VoIP Jobs

Hello, the project is based in develope a configuration into a CISCO ASR1002 to customize a SIP/SS7 signiling switch. The SS7 signiling is received in TDM/E1 using a Channelized STM1 Interface connected directly to the Operator, inside the STM1 the first 8 E1's has the SS7 link & CIC's. The system will have to convert to an ITP / Point Code for SS7 signaling as a part of the final s...

I need software application for voice broadcast,survey,multinant with billing.

I'm looking for someone with a high level of English to do some cold outbound email outreach to try and sell advertising space on a website I have. Could involve calls or emails. But they would need to compile lists of potential customers, devise scripts and email patterns and then go out and sell the ad spaces.

I'm looking for a resource to develop a script to Auto dial given number list from a CSV file and play a voice clip. I need to get the list of successful numbers and failed / no answer numbers report also.

This project is about "ISP CRM" With freeradius, Asterisk and Network Monitoring Capability which Support Vendors [Mikrotik, Huawei, Cisco, Juniper and Star OS]. Required Features and Modules # Admin Control Panel (Web + Android ) * User Management * Plan/Tariff Management * Finance Management >Invoice generation and payment processing >Recurring billing en...

i have subscribe for the click2call from issabel pbx and the script working fine and i can receive calls from the website but there is no voice i can receive calls from out of saudia arabia i need to know what is the problem BR

i am looking for voip setup to make call for premium rates number

Searching for the valuable software house who has exp on developing iOS and Android Mobile application on SIP protocols with the help of Admin panel and Must have experience on asterisk or elastix. Deep Project details will be provided over the PM.

looking for a cold caller for B2B calls

I have a couple of Android POS devices (Android 4.4) that will be use for sales. I, however, want to restrict the following android functions before I deploy: Airplane Mode Debugging Features Factory Reset Outgoing Calls Safe Boot Uninstall Apps USB File Transfer

I would like to get the SIP trunk configuration for goTo and Nextiva sip provider

We have installed Freepbx , Now we need setup , complete configuration We are using Dinstar GSM gateway , will be using this with freepbx Experience in GSM gateway configuration will be preferred

Configuration and support for PBX Siemens HIPATH 4000 - for migration and interconnection with PBX Asterisk

I have installed OpenSIPS 3.1 server with UAC, CP and rtpproxy on Debian 9 server with 2 network interface. (Asterisk server - OpenSIPS - Service Provider) Now I am able to establish calls and also registering to my service provider. I need someone's help to fix the rtp traffic and routing.

A list of apartments and their owners on a screen. By pressing a phone button, you will call that person. If the person wants to open the door, he presses *9 or something like that. All on a web page. There are JavaScript sip clients. The phone part is by asterisk installed on the same raspberry pie using google voice. [login to view URL] [login to view URL]

Hi, I need to build a sytem that can detect IVR when you call, record a part of the ivr and detect which messages is, with previus record files into the server. The target is to to avoid IVR balance, IVR of blocked sims and detect which record has been detected. We need the build in Python, asterisk.

Need to Upgrade Freepbx from 13 to current and install: SIP Poe Mullion Video Keypad Intercom 8039 8063 SIP Interface Module (IP Relay controller) for Algo door control only. 8063

Welcome messages to be uploaded to a PBX for a client Requirements are as listed. Format: WAV Channel: Mono Bit rate: 8 kHz Sampling: 16 bit If you can not provide them in this format, I can not hire you. The messages will be as follows, actual business name and actual business hours will be provided once I hear some of your samples. Message 1: "Thank you for calling [business name], some...

Dear Freelancer i am not expert in 3cx, looking for someone sove the issue of inbound calls, the Outbound calls are working fine not issues, but when someone calling from out its simple disconnet the call and not reach to 3CX

I have a app that basically a calling app and I am using a asterisk server which is setup at my system. I face some issue during integration of calling api to in mobile app. I need some one who have experience with asterisk server and please also share your past apps you have done . If you did this i have lots of other work in queue for you. :) Thanks

looking for developer who have experience in push to talk over internet platform including application (ios-Android) using GPRS, 3G, 4G and WiFi network infrastructure, i want to build solution provides core functions of traditional two way radio: instant voice communications, integrating with advanced features like multimedia messaging, video and photo uploading, live video streaming, integ...

I need a Goautodial V4 or V3 based dialer with an ivr. with hosting

We have a social media app using open source linphone SDK for android and ios SDK for voice. Need voip developer to enable video in existing system and autoscalable solution. Using asterisk on backend rabit MQ

Se requiere el desarrollo de una aplicación web que interactúe con un conmutador IP dentro de la red de datos (Asterisk o Freepbx) para realizar la marcación de forma masiva, por medio de un archivo txt o con los teléfonos a marcar e incluir la opción de reproducir un audio editable. es necesario poder especificar la hora de inicio de marcación y fin, co...

Necesito alguien experto en Freepbx para configurar una centralita que ya tenemos montada pero que está con diversos problemas.

Необходим человек который оперативно сможет рассказать как и что настраивать в VOS3000. В первую очередь необходимо установить SIP Header Diversion для поставщика услуг