VoIP Jobs

We need to configure a new SIP trunk on our Avaya IP Office Server Edition. We have the SIP trunk from our SIP service provider. We will need: 1. Test user setup 2. LAN details updated 3. SIP line setup 4. Call flow testing and troubleshooting

Connect Sinch VoiP DID to Fusionpbx and Integrate to ensure it meets our requirements.

As a developer, I am looking for a skilled professional capable of providing development services focusing on FreeSwitch configuration. No existing code will be provided, so the individual chosen should be proficient in working with the platform without any starting points. I am counting on the ideal candidate to bring knowledge and skill in the development space to make this project a success. I need candidate to develop WEBRTC Server with a webrtc client to integrate with different CX platform. Post completion of the project, I will need proper documentation in order to install WebRTC server internally for my project.

I need someone who experience in Linux and script development knowledge for Asterisk /freepbx patching. I will integrate existing asterisk PBX to Kommo CRM. The Asterisk/freepbx V18 PBX and kommo are ready, but don't know how to run the script development as below link:

I am in need of an IVR with SFX and E1 cards that have an interactive voice response and use both speech recognition and text-to-speech to process E1 cards will be Ethernet based. I am looking for developers who can create a fully functioning IVR system that provides a user interface with an automated greeting and menu options, as well as personalized instructions for customers. The IVR should include features such as call forwarding, automated call recording, call transfer, and the ability to customize speaking voices for different calls and caller circumstances. an ideal candidate will have a strong knowledge of the technologies associated with an IVR system, including E1 cards, integrated systems, signal and speech recognition and text-to-speech. They should also be familiar with progr...

I am currently experiencing a TFTP timeout error with my Cisco CUCM server. I have access to the CUCM server and do not know if a network firewall is blocking CUCM traffic. I need help discovering the issue and resolving it.

Hi looking for voip engineer with following experience Freeswitch Opensip Attsp

Need enterprise level vicidial customized installation

I am looking for an experienced developer to help me design and implement a full Cloud Pbx system with mobile application (example: 3CX). I have decided to use a multiple-server architecture, and I do require specific hardware and software solutions. I also have a deadline in mind, so this project needs to be completed quickly and efficiently. If you believe you are a fit for the job, and can help complete my project in a timely manner, please contact me. Thank you!

Require voip expert for installing fusionpbx and configuring it with our termination provider ip address.

Soy de Brasil, como vai? Representamos a empresa VOIP, ramais, sip video conferencia e coloquei dentro do CRM Bitrix um Sip Trunk mas ainda falta alguma autorização da Goto (antiga LogMein)......por enquanto nao fará parte do escopo. Preciso Integrar CRM Bitrix via API com meu sistema ominichanel simples, jogando as conversas chat para dentro do CRM. Projeto perfeito seria : VOIP que ja ta no marketplace Bitrix + ominichanel jogando los protocolos, pesquisas de satifacion e auditoria que ja estão desenvolvidos (frame flask phyton) .... acredito que será tudo por API. Projeto mais que perfeito :VOIP + Gravações integrados no CRM,call center e gestão de projetos Bitrix + chat GPT para acompanhar historicos tarefas cronogramas e...

Setting up a Wan ethernet connection and aggregate 2 WANs Setting up lte 4g as a failover alternative Set up rdp with port forwarding Security on users entry Vpn between rdp users Vpn to othersites Setting up a vlan Prioritise Voip traffic with

Tasks: 1.- Define hardware specifications for a VM, linux distro and version and we'll provide corresponding ssh credentials. 2.- Setup GenieACS 3.- Create initial config of GenieACS 4- Create provisioning templates for AVM Fritz!Box (2 models, 7530 and 7590) 5.- Specific to routers: Being able to monitor these parameters from the devices: ping, manage WAN settings (PPPoE dialin or DHCP), manage PPPoE credentials, manage vlans, manage multiple VoIP accounts and VoIP PPPoE login Provisioning data will come from an external RESTful API with JSON response. Query parameter is the provided serialnumber of the specific device. It will return all credentials (PPPoE, VoIP, etc.) at once. A sample will be provided. We need to have the project completed within 2-3 weeks. This project is...

I am looking to set up a Twilio Ringless Voicemail System with VoxDesk for 1-3 users. This system should be able to drop voicemails without ringing the line.. I realize you've probably never used VoxDesk so we're looking for someone who can test it out. It is a no-code platform but you need to be familiar with Twilio. We will also want our service to forward to a different telephone system. We have a $150 budget for this. Thanks

I am looking for a freelancer who can help me build a cloud PBX telephony software that is supported by Twilio or Asterisk, using the SIP protocol. This telephony software will not need to be integrated with an existing management platform.

voice broadcasting, automations call , IVR set up software with help of SIM card IVR device

we need an expert in flutter to develop flutter app that will combine 2 plugins #1 flutter gsm dialer # =sip client the app will rout the flutter gsm dialer mic audio to the flutter sip client speaker and rout the flutter gsm dialer speaker audio to the flutter sip client mic To route the microphone audio in a Flutter GSM dialer with a Flutter SIP client speaker, you can use the Flutter AudioRecorder and AudioPlayer plugins to capture and play audio, respectively. You will also need to use a SIP client library to establish a SIP session and communicate with the SIP server. Here are the general steps you can follow: Add the flutter_sound and flutter_sip plugins to your Flutter project. The flutter_sound plugin provides audio recording and playba...

I need an expert in installing FreePBX system on our server. I have specific requirements and I want these requirements to be fulfilled before setup. If you think you are the right fit, please let me know – I am looking for someone to start this project as soon as possible. Thank you for your time!

Hi, I have goautodial 4 setup on server with ssl/tls everything is working great except for the carrier dial plan that does not make calls out. All i need is assistance with the dialplan to make sure that the calls can be made from the webrtc phone built into goautodial 4. The software is setup already the solutions looks like it works fine. I assume the problem is with the dial plan. I need someone that can go through the system to make sure calls can be made out using the web phone built in.

the project have 2 api First api --- route audio from whatsapp mic to sip client speaker and from whatsapp speaker to sip client mic second api this api will reside in android and will be connected a to remote server that will be able to instruct the api in the android to initiate call in whatsapp and to monitor the call stat like ring dial . call connected and end call and to be able to terminate the call both api will work on android devices with android version 7 and up

we are looking for a exceptional good expert about SIP protocol and SIP states. Your task will be to help us to identify things on SIP to be able to discuss with our developers (with low expertise on SIP). The developers have to implement some features, but do not understand the SIP protocoll well to find the correct paths. Your task will be to help in the discussions about low level SIP featuers like: - how to list all registered SIP devices (softphones, smartphones-apps, desktop-apps, physical phones, ...) - how to identify how many onging calls are running in parallel? - how long is each call onging? - who (device) has taken the call, what time ended the call, ... - and many more Your task will be to consult only, except you are a developer too. If this is the case, you are welcome t...

Hello, We have server on which is installed FreePBX. We want to buy Yeastar TG100 GSM and Cisco SPA502G. We need to configure this 2 devices so when call is made to GSM SIM cart that is in Yeastar the call to be trough VOIP to Cisco