VoIP Jobs

Estoy buscando alguien que me ayude a terminar de configurar el Freepbx que tengo con una troncal SIP. El servidor tiene que conectarse a 2 interfaces. La red local y la red de la troncal SIP (tiene configuradas las rutas necesarias). Las llamadas luego de cierto tiempo se caen y hay que reiniciar los servicios.

nesecito a alguien que me cree una cuenta en vitelity, esta es la url de la pagina web [iniciar sesión para ver URL]

nesecito que me recartue vitelity, esta se recarga con paypal, el monto a que me recarguen es de 200 US. esta es la url de la pagina [iniciar sesión para ver URL]

Necesitamos crear un sistema operativo personalizado basado en Android Open Source Project que cuente con Voip para llamadas y contacto con mensajes por datos, y VPN para conexión a datos, sin importar lugares de ubicación con simcard sin planes contratados.

Hello Freelancer I'm in need to Asterisk professional to support for long term on our call center setup. Interested for long term support ping me

Candidates to manage a high volume of inbound and outbound calls within a call center environment

Требуется разработчик с глубоким пониманием SIP, VOIP уведомления для Apple в приложении Flutter. Задача следующая: есть связка ip домофоны - сервер Asterix - мобильное приложение для ios, android. Уперлись в следующую ситуацию: не получается разбудить приложение на ios.

Hello We are looking for someone who been working with [iniciar sesión para ver URL] flex. we need our account updated and we someone to manage the communication hub for us. only bid if you looking at joining a team and per hour on time tracker.

need someone to help me set up my proxy on undetected browser.

I am looking for someone who has experience working with VoIP and a number of different PBX systems. The PBX will most likely be installed on Amazon AWS unless you have a good reason to use a different host. ========================== Wishlist for the PBX is: - SIP based - Free or low cost licensing for the whole thing (PBX and modules - Free is better!) - Ideally on some kind of linux os (M...

I am looking for someone who can configure and provide the PBX.

We are looking for English speaking freelancers to act as our "prospective Client" and test the quality of phone call responses from our customer support / front-desk team. We will provide a checklist of items to observe during the phone call. The testers should carefully observe the items mentioned in the checklist, complete it and send back the report to us every week and on-demand (...

Hello, I need a voip expert to setup asterisk with didwww need it urgently now,

we are using pabx UCM6208 and using jami as a sip client problem is that it JAMI or our sip server only allow ONE registration from a single LAN IP BUT we are using remote desktop - so multiple users have single lan ip and now only one jami client user can be registered at one time hopefully it will be a small issue for a knowledgeable person

We are looking for English speaking freelancers to act as our "prospective Client" and test the quality of phone call responses from our customer support / front-desk team. We will provide a checklist of items to observe during the phone call. The testers should carefully observe the items mentioned in the checklist, complete it and send back the report to us every week and on-demand (...

There is a simple Cisco Packet Tracer Activity. Please contact me for more details. Thanks

I have installed fustionPBX at the cloud server & configured GSM gateway openvox gw1600 (watch atachment). Need to connect gateway to FusionPBX for incoming & outgoing calls , I have 3 sim cards on the gsm gateway

Hi, we have new installation of freepbx and asterisk. We created extensions and used sip app to connect. One phone can call another phone, but both parties cannot hear each other. Error is "Couldn't negotiate stream 0:audio-0:audio:sendrecv"

we already have a running and functioning Asterisk system installed. we just need someone to look at the code and find the reason why some calls are getting dropped (2-3 %). and some other simple issues.

Need someone with better experience in voip billing system. For configuration for international premium rate numbers. To ring customers.

Need to implement Linphone on ios, to be able to receive Incoming Call VOIP Push notification from PHP (Freeswitch --> PHP --> Linphone ios to be precise). _____________________________ I have a very specific need. As my team failed to make this work. My team was able to compile the softphone from Linphone source code. Was able to write push notification script, and able to send and re...

I need a asterisk phone server set up

Need somebody who can configure a Cisco C60 Codec as Standalone device (No Video Communication server or similar) in order to join Webex meetings via SIP call. A free sip registrar like [iniciar sesión para ver URL] should be used, alternatives are appriciated. If possible there should be no reucuring cost in the solution (e.g. Hosted SIP registrar/H323 Gatekeeper like pexip) As deilvery th...

This is a new Asterisk 18 installation from scratch, preferred a Docker/Podman container. For this project, you need to set up a new IAX trunk with one of our partners and set up the dialplan as per the specifications received. Basic calls routing from incoming to less than 10 VoIP SIP phones.

This project contains 4 milestones: Milestone 1 Setup fully working Asterisk 18 as Docker container with ODBC MySQL remote connection: Dockerfile Asterisk config files Milestone 2 Migrate existing Asterisk application (dialplans, extensions, trunks, bash scripts containing Asterisk commands etc) from Asterisk 11.7.0 to Asterisk 18 container from Milestone 1 with MySQL connection to a remote My...

We are building a SMS platform. We are looking for an expert with advanced knowledge of Kannel or Jasmin and SMPP gateway. The platform has capacity of handling 500k sms mesages per day.

Looking for someone to hookup the interconnect from FreePBX to flowroute.com.

Hi there i do e-commerce in GCC countries (Kuwait-Bahrain-Saudi Arabia -Qatar-Oman) i was working with a guy and he gives me a platform and I can call any customer with any number i want let me know if you have an idea about that and how it works, its called number spofing! let me know if you have any solution for me thank you

THE TASK IS TO BUILD AN IVR FOR OUR COMPANY IN TWILIO STUDIO, WE HAVE AN EXISTING FLOW WE WOULD LIKE TO USE AND THE SYSTEM WILL FEED INTO OUR ODOO PLATFORM.

Projeto similar ao mizu voip para fazer chamadas através de um navegador

Setup GSM SMS Modem in my laptop, i have AT Command Tester tools which is port is runing need to setup dialup and another tools need to send SMS. if someone can setup let me know GSM Model : MF825. its urgent please help me to do this.

Setup GSM SMS Modem in my laptop, i have AT Command Tester tools which is port is runing need to setup dialup and another tools need to send SMS. if someone can setup let me know GSM Model : MF825. its urgent please help me to do this.

Asterisk dialmacro for running php script when user pickes up phone. WARNING !!!!, the work needs to be done in a sandbox environment at youre own location. No access to the server where the diaplan needs to go.

Webphone que funcione através de navegadores, ex Chrome, Firefox e etc... Usando qualquer conta voip Similar ao mizu-voip

Kannel sms gateway and playsms with dongle configuration with local