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We are looking for an Avaya Certified Implementation Specialist (ACIS) or Avaya Certified Support Specialist (ACSS) to configure & upgrade an Avaya Session Border Controller Enterprise (ASBCE) for Avaya Equinox OTT Service Provider solution 9.1.11. Inc. SSL certificate installation, reverse proxy configuration and call flow rules for WebRTC client, SIP audio trunks and SIP Video endpoints. The ASBCE is deployed on version 7.2.2 and will need to be upgraded to latest version 8.1.2. Remote access is available via VPN.

Set up Huawei Smartax ma5620 pri. Requires manual documentation and diagrams. Using asterisk. Need instructions on the cable connection.

We are planning to build a Automated IVR system, we would like to integrate the below systems 1- Implement Media Server using UniMRCP 2- Integrate / Enable ASR (Google Speech ) with UniMRCP 3- Integrate UniMRCP with our Conversational AI Platform 4- Integrate Asterisk / UCCE(Unified Cisco Communication Enterprise) for a end to end voice journey

I need to build a proof of concept python multithreaded application that will connect to one or multiple Asterisk servers using AMI protocol (preferably using a library) and trigger a callback function when receiving events. At the same time, one or multiple clients can connect to this python application using a predefined port (like 5039) and using the AMI protocol receive the events from the asterisk servers and send commands to one or multiple clients. Each connection to the Asterisk…

VOIP calling app which can make calls like app to app & app to mobile phones/landlines

Hi guys, we run fusionPBX on AWS with an end user via grandstream ht801 ATA. She is US based and complaining about excessive Post Dial Delay (PDD) - meaning she dials the number she wants to connect to...then hears silence for 4, 5, 7, sometimes 10 seconds, then hears the ringback tone begin. We use BulkVS as our main provider for outgoing, however also use Alcazar and can switch to others. I have a VoIP phone on my desk connected…

Hello! I have installed a Cloud FreePBX server [Fully configured] SIP with TLS But the user's location is in a country that has SIP Traffic restrictions (Blockage) Everything is working and the calls go through 2 SIP Extensions but there is something wrong with the voice! The voice is like shaking! Note 1: If I use a VPN to avoid the SIP traffic restrictions, the voice is so clear and there is no such problem in the call I think…

I want the smscaster keygen, does someone has it??

PROJETO LIBERAÇÃO DE EXAMES. Desenvolver um programa que seja capaz de entrar no sistema web, abrir um cadastro de um cliente e verificar o status do exame do cliente. O programa deve ter uma interface web simples com login de acesso, com as opções de cadastro de código de clientes e cadastro de código de exames e opção de agendamento da liberação com os campos dia data e hora mais as opções de editar e excluir o cadastro. Essa parte…

I need to have e GoIP GSM Gateway configured to work with SIP so that I can call through a SIP softapp routing the calls through the SIM card in the GSM gateway. This is for personal use, not a business so not sure if a PBX is required.

One mobile app to record voice in wav format and send it to remote server. Shall be able to configure voice sample rate 8khz, mono or stero and ruration of sample in seconds starting from 1 second. Shall be able to configure IP , port, loging, password of remote SFTP server.

Hello I am looking for someone who knows the programming of astpp freeswitch scripts. There is PHP script , lua script and XML script. Looking for someone who has already worked before with those files. I am not looking for web interface programmer. Looking for the freeswitch part programming

Dear all, I am seeking someone who is an expert in FreePBX and CISCO routers as its used as FXO I have managed to configure both and all is good except for the follow me , keep getting issues

Dear all, I am seeking someone who is an expert in FreePBX and CISCO routers as its used as FXO I have managed to configure both and all is good except for the follow me , keep getting issues

I want to setup astpp and direct calls from freepbx to astpp