Voice over Internet Protocol or VoIP refers to a host of transmission technologies meant to deliver voice communications over IP networks that include packet switched networks and Internet. VoIP systems include communication services such as facsimile, voice, and voice messaging applications. A number of firms are shifting to this system to promote a cost effective medium of communication. You can hire an expert who can help your firm to leverage the potential of this new system. Just post a job today! Ansett VoIP Developers

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    48 jobber funnet, priser i USD

    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

    $73 (Avg Bid)
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    SIP register with Custom software 6 dager left
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    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 [logg inn for å se URL] i want to run this program on a router Router Link [logg inn for å se URL] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580 Flash MB: 8 RAM MB: 64 server send call to router >>router se...

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

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    Multichannel Phone Dialer 6 dager left
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    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...

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    Asterisk 16 Queue Call Back 6 dager left
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    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 [logg inn for å se URL] will automatically connect them to the caller. This process will save your age...

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    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...

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    Cloud Telephonic and IVR Solution

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    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

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    sip to Native dialer APP 5 dager left
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    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...

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    Outbound Appointment Setting 5 dager left
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    I need an Outbound Appointment Setting agent to schedule Phone Conference Meetings. I will supply the List, Appointment Setting Script and Process...

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    need someone to fix the extension file on the asterisk script

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    Upgrading Linphone for ios 5 dager left
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    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.

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    Upgrading Linphone -- 3 5 dager left
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    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.

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    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...

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    VOIP, SIP Expert 5 dager left
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    I’m looking for an expert of VOIP, SIP, Hack for educational purpose

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    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...

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

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    A2billing Configuration 5 dager left
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    Hello I need someone to complete a2billing configuration for an customer .

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    Upgrading Linphone -- 2 4 dager left
    VERIFISERT

    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.

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    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)

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

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    ivr script 4 dager left

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

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    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)

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

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    Connect IP PBX to Twilio WebRTC 4 dager left
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    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.

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    Upgrading Linphone 4 dager left
    VERIFISERT

    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.

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    SIP Proxy Server install 4 dager left
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    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

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    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.

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    IVR Chatbot 3 dager left
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    Require someone who can develop a chatbot for IVR/Call for Indian company and users. Not looking for Google Home and Alexa bots.

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    Asterisk Web development -- 2 3 dager left
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    Need Asterisk Web Developer, Need Application like VOIP Service, Miss Call Alert service, You can check demo like myoperator , knowlarity etc

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    Ringless Voicemail Software 3 dager left
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    We're looking for freelancers/companies who have already built "Ringless Voicemail Software" or who can build it in short period of time.

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    Secure Issabel PBX (asterisk) 2 dager left
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    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

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    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...

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    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...

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    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...

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    cisco engineer for switch verification

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    Need cisco engineer for cisco hardware testing

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    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.

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    Twillo or Nexmo expert needed 1 dag left
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    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

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    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.

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

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    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...

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    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]

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    Connect Digium D60 to FreePBX 9 dager left
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    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.

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    We need to build script via php or some other language where We can compare the hang up call + live Call acd and based from configuration we can hang up the call

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    i need a custom voip switch like astpp which can handle above 1000CPS and CC with DID Management and recording with security

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    required a AVAYA Call Exchange PHP scripts for SIP, call recording, call records and so on ... along with DB

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    ASTTP Installation Avsluttet left

    We need to install the following; 1 Asttp Billing 2 Import backup 3 Install Codec G729

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