VoIP Jobs

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

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

Situation: We have create a simple app based on Python PJSUA library ([conectează-te pentru a consulta linkul URL]) for a Voice bot solution. Simple diagram of a application is in attachment. Application registers via SIP to a telephony (Twilio or a Asterisk) and accepts incoming call from soft-phone or connected phone-number via telephony system. Then it streams audio via web-socket to another a...

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...

Hello all, I have Freepbx installed on a VM with a public IP address. I am testing with 2 softphones which can register with PBX and call each other. But there is no audio at all. I would like your assistance to debug and fix the audio issue. I will happy to discuss my setup with shortlisted candidates.

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

Given the important restrictions currently in place almost everywhere, getting to one of our showrooms isn’t possible right now. To help you get started on the shopping journey, we need to offer video appointments with us. So I need a banner on the starting page, calendar and time option, and details confirmation page. Access of Zoom with my Zoom business details. Who can do for me? Attac...

En la eth0 esta configurada la LAN con salida a Internet para Datos, necesito configurar eth1 para salir a internet para Voz IP

I want to build linphone desktop app on my PC. Windows or CentOS, I prefer windows No need to change anything, just build

i want to AUTO BLOCK DNC number when agent put call disposition to DNC on dailer it will Block that number Automatically in camp NOT in DID or in Group i have used below setting still not did not work Use Internal DNC List: Y Use Campaign DNC List: Y Other Campaign DNC: i am useing VERSION: 2.14-783a BUILD: 210103-0856

Freight Broker, Freight Agent, Sales calls, leads, customer service, cold calling, warm calling, Freight Agent training, freight Broker Training.

HI, I have a IP phone yealink - T21P-E2. I need to know its SIP credentials, mainly the PW. I have local access to the device and can give teamviewer or anydesk to retrive the info.

setup failover for free pbx the task must be completed in 48 hours

HI, I have a Yealink T21P-E2 Ip Phone with local access. I need to recover its settings via config file or any recovery tool

I want to make a new Network with cisco switches, a firewall and IP Phone. Also a Few Windows Server...

I would like a multi-class SIP tutorial, at first with the most basic concepts and increasing the level with each class, begining with a basic level and finishing with more advanced concepts, to learn and understand completely how working SIP networks. The minim total duration of the videocourse with all of classes is 30 minutes.

Current VOIP stack Setup: 2 opensips servers (clustered), 3 rtpengine servers (clustered), 2 database servers (clustered), and would like to host multiple PBXs using opensips as a with different domains. Would like to verify current setup and also get webrtc functioning correctly. I am looking for an expert in OPENSIPS and working with webrtc experience. Thank you, Convergence

I have to connect an 3CX pbx to an T-System SIP-pure Trunk (sip-trunk based with 3-way authentification). I want to use kamailio as sbc - but i have no experience with kamailio sbc-config. I need someone to write an [conectează-te pentru a consulta linkul URL] to archive the following: - register to T-System Trunk (3-way auth) - connect to 3CX (sip-trunk) - rtp engine to convert g711 - t.38 an...

Hello I have a centos8 running kamailio asterisk and a2b, the service has chat with file transfers and of course voice and video calls, I have built it with several developers and I have also android custom built clients and apis, Im looking for someone to start installing the required above explained prefferably on debian 10 but 9 would do its a cooperative work between you and me to copy my curr...

Simulate SIP vulnerabilities that include DDoS, DoS, password guessing, Man in the Middle attack, and SQL injection using Omnet++ in VoIP architecture. Take a video of each step of the process. The video should have no voice but should be clear in terms of pictures. Also, include how to mitigate each attack mentioned above in the simulation and capture it in the video.

Desenvolver uma extensão que ao ser acionada ligue para uma número e faça a conferência com o canal que acionou a extensão.

We are in need of an updated looking soft phone for our call centers. This developer will be using [conectează-te pentru a consulta linkul URL] as the library. Video is not required for this. But implementing SIP Messaging is. We may also require you to call a REST service to autocomplete company directories and such. We are open to feedback for that part of it.

Estou precisando de um desenvolvedor para o Desenvolvimento de uma URA em Asterisk com integração com CRM através de Apis (rest / soap) e agi, ura deverá se integrar a ferramenta de vocalização tts, o Asterisk é puro e já está instalado e em funcionamento, sendo necessário apenas a construção da URA.

i need the numbers in twilio to be sip accounts and to create an ivr for incoming calls.

We have Mautic installed on our server, but we need to configure it so that it sends via Amazon SES and also Twilio. This should be an easy project for someone who has done this before. Please apply with "I can config Mautic for you!"

Hello, I have installed an Openvox UC 501 Asterisk Based IPPBX at my home. I have a BSNL FTTH Connection with a Landline Number. The landline is currently working through a POTS port on my Syrotech ONT Modem. I want the Landline (VoIP) to work as an SIP trunk on my IPPBX. I live in Raipur, Chhattisgarh, India. Thanks.

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I need a 3cx expert to help me debug the configuration with a patton smartnode 4171, the switchboard in outgoing calls from error 503

I need a 3cx expert to help me debug the configuration with a patton smartnode 4171, the switchboard in outgoing calls from error 503