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.Asterisk PBX Developers anheuern
Looking for a developer having experience in connecting VOIP server to an Android Application. Need a Connection to VOIP server into my Android Application for Audio Calling. I already have a VOIP server ready. We use XMPP Server prosody for chatting Only Freelancer.
push and pull of data between zabbix server and zabbix agent/snmp agent communication between zabbix server and agent using tls certificates monitoring requests and queries create charts to see perfomance, monitor behaviour of resource usage and make deductions from this
Experience Level: Expert General information for the website: We need to develop a SIP to Viber gateway. Kind of development: New website from scratch Description of every page/module: We need to develop a SIP to Viber gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber to complete the call to the called party number. The development platform/op...
We need to install WebRTC to be available to users logged into fusionpbx. Also, need to setup call park with ability to add and retrieve multiple calls. We need full documentation on how this can be replicated in the future. A step by step video of the solution being implemented will be preferred.
You will have my 3CX server assigning a predetermined conversation lenght to a predetermined telephone number after which the call must automatically be cut off. If the phone call ends before the maximum predetermined time amount, the telephone number which the time of conversation was assigned to must be able to call again for the lenght left from previous calls. Example : The number, I want to ...
I am seeking someone that can setup my IVR broadcasting campaign, so that customer will have the option to ignore the message, report the message and Respond to the message if he is looking for a service. Kindly share what all you required to get started?
hello i need to make direct inward system it's calling system for receiving calls for more detail you can google it i need receive calls on one number i will login this number on 10 system it has to come call on all system for e.g I need USA virtual number with 10 channel DID for Inbound service I just wanted 1 virtual number of USA 10 channel means ... If any call coming on that virtua...
We require an IVR phone survey module that can be used to evaluate the quality of customer support calls. The list of customers (phone numbers) will be uploaded from a csv file. The customer will receive call after customer service and will be asked questions and asked to press the appropriate number key to indicate their respons * Must provide dialplan and AGI for asterisk. * Will be 5-6 ques...
Want to integrate soft phone in my application made in php Laravel framework. i want a dial pad with the back end connectivity with sip/softphone to make calls from my application itself. can anyone help me with that? I already have installed GSM gateway, FreePBX and my PHP CRM (with softphone front-end) on a single server
Looking for a developer having experience in connecting VOIP server to an Android Application. Need a Connection to VOIP server into my Android Application for Audio Calling. I already have a VOIP server ready. Only Freelancer.
I am looking for someone who can deploy a FlexiSIP server to send iOS push notifications to an app we have. It will sit beside an SBC which will route the traffic accordingly to the desired FreePBX server for authentication etc.
hi i want a freelancer which r expert in voip project sip trunk and freepbx so bid only those guys which r exepert and work for long term another project discuss on pm and hire only those guy which r expert in this project new freelancer r also welcome
Very simple task, we are a business that handles simple phone calls for our clients. This is mostly outgoing calls and requires mostly a simple script and American accent. We have many clients and the work is ongoing.
I am seeking best Voip and Did number providing company that can give me billing 1/1, you need to give me its carrier settings, Ip address, username and passwords, Prefix etc. The cheaper the minutes cost and DID cost the more chances of getting your bid approved, my channels will not be more than 30.
I want to limit the extensions to only view their billing report in Issabel(former Elastix) Like the way CDR report limits the users to only view their own CDR report.
Server side OS debain 9 & also Should support Cent os 7.X install asterisk install openvpn install Monast Ass Sip clien to send traffic , that traffic should push to client side using prefix client Side OS debain 9 & also Should support Cent os 7.X install asterisk install OPENVPN Install Chen_Mobile Connect Server With iax to optimise bandwith Register gateway
Hi I need an application that can receive a call through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. The app should be able to run on cheap devices (+-100$). I was thinking on a way to emulate a headset and set it as default, so just need to initiate a call on gsm networ...
Company is planning to setup a FREEPBX and Asterisk support center engineer in Malaysia and Singapore. 2. VOIP Technical Support Engineer Job Description (According to what I have posted) About the Network Engineer Role: To provide assistance and support for ongoing and new projects in our Customer Support, Help Desk, Remote Support, Provisioning Services, and Fulfillment efforts. Key Respon...
Fix of no audio/video issue on Freepbx and write simple insert query to create and test extensions programmatically using mysql query on the asterisk db without having to use the FREEPBX UI AT ALL. I am using asterisk 15/Freepbx 14.
We are an ITSP business engaging in viop termination. What I want to achieve is to compress voip packet going through our LAN which should largely result in a better performance of our internet bandwidth.
Fix no audio issue on freepbx, fix chat, extension to extension calls and video(extension to extension). This is just basically only updating the settings on freepbx and testing