Freepbx freeswitch jobs


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    3,299 freepbx freeswitch jobs found, pricing in PHP

    I need an experienced freePBX/ Astrisk person to implement the following things into my pbx: When a call comes in, I need it to send the phone number, time, date, the did number, to a url on my site, so I can collect the variables and do with them what I need. I need a agent group, where agents can call into a did and log in and out via their

    ₱10955 (Avg Bid)
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    8 bids

    Configurar Freepbx y gateway Grandstream HT503 como Trunk para salida de llamadas por línea analógica.

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    I want to build my personal SIP server. I have deployed raspbx image and followed the instructions on [url removed, login to view]@ety001/use-the-raspberry-pi-to-build-an-asterisk-service-to-receive-calls All NAT settings handled but I can not connect.

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    ...project if you live in: Eastern Europe The Balkans Russia ***PROJECT SUMMARY*** FreePBX seemed like it might be a good solution for my needs. However, I would be willing to consider using another solution such as Elastix. I tried to install FreePBX on a VPS (Virtual Private Server) at [url removed, login to view] I followed the instructions in this video

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    we have: asterisk 11 on freepbx 2.11 on Centos 6.5 server. all our extensions are recorded by default. (include internal calls and calls from queue) we need to enable the agent (extension) when he is on a call to be able to press on DTMF like (star,2) and pause the recording. when he wishes he can press again (star,2) and continue the recording

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    Hi Anton F., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I just bought a Freepbx appliance and I am looking for someone to install it

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    Want to write CRD records from a FreePBX Queue to PBX Manager in vTige. FreePBX 14 Asterisk 13 vTiger 7.1 Call Status Customer Number Customer User Recording Duration (sec) Start Time

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    Need someone with openSIPS/freeswitch experience the initial task would be to help us setup and configure a high availability deploy of OpenSIPS ->[ Freeswitch1, Freeswitch2, Freeswitch3, … ] the other aspect is I want to setup OpenSIPS to handle … - [url removed, login to view] - [url removed, login to view] Where each subdomain has different ip whitelists

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    hello experts, We have a freash installed freepbx on aws with public IP and one dinstar with private IP but internet connection. so you need to do simple: register dinstar as sip trunk on freepbx with username and password based. keep in mind dinstar is not having public IP and and if you are not expert in this sip trunking, dont waste your and mine

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    I am having trouble setting up my Free-PBX system. I have an ESXI host, PFSense Firewall, and running FreePBX on a Virtual Machine. I have inbound calls working, but when I call outgoing I get "All circuits are busy now"

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    I need help to create a trunk between Freepbx and my other SIP server. There are E1 cards connected to FreePBX and we assign DIDs from Freepbx to end users. They register by ID and Password to receive incoming calls. We don't have any issue there. But now we have another request to forward bulk DIDs to another SIP server and the authentication will

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    We are looking for a Freelancer with good experience in OpenSIP configuration with Freeswitch in a fail-over/ Load balancing environment with session recovery. The Freelancer need to work through Team Viewer and need to provide documentation on Environment/Prerequisites and configuration performed for a successful delivery of project.

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    * Creating campaigns, and after answering, connecting to a queue, or playing a message. * Manage contacts. * Upload audio & contact list. * Reports.

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    Hi android developers!! I am looking for sip android app which can connect with freepbx server. Which should have following features 1. audio / video calls 2. instant messaging. Reference app Linphone Please contact for full description. Thanks.

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    This project should support Asterisk 11 on top of FreePBX 13. This is why I am asking for AMI to be used, not ARI which appeared in Asterisk 12. Keep in mind this is a fairly simple project which should require less than 75 lines of code. Please specify which language you plan to write it in to be considered for bid. I would like to have you make

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    Use a raspberry PI 3 as a PBX 1) PBX system (FreePBX?) with SIP for google voice (& callcentric) 2) 1-3 extension to go to a rj11 jack (preferably 3 if possible) 3) 1 extension to send a low voltage (<5v) signal from the PI, only after entering code (for example when selecting extension 9, it should prompt for a password/code, & if the code is verified

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    Hi Guys, I need a an opensource sip platform which can enable video calling and audio call conference with Mizutech webphone. Webphone works with webrtc so the sip platform must support webrtc. Payment will be released only after successful testing.

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    ...and think voip could be a better solution for us than the traditional landlines we have currently. I have researched the systems a little, and it seems like a cloud hosted FreePBX is the best choice for me. But I am open to other options that you might suggest. I would like to have one system that all three stores, but I am not sure if that would be

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    ...have a GSM Gateway from DBL Tech [url removed, login to view] I had managed to get it working with 3CX but unable to get it working with FreePBX 14 Want to get the incoming and outgoing working. I can provide ssh and web access to the PBX and also access to the GSM gateway from a RDP Server as they are behind a local

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    Looking to create a web/cloud based website that will talk directly to an asterix VoIP install with freepbx, that will allow clients to edit their forwarding numbers, account management etc. Want it to be fully automated so as they can purchase numbers using a third party and then manage their pre-paid minutes through us. Examples would be sendmycall

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