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.

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 ...

I need to customize some litle aspects of the GUI of the ASTPP Billing. This is the customization that I need: 1 - Show active calls on clients login 2 - Show negative "balance" for post-paid clients, exactly like the PRE-PAID clients, not the "credit limit" information. 3 - Create a random callerID for specific sip devices, like that: If I call to the number 55 48 996843116,...

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: [登录来查看链接] 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 [登录来查看链接]

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

install sms features in Twilo just receiving

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 ...

We are seeking an experienced programmer to develop and manage our Freeswitch and ASTPP billing platform currently housed on a Digital Ocean server.

My requirement is to get a high-level definition document created which talks about design and implementation of a secure VoIP calls in a local area network with appropriate dial plans in a fully virtualised environment More details in the attached document The document is based on hypothesis and would be roughly 20 pages. Details attached Timeline- 8 working days from date of allocation to th...

ive issabel asterisk 11 and vtiger 7 installed and configured i've even configure Vtiger connector.. nowi just need you to configure the dialplan so the users will start getting the pop on on vtiger Inbound +Outbound (Click 2 Dial or Click 2 Call) these 2 should work

I want to create a kamailio dispatcher but does a lookup in dB to locate the asterisk server which accounted is located on.

Hello I am going to start a hosted pbx system business. i need IVR for my clients. female voice UK accent . if you are available and if you can help me all the time please apply for the job thanks

Vtiger VOIP4 expert, need one time setup on UBUNTU 18.04 In VTIGER 7 you can use 3rd party modules or maybe you have scripts/modules that already working with vtiger - DONT ASK ME ABOUT modules - i dont know, you need find a way or possibly you have already those in your vtiger voip installations, use github, google, 3rd party voip free etc modules for vtiger. I need in VTIGER - voip features:...

help me find unlimited flat monthly rate pbx trunking to the Caribbean

In a scenario where there is: - A Asterisk VoIP server (provided to you) - A phone extension connected to the Asterisk server (provided to you) You should build an Android APP that connects to the Asterisk Server and is able to do the following: A) Receiving a GSM Call and Making a VoIP CALL 1) Automatically answer phone calls 2) Take GSM Audio call and create a voip IAX2 protocol call t...

I need a programmer to install an new instance of Freeswitch and then configure the PBX to match the PBX I have built in Freepbx.

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

i am looking for someone who may have a source for sip phone / client for windows

I have a asterisk server maintained by a freelancer, he is not responding from a few days ,I need to make changes, dial plan have all the required settings just it is disabled , I need some one to help me enable and disable dial plan , my budget is 10$ , you do not have to write anything just guide how to disable and enable dial plan.

Unable to play music file in FreePBX [登录来查看链接] error "Unable to find an intermediary converter for /home/asterisk" And FXO Tele Routing & FreePBX Inbound Configuration need to be setup

1. installation for huawei FW switch router. 2. Basic configuration and intergration 3. Perform one accept with customers 4. familar to equipmen of Huawei FW router switch

these are the details 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 sys...

I need some help with finding some leads. Please contact my leads database and follow up with potential customers.

I do have existing cisco 2800 router with 1(one) analog line connected 10 cisco ip phone 7965. I need to add another phone line but it is digital line. I need to configure and change all the phones name and extensions. And also, adding 2 (two) more phones. And i need to call out each line with different company caller ID and when we dial out as 9 Company A, Company B dial out 0. current fax line ...