees. Anyone got any skeletons on them? Thanks Alan -- The way out is open!

Hi, I'm having trouble getting caller*id to appear on my phone connected to an ATA186, and being called from Asterisk. Does anyone out there successfully see callerid on their ata186-connected phone? The "From:" header in the INVITE to the ATA seems to have the "right stuff" - eg From: "Study phone" <sip:6002@195.217.255.45:5062>;tag=as412db061 But

Get 20% off selected Cambridge books for the R-help community. Until the end of June, R-help users can get an exclusive discount on the following Cambridge University Press books that use R or S-plus! Simply visit http://www.cambridge.org/rbooks and click the appropriate link to get more information about the book(s) that interests you. To get your discount, you will need to click the

Hi everyone, Today I tried to use the CD-ROM that came with my "Cambridge Grammar of English" on Ubuntu 8.10 and Wine 1.1.14 The install went ok, but when I try to run it I get the following error: Code: $ wine start 'C:\Program Files\Cambridge\Cambridge Grammar of English\cge.exe' fixme:exec:SHELL_execute flags ignored: 0x00000100 vwilson at

Hello again LLVM people in Cambridge (or London, or places similarly easily accessible on the train), We plan on holding the next Cambridge LLVM social on Wednesday the 20th of June, from 7:30pm onwards, in The Cambridge Blue: http://www.the-cambridgeblue.co.uk/blueroadmap.html The pub does food as well as beer, so at least some of us will be eating there. I hope to see you there, David

Hello any LLVM people in or near Cambridge (the original one, not the knock-off in Massachusetts), Chandler's closing speech at EuroLLVM indicated that LLVM developers on this side of the pond don't spend as much time in the pub as our Californian counterparts and that this was an important thing to address. To address this, I propose that we start a regular Cambridge LLVM social. The

I've got the following setup: IAXy -> Dect Base Station. When you dial from a SIP phone (cisco 7960), the rings with very little delay. However, if you hangup it takes 3-4 rings after hanging up before the dect base station phone stops ringing. The same applies when an incoming call is directed from PSTN FXO -> Dect Base. Is there a fix to this I've looked about on voip-info but

The Boathouse is nice - on the river and with reasonable beer and food. The Old Spring is also quite nice, has big tables outside, and is usually quite quiet midweek. David (Who has not yet completed the process of performing an exhaustive survey of Cambridge pubs) On 19 Jun 2013, at 16:46, Steve Montgomery <stephen.montgomery3 at btinternet.com> wrote: > Just cycled past The Fort.

Hello Everyone - Well, I think I'm getting closer with the asterisk connection. This is my setup and I keep getting this error below in ,my /var/log/asterisk/messages file. I have opened 5060 port on the firewall box. I would this is Warning which I can ignore! But I see the connetcion coming but NO DIAL TONE on mt ata186 box sitting in my 192.168.200.x site! I'm using ATA186(cisco

I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn on the mark echo canceller. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, April 13,

The tentative date for the next Cambridge LLVM Social is Wed 22nd October, 7:30pm at the Cambridge Blue. http://www.the-cambridgeblue.co.uk/blueroadmap.html If you haven't done so yet, you can subscribe to our calendar to know when the next one will be: https://www.google.com/calendar/embed?src=as5fh066i94qthrkct1lv7rojo%40group.calendar.google.com&ctz=Europe/London Most people feel