similar to: FYI: BT Caller ID.

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

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

Hello I was wondering whether anyone has managed to get an X100P card to receive Caller ID information in the UK (connected to a BT line)? I guess that it probably won't work at present because BT sends the CID information before the first ring - hence there is no event to wake up the channel. I can hear it using ztmonitor, but it is far too soon for anything to receive it. According to the

Hi Just moved offices in the UK and moved our Asterisk box from old one to new one. Using idefisk softphones, Junghanns quadbri card for ISDN 2e interfaces. At both offices we had one standard number and a DDI range, routed with Asterisk. We'd set up the configuration so each idefisk set its own caller ID which then got sent by the extensions.conf script. Worked fine at old place but in

Hi all, I can't seem to find any info on this anywhere on the web, except that BT caller ID doesnt use the standard bellcore system in use in the US. So, if anyone here in the UK is onlist and using the x100p successfully, please let me know. TIA, Dave

Any body used patch, http://bugs.digium.com/bug_view_page.php?bug_id=0001719 to get the callerid for BT Line. I applied the patch successfully but could not get it to work. Any help. Here are the logs: -- Starting simple switch on 'Zap/1-1' Jun 17 18:22:31 NOTICE[426000]: chan_zap.c:4811 ss_thread: Got event 2 (Ring/Answered)... Jun 17 18:22:34 NOTICE[426000]: chan_zap.c:4811

-----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 all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : "TDM400P & Not Detecting Hangups: Got a TDM400P installed and having problems with Asterisk not detecting

-----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)

I don't know if this has been mentioned here or even thought of in the grand scheme of things and I'm just sending a one time email about it "just in case" I recently swapped my server around and I'm using dovecot now....LOVE it! Could use a bit more intuitive error messages, but otherwise, it's great! But there is one flaw that I've run into with my users.

Hi, got a Sangoma A200 with a bunch of extension cards and having real problems getting it to deal with a normal single BT line Symptoms are that incoming calls are fine. Outgoing calls ring the far end, BUT asterisk never sees that the call is answered (ie no message in the logs files saying so), as a result the remove end can hear the PBX side talking, but there is no audio back from the

Just hit Slashdot: http://slashdot.org/article.pl?sid=06/06/08/1725215&threshold=1 http://money.cnn.com/2006/06/07/news/companies/pluggedin_fortune/index.htm >From TFA: "But what's really cool about what will happen in Cardiff - and eventually the rest of the U.K. - is that BT is creating an open, standards-based platform for which anyone can develop new applications. In other

Have anyone tried to interface BT's Broadband Voice with asterisk? Kannaiyan

As a general rule of thumb it would be good to make the distinction between 'Credit Card' and 'Debit Card' too. If possible, never ever use a debit card for online purchases. It taps directly into your account and removes REAL money. Credit cards are 'virtual' money in that they are credit and your cash in account remains untouched. That way if your credit card account

Hi all, I've been playing about with Asterisk for years now on and off, just SIP to SIP calls, using FWD and suchlike. I'm moving house at the beginning of September and have decided to build an Asterisk based system for my home office. I'm in the UK and wonder if anyone can give me advice on lines and hardware to use. Had planned to go with an ISDN2e line coupled with a BT

Howdy, My first forray into using res_mysql.conf for realtime access of sip users and extensions. I have the following relevant section of extensions.conf: --- [trunklocal] exten => _NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [local] include => trunklocal include => trunktollfree [longdistance] include => local include => trunkld [international] include

I'm considering purchasing a Digium TDM400 card with FXO module to connect my asterisk box up to BT's PSTN. I've searched Google and have found that in the past there have been problems with this card detecting the calling party hangup. I am interested to know if this issue has now been resolved and if UK caller ID is now fully supported. Also if anybody using the card in the UK is

Well, I did a test install of 3.0.0 today (different base from the old 2.2.7a incase it all went wrong). Everything seems to work (minor glitch failing to read the documentation carefully enough), but the roaming profiles appear to be broken. What I'm seeing is that the profile isn't loaded and a vaguely default profile is provided instead. I'm not seeing any error messages

Hello ppl, Sorry to re-post it, but kinda these issues are getting on my nerves. I tried Set(CALLERID(num)=7329) on 1.2.12, which works fine, but not on 1.4.4. The problem : 1. I receive call from caller 'AAA' on my number, 'BBB' which is on my Asterisk box. 2. I have to redirect the call to some other number, say, my cell num - 'CCC'. 3. My PSTN provider wants the

Is it possible to reject all incoming calls that do not have a CID? Could I do something like that (modified version from the book): exten => 123,1,GotoIf($[${CALLERIDNUM} = ]?20:10) exten => 123,10,Dial(Zap/4) exten => 123,20,Playback(abandon-all-hope) exten => 123,21,Hangup( Alternatively, what's the privacy.conf file for? What does it mean for a user to have to chances to