Asterisk Freelance Jobs

 

Asterisk AMI Expert 

Fixed - Est. Budget - $100.00, Expiry - May 7, 2020, Proposals(0) - posted at one week ago
We need a Asterisk expert to assist our developer with nami / ami for node js. We're building another console and need actions on how to Answer / Hold / Blind Transfer / Attended Transfer (without parking) / Stop Transfer / DTMF Dialing. This project will last less than 1 week with you helping dev with correct code for each and...read more

We need to develop a SIP to Whatsapp gateway. 

Hourly - Est. Budget - $24.00, Expiry - May 5, 2020, Proposals(0) - posted at one week ago
We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass recorded audio as voice calls 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 w...read more

Looking For SBO Solution Voip 

Fixed - Est. Budget - $300.00, Expiry - Apr 10, 2020, Proposals(11) - posted at one month ago
Hi, i am looking for SBO solution for voip, which can be build with Asterisk (PBX).I will share some refernce link with shortlisted candidatesPlease let me know if you can 100% sure you can do this job and you did that before.Best Regards,

Vos3000, FREESWITCH or ASTPP Expert 

Fixed - Est. Budget - $100.00, Expiry - Mar 20, 2020, Proposals(0) - posted at one month ago
Looking for VOIP Experts who can add multiple voip providers, Clients, setup gateway, setup routing.. setup extension.We are looking for someone who has extensive knowledge in SIP Trunking and PBX softwares.Please bid only if you know much about this.Thanks.

Open vpn module gui for freepbx 14 

Fixed - Est. Budget - $150.00, Expiry - Feb 29, 2020, Proposals(8) - posted at 2 months ago
We want customized Open vpn module client to install in FreePBX 14 to upload an Open VPN certifcates like yeastar s series.This our conditions for this project:1. The module package should be installed in any Freepbx 14 machine.2. Professional document on how to install the module and use it step by step must be provided.

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Asterisk programmer for new service 

Hourly - Est. Budget - $18.00, Expiry - Feb 27, 2020, Proposals(0) - posted at 2 months ago
I need someone who is very experienced on asterisk and linux who has extensive experience in setting up a VOIP network.

Communications / Networking Engineer needed to implement an online PBX protocol 

Hourly - Est. Budget - $24.00, Expiry - Feb 25, 2020, Proposals(0) - posted at 2 months ago
- Set up desktop dialer system with IP PBX- Develop a system using off the shelf IP PBX packages to enable a user to select name of another user on a website and talk to them over conventional phone network, i.e. desk phone or mobile phone (and not from their computers). System would have preferred phone numbers saved for both u...read more

OpenSips Load balancing for Asterisk 4 Instances 

Fixed - Est. Budget - $20.00, Expiry - Feb 18, 2020, Proposals(0) - posted at 2 months ago
I need someone who can enable/configure incoming calls load balancing on OpenSips among 4 Asterisk instances, I will share my current opensip conf with you soon.

Fix Avaya 9608 ip phone stuck in discover screen and with pbx in a flash as server 

Fixed - Est. Budget - $100.00, Expiry - Dec 2, 2019, Proposals(0) - posted at 5 months ago
Need someone to troubleshoot and fix an issue with two Avaya phones 9608 that get stuck on discover screen. I am usin pbx in a flash as the pbx and all my settings are local. I don't have the IP Office manager software since i bought those phones on amazon

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VTiger with Asterisk Integration 

Fixed - Est. Budget - $50.00, Expiry - Oct 8, 2019, Proposals(1) - posted at 7 months ago
We need someone to configure the vTiger-Asterisk API to integrate both the systems. We are using vTiger v7.1 and Freepbx.Under organizations/contacts/leads of vtiger, the call history and call recordings should appear. For dial out, once the number is clicked, the hard-phone (cisco) should ring of the user and call should be con...read more

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