similar to: StopPlayTones() after first digit?

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'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, What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. I tried to look at WaitExten and PlayTones, but they seem to not work together - WaitExten doesn't interrupt going on PlayTones. Is there any way how i could do that - so that it looks really natural? It would be silly to create long-long-long

Hello list, I've got a problem (maybe only a problem of understanding how * works) with IAX and WaitExten. To simplify the problem I've brought it down to the following scenario: - 3 Asterisk Server A,B and C (central). - A and B both register with C. Now I want to be able to dial an extension at A to become connected to C and there I want to dial an extension to become connected to B.

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

I had to switch quickly to 1.4.9 from1.2.4 and now I can only get 2 digits into the dialplan. error -- Invalid extension '81' in context 'impact' on SIP/207.174.111.34-b77167f8 I pressed 8107 and ideas my dial plan is (part of it) [impact] exten=>s,1,Answer() exten=>s,n,Set(CALLERID(name)=Impact - ${CALLERID(number)}) exten=>s,n,Background(IMPACT)

OK here is a possible tricky one. I have a rocom door entry system which connects to an FXS port on my TDM400P card. When the door button is pressed it initiates an 's' extension which dials a number of SIP extensions. When a SIP phone is picked up the user can speak to the person at the door and press the 7 digit which sends at DTMF tone to the rocom unit opening the door. All this

I'm now up&running with - mISDN with avmfritz driver for Fritz PCI card - chan_capi from debian recompiled with a patch (see below) - EuroISDN with Point-to-Point (ptp) mode (Austria) - With Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k from debian sarge But am having some problems: 1) I needed to patch chan_capi.c from debian sarge (see below) to give the new channel to asterisk in state

Hello all, HI i am very new to asterisk and my boss needs me to investigate setting up asterisk for a new client. I have downloaded and installed (make, make install and make progdocs)asterisk on my personal computer and when i try to run it (./asterisk -vvvc) i get the following output below: NOTE: i am running REDHAT 9.0 on a 796MHz cpu machine: I am excited to be able to work with asterisk

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

1) I'm working on a quick replacement for DISA, and I ran into the following snag: When I specify "Playtones(dial)" I can only get around 7 seconds of wait time before the dialtone stops, and the context goes to the "h" extension. Is there a way around this fixed timeout? The DigitTimeout setting doesn't seem to have any effect at all on this hangup problem. I

When I do a 'asterisk -vvvvvc' I get following, but asterisk does NOT stay up: == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found

Hi: I deleted old modules in /usr/lib/asterisk/modules before make install. I built zaptel and libpri before asterisk. Modprobe zaptel and modprobe -v wctdm executed witiout complaint. Also, ztcfg shows 4 channels configured, as it should. Starting asterisk produced the output below with several warnings and a failure. Can someone help, please. I double-spaced the warnings in the text below

Calling all Polycom gurus: I am using Polycom IP601 phones with Asterisk 1.4.21.2 In all Polycom phones, I set the following in sip.cfg. <dialplan dialplan.impossibleMatchHandling="2"> </dialplan> (I leave the digitmap unchanged because I thought setting impossibleMatchHandling will ignore the bitmap) ...so that I could dial any number by entering a variable-size

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.

I am trying to setup an interactive menu where a caller hits the main menu and can then dial an extension. As far as I can tell the "Waitexten" app is failing to read 3 digits and just reading the first and then announcing that it is invalid since all extensions are 3 digits. How do I make Waitexten wait for 3 digits? I have setup the extension "100" for users to reach the

Hi all, I have just setup a trixbox system and I am implementing hylafax/iaxmodem solution for the faxing. When i send a fax to it by phoning in listening to the IVR and manually pressing start to initate the fax, the call gets picked up correctly as a fax and everything works well. When I try sending a fax by entering the phone number and pressing start to initiate dialing it sounds like

Hello group I just update to the newest CVS now I'm not able to get asterisk to start. No error during the make or make install I did a make clean before the make;make install Any help would be great!!!! Here is the output asterisk -vvvvvgcd Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf == Binding realtime_ext to mysql/realtime/extensions_table == Binding

Hi All, I am a newbie and having some trouble getting Asterisk to work. I have checked out zapata zaptel libpri and asterisk (both v1-0_stable and regular--in separate directories). All built according to the documentation I have found and installed correctly. I have modified/created zaptel.conf zapata.conf as attached. I get the following messages/error during asterisk startup. ==

Hi: I deleted old modules in /usr/lib/asterisk/modules before make install. I built zaptel and libpri before asterisk. Modprobe zaptel and modprobe -v wctdm executed witiout complaint. Starting asterisk produced the output below with several warnings and a failure. Can someone help, please. I double-spaced the warnings in the text below. The first warning is about music on hold because it