Asterisk web screenshot jobs

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    11,779 asterisk web screenshot jobs found, pricing in USD

    I need a web application make in php, mysql, ajax with nice css style. A form that allows the inclusion in a table (which is updated with ajax) of the various practices made by a user of a team. USER LOG IN - Drop-down menu from which to choose the customer - Drop-down menu in which to indicate the name of the practice performed - Time taken - Small

    $271 (Avg Bid)
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    Hi Arshad, we're looking for someone skilled in FreeSwitch or Asterisk to build a ringless voicemail system. We operate [login to view URL] which is a popular ringless voicemail platform in the US. However, we use other provider's APIs to do the delivery. This isn't cost effective for us. We need to operate our own delivery infrastructure. Is this something

    $20 / hr (Avg Bid)
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    ...website and take screenshot of the webpage in high resolution both for desktop and mobil mode and save it on the server. We need someone to finish the work in one week. The idea is: 1. Upload an excel file with eg. 1.000 domains 2. If the domains/website contain special word we take screenshot After that we manually look at each screenshot for an extra

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    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....

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    ...OVERVIEW I have to shift my existing asterisk vicidial server files ( my dial plans, modules , database ,IVR , custom code on the server, user details, ALL CONFIGURATIONS. etc ) to a better hardware /server with Xeon processor 3.2 GHz.& 16GB RAM NEW VICIDIAL server details & its configuration ● Centos7, Asterisk, FreePBX & VTiger is already Installed

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    Screenshot Edit 4 days left

    Hi, I need someone who can make a little change just need to update few words on a screenshot This is emergency basis , Please write me if you are ready to do now immediately

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    ...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

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    After login, receive message "cannot connect to asterisk"

    $21 / hr (Avg Bid)
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    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).

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    Hi, I look for Freelancer who can implement Asterisk with my MGCP provider. Not SIP. Can somebody do? Thanks

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    I have requirement of asterisk PBX developer

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    ...Inventory/Asset Management Tools ([login to view URL] with GLPI) e. Visitor Management Tools ([login to view URL]) f. Printer Management Tools g. ID Management Tools (Open LDAP) h. VoIP Tools (Asterisk, FreePBX, FreeSwitch) i. Messenger Tools (Jabber) Basically the idea is to help new IT Managers and IT administrators to manage IT infrastructure using opensource tools to

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    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

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    ...>>router send to voip gateway we install our program on router and our router OS is openwrt like server to router resisted and router to Voip gateway all connection By SIP asterisk server to creytiv/re router then creytiv/re to VOIP gateway some one already made it they can run more then 64calls but if you can run 32calls i am Happy need a easy way

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    im looking for asterisk pjsip developer for support our system

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    Multichannel Phone Dialer 3 days left
    VERIFIED

    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

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    Asterisk 16 Queue Call Back 3 days left
    VERIFIED

    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

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    Update the HTML file to look like the output on the [login to view URL]

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    need someone to fix the extension file on the asterisk script

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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

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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

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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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    we have a java script which is connected to asterisk we need someone to help change the settings

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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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    ...portal, generate a voucher of the customers choice, enter the customers number, then click buy. The consumer will get a SMS containing Voucher number. (SMS sent out via local Asterisk Server) Major features for the initial design include: ● A minimal UI than allowed vendors to generate a unique token for client use. (UI needs to be made modern) ● App will

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    SIP Proxy Server install 1 day left
    VERIFIED

    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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    i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for

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    Asterisk Web development -- 2 Ending left
    VERIFIED

    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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    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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    ...SIP accounts (with hunt groups). Of course system needs configuring Asterisk for custom DID:s, call forwarding, internal calling, voicemail, ability to activate/deactivate service at any time, etc. Iam looking for a relationship with developer who will handle full deployment of asterisk with billing,GUI , with these functions : call forwarding, voice

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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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    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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    Take screenshot automatically without any icon on screen and save it in device storage

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    we have 2 extension files we want someone to copy the dial out routine from one to the other

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    Edit a screenshot and make into a PSD so all information can be edited.

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    ... [root@pbx]# fwconsole restart Asterisk not currently running Running FreePBX shutdown... UCP Node Server is not running Running FreePBX startup... Starting Asterisk... [--------------->------------] 13 secs In [login to view URL] line 187: Unable to connect to Asterisk. Did it start?

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    ...Push-To-Talk button + Mute button + Speaker button. - Group can be created by users added as contact (or connected into one group in back-end). - Developer can choose between Asterisk / FreePBX -or- XMPP / OpenFire as user management. - Communication must use Over-The-Air must be always 256-AES encrypted. (All live Push-To-Talk recordings must be 256-AES

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    ...Push-To-Talk button + Mute button + Speaker button. - Group can be created by users added as contact (or connected into one group in back-end). - Developer can choose between Asterisk / FreePBX -or- XMPP / OpenFire as user management. - Communication must use Over-The-Air must be always 256-AES encrypted. (All live Push-To-Talk recordings must be 256-AES

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    We are looking for Java developer who can develop CTI & IM desktop software for our asterisk based PBX. for instant messaging we are currently using Openfire server but other XMPP servers are also welcome. We already have design and initial code to start with. Details of project will be shared on request.

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    ...-assistant-using-official-docker-images/42860 . (i want to use this dashboard/ analytics to see the analytics of my server , i have cpanel/whm server on centos 7, i have asterisk voip server, plex media server, i want to follow the performance of them, and i have many website on my cpanel/whm and i want to be able to see unique and realtime visitor

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    Trophy icon Asterixx Entertainment new logo 1 day left

    ...business). The business name is "Asterixx Entertainment". I have attached my current logo, but keep in mind this is not what I am after. If you are able to incorporate an 'asterisk' into the logo that is a bonus but not necessarily going to be the winning design. This logo will be featured on business cards, future website and facebook pages/promotional

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    x code with third party project So when we boot the computer, the ...application start with opening another application ( third party ) Then the user can screenshot the preview . So we need a review screen of thé screenshot and a print Button To print the picture . We need a review screen with print button and a countdown to take the screenshot .

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    ...${ARG1}) and php agi write all incoming calls to database and web php read from DB but i dont know where is prob i saw some people they read active call every second same like console CLI real-time i need something realtime like it have to show all my active dial and all calls on my web panel real time my current one i can do but its take too much

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    PLEASE DO NOT OFFER TO MY PROJECT BEFORE READING. IF YOU READ THE PROJECT PLEASE TYPE "YES I READ YOUR PROJECT" ON YOUR OFFER, AND D...will not ask Twilio for the same number if it is already asked before and already the database. P.S.; definitely we do not want to use the asterisk. We would like to run it on VoipSwitch. please do not offer asterisk.

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    Asterisx PBX application - Receive call, check filesystem (configured UNC path) to determine if audio files needs to be played. - Transfer (bridge) and record call. - When call ends, send recorded audio file to a http endpoint (HTTP). - Configure all into a Hyper-V virtual machine (Windows 2016 server)

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    I want to customize my A2billing with a more current theme, and configure an asterisk live, to have the activated of the graphic CLI.

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    Hello, You are supposed to configure the FreePBX and Asterisk with FXS and GSM Gateway. Freepbx and Asterisk is already installed on the server. Need immediate start on this.

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    Edit a screenshot: Remove text on the top Remove text on the bottom Fix the shield on the hand where the hand is popping out of the shield due to bad physics

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