similar to: Play tone till first digit read

Hello, I've been working on an * dialer application, whereby a requirement is that if no one answers the call, a message must be left on voicemail. I've been using the record(tmp.gsm) function with silence detection enabled to wait for the greeting to finish before speaking. However, on voicemail systems where you can interrupt the greeting with a pound (#) key to access your voicemail

Hello - I'm using ATA-186 devices, with RFC2833 DTMF encoding. I am having problems with routines that input long strings of numbers, in that I am getting more than a small number of double digit entries. As an example, I have a section that asks for the user to enter a call forwarding number, and then puts that number into a database. Almost always, there are double digits when the

If you're an asterisk 1.6 user, and use the 'Directory application', have you noticed that the first keypress is always missed if you press it during the part of the announement where alison says "using you touch tone keypad" If this includes you, have a look at mantis bug https://issues.asterisk.org/view.php?id=16409 If you are keen, please apply the patch and report

I'd like to write a snippet of dialtone that plays dialtone and collects a specific number of digits into a variable. Sort of like READ but with a generated dialtone. Naturally, I want the dialtone to stop playing after the first digit. I can't find this anywhere. Only thing I can think of is a no-password DISA. Is this the correct method? Is there a better one? </edg>

Apologies if this has been brought up before, but extensive googling and digging through my list archive didn't turn anything up. Basically, I'm working on an AGI web app and need to read some digit input. I'm having multiple issues with asterisk interpreting agi commands at the moment, but I figured I'd start with this one. when I call GET DATA or STREAM FILE I don't

Hello all- So I have * up and running and connected to a legacy system via em_w lines and have no trouble dialing from * through the tie line but from the PBX across the tie line I am having intermittant receipt of the DTMF. T-Berd testing is showing that the digits are coming across but * is either missing the first digit consistantly. This seems to me to have something to do with start timers

Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and to only be answered by the Sipura when the extension answers. Has anybody made this work?

Hi, I'm looking fore a way to play a dial tone before our IVR platform answered the phone line. I want to use for the following reason: When a caller calls our Voice Platform, the call will direct dial out to a number. I want to dial out before the inbound call is answered. But now the inbound call here's nothing. When the outdial call is picked the inbound call will here

Hi, After some testing I've found out that my client's hardware recognizes DTMF only if digits are sent 50ms apart with 50ms of tone duration. This was tested using a test device which generates DTMF. Now asterisk doesn't do it by default because digits going out from Asterisk are not being recognized. Using command sendDTMF, I can control inter-digit duration, and using

Hi. I am using *1 in features to initiate a mix monitor recording. However, when I hit *1, the callee hears the courtesy tone which I have, so I know when the recording is started or stopped. This is a problem, particularly in automated system where the beep is mistaken for a tone or other problems. Should I file a bug, or is this going to be fixed? -- Your life is like a penny. You're

Ok, I followed the advice of the list and moved more code into my model from my controller. When developing tests for this new code, I ran into a problem... My model code creates a receipt object and sets some values on it: @receipt = Receipt.new do |r| r.x = 1 r.y = 2 # etc end I wanted to be able to stub out Receipt.new so that I could set expectations on the methods called on the

Hi everybody, Every time callers reach my FXO port, asterisk produces one ring tone just before it executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto -------------- next part -------------- An HTML attachment was scrubbed... URL:

I read the wiki and looked at the examples, but I'm still having problems. I have a Digium 4 port card with POTS lines plugged into all four ports. How do I play the congestion tone the the caller when they try and dial out but all the lines are in use? should something like this work? [dial-trunklocal] ; Local calls ignorepat => 9 exten => _9NXXXXXX,1,Dial(${TRUNK1}/${EXTEN:1}) exten

R-helpers: What would be the absolute fastest way to make a large "empty" file (e.g. filled with all zeroes) on disk, given a byte size and a given number number of empty values. I know I can use writeBin, but the "object" in this case may be far too large to store in main memory. I'm asking because I'm going to use this file in conjunction with mmap to do parallel

Hi all, The thing that has bugged me about Asterisk since I first started playing with it, is the fact that the pound sign/hash/octothorp doesn't resolve digit conflicts or cancel timing on a variable length string such as a tie line code or when you call numbers in a country whose length can be different between numbers in the same plan. In North America, we see this when calling

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 guys, I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P. Input calls VOIP Proider ---> Asterisk ---> Alcatel Output Calls VOIP Proider <--- Asterisk <--- Alcatel In alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems: 1. When users dial 2 on phone (alcatel) they don't received a dial tone,

Hello, I did a fresh install of ruby182 and gem rails --include-dependencies Now when I do: rails receipts cd receipts ruby script\generate scaffold receipt receipt rails does not create the views or controller. What can I do? Thanks Frank

Hi Madushan Maybe I was not clear ?. After SIP negotiation and SDP set up on the VoiceMail Server ?. Is there a file to specify a MGw (the machine that deliver RTP packages to end user)? From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Madushan Geethanga Sent: 15 July 2016 13:00 To: Asterisk Users Mailing List - Non-Commercial

I need to set up an office full of Cisco 7960 phones behind NAT with the server out in Colo. The first test phone registers fine, but the second one does not register. The first phone's registration looks like so: /SIP/Registry/3115552368 :64.169.xx.yyy:38836:3600:3115552368:sip:3115552368@64.169.xx.yyy:5060 When the second phone tries to register, it gets back a 404 not found. Not a