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 Developers
Configuration and support for PBX Siemens HIPATH 4000 - for migration and interconnection with PBX Asterisk
Hi, I need to build a sytem that can detect IVR when you call, record a part of the ivr and detect which messages is, with previus record files into the server. The target is to to avoid IVR balance, IVR of blocked sims and detect which record has been detected. We need the build in Python, asterisk.
Hello all, I have Freepbx installed on a VM with a public IP address. I am testing with 2 softphones which can register with PBX and call each other. But there is no audio at all. I would like your assistance to debug and fix the audio issue. I will happy to discuss my setup with shortlisted candidates.
Dear Freelancer i am not expert in 3cx, looking for someone sove the issue of inbound calls, the Outbound calls are working fine not issues, but when someone calling from out its simple disconnet the call and not reach to 3CX
I have a app that basically a calling app and I am using a asterisk server which is setup at my system. I face some issue during integration of calling api to in mobile app. I need some one who have experience with asterisk server and please also share your past apps you have done . If you did this i have lots of other work in queue for you. :) Thanks
We have a social media app using open source linphone SDK for android and ios SDK for voice. Need voip developer to enable video in existing system and autoscalable solution. Using asterisk on backend rabit MQ
i want to AUTO BLOCK DNC number when agent put call disposition to DNC on dailer it will Block that number Automatically in camp NOT in DID or in Group i have used below setting still not did not work Use Internal DNC List: Y Use Campaign DNC List: Y Other Campaign DNC: i am useing VERSION: 2.14-783a BUILD: 210103-0856
We need to modify our Asterix phone system to recognize codes digit by our customers and to provide information using data from the DB. In the same time we need that call center operators can know from wich lines the customer is calling. We will provide a copy of the system to work on. In this way nothing can affect the production one. In the same time we will provide a remote desktop connection ...
I would like a multi-class SIP tutorial, at first with the most basic concepts and increasing the level with each class, begining with a basic level and finishing with more advanced concepts, to learn and understand completely how working SIP networks. The minim total duration of the videocourse with all of classes is 30 minutes.
Hello I have a centos8 running kamailio asterisk and a2b, the service has chat with file transfers and of course voice and video calls, I have built it with several developers and I have also android custom built clients and apis, Im looking for someone to start installing the required above explained prefferably on debian 10 but 9 would do its a cooperative work between you and me to copy my curr...
En la eth0 esta configurada la LAN con salida a Internet para Datos, necesito configurar eth1 para salir a internet para Voz IP