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 DevelopersHello, We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows WhatsApp executables, or by using the Android / Windows Phone mobile ver...
Program Needed which can convert .asf Files into Mono Wav File. Please check about .asf files before applying Include mwv in the description .asf Files are Windows file and run on WINDOWS Media Player
want to create an angular module for Linphone based on their SDK. Both iOS and Android. The UI will be customized based on provided Wire frames (12 related screens)
Development of a SIP to Whatsapp gateway required. Either standalone system or module for Asterisk/Freeswitch.
I need my own PBX/Voip system like Vonage , Twillio dot com or Magicjack. This is for the call center. Need developer who can develop that. Must have prior experience doing this. Must have similar portfolio. The main requirement of the system:- 1) Outgoing/incoming call 2) Record the calls, track stats like duration of the call, response etc. 3) Software to access the stats, allow, block num...
To make customized Invoices to bill customers for incoming calls through our PBX and Credit Notes to pay telephone agents for time worked. Min. experience required: 1) Asterisk PBX - CEL 2) AstPP billing software 3) LibreOffice Calc 4) MySQL DB 5) KVM qcow2 N.B.: What is experience ? -- It is when You have already worked with it... You don't have to google around ! For invoices and Credi...
hello, i have 2 ready freepbx servers + php code to fire calls throw asterisk api i need to add trunk on 2 freepbx + run my php code on it
Hello, I tried to get GV number but there is error message appears to me "Something went wrong We had a problem with this. Please try again." I need some help to got GV number.
i need help with my AGI dialplan ,if you complete project will give you more work for same, my budget is 50$ for this project .
I 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
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 ...
hello I would like Trixbox / Asterisk [login to view URL] installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks
Hola necesito un dos para un IVR con una voz femenina los audios tiene es de aproximadamente 1 minuto
