similar to: Call Screening Not Working

I am trying to use a macro to screen calls by calling several different phones at the same time in different groups. Find me will not work and queues will not work either. Trust me, I've tried them both and they don't work like they should. Here is what I have: A call comes into 6084 and does the following (in default context): exten => 6084,1,NoOp(test) exten =>

I am following the example at http://www.voip-info.org/wiki/view/Asterisk+tips+findme but I find that no matter what, the call is connected. Can anyone confirm that config is working for them? Any suggestions appreciated. I need to transfer calls to a list of cell phones, ring all of them, allow them to screen the call, connect the call to the first number that accepts the call, and allow

Hi list, We're trying to set up a phone number that customers can call to get a hold of anyone of a group of sysadmins (and not their voice mails!). We found the findme example ([1]) that makes the callees press 1 to accept the call. It almost works, but it doesn't work correctly when one of the callees, the sysadmins, hangs up after accepting the call. We're using this

All the aforementioned techniques need change everytime on the dialplan. I need the office secretary to edit a file (call file) and place it in a particular folder in their windows PCs. this folder is the outgoing folder of LINUX shared through samba in LAN. i need to make it as easy as possible, please. On Tue, Mar 20, 2018 at 5:41 PM, Frank Vanoni <mailinglist at linuxista.com> wrote:

Greetings all, I've been working on having Asterisk put a call through to two different numbers, and give the call to the first one that acknowledges by pressing the 1 key. I found an example on the wiki, but I can't get it working. When I answer the call I hear the message telling me to press 1 to connect, and as soon as the message is done, the call is connected. In other words, it

Hi List I have had a look at the various posts on this and seem to be more confused than ever - but then again that's not hard ;-) I am using Version 1.4.33.1 build from the Debian "lenny" distros I am trying to implement a simple screening [macro-screen] exten => s,1,Background(press1) exten => s,n,WaitExten(5) exten => 1,1,NoOp(accepted) ; Dont set a reply so dial

A call comes in and goes into the queue, the queue dials a sip channel using a macro. The macro plays a set of options to the callee and if the callee presses 3 it sets MACRO_RESULT=CONTINUE and the macro ends. For some reason the caller goes back into the queue rather than continueing on in the dial plan. Why is this, i could have sworn in 1.2 if i set MACRO_RESULT=CONTINUE that the

This is a followup to the posting earlier about Hunt Groups with Call Screening. I have implemented the following macro and for some reason the Macro does not exit and continue the context it was called from when the called party rejects the call - It always drops through to the NoOp at the end and connects the call. Below are two examples of the dial commands I am using to call the macro.

Hello list, I'm using the following macro when calling an external callphone/GSM number : [macro-press1] exten => s,1,NoOp() exten => s,n,Playback(/var/lib/asterisk/sounds/prompts/press1) exten => s,n,Read(INPUT,,1,1,1) exten => s,n,NoOp(input : ${INPUT}) exten => s,n,GoToIf($["${INPUT}"=="1"]?exit:hangup) exten => s,n(exit),NoOp(call accepted) exten

Hi, everybody: As part of a paging macro I'm using SendDTMF to send digits to the called party. The section looks like this: exten => s,1,Wait(0.5) exten => s,n,SendDTMF(9531290) exten => s,n,Wait(1.0) exten => s,n,Set(MACRO_RESULT=CONTINUE) To test I direct the call to a live extension just to hear what's happening -- what actually happens is that only the 9 is sent, and

I am running into two problems: 1) The ringing stops during call screening once the extension picks up (but has not yet approved call) When a person calls and choose an extension, the Dial link is called and the person hears the ring -- but as soon as the receiving caller picks up (even though they have to approved the call), the ringing stops... the person calling hears silence. This is

I am trying to make a dialplan that when I dial 90 I can go round a whole set of extensions and leave them a short message, hangup and go on the next one. I use the M facility of dial, with something like this [messages] exten => 90,n(calcnextchan),Set(DIALCHAN=...) exten => 90,n,Dial(${DIALCHAN},30,M(domessage)) exten => 90,n,Goto(calcnextchan) [macro-domessage] exten =>

I am setting up an after-hours on-call system. Someone calls in and requests service, and while they listen to music on hold, we dial out to several people's cell phones and home phones. We don't know if they will be answered by the employee, or by voicemail or a spouse/relative/child/pet. So we play a message that says "press 1 to accept the call" and ask employees to train

Hello, I use the following macro to screen calls when they come in. Priv-instruct-custom says "press 1 to accept, press 2 to reject" However, when no input is made (or the call goes to my mobile's voicemail and therefore no input is made), the result is that the ACCEPT variable is not set and then the GotoIf statement fails. How can I resolve this? Can I simply preset the ACCEPT

Sorry for the crosspost (this was also posted to asterisk-at-uc-dot-org) but I haven't got a response. I have a cell phone added to a queue as a local extension (member => Local/299). I want the cell phone to be able to reject calls to the queue without the person sitting in the queue being hung up on, etc. The way my dialplan is set up, the person hits 1 to answer the call and any other

Is there any way to see the called number when a call gets redirected to the 'a' extension from voicemail? Say x123 calls x456 and it rolls to voicemail. x123 hits * and gets dumped into the 'a' extension in the original context. I need some logic in 'a' to do a database lookup based on the original called number (x456). Any ideas? When I do a test, it appears

Is anybody using 1.2 (or 1.2.1) in a production network using Realtime (voicemail, sip or extensions) with 100+ SIP phones? If so, what are your experiences? We've been running 1.0.3 for about a year and it's been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm afraid of killing our stability. Obviously, we'd do it in stages (upgrade to 1.2, then realtime

Hi. in my system i have a conference room where someone can call it eg 698 dial the PIN eg 1234 and enter the room as a user. The admin enters in through a different number and PIN. I would like to have a call file and call all participants eg 610-619 at certain time of the day and give them access to the conference. During my try i managed to create a call file where it calls the a SIP phone and

I have 20-30 GXP2000's connected to * over a T1 line. Neither end is NAT'd and there is plenty of bandwidth available over the line. The GXP's are 1.1.5.15, which is the latest. I have a problem where the phones keep dropping off of * and I get a "failed to register" message in the log of *. Sometimes they eventually connect and sometimes, I have to reboot them to

Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones....