similar to: pstn call waiting and zap

Hi, I've installed asterisk 1.2 on debian/unstable and i would like to add the chan_celliax channel to my existing configuration in debian, is there a way to do that?? -- patoVala Linux User#280504 Hablando en http://www.elprimoalcahuete.com "<markm> c++: the power, elegance and simplicity of a hand grenade"

Hi, sorry for the question, i've been searching for a real time billing system for asterisk with zap/sip support, for use in post paid systems like "locutorios", do you know of or use any ? thanks -- patoVala Linux User#280504 Hablando en http://www.elprimoalcahuete.com "El tiempo cura los dolores y las querellas porque cambiamos. Ya no somos la misma persona. -- Blaise

Hello all, I have been trying to solve this problem for days, with no luck. When I run an AGI script from my extensions.conf, it seems no matter what I do, the "WAIT FOR DIGIT" command will not work. The system just flies past it without waiting a single millisecond, and of course my script crashes because it doesn't have the input it needs. I have run 3 different versions of

Hello, Is it possible to detect a hook flash in asterisk. I want to be able to perform some functions an hook flash. I have the following entry in features.conf which executes a Macro on detecting key press '**'. [applicationmap] test => **,caller,Macro,testflash Is it possible to do this action on hook flash? -------------- next part -------------- An HTML attachment was

Problem resolved. This makes it nice and simple to 'flash' an incoming POTS line (ZAP channel) as opposed to the dialplan scripts that I have seen that require tranferring the call, hanging up, and waiting for a call back. That was too confusing for my wife. Now all she has to do is pres *3 and it is done. No transfers. No hanging up. No dial back. extensions.conf [context] exten =>

Hi everyone, our setup includes a NEC PBX connected to our asterisk via bri lines. The NEC has a doorphone feature, which is just an extension that calls you when someone rings. When connected to this extensions, a 'flash' signalling opens the door. So now, i'd like to trigger this from asterisk, too, but unfortunately wasn't able to do so. Setup: asterisk

So far no one I have talked to has either had this issue or does not know a answer. I currently run asterisk 1.0.9. I have two issues to deal with. 1. The caller waiting caller ID does not show on the uniden hand held that is hooked to a sipura spa 1000 or my Polycom ip500. 2. When a caller is calling in and I hear the caller waiting beep on the line when talking with somone, is there no way

Hello, I''m using these rules (attached) to control traffic going out from ip 192.168.0.100 which is acting as a p2p server, but when I have these rules on and mldonkey running I experience some latency in web pages, which Iwould like to eliminate. I''ve read that this is where the burst and cburst (even quantum) parameters are useful, but I still can''t understand

Hi Guys!! I wrote a little patch for asterisk 1.2.5 and I will maintain it for future release unless somebody explains me how we can ask people at Digium to add it to the source tree... We are planning on using Asterisk as our main PBX for the office over the next few weeks. Our current setup uses TDM400 cards to bring our 8 lines into Asterisk, our Telco provides us an option for three

http://wiki.dovecot.org/Plugins/Zlib says pop3 and imap can use this plugin. Does deliver also use it? -- Ralf Hildebrandt (Ralf.Hildebrandt at charite.de) snickebo at charite.de Postfix - Einrichtung, Betrieb und Wartung Tel. +49 (0)30-450 570-155 http://www.arschkrebs.de I'm looking for a job T-Shirt I need to see - SARAH PALIN: IN ENGLAND WE

Hello people, Since the invention of this storage called dbox, I have never quite gotten round to understanding it, nor getting to use it. It seems so strange to me and I must ask a few stupid questions about it. I use the MTA to deliver mail to Maildir, either in ~/Maildir or /some/path/%d/%n/Maildir. It has always beat me how dbox comes into play under such circumstances. From the wiki, I see

Hello. After doing much web searching and searching archives of this mailing list, I see that my question has been asked at least 6 separate times but no answers have been attached. In a nutshell, is there a way for a SIP phone to easily hook flash a ZAP analog trunk mid-call? (This is important when trying to make use of features on a PSTN analog line such as call waiting, call forwarding,

Greetings people; The section of drakeconf that pclos uses has two problems. First is several screens full of complaints about deprecated syntax in "/etc/udev/rules.d/70-nut-usbups.rules" at each reboot. So I pulled the current 2.5 svn trunk, then found I was missing quite a few build tools needed here that aren't used when building a kernel, currently running a 32 bit

Hi, I have a problem with a TDM400P card configuration. Incoming calls are answered by asterisk, asterisk place the call on the destination ATA/analog-phone, the phone begins to ring and when our recepcionist pickup the phone to play a welcome message, she nothing hear on the line during five or six seconds repeating the message three or four times until someone appear on the phone. I was playing