similar to: Modems compatible with NTL caller id

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

Here's a bit of a twist to the common UK Caller ID question... (Which I've got working nicely thanks to some slight changes in Jonathan McHarg's scripts off the asterisk-dev mailing list, and a Pace modem from ebay!) Can a standard BT phone that supports CID (Such as a BT Decor 310) pick up the CID information that asterisk passes out to analog lines or would I have to get an

Hi, Does anyone know anything about the callerid format that NTL uses here in Cambridge, UK. This is the former Cambridge Cable, who is sometimes different from the rest of NTL. They did say that only some equipment works with their switch. I hoped that they might use US-style CID, which would work with the X100P, but it doesn't seem to come through. I do notice that my DECT phone bought

I'm using a macro such that if a call is forwarded to my mobile, I can choose to accept it or not. This is done with the following Dial command: Dial(IAX2/voiptalk/44${ARG1:1},15,trM(screen)ow) and the following macro: [macro-screen] ; Allow called user to accept/reject the call exten => s,1,Wait(0.2) exten => s,2,Read(Accept|loho-support-choose|1) exten =>

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)

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm just learning about VoIP and Asterisk. I've got a developers kit on its way and I've managed to get hold of a couple of cheap Cisco VIP 30 phones. I've trawled the web and found a few snippets of information on these phones but I still can't get them to work. Does anyone have any config files or any idea on how (if I can)

Hi, I'm writing a Java client/server application that talks to the Asterisk manager interface via the asterisk-java stuff. The idea being it will give you an app to run on your desktop that monitors your phone essentially. Once I've got something vaguely working it will be released under the GPL and hopefully people will contribute to it etc... As part of this, I'm currently

Hi everybody, I would like to set asterisk up such that to use the call parking feature, instead of transferring a call to the extension set up in features.conf, you just dial a code (e.g. *3) and this then parks the call. The main reason for this is that a number of the phones I use have transfer buttons that I can't reprogram to use Asterisk's own transfer functions, therefore you

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've got my developers kit from telappliant and got a machine up and running to become the house phone system. Most things are working now, such as incoming calls, call transfer, call parking, voicemail, etc. The one thing I can't do is make my analog phone ring! I can see the call coming in on the asterisk console and can then pick

Hi, I've got a recent copy of CVS (09/08/04) with the UK CID patch on and working, but I'm having trouble receiving SMS messages. I can send messages without any problems but whenever an incoming message is detected, nothing is left in the /var/spool/asterisk/sms directory. Can anyone help? Thanks, Darren. extensions.conf: [default] exten = s/08005875290,1,SMS(testq,a) exten =

Hi, I have a load of IAX extensions, which I'm trying to set up a standard macro to dial them, which gives unavailable or busy voicemail if there is no answer or the phone is in use respectively. The macro I have at the moment is: ; std-exten macro, ${ARG1} = Device to call, ${ARG2} = voicemail box [macro-std-exten] ; Call the user for 20 seconds exten => s,1,Dial(${ARG1},20,tr) exten

Darren, yes, I'd be happy to help. I'll contact you off list to sort out the arrangements. I should warn you that it may be a wasted journey for you, as I really dont know if it will exhibit the problem. Tim. "Storer, Darren" <starusers@comgate.tv> wrote: __________ >Hi Tim, > >TP> So it _may_ not be a problem for me as NTL is a patchwork >TP> of

Hello again, I'm not attempting an install of Samba 3.0.4 from source. I want to specifically compile in ads and winbindd support. So, I already compiled and installed OpenLDAP 2.2.13. Now the configure script is hung on krb5 dependancy: checking for krb5.h... no configure: error: Active Directory cannot be supported without krb5.h So I downloaded and attempted compile of krb5 1.3.4, but

Hello all, Forgive me as this is probably a common question. However, I could not find an answer while searching. I have a few Linux servers running Samba 2.2.x that are hosting open shares with guest read/write access. I would like to have these become a member of the active directory and allow pass-thru authentication and authorizations to the shares hosted. I find where I can join the

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,

Advance apologies for the length of this mail; 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) and all my SIP extensions (which all work) along with

Kelly Lee Myers: > Depending on what you are broadcasting, for 28.8 I would go 16kbps and be > really safe... well, you could consider not broadcasting at all, that would be 0kbps, and would fit my 9,6kbps gsm connection ;) scnr uno > ----- Original Message ----- > From: "John Griffiths" <john@capmon.com> > To: <icecast@xiph.org> > Cc:

ok so 24kbps for 56k modems... can i go any lower and get the 28 k modems? (still a lot of them about) or will 24 be good enough fo that? At 04:57 PM 4/4/2001 -0700, Jack Moffitt wrote: >> > >> > the bitrate on the mp3's is 128 (great for listening off the HD) >> > >> > whats a good bitrate to encode for streaming audio? >> >> I think this