similar to: setting up AMD

I'm trying to get a script working on a website to send out automatic email reminders to customers reminding them monthly to change furnace filters. I haven't got one running successfully, yet. That made me think - could it be done with a phone call using Asterisk? A monthly automated phone call to remind people to change their furnace filter? I have no ability to figure this out

Though asterisk support AMD which is based on silence detection but I did not found support of tone / beep detection in asterisk to record a voice message for answering machines after detecting tone Will appreciate any help in this regard Best Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT Unified Communication Telemarketing

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

Hi, I have an Alteon in test (a sip/rtp load balancer). This Alteon sends to the asterisk box a "SIP OPTIONS" to know if asterisk is alive. However, asterisk sends me a 404 message and not a response like, for example, a Thomson (200 + SDP) I wrote a very little script (you can find it at the end of the email) to send an Options message to asterisk/phones to try. It works

Hi All, I have been struggling with the auto dial out in asterisk. I am trying to get a call to be auto dialed and play back a message once the line is answered. So far I have been unsuccessful. Currently what happens is I have my .call file. I mv it into /var/spool/asterisk/outgoing. The call is initiated and that all works, my problem is that it does not wait for the line to be

I'm in the process of writing a simple autodialer to dial a list of numbers and play a message. One of the options I want to give them is a way to "dial X to have a customer service representative call you" Looking for a simple way to pass the number that I dialed to a script in extensions.conf... something like this: [serviceinterruption] exten =>

I've set up an outbound .call system for customer callbacks and the like. Calls are going out over analog lines and I'm trying to use the WaitForSilence routine to make sure the phone has stopped ringing before starting message playback. The problem is that if I set the first argument of WaitForSilence to anything other than 1, WaitForSilence never exits. Some general info on my setup:

Hi All when I try to use auto-dial to connect to outside phone , my applications get executed before the caller attend the calls , this happens only when I call outside no , ie when I use Channel: ZAP/1/0507451111 in my sample.call file , if I use Channel:SIP/326 , it works fine my ?sample.call? file contains Channel: ZAP/1/0507451111 Callerid: Asterisk MaxRetries: 2 RetryTime: 10

Hi all, i'm trying to make AMD, Answer Machine Detection, to work on my outbound context but i can't get it to work, just on inbound context like whe i use the application Answer before AMD, but i need to make AMD to do the detection on an outbound predictive dialer integration. Follow are the inbound and outbound examples. My current environment is Asterisk 1.4beta3 and a Digum

Hi All, I am trying to build a command line with spaces in the argument. This demonstrates what I am trying to do. Clearly the first two commands work fine. However, the last 4 lines to not. /opt/libreoffice5.4/program/soffice.bin --headless --convert-to csv "/tmp/file.xlsx" /opt/libreoffice5.4/program/soffice.bin --headless --convert-to csv "/tmp/file 2.xlsx" MSG="file

Hi, Having some issues here with getting asterisk realtime for the dialplan (extensions.conf) setup: mysql> desc extensions_table; +----------+--------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +----------+--------------+------+-----+---------+----------------+ | id | int(11) | NO | MUL | NULL |

I am originating a call to a Local channel using an Originate Action: Action: Originate Channel: Local/dial at outdial Context: outdial Exten: answer Priority: 1 Timeout: 45000 ActionID: some_id In my dialplan, I have this: [outdial] exten => dial,1,Dial(${DIAL_STRING}, ${DIAL_TIMEOUT}) exten => dial,n,NoOp(Dial Status = ${DIALSTATUS}) exten =>

I saw a couple of posts about this in the archive, but none seemed specifically to address the problem I am having. If I missed something please let me know. Right now I would classify myself as "novice," and there is probably really nothing so trivial that I couldn't possibly have screwed it up. :-) I'm trying to use the AMD command to detect answering machines, and have

Hello all, We have a script that does a nightly backup of a windows server to a linux fileserver. The linux system uses smbclient to make a tarball of the share. I get this error when running smbclient: Domain=[OURSERVER] OS=[Windows 5.0] Server=[Windows 2000 LAN Manager] tar: dumped 56781 files and directories Total bytes written: 9800680960 write_data: write failure. Error = Connection

Hi Folks, I've been playing with call recording for our support department which was kinda going ok until I spotted something odd in the CDR. None of the support calls are being entered into the CDR properly. I'm using mysql as the back end and Areski's web based front end and all was going fine. The problem seems to be that the CDR doesn't get populated with the destination

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26

I am having a bit of a problem getting AMD to work on a new server. On my regular office server it works like a charm. I am running Asterisk 1.4.13, Zaptel 1.4.5.1 on both machines. Both servers run CentOS 5 and I am using a SIP trunk to send out calls (the same one on both servers). Here is the output of a call on my office server: -- Attempting call on Local/0445540881644 at CC2 for

Hi I have the following formula (I hope it is clear - if no, I can try to do better the next time) h(x, a, b) = integral(0 to pi/2) ( ( integral(D/sin(alpha) to Inf) ( ( f(x, a, b) ) dx ) dalpha ) and I want to do an mle with it. I know how to use mle() and I also know about integrate(). My problem is to give the parameter values a and b to the

I want to be able to take an existing pre-recorded audio file, and dump it into an extensions voicemail box. My application is that someone/thing has dialled an extension which has triggered an event - lets say it's turned a lightbulb on via a System call in the dialplan, and I then want to send the owner a voicemail message (pre-recorded) which says "Lightbulb is ON". Now, I