similar to: *66 auto redial emulation?

I'm trying to find a way to generate an 'internal extensions' tonelist but I can't seem to find anything on how to do this. My idea was to start a Playtones(intercom) tonelist and not indicate ringing to the line (dead air). But then, somehow StopPlayTones needs to be run once the ringing telephone picks up. This seems like a dirty way to do this. I envision an option to the

This is a new post, but its really a three-time retread. I hope someone has a clue on this, as it could be helpful in many circumstances: I am looking for a way to dial 'special' an extension (in house, like 102), which are all Polycom IP. I'd like to ring the extension as normal, but have the option of, while the line is ringing, to press a digit, hop out to a new context and/or

Hi everyone, I'm in the very early stages of rolling out an asterisk box at work, and one of the things I'm setting up is a trap for telemarketers >;) What I want to do is have a sipgate number in the UK here which rings for 10 seconds without calling a hard or softphone, then goes to a voicemailbox. The problem I'm having is that Playtones doesn't seem to be sending any

exten => did,1,Answer exten => did,n,Playtones(ring) exten => did,n,Wait(10) exten => did,n,StopPlaytones() exten => did,n,BackGround(sound file) did = the DID number as presented and note the '1' before Answer. This works for me. exten => 820055,1,Answer() exten => 820055,n,PlayTones(ring) exten => 820055,n,Wait(5) exten => 820055,n,StopPlayTones() exten

Hello list I had the following echo-test extension on my Asterisk 1.2 setup. exten => 1003,1,Wait(1) exten => 1003,n,Playtones(!1050/1000) exten => 1003,n,Wait(1) exten => 1003,n,StopPlaytones exten => 1003,n,Echo exten => 1003,n,Hangup After migrating my testing server to Asterisk 1.4, and a minor extensions.conf update, everything works just fine. Except for the Playtones

Does anyone have a quick-n-dirty context to implement *66 automatic callbacks? I have a few people who like to have no call waiting on their phone (can you really blame them?) It would be nice to have something like *66, and also like 'Camp On', but instead of waiting something like 30 seconds, monitor the channel until it becomes available, then immediately ring back your phone to

Hi, I've successfully used the 'd' flag in Dial() so that when I dial into my phone system from out there in the PSTN network I can press the 2 key while the phone is ringing to listen to my voicemail. It seems that one issue is that the public providers do not deliver DTMF, or anything, until the phone is answered. This is for security reasons and sounds like a good idea to me.

ðÏÞÅÍÕ ÂÙ É ÎÅÔ? ÷ÏÐÒÏÓ ÔÏÌØËÏ × ÔÏÍ, ËÁË Õ ÔÅÂÑ ÂÕÄÅÔ ÓÏÅÄÉÎÑÔØÓÑ ÐÁÎÁÓÏÎÉË Ó ÁÓÔÅÒÉÓËÏÍ. Date: Wed, 30 Jul 2003 20:06:17 +0400 From: Pavel Zheltouhov <pavlo@comlink.ru> To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk,ata186 and Panasonic TD1232 Reply-To: asterisk-users@lists.digium.com I have Panasonic TD1232 pbx, few cisco ata186 and linux box with asterisk.

Does someone have a link to step-by-step instructions to making the Line-In on the console sound card a MOH source? I know this has to work somehow. Chris Coulthurst <mailto:chris@shuksan.com> chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050512/4a3c3025/attachment.htm

Get the new firmware - it's supposed to have changed the callerid display presentation to include name and number. _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Coulthurst Sent: Friday, May 20, 2005 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Displayed CallerID on

Is there a way to make an outside call hear "The person at phone number XXXX is unavail", but when an internal extension calls another extension, they hear "The person at extension number XXXX is unavail"? I swear I've read this somewhere before but I'm not typing in the right search. I probably found it before by complete accident. Of course, we want the outside

Hi, is there a way to enable dtmf detection on zap channels? I am trying to pickup, play a ringtone and the dial out. I.e. exten => s,1,Wait,1 exten => s,1,Answer exten => s,2,Playtones(dial) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => _X,1,StopPlaytones exten => _X,2,Dial,Zap/g8/BYEXTENSION|10

Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension

I have Panasonic TD1232 pbx, few cisco ata186 and linux box with asterisk. Can I dial from asterisk into ata, then indicate phone number playing tone (use DISA feature at panasonic) and connect to any analog phone connected to panasonic ? I think some of Playtones application within Dial application can help me. But I don't know how. -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia

I?ve got an interesting problem and am wondering if anyone can shed light ? I am running Asterisk on RHEL Server release 5.2 connecting to an Avaya Definity G3R via a Digium TE220. Asterisk 1.4.20 Zaptel 1.4.4 Libpri 1.4.4 MySQL 5.0.45 Festival Speech Synthesis System: 1.95 We have about 4200 accounts in a MySQL db. Asterisk retrieves the user information from the database, festival tts says

Does anyone have success in upgrading Aastra/Sayson 480i CT firmware? All I get, no matter what I've tried is "Unable to upgrade firmware". tftpd is working because the dialplan freshens, and I have aastra.cfg whatevermacaddressfile.cfg and firmware.st in /tftpboot Am I missing something stupid? Is there another way to upgrade it? Chris Coulthurst chris@shuksan.com

Does anyone know how to change the display format of caller id on the screen of a polycom 300/500/600? When I call FROM my 'shop phone 203' TO my 'office phone 201', a Polycom 500, it only says 'Shop' as the calling party. More specifically, the two lines look like this: Incoming call from: Shop I'm looking for a way to make it use both lines for caller id,

Can anyone explain the Polycom Text Messaging features built in to the IP 500/600? Can Asterisk (or something else) talk to it? I've seen vague references to MSN Messenger, and somehow that's mentally disturbing. Chris Coulthurst <mailto:chris@shuksan.com> chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Guys, I'm looking for a light on the next problem integrating an Asterisk server with a Panasonic PBX using a PRI. The follow config has been working for two weeks and works almost fine, except for some little problems... 1- PSTN provides the E1/PRI (EuroISDN) 2- Asterisk receives the PRI 3- Asterisk provides a second PRI to Panasonic 4- Panasonic receives the asterisk PRI People can

I had CC readers going over the internet (with pings over 80ms) connected to Linksys PAP2. It was only successful once every 3 attempts. I had 100% reliability when it was connected on LAN. Timing is an issue, if you doing everything on LAN it should not be a problem. Just make sure you use G.711 protocol. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com