similar to: New features released in ICTBroadcast

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

Why not to use Fail2ban https://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com http://www.ictbroadcast.com Leveraging open source in ICT On Tue, Feb 21, 2017 at 12:28 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: > On Mon, Feb 20, 2017 at 11:36:24AM -0300, Victor

Sorry , I forget it for another monitoring tool monit that we have used in our production systems to restart asterisk in case of asterisk crash or halt. I have attached a monit configuration for your reference. it will work almost in all cases This configuration will check Asterisk for following 1. will check for Asterisk process. 2. will check Asterisk via AMI 3. will check

Interested to know a good wholesale sip providers for 15k concurrent calls regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20231120/75b0e652/attachment.html>

@Olivior I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations. Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is

Hi Olivior, We have recently worked on a WebRTC based agent panel. As based on my experience I think that WebRTC based phones are far better and cheaper then those soft / sip phone. the big plus is that they are easy to customize and developer can use the power of browser and web to build / offer features which are not possible with regular phones. Regarding your concern about BLF or call

Hi, Oliver. Maybe something like this (add this script to your crontab): ------------------------8<-------------------------- #!/bin/bash # # File: asterisk-watchdog.sh # Date: 2015.05.26 # Build: v1.0 # Brief: Secuencia para monitorizar procesos. # # ${PATH}: Variable de entorno con las rutas a los ejecutables. PATH=/bin:/sbin:/usr/bin:/usr/sbin # ${DAEMON}:

On Monday 20 November 2023 at 12:14:11, Tahir Almas Dhesi wrote: > Interested to know good wholesale SIP providers for 15k concurrent calls You might want to specify a bit more detail, such as: - which country are you located in - do you require inbound DDIs (if so, in which region/s)? - which countries' Caller ID/s do you need to present? Antony. -- These clients are often

We tried to use the CURL fn to POST json, but it's sent as form data and there seems no support for changing the Content-Type header. We switched to invoking curl in the shell. All the documentation I could find says there is just one parameter for the url and an optional second for POST body. Is there an undocumented way to set Content-Type? -------------- next part -------------- An HTML

I'm trying to add MySQL CDR recording in Asterisk 1.4.6. I'm following the instructions posted here: http://www.voip-info.org/wiki-Asterisk+cdr+mysql I have MySQL installed and it works fine - starts on stratup, I can create DBs, tables and so on and I can connect through php. rpm -qa indicates: MySQL-server-5.0.22-0 MySQL-devel-5.0.22-0 MySQL-client-5.0.22-0 However I still get XXX

Hello, I'm currently developing a simple asterisk application using SFS (Skype For SIP) which tries to call to an outbound number, play a message and read DMTF digits. My first approach used the Manager to originate calls and then called an agi script to deal with the rest. Anyway, this ended up being not so clear because the call did not start on the Originate extension that it was supposed

On 7/8/2023 5:32 PM, Federico wrote: > > I am using Asterisk 16.30 inside Freepbx, with commercial modules, > purchased from Sangoma and Symphony. After a few hours my memory usage > reaches 900 GB, no kidding, in a box with 1 TB of RAM.  The question > is: how can I determine what is causing the memory leak? Can somebody > send me instructions to find out what module is

Hello List, I'm working on an autodialer project. At the moment I use the Originate application then I "throw" it to an extension where I Dial() the other party and then both legs are bridged. The problem is that the Dial() will only run after the Originate finish its bit and I have lots of wasted time or even worse, the remote party hanging the call because instead of a human

Thanks for sharing the link. Actually I'm looking for a different approach without installing/using third party i.e. a user sends an email to Asterisk (which is also running mail service), as Asterisk receives the mail where the mail contains attachment and subject contains destination number, Asterisk will download the file and capture the number and later send fax to destination number just

I'm looking for an easy way to make asterisk perform as a basic (broadcast)autodialer. Basically all I want to do is give it a list of phone #'s and a pre-recorded message and have it call each one and play the message or leave it on the person's answering machine. The people I'll be calling are all our customers, etc. so I don't need to do any do-not-call checking. Just

I have a * serving 15 sip clients. I use the digium 4 port t1 card. We have an autodialer that calls and reminds clients of there appointment. it uses a pri t1. I would like to plug its t1 output into asterisk to use voip. I am very new to * and am confused. Any help would be appreciated. _________________________________________________________________ Express yourself instantly with

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 =>

> We once developed a reminder system for a customer. He's a cleaning > company, cleaning homes and offices. He was spending two hours a day calling > his customers to remind them of their appointment the next day. Two hours a > day equates to 40 hours a month that he saved with that system. He's been > using it for maybe 6-7 years now and not once was a customer upset

Hello, I want originate a call to some destination, and when B side answes to play a prompt. Asterisk version is 1.6.0.5. But also I need to insert a SIP header to Invite, that's why I'm using Local Channel. This is my extension.ael: context autodialer-local { _X. => { SipAddHeader(P-Asserted-Identity: <sip:${CALLERID(num)}@xxx.xxx.xxx.xxx;user=phone>);

Any ideas on how asterisk could accept an email (such as an email to SMS or "number at mybox.org" sort of thing) and make a phone call to a specific number and make an announcement? I imagine the first part is the big question. joe a.