Asterisk PBX Jobs
Asterisk PBX, like any other PBX, is a complicated subject that is best handled by experts. If you are a pro in this field, then you should bid on the many jobs at Freelancer.com.
Asterisk PBX (private branch exchange) is implementation software. Created by Mark Spencer in 1999, the software simply allows connected telephones to make calls to each other and also to connect to other services. The name is based on the symbol asterisk, (*). For Asterisk PBX to function as it should, the configurations must be on point, which is why this should be done by an expert.
Asterisk PBX is a topic that needs skill and if you are an expert in this, then you should earn money through what you know best. There are thousands of jobs posted on Freelancer.com related to Asterisk PBX and if you at a pro in this particular field, then Freelancer.com will offer you a chance to work on projects you understand. The site attracts some of the best-paying clients and offers an easy-to-use platform, where freelancers can browse and bid on jobs they are interested in. You can simply start your career in Asterisk PBX at Freelancer.com today.
Hire Asterisk PBX DevelopersI 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.
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
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 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 [login to view 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 there is either a c...
I am looking for someone who has enough Experience with Zoom APIs. I want to export all audio recordings of my tutors from the Zoom servers to my external dedicated server hosted in the UK. This is due to the very limited space available in the Zoom servers. 1: This action will be performed every few hours (4-6 times, every day). 2: I need to know what parameters, options we have to export aud...
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.
I desperately need someone to help me make a voice module with the mass effect sovereign voice programmed into it l, I do cosplay and roleplay and intend on installing the module to a helmet but I know little to nothing about programming a module I only know how to soldier so please help I'd be willing to pay for it as well
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.
Kannel sms gateway and playsms with dongle configuration with local
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 [login to view 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 [login to view 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.
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