VoIP Jobs

Coral Telecom is a designer manufacturer of PABX systems and we use Freeswitch as our core. Require support in running / activation of 256 AES encrypted VoIP calls. This is just an implementation support for setting up SSL certificates and activation of TLS / Zrtp on Frew switch. Deadline: 2 days.

I have a FreePBX server that I want to be able to log and see live in a nice GUI" - How many operators are on the line - How Many are avail - How many calls in the que - Status for each extension - Ect. - must have nice GUI - - must show live info

We are looking for an asterisk expert, can help with a minor thing via anydesk. We must have transferred some data to our database with each call

thanks for reading, i am after a specific type of script that will allow me to send outbound calls from a web view , a simple click to call will suffice the foundation is asterisk / php

We are a Networking company in the UK with a number of large projects and looking for an advanced remote Cisco resource to assist with Design, Config and potentially assisting engineers onsite with commissioning on an ad hoc basis but planned and scheduled. Skills required - 802.1x, QOS, Multicasting, ASA, WLC, Mobility Express, Routing/Switching

hi and thank you for reading . is there anyone who can write a php code that can send out calls from the web , want the code to monitor status of call . i have a dial plan and extension already that would connect to it

i am looking for a simple set up for the mikrotik model LtAP mini 4G kit on public buses we want to provide free internet for users on the bus, i am not sure of the best set up we need to limit : -upstream and downstream for users -time that can be connected to free wifi on bus -how much data each user can used . -user will search for the wifi connection name and enter a password ( user w...

we are looking for cluster freeswitch server with load balancer with given below features. A- API Documentation B- Deployment Diagram if available C- System Installation & Configuration guide 1- API Server 2- Freeswitch D- Complete Source code E- Funcationality 1- Live calls 2- amd beep detection 3- Play messages on press 1, press 2, press 3 etc... 4- after press 1, 2 send call to sip extens...

We need to create a custom asterisk channel, that work like chan_console, but instead to use local audio card it will use a spice connection to a kvm server for bidirectional audio. What is played from the virtual machine must be forwarded to the caller and audio coming from the caller must be streamed as microphone of the virtual machine. *Asterisk version 16 *Operating system for asterisk Linux...

I need to send bulk/campaign SMS messages to mobile numbers all over the world. I don't want to buy SMS API/Gateway. Just I want to configure SMSBlast coding to my PBX. Is it possible and can anyone of you quote the best price. I will look into it. Office is in Hyderabad, PBX server is also in the same place.

We have an Entry Phone App which we need to improve. We wish to expand out the number of features offered to user of the app. The current interface is too limited and cumbersome to offer our proposed feature set. We wish to develop a smartphone APP in both the iPhone and Android platforms. This is a large project, but for now, we're only looking for a freelancer who can complete the first ...

Looking for a network administrator, who can setup the entire network, VOIP phones & Firewall.

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Hi, I need somebody who is familiar with Twilio to set our IVR up. Pretty simple but I have no idea what I'm doing and the Twilio manual is gibberish... Cheers Richard

Needs integration asterisk server with odus. Ai

We need to develop a SIP to Viber/Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber/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 Viber/whatsapp executables, or by using the Android / Windows Phone ...

we need to integrate to video call in kamailio, we already have voice call set up.

help with error receiving calls to pots

Hello i am looking for an expert user on freeswitch pbx system. i need to learn lot of things on this freeswitch depend on my time time need. at the moment i am using fusionpbx for the multitanent. but i need to learn few things like inbound route outbound route dialplan how works etc etc etc. i am a advance user so i am looking for expert user. person will be help me on the anydesk on my laptop...

Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support any client Asterisk instances that we man...

SCREENSHOT: [fazer login para ver a URL] FREEPBX 15 location on server: /var/www/html AJENTI PANEL (UBUNTU 18.04 lts, php 7.2, fpm, sql etc) installed You will have access to: ajenti panel to fix nginx configuration for FREEPBX (and also ssh if needed) Extra help for you [fazer login para ver a URL]

I need to direct a DID to my server, Create a new ring group and extentions , and register 3 softfones.

We are implementing a VIOP system. We need a consultant in OpenSips or similar

Need English speaking outbound call center agents with their own dialer software for outbound USA calls. Dialer must be able to do warm transfers.

We set up SIP Trunking and need tech support on Configuration and occasional troubleshooting for Adtran TA908e Routers for SIP trunking into TA908e and out to PBX as PRI T1

I need a voip platform where users can make international calls

We want to create a new chat system based on xmpp and ejabberd. We want a people that help us to configure the system and push notification for device based on GCC FireBase.

FreePBX (Asterisk) Freelancer in bangalore, India

We need a sip-bridge application which will work in android. 1. Calls will start from asterisk with sip protocol. 2. The sip-bridge application which will work in android will take the coming sip requests and make the calls via skype/viber/bip. Calls will be transfer to the mobile number, not his skype/viber/bip (to the called person's mobile number). The info about which number is gonna be ...

Details will be shared winning bidder. please bid if you have the experience.