VoIP Jobs

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 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...

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...

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]

Hello I need someone to complete a2billing configuration for an customer .

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

We're looking for freelancers/companies who have already built "Ringless Voicemail Software" or who can build it in short period of time.

I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

Customer should Signup with Hosted services and chose from the ready made IVR Applications or get one developed at effective cost Click to call features: 1) When number clicked on webpage, viewer types in his/her phone number and submits to call other party, Click to call calls and connects both parties 2) Ability to add an audio file message (in wav or mp3) to the placed call, greeting both pa...

The system is a very simple auto attendant with 5 menu options. Four of them just go to simultaneous rings and collect a voicemail if no answer (so it must prompt on outbound to ensure it isn't transferring to a personal voicemail). One goes straight to voicemail. All voicemail collected must be texted or emailed (with link to voicemail recording). Any code that cannot be hosted by SignalWir...

We are looking for a company who has 24/7 server administration. We have between 4-6 servers running at the same time, and we are looking for a new company, who can be in charge of everything concerning the servers. It is important that there is 24/7 support, and that there is system admins attached, if it is not only system admins. If every case goes through level 1 or level 2 support first, the...

cisco engineer for switch verification

Need cisco engineer for cisco hardware testing

I would like create agent survey on Issabel/Elastix. Need modification in UI as well like adding menuitem under call center reports section to show Agent survey results page within elastix/issabel UI.

I am looking for an expert that can set up SMS sender through twillo or nexmo , you must be an expert at this for us to proceed. I am not looking for people without experience

We are currently doing a localised speech to text for SEA. We need a UI designer to help us to design the output from our ASR. An image to show the example of it. We will need to insert time code as well.

Build online internet banking and telephone banking /Twilio ivr with real time transaction to any bank in the world .

Hi, I deliver VoIP services to my customers and i use 90% of the time the Yealink SIP phones. The problem in 50% of the cases is that when the client is turning on their voicemail, they can't see it anyway if it is turned on yes or no. I know it is possible to use something with XML and a webserver to create a connection with the phone which can enable / turn on a led to indicate a button i...

We have developed a complete workflow management software the allows clients to use the CRM that is integrated with our VOIP phone system. It is built in asterix. We need developers to maintain and manage our asterix phone system within US timezone. This will be an ongoing project. Our website is [removed by Freelancer]

We need someone to connect our Digium D60 to FreePBX and twilio. Provide detailed instructions of setup so that we can replicate. We already have a Twilio account. A FreePBX instance set up and our phone connected to the server. We can't figure out what's missing. Start your bids with your favourite color.