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 want to use the VoIP function in the Arris TG862S and I need a *.cfg config file. Modem Arris TG862S firmware [login to view URL] Without VoIP I use the "[login to view URL]" config file and it works fine. I tried to find a solution myself and I think the code from "default_voip with error" must be used. I tried to adapt, but I always get an error when compiling. So I ne...
I have a bigbluebutton server with greenlight with 0 users in room, freeswitch starts using 20-40% cpu and X minutes after raises to 100% for some minutes and then go back to 20-40% with no reason at 0 users. This is a weird error/bug due to I have other 30 servers with the same configurations and all of them are okay but sometimes a server starts with that same error, so THIS HAPPENS IN ANY SER...
I have a bigbluebutton server with greenlight with 0 users in room, freeswitch starts using 20-40% cpu and X minutes after raises to 100% for some minutes and then go back to 20-40% with no reason at 0 users. This is a weird error/bug due to I have other 30 servers with the same configurations and all of them are okay but sometimes a server starts with that same error, so THIS HAPPENS IN ANY SER...
I have a asterisk phone system setup working using Magnus Billing. I am seeking a reliable person that can update it and make changes as I need them.
To make customized Invoices to bill customers for incoming calls through our PBX and Credit Notes to pay telephone agents for time worked. We make and supply the background images for the LibreOffice Calc Spreadsheets. We use a small spreadsheet with background image for the "center part" of an invoice-template and an interchangeable "top" and "bottom" part on the sam...
s it possible to forced change RTP IP and Signalling IP by using asterisk and / or other application ? example ; 1. Source RTP and Signalling IP is [login to view URL] and trigger a call to asterisk 2. Asterisk forced change to [login to view URL] and send the calls for termination. please let me kow Thanks
Development of a SIP Whatsapp gateway required. It can be standalone system or module to Asterisk or Freeswitch.
need help creating a docsis 3.0 file for SBG6580. Also, a config file for TG852. The modem is a eMTA, we need to setup the modem and guidance on how to setup our existing VoIP gateway system to it.
I need you to check my zabbix installation and make me suggestions what to change. I need you also to create a voice alerting - you can suggest me a provider what ever. and 3rd is to help me with my grafana.
Hello , I want someone to automate the process of generating 100% valid ANI and set to send dialed number matching the area code just FYI its a Press 1 camp of one our client
Need to make Cisco504G, Linksys 942, Fanvil X7 Series and Grandstream phones compatible with FusionPBX. Set Up Provisioning to manage… Sip Registration Local Date & Time as per Client’s Location LCD Screen & Screensaver (On/Off/Timer) Idle Key List Missed Call Key List Off Hook Key List Dialing Input Key Progressing Key Connected Key Start Xfer Key Start Conf Key Conferencing...
Asterisk server docs We need below points on Asterisk server. 1. Call will be generating from our web app by a user from his browser so we need the correct codecs to be setup on our asterisk server. 2. Once lead picks up the call then we check if it is a human or an answering machine (AMD system) If it is a human then the call will be connected to browser and if it is a machine then we simply hu...
Looking for twilio expert who knows how to create sip end points remotely. or knows how to work with twilio programmable voice api's
We have opensips running on our cento7 server, we need someone who can install cgrates & cgadmin and integrate it with opensips for real time billing. your bidding price will be deemed as final price.
I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX [login to view URL] that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: [login to view URL] [login to view URL] [login to view URL] I may probably also need help with G729 codec configuration
I need someone who can create custom reports as per our requirements for vicidial and send via email. For example some of the reports would be daily reports Some of the reports weekly reports monthly reports and yearly reports. Attached sample in excel. But need it neatly designed in html and sent as a email on a condition. THIS IS NOT AN EXCELL JOB ONLY WHO WORKED ON VICIDIAL PREVIOUSLY CAN ...
i need some one to help me set up dial pattern for my voip , i have to outbound routes one for international, and one for local just need someone who understand how to set it up correclty I am using vitalpbx (asterisk)
I need Odoo developer familiar with Asterisk distribution like Issabel , freepbx ...etc .
Hi. Looking for a professional experienced in setting up remote call centers. Needed: - autodialer - ability to transfer clients while staying on the line with them and using client's caller ID - connecting hundreds of low cost local US numbers and use them to show local caller id when dialing US leads Please reply with your experience, what kind of setup are you familiar with, etc.
Hey I have a 17 user environment that I need help setting up. My preference is 3cx and I will be hosting it in google cloud. We have 17 users spread over 2 sites currently, however, we have 32 sip numbers. Each employee has a direct line, then there are 2 receptionist phones (as well a 1 wireless phone for when the receptionist has to leave her desk) and the rest are on standby for expansio...
Hi all, I am looking for a system admin to install A2Billing on my CentOS 7 server. I have tried installing it using the following guide but it did not work: [login to view URL] I am receiving the following error when I make a call from a DID number: pbx.c:4458 __ast_pbx_run: Channel 'SIP/DEFAULT-TRUNK' sent to invalid extension but no invalid handler: context,exten,priority=a2billing-...
Freepbx 15 installation on DEBIAN or CENTOS OS. I tried many times but it never worked (calls drop after 32 seconds, or I hear nothing) I even would like to use API REST or other module to call url when incoming call to specific number, then this url respond with number that asterisk needs to callback.
need to make a tunnel app for android which work with our voipswitch
hello, we are looking to implement AMD on asterisk using give below case 1- We will send call from freeswitch to asterisk 2- Asterisk will send call to route. 3- once call is answered by called party 4- asterisk will run AMD 5- if AMD is detected the asterisk will hangup 6- if live person is detected then continue the call Thanks
Hello. I have a flowchart done for twilio sms chatbot. The flow chart is very detailed so you can estimate the price for this gig. I ha e a twilio account and phone number already. Are you good with twilio sms chatbot and interested?
Hi developers, We are looking for people who have experiences in the integrating Zendesk with SIP Softphone. We are in the midst of development and would like a hand to help speed up the development. Please refer to the requirement doc attached.
Cisco webex/Zoom is used for conducting webinars. Attendee can register for the webinar and gets the confirmation mail. What we want is: If the event is paid, attendee registers for the webinar and make the payment(This is required) and gets the confirmation mail. We want this payment gateway to be integrated in the webinar Registration. Please note : Its not just integrating payment gateway. The...
Hello We have an urgent need of a person who can handle Issues / Queries and enhancements in Vicidialer and also able to handle Connectivity with SIP/EPBX/GSM Gateway Connectivity with VICIDialer
Hola necesito un dos para un IVR con una voz femenina los audios tiene es de aproximadamente 1 minuto
Hola necesito un dos para un IVR con una voz femenina los audios tiene es de aproximadamente 1 minuto
Necesito un experto en VICIdial, para activar un servicio de Dialer desde zero para una pequeña empresa de Telemarketing. Que posterior a la configuración inicial pueda y seguir dando mantenimiento y actualizaciones al sistema.
Buenas, Necesito alguien que me ayude con el proyecto de montar una centralia con 3CX y TWILIO. Gracias.
