Here are some essential steps that you should not overlook when preparing to launch a new business from your home.
Make calls through telegram which will go through a SIP provider and asterisk api. You will have to use IVR and DTMF on the project as well. The necessary skills, langages and packages will be. Sip trunks configuration Python Asterisk config DTMF FREE PBX
Build a program which can make outgoing calls using a SIP trunks provder which relays information back to telegram. You must have knowledge on python - asterisk - freepbx and understand SIP service
Hi there We have set up a Free PBX Server and it has been running without issue for over a year. but I'm aware my knowledge isn't quite there and I need someone familiar with both FreePBX and PFsense to sanity check my config and make alterations for better security and performance where required.
Good routes for VOIP Dialing in UK and US
Hi All, I am looking for VOIP calling solution. If any member have experience or have solution. Kindly contact. If any one have implemented any API with existing solution providers and have code, Kindly contact.
i have FreePbx installed with Synway E1 card / TEJ-101. however it not detected by the Freepbx GUI. i would like to to configure my freebpx so that e1 card is visible on GUI.
We need an account at an australian VoIP provider. He requests an australian mobile number for authorization. We don‘t have one. We ask you to give us your australian mobile number and forward the details you get from the provider so we can activate the account.
Need to Configure a voice Mail Drop Server. I need a technical person who can work in Cisco and experienced building servers. you need to complete the following steps : [đăng nhập để xem URL]
Requiero el desarrollo de una aplicación, que me permita traer los datos de un IVR organizarlos en una estructura especifica para armar una base de datos , luego permita generar gestión sobre dichos registros, guarde el histórico de la gestión y emita un informe de la mejor gestión realizada
I have an android and iOS app and I am looking to integrate a SIP/ VOIP SDK such that users on the app can make voice/video calls to another user on the app. This is a hybrid ios-android app built in react native. As outcome, I would like the following: Testing if the calling functionality works for both android and ios Pricing to give Twilio/Zoom (or another 3rd party SDK) based on the servic...
We are a startup and looking for a VICIDIAL expert who can work with the team to answer their queries and also help them in troubleshooting any issue they face in installation / configuration / setup of vicidial systems for our clients Engagement model - 2-3 hours session on weekends for about a month - 5 hour effort per week to help troubleshoot the issue
As the (verbose) title suggests, I'm using Asterisk 16 with the old chan_sip, apache2 on debian with a certbot (letsencrypt) certificate, the SIPml webphone and everything is working decently. Currently I generate and deploy a certificate for each installation and each installation is a subdomain of the same domain example: [đăng nhập để xem URL] and [đăng nhập để xem URL] For this project...
I want to use an Open Source Software solution to create my own sip server that I can use to make calls from. I don't want to have to use any external service, I want it all on my own server.
I need an expert in Asterisk 11 to modify the Call Center module according to the following problems I want to call Outgoing Campaigns with numbers in order according to the uploaded file. The file will be uploaded with the mobile number and name only without leaving numbers that cannot be called. In Campaigns Script, the Arabic language is not displayed in the Agent Console. When Outgoing or Ing...
We have a VM running Elastix. We will need you to develop a module in Elastix (or modify an existing open source one) to allow import of a CSV to control calls. Upon connection Elastic will play a wav/mp3 message which can be selected from uploaded files (also required) The CSV will state: call start time, call end time, destination number, sim to be used, recording to be used You will need t...
Somos una empresa con mas de 10 años de experiencia en implementación de soluciones de comunicaciones unificadas. Estamos en búsqueda de un experto en Asterisk que nos apoye por 3 meses en la implementación de dos proyectos en Colombia.
I want Beuild Goip / SK Gateway to gateway Intercom System. Ex. Gateway 01 Port 01 Dial Call 5 When Complete it coming Incoming Call Gateway 02 Port 01 (Automated System)
I have setup a FusionPBX which is running on FreeSWITCH and I am using a branded Linphone as a softclient. However the iOS app cannot receive call while running on the background as it is expecting a push notification from the server. I need someone to setup the push notification on the server for iOS and android.