Freepbx Jobs

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Fixed-Price - Intermediate ($$) - Est. Budget: $200 - Posted
I am looking for an engineer to help me configure some items in FreePBX. I need step by step instructions on how to configure and provision each item along with commands/steps to verify that it is functioning properly. ... Here are the items: SRTP media encryption - Full encryption from the phone to the FreePBX server Endpoint Manager firmware upgrade - Help putting on the firmware and provisioning endpoint manager to have the phones upgrade their firmware T.38 Faxing from the endpoint (Cisco SPA112) to the server and then to the carrier.
Skills: FreePBX Asterisk
Fixed-Price - Expert ($$$) - Est. Budget: $50 - Posted
We are looking for someone who can setup a PBX for us. Our current PBX has failed and we've lost the data on it. 1) Install PBX software on a new CentOS 7 server 2) Setup our trunk provider 3) Setup 4-5 extensions on it with DID from our trunk provider 4) Setup the dial-in with the provided MP3s for the menus, etc 5) Setup voicemail for all servers 6) Setup tftp server for configuration of 4 Cisco 7940G phones via SIP Looking forward for offers, please let me know the software you'll be using. It should be a quick job if you know what you're doing Thank you, Mohammed
Skills: FreePBX VOIP Administration VOIP Software
Fixed-Price - Intermediate ($$) - Est. Budget: $200 - Posted
I will pay an Asterisk/FreePBX expert to document and teach me how to show BLF lights between multiple systems interconnected with IAX2 trunks. ... Site "Romeo": Extensions numbered 2xx Site "Sierra": Extensions numbered 3xx Site "Golf": Extensions numbered 4xx For example, I want extension 201 at Romeo to see if extension 404 at Golf is on the phone. FreePBX systems are in place and IAX2 trunks are working. ... The phones in place are Digium D70 provisioned with DPMA. I am comfortable with editing FreePBX custom configuration files if necessary.
Skills: FreePBX Asterisk
Fixed-Price - Expert ($$$) - Est. Budget: $150 - Posted
I need Goautodial configured and integrated with Elastix/freepbx per this tutorial: http://viciwiki.com/index.php/Dial_Through_FreePBX_with_Vicidial The Goautodial/Vicidial machines must be able to use the Elastix/freepbx machine to dial in and out and somehow rotate the outbound trunks, as the single carrier (Sangoma) is complaining about abuse. ... I need Goautodial configured and integrated with Elastix/freepbx per this tutorial: http://viciwiki.com/index.php/Dial_Through_FreePBX_with_Vicidial The Goautodial/Vicidial machines must be able to use the Elastix/freepbx machine to dial in and out and somehow rotate the outbound trunks, as the single carrier (Sangoma) is complaining about abuse.
Skills: FreePBX Elastix vicidial VOIP Administration
Fixed-Price - Expert ($$$) - Est. Budget: $50 - Posted
i want soft phone application that support vpn connectivity like the following http://www.linphone.org/ http://www.portsip.com http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software http://www.c-sharpcorner.com/members/csilla-szabo-kozell http://voip-sip-sdk.com/p_332-voip-sdk-developers-guide-voip.html http://techipost.com/2010/11/12/open-source-pbx-soft-phone-in-c/ use one of above
Skills: FreePBX .NET Framework Asterisk C#
Fixed-Price - Entry Level ($) - Est. Budget: $99 - Posted
Hello for some reason the caller id is not publising when the calls are comming from a press one. Only the extension Que that is routing calls to agent. Current methos call dials out when press one call goes to ext (que) then dispurses the call.
Skills: FreePBX Asterisk vicidial
Fixed-Price - Expert ($$$) - Est. Budget: $5 - Posted
PBX software should preferably be open source, but I am open to see paid solutions Requirements of the job: - setup of virtual machine in the cloud (I would prefer windows azure virtual machine) - select a pbx solution and describe why you prefer it (I know asterisk and freepbx) - install and configure pbx into virtual machine - Configure PBX to connect to Mexican SIP trunk - Handle incoming calls received via SIP trunk - Make outgoing calls routed through SIP trunk - Automatically record all inbound and outbound calls and save file - Deliver document with configuration details and any information necessary to keep the solution working.
Skills: FreePBX Asterisk Call Handling Linux System Administration