similar to: Ringing for incoming call

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

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.

Greetings all- I've got a rather odd situation and would like to know if anyone can shed some light on the issue. Some background- I've got an * system running 1.4.11 (yes I know it's older.. upgrades are planned at some point...). I also have a remote user with a cordless phone connected to a Draytek ATA device. When this user is on a call and receives another call via call

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

Has anyone ever tried to roll out a *66 auto-callback-redial feature on asterisk? I'm sure that implementing this for outbound Zap calls would be a nightmare, but what about something easier, like internal extensions? On my old Panasonic key system, it used to be such that, if the called extensions were busy, you could press 6 while hearing the busy signal, it would beep twice and hangup.

I'm struggling to get my dialplan to work with a simple analog fax machine. I have TDM400B zaptel card with an FXO and FXS port. I have the FXO port connected to the POTS machine and the FAX machine connected to the FXS port. The FAX machine itself works fine, I can FAX outgoing messages fine. I can also dial the FAX extension from the internal context, the FAX machine answers and I

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

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?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

The problem I was experiencing is still occurring, and it is getting worse. There are several names that Festival "gets stuck on". I don't know if it is a Festival problem or an Asterisk problem. The scenario, a call comes in goes through the dialplan (shown below in original message), and either reads the value ${FULLNAME}, but then hangs. It doesn't go to the next command

With softphone I mean linphone csipsimple or whatever. How should a dialplan lokks like? On 31.01.19 11:26, Antony Stone wrote: > On Thursday 31 January 2019 at 10:59:01, basti wrote: > >> Hello I use this dial paln: >> >> [o2-in] >> exten => o2,1,Answer >> exten => o2,n,Playback(hello-world) >> exten => o2,n,Ringing >> exten =>

Hi, all. I'm probably doing Something Dumb(tm), so please feel free to point out whatever I'm missing, no matter how stupid. Anyway, I've got IAX set up to Vitelity. When I try to call my DID, I get: Rejected connect attempt from 64.2.142.19, who was trying to reach '6031234567@' This leads me to my first question -- why doesn't it show a context? (My second is,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am having problems with the PlayTones application and VoIP softphones. I have the following in my extensions.conf: exten => 123,1,Answer exten => 123,2,PlayTones(Busy) exten => 123,3,Hangup But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. I get the following on the console: --

I configured our SIP gateway to automatically dial extension "s" when a phone is picked up. I want Asterisk to play a dial tone, wait for an extension to be dialled, and hangup on timeout This works great, but I also want Asterisk to *stop* playing the dial tone after the first digit is pressed So far my extensions.conf contains, [internal] exten => s,1,Answer exten =>

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

I've searched around and found a few similar situations where the phone will call out when using a Asterisk server but not receive inbound calls. My issue is a little stranger. If I call out from the phone then the phone will receive the next inbound call. The phone will not receive another inbound call until a call out again from it first. Any ideas? I am using SIP and am using the latest

Hello I use this dial paln: [o2-in] exten => o2,1,Answer exten => o2,n,Playback(hello-world) exten => o2,n,Ringing exten => o2,n,Dial(SIP/10&SIP/20&Local/s at no-op,25,rt) exten => o2,n,Playtones(425/1000,0/4000) exten => o2,n,Wait(30) exten => o2,n,Hangup() All is fine. Hello world is Playback and I hear a ring tone. If I remove the Playback hello-world. No ring

Hi there, I've got a small problem with the zaphfc channel. No MSN of an any incoming call which comes trough the ISDN card (Acer ISDN, with HFC chipset and zaphfc driver) which will be forwarded to the SIP-Phone will be displayed. Always it will be shown "asterisk" an the Display. --- snip (zapata.conf) --- [channels] language=de switchtype = euroisdn signalling =

I have a Asterisk server setup and I am able to connect to the server using a soft client 'x-lite' and call and leave a message on my second extension 102. I have setup a Vitelity account and add what I believe to be the correct information to my sip.conf and extension.conf. I would like to setup incoming and outgoing calls with voicemail support. I've searched all over but many of the