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.
Hi, we have a new Windows VPS and need someone to:
Install VoIP switch latest versions (please specify)
Add all modules, like PBX, reseller, etc.
Install portuguese and spanish languages in main GUI ,
Not use old Flash GUI (we already have it and giving a lot of problems)
Help migrate DB from older freswitch
Initial configurations and suggestions / help.
You make one time install per s...
We have a running installation of Asterisk and we want to develop a prepaid billing(registration) platform for our customers.
Billing & self service CRM Module features:
User/number registration in(from) our existing mysql database connected to Asterisk(for example: we have 1000 free numbers in the database as a registered user gets(on email) first free + pregenerated password from the s...
Need a module that would send an email for an abandoned or timeout call to a specific queue. Would prefer this to have a setting on each queue to be able to turn on or off.. Full call details in email body preferred. (caller id, phonebook entry if present, timestamp, ring duration, status (timeout or abandon).
We need a Voipswitch expert who can configure our 3.1.12 system for Carriers, Clients and Call Flow. This is a max 2-3 hr job and need an Expert only who is Efficient and Smart. We may have frequent and long term requirement later.
Years ago we used Vicidial (hosted) with FreePBX (hosted) and it suited our purposes. We discontinued using Vicidial when we built our own automated CRM dialer (using RingCentral Voip) which was easier to manage and suited our purposes.
Our needs have changed and we are looking at going back to a Vicidial / Asterisk type solution. We will either have an in-house server to host the VOIP solu...
WebRTC and Asterisk.......VOIP Click To Call For Lead Generation Website
I have recently build an insurance lead development website. It is almost functional, but the website developer is unable to implement a click to call feature on my website where prospective buyers can click and be matched up to an agent for a live phone transfer to that agent. He recommended that I use Asterisk. Looking for an affordable expert with experience doing this that can get...
We're running outgoing call-center, Elastix built-in functionality combined with external surveying system (this is also standard).
The problem is a lack of feedback from surveying system back to Elastix.
We need to reach out 50 males and 50 females. Our campaign consists of 200 numbers, 100 males and 100 females.
As soon as 50 males are reached, we want to exclude them from di...
I am specifically looking for an experienced person who can configure our Voipswitch ver 3.1.12. This must be just about 2-3 hrs job. Need this done Urgently. Once again - please apply only if you have experience in Voipswitch configuration and can confident about it.
Application to run a phone-book and connect with Asterisk/FreePBX -- 2
We require an application to run on windows (windows 10, 8 and 7 - 32bit and 64bit) that can connect to our hosted PBX system.
Our system is a asterisk PBX running FreePBX.
We need to be able to dial from our PC's (but still use the handset) but also have a central diary of all our clients so that we can dial them directly. If we need to get a opensource CRM and then connect it, that is...
I need a Web Phone interface plus IP-PBX solution (asterisk/freepbs/resiprocate/...) made for use in a docker infrastructure with docker-compose.
The softphone needs to be embeddable in other pages.
The softphone needs to both make and receive calls using the browser's mic and audio output.
The phone working example can be done using any Web server or front end solution.
Asterisk chan mobile is not able to work with more than 16 devices, even though i revised hcitool dev to show more devices
I would like for someone to revise chan_mobile to show up to 64 devices
I can provide test environment
We would like you debug and fix our sip trunking setup to work with our providers so that we may use Elastix MT 3.0 to the fullest with multiple organizations and users per organization.
We are currently able to have our handsets receive registration, and our trunks are showing registration from our sip provider, however we are not able to make or receive phone calls through the end points (Poly...