similar to: X100P call progress detection in UK

Hi- Having a little trouble with the SMS application in the UK. I've read the wiki topic and also the BT documentation SIN 413, which describes the service technically. I've been able to successfully send messages from my customer's ISDN landline to various mobiles, but cannot get the mobiles to be able to send messages back to the landline. I've tried sending from both Cellnet

Hello This is a fun one for the list... Twice now, the Police have contacted us to say they have had a silent call then hangup from our landline number to the 999 service. As a matter of course, they follow up these calls in case someone is in distress. Nobody here was in distress - well, no more than normal! The Police aren't hugely happy when we tell them it must be a mistake. Thing

Hello, I need to write a script which prompts the callee to type a number, and then read it back to them as confirmation: ======= extensions.conf [robocall] ;Expect 10-digit number excluding final #, 2 tries, 20s time-out exten => s,n(nbr2call),Read(NBR2CALL,please-type-number,10,,2,20) exten => s,n,GotoIf($[${LEN(${NBR2CALL})} != 10]?end) ;exten => s,n,SayDigits(${NBR2CALL}) exten

All, I'm running CVS-HEAD as of 15thJune with an x100p and the x100p callerid patch. I'm trying to use app_sms to recieve sms to my landline but get the following response. Any ideas. -- Executing NoOp("Zap/1-1", "Testing without Answer 08005875290") in new stack -- Executing GotoIf("Zap/1-1", "0?s|5") in new stack -- Executing

Hi. I tested to use samba-2.2.1a at my environment. the strange phenomenon appared that smbd consumed continuous cpu resources. I tested linux-2.2, linux-2.4 and linux-2.5 environment, results are same. I use two PCs, one is a linux-box , other is a win2000-pro that is updated all patch from microsoft windows update. I stored a lot of files in my home directory in linux-box about 10 GB. I use a

We're trying to forward an incoming SIP call from voipfone (UK ITSP) that originated from a UK landline back up a SIP trunk to the same ITSP and on to another UK landline number. UK Landline->voipfone->asterisk 1.4->voipfone->UK landline About 1 in 3 times the call at the final landline is silent and we see "RTP Read too short" scrolling on the console log. Where do we

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Hi, I'm running * 1.0.2 . Which patches can I used against this version to get BT callerID to function please? Thanks Mike

Hi A quicky - has anyone had success with an X100P in the UK ? I'm seeing some oddness - when I plug the wall socket into the card, I lose dialtone. If I unplug it, it comes back. Does this sound familiar to anyone ? Cheers Nigel Nigel Taylor Technology Director ITAzure Limited Dunn House Warren Park Way Enderby Leicestershire LE19 4SA 0116 286 3016

Hi, I'm currently running asterisk 1.0.9-r1 and zaptel 1.0.9_p1-r1 (the current gentoo ~x86 versions), with the UK CallerID patches from http://www.lusyn.com/asterisk/patches.html applied. The Zap interface itself seems to work fairly well - although it's a little quiet, there is no echo. Unfortunately, there's also no CallerID. My zapata.conf is as follows: [channels]

Hi, i'm currently using Originate command on AMI, i can call a certain channel like a SIP user SIP/1000 then once 1000 is answered it dials out to amobile or landline. Would just like to know if i can use AMI to dialout to a mobile or landline first (instead of SIP user) and once answered, dial another mobile or landline again. If not is it possible to call a macro from the AMI? i think

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

I was wondering if anyone has put together a comprehensive list (that is reasonably maintained) that lists country codes, landline numbers, mobile numbers, etc. The particular requirement is for a dialplan to know what is going to be charged to whom. For example, mobile and landline rates are the same in the US the US has a unified numbering plan of 1NXXNXXXXXX, while the UK has: 441xxx

downloaded and compiled today's CVS (04/08/2004) tried using callprogress on Via mini-itx (running RedHat Linux 9) if callprogress is set to yes on x100p, an i call the line connected to x100p, asterisk would execute the first app and will wait forever. anyone had success using callprogress? thanks. __________________________________ Do you Yahoo!? Yahoo! Small Business $15K Web Design

Hi, I've got the latest asterisk@home setup running with SIP. I've been trying to tie it in with the PSTN off and on for a while with no success. I gave up on ISDN and purchased 2 x100p cards from x100p.com. I've got the card installed, the machine can see it. The problems I have at the moment are First time I start up the drivers and asterisk, I can call the pstn number. The

The following is my dialplan for outgoing international call. What I want are: - when people dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, use voipstunt to dial out - otherwise, use my pstn to dial out. What I've found is when i dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, it always use the pstn to dial out. Anything wrong with my dial

Hi, I need to use the trailing 5 digits of a callerid. callerid may be anything from a length of 4 to 10 digits in this case. Using this: ----------- SubString,cid=${CALLERIDNUM}|-5|5 Works great, BUT shows this message: "The use of Substring application is deprecated. Please use ${variable:a:b} instead" So, I try --------- SetVar(cid=${CALLERIDNUM:-5:5}) The result is a empty

Hi, Using latest cvs. A comment-line begins with semicolon ; However - if the line contains ;-- or like this ; -- blabla bla -- You get this error and * stops reading that file: Feb 21 13:47:12 WARNING[17393]: config.c:664 config_text_file_load: Unterminated comment detected beginning on line 0 Shouldn't Asterisk skip any line beginning with a semicolon? Or should a comment now be

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

This is pretty much exactly what I'm doing for my Uni project.....but just can't get Icecast and Ices0.3 to work. :( <p>>From: "Stefan Neufeind" <stefan@neufeind.net> >Reply-To: icecast@xiph.org >To: icecast@xiph.org >Subject: [icecast] Solution for automatic radio? >Date: Sun, 14 Mar 2004 16:05:35 +0100 > >Hi, > >I've read various