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 DevelopersSet up cloud based Install of elastix v4 in AWS instance Setup Elastix 4 firewall, iptables, fail2ban, etc. Setup trunk and 5 extensions.
We require someone who has experience using / programming in Asterisk to provide a white labeled app for use on Iphone & Android. We require all features as noted here: [login to view URL] We also require call recording (to be stored on server or customers server) as well as Caller ID forwarding.
Need a simple Zello to SIP interface that will allow a phone on an Asterisk PBX to call into a Zello channel. When talking on the phone it will transmit/receive over the Zello channel.
We have a freepbx server, and we need to make calls based on csv files. We may can use our server, or maybe another third part service, if suggested/approved.
Hello I need to setup the Bitrix24 telephony system. Attached to 3 different SIP servers. Thank you
60 to 70 percent work is already done on freepbx but something's still incomplete in my project. First i want to explain what is my requirements. I want a asterisk base caller/callee filter solution on PHP which can filter incoming traffic and create whitelist and black list base on condition of repeated callee/caller and non repeated callee/caller with VoIP call bandwidth optimizer. It wil...
need to configure my vos 3000 softswitch with my goip 32 and make script working for InterCall between ports in goip
I have Isabel installed, i need the following 1. configure 1 number and 3 simultaneous calls? 2. configure power dialer and tell me how I can load my clients' contacts 3. manage predictive calls on any softphone
I have Isabel installed, i need the following 1. configure 1 number and 3 simultaneous calls? 2. configure power dialer and tell me how I can load my clients' contacts 3. manage predictive calls on any softphone
I need to connect an 8 channel DINSTAR gateway to my asterisk server and create extensions to configured and ip iphone.
Looking to develop VoIP call forwarding Portal.
delay the buy signal on asterisk or opensips, when the call is hangup delay between 6 or 7 seconds to receive the bye
Hello, I am using Issabel 4 with asterisk 16. At the moment I can only upload audio file at 8000 hz . I would like to know if it's possible to use .wav files at an higher rates like 16000 hz or 48000 hz in order to increase audio quality. Audio files should be used for all the audio parts: IVR, Announcements etc...
Looking for someone experienced in mainly Mirta PBX or Asterisk by itself is less preferred but might be an option.. you would likely be a voice Engineer and will need to train people in use of and maintenance the PBX.
I need to escalate my ASTPP server, putting an opensips in front to send calls to more freeswitch servers. I need to handle a minumum of 150 CPS with this setup. Maybe 8 freeswitch servers or more.
I'm using freeswitch and want to configure mod_gsmopen so calls can go via sim
Dear all, I have CISCO router act as a FXO gateway for Asterisk server, it was working find upto the point where the incoming calls are not being ended when hangup , it keep active which make the line always busy,
