similar to: MySQL components in asterisk-addons not being built

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

Following new features are now supported by asterisk based telemarketing software Auto subscription / registration after call recipient press a key in voice broadcasting https://www.ictbroadcast.com/Subscription-Campaign-to-automatically-register-customers-at-website-with-Voice-broadcasting-Autodialer There will be restriction to call a number in off time accordingly to timezone of

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>

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

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

Hi, I'm having trouble compiling zaptel 1.4.4 on SUSE 10.1. I'm really only interested in getting ztdummy to work because this is a dev machine with no zaptel h/w. SUSE 10.1 is a 2.6 kernel: asterisk-dev:/home/hugh # uname -r 2.6.16.13-4-default It seems that my problem is related to autoconf.h - I cannot find that file: asterisk-dev:/home/hugh # find / -name 'autoconf.h'

Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not

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

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

Hi, I'm installing * 1.4.9 and a couple questions have come up: 1) I read here<http://www.voip-info.org/wiki/view/Asterisk+installation+for+CentOS+4.x>( http://www.voip-info.org/wiki/view/Asterisk+installation+for+CentOS+4.x) that if you are using E1 cards you need to install LIBPRI. I'm not using any cards on this system, so does that mean I don't need LIBPRI? Asterisk built

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}:

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.

@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

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

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Hi everybody, >From asterisk CLI we can add extensions in dial-plan dynamically using "dialplan add extension" command. but how we can dynamically create a context in dialplan. is that possible? Nasir Iqbal

Hi all, I thought I'd share I script I made (based on some of Leif's works) that lets you download, compile and install Asterisk all in one go; and then removed the dev tools used. We use it quite a bit to provision systems using Ansible, but it is easier than remembering everything every time even if you are using a shell. At the moment I have scripts for Centos 7 and Asterisk 13, but

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

Respected Sir, I am a MS CS scholar of Virtual University of Pakistan, I want to participate in GSoC 2016 for LLVM. Data Science, Networks, Information security, digital forensics and ethical hacking are my core areas of interest. Currently, I am working on a research project on live forensics of GPU and volatile memories like RAMs and Caches. I am looking forward your guidance to start my