VoIP Jobs

We are looking for a technical person who has good telecom experience and can help us developing ringless voicemail broadcasting system. There are very few companies are doing this and we have no idea how do they do that. Here is detail about companies that provide ringless VM : [login to view URL] [login to view URL] Http://[login to view URL] You can read about it and tell me if you can do it....

We are looking for a solution, to build a Push to Talk network, using 2 way handheld device, smartphone device over Mobile Carrier Data. along with a Dispatch, or base station. Please look at zello as a reference. Only serious party, as this will take a longer time to create and deploy.

I need a 3rdparty Cisco webcti finesse gadget with following functions: [login to view URL] call (agent reserved) (phone ringing) 2. Hold a call (agent in taking ) call on hold .. retrive button and time for hold 3. Duration of call./ Wrapup timer 4. Transfer a call 5. End call 6. Make a call to certain numbers

I need someone to configure threat feed/ips signatures updates, HA conf, setup ACL and setup policy map to set inline deployment of firepower module

we just install configure GOAUTODIAL 4 we need some explanation and fix some issue we are facing we can connect the agent to the server ?

we want to customize the agent interface of the goautodial

please offer you the best-discounted price for the Installation & Configuration of the following BOQ. S/N Part Number Description Qty 1 R-UCL-UCM-LIC-K9 Top Level SKU For 9.x/10.x User License eDelivery 1 2 CON-ECMU-RUCLUCK9 SWSS UPGRADES Top Level SKU For 9 1 3 CUCM-VERS-11.5 CUCM Software version 11.5 1 4 LIC-CUCM-11X-BAS-A UC Manager-11x Basic Single User License 9 5 CON-ECMU-LICXBASA SWSS...

Hi I need to modify ASTPP so I can have weekly billing. I need you to know ASTPP, I have started to make the modification and it is almost ready, so I can't afford losing time explaining further. If you know ASTPP and know how to do it (its an easy few lines modification). Please apply. Below the exact modification i need to get: - I should be able to chose weekly billing and pick up the...

I need a modulation algorithm, which can modulate any binary bit steam, to let the bit steam go through VOIP compression and decompression, and then be recovered with minimum distortion. VOIP for example is skype, or wechat, etc.

Billing customization , Callback customization., CDR customization , Multi owner ,Reseller solution Clustering / Backup / / Monitoring , Security / Hardening Pls send me your lowest & best quote by remote access for the below requirement OVERVIEW I have to shift my existing asterisk vicidial server files ( my dial plans, modules , database ,IVR , custom code on the server, user details, AL...

A mobile VoIP app solution to make calls from App to phone numbers all over the world

we have cisco ASR router need to integrate it with freeradius, to provide connection rate/speed limit based on the plan provided by freeradius. integration between freeradius and cisco ASR has been completed successfully but we still need to apply connection rate/speed between freeradius and cisco asr router

This current project shall be awarded to individual who can install IP based PBX alongwith open source billing module ([login to view URL]). The IP PbX shall be asterisk based on linux platform. VPS for installation will be provided. The payment terms include 50% payout at the time of 50% work completion and rest at the project closure. Please note that we do make any advance payments. Thanks

Looking to have installed a Vicidial Cluster with High Load, High Availability and viciphone using AWS (or similar). May also need ongoing support.

After login, receive message "cannot connect to asterisk"

small office of a company complex network design is required, pkt i cannot open, somebody tried to do it before, maybe is fake the pkt..file

Dear all, I am seeking someone who can write a script that will install LAMP, asterisk, a2billing 2.1.1, and FreePBX automatically on CENTOS 7 by firing one or 2 command line(s).

connect vtiger 7 to elastix 4 can see recording when call popup that call incoming

I have requirement of asterisk PBX developer

Hello, We need to integrate Ring Central with Bitrix24 CRM telephony services. Need the support on urgent basis. Please contact only, if have experience in same,

I have an ios VOIP app customized using Linphone source code. I need to upgrade using the latest Linphone version. I want to keep existing rules and customization but modify the audio options and turn on a few more options. My app name is VoiceHi which can be downloaded from apple store . Please see attached file. Please use the logo, app icon and startup screen from the ones in the source code? ...

I have an ios VOIP app customized using Linphone source code. I need to upgrade using the latest Linphone version. I want to keep existing rules and customization but modify the audio options and turn on a few more options. My app name is VoiceHi which can be downloaded from apple store and google play. Please see attached file.

Hi -I want to install freepbx and configure termination gateways on my server using vpn. I already installed asterisk and a2Billing -configure a connection with clients

hi i want to register sip to sip and pass call g729 problem is i dont want to use big pc and lots of HW so i found a program its allow to do that i want [login to view URL] i want to run this program on a router Router Link [login to view URL] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580 Flash MB: 8 RAM MB: 64 server send call to router >>router send to voip gateway w...

Looking for a good consultant with experience in Fusionpbx / Freeswitch who has worked in supporting production environment and has done customization.

In need of software or asterisk/freepbx hosted solution for "Voice Broadcast" or "Robo Dialer". - Connects to SIP Trunk - Alternate Caller ID; Area Code Caller ID campaigns with alternating numbers - Ability to play and add audio files messages that are randomized - Multichannel lines (100) to make hundred or thousands calls per minute - Choose number of outbound calls per mi...

We have 2 platforms: 1. Freepbx 2.11 and Asterisk 11 2. FreeBpx 14 and Asterisk 16 CallBack makes callbacks simple. When your agent receives a callback request, they will see all of the caller’s information in the screen. They can click one button to accept the callback request and [login to view URL] will automatically connect them to the caller. This process will save your agents a cons...

Cloud Telephonic and IVR Solution

hi have a ip500 with ip office ver 10. with analog phone line and zip trunking were install some time back 40 constant calls only can see 24 of them 10 1616i phones with bm32 module so I need some body with Experince in avaya platform also will like to connect a remote phone,voice mail

Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for this app will be , our remote server send a c...

I need an Outbound Appointment Setting agent to schedule Phone Conference Meetings. I will supply the List, Appointment Setting Script and Process...

need someone to fix the extension file on the asterisk script

I have an ios VOIP app customized using Linphone source code. I need to upgrade using the latest Linphone version. I want to keep existing rules and customization but modify the audio options and turn on a few more options. My app name is VoiceHi which can be downloaded from apple store. Please see attached file.

I have an ios VOIP app customized using Linphone source code. I need to upgrade using the latest Linphone version. I want to keep existing rules and customization but modify the audio options and turn on a few more options. My app name is VoiceHi which can be downloaded from apple store.

i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many peppole allready made it you need to make a softwar...

I’m looking for an expert of VOIP, SIP, Hack for educational purpose

We have installed OpenAnswer, FreePBX, and Asterisk on a AWS. We are looking for someone familiar with the software to finish fully setting it up for us. This includes changing all the default passwords. After setting it up we want to upgrade to PHP 7.2, and NodeJS 8. We are also looking to remove all test files, and default configuration files that are not needed. The system will already keep tr...

We're looking for someone who's able to get these key features to work between the 3CX PBX and Odoo 12: [login to view URL] If incoming call hits the 3cx server we need to open the contact from odoo linked to that phone number. Like here [login to view URL]

I have an ios VOIP app customized using Linphone source code. I need to upgrade using the latest Linphone version. I want to keep existing rules and customization but modify the audio options and turn on a few more options. My app name is VoiceHi which can be downloaded from apple store.

I have a vps server centos7-64 I would like to install the latest version of the following -asterisk/freepbx/a2billing/fail2ban/vpn -a2billing load balance traffic across termination gateways -IVR filter to drop test call generator (press DTMF to continue call)

i am having trouble using a dinstar simbox for marketing purposes .i need some help

we have a call out page already , need someone to change the extensions to an ivr based answer

I have a vps server centos7-64 I would like to install the latest version of the following -asterisk/freepbx/a2billing/fail2ban/vpn -a2billing load balance traffic across termination gateways -IVR filter to drop test call generator (press DTMF to continue call)

I have setup an Asterisk Cloud deployment and need some help tweaking the final steps of the setup

I want my IP PBX to be connected to Twilio WebRTC so that can be able make calls using Twilio SIP account. Calls from my PBX will come from an IP so I need someone who can make the set up for me to start using Twilio voice trunk.

I have an ios VOIP app customized using Linphone source code. I need to upgrade using the latest Linphone version. I want to keep existing rules and customization but modify the audio options and turn on a few more options. My app name is VoiceHi which can be downloaded from apple store.

i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know

you have to install skype for business 2019 and set it up. we have multiple sip phone numbers which have to be also connected to skype for business 2019 for in and outgoing calls.

Require someone who can develop a chatbot for IVR/Call for Indian company and users. Not looking for Google Home and Alexa bots.

Need Asterisk Web Developer, Need Application like VOIP Service, Miss Call Alert service, You can check demo like myoperator , knowlarity etc