Asterisk bandwidth compression jobs

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    24,236 asterisk bandwidth compression jobs found, pricing in CAD

    I have requirement of asterisk PBX developer

    $28 - $232
    $28 - $232
    0 bids

    ...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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    ...gateway routing protocol Evaluate the QoS (quality of service): • Delay • Variability (jitter) • reliability • (latency) • Lost packets • Package traffic (sent and received) • Bandwidth CONDUCT A SUMMARY OF THE ANALYSIS OF THESE ROUTING PROTOCOLS AND SUGGEST THE MOST APPROPRIATE FOR MOBILE NETWORKS AD-HOC Comparison of routing protocols: METRICS OF ROUT...

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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

    $44 / hr (Avg Bid)
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    Multichannel Phone Dialer 6 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 6 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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    ...(will be provided in the format required) > Google analytics, tag manager & other analytics enabled (tracking codes will be provided) > CDN needs to be implemented > GZIP compression needs to be enabled > HTML/CSS/Javascript needs to be minified > Proper author profiles need to be there > Schema needs to be implemented The design for the blog is ready

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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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    Develop Procurement platform 4 days left
    VERIFIED

    ...purchasers will post a clear RFX and all supplier approved or not approved by eh purchasers will send offered for items listed by purchasers , using latest technology of compression/rating the supplier bids , and managing products catalouge of every suppliers for upload and export the list of products need to have the following as basic : supplier

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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 4 days 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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    edit image -- 3 3 days left
    VERIFIED

    want to edit this image [login to view URL] (laptop frame), ...same as on [login to view URL], i have both image laptop and screenshot from app (see attachment) dont resize laptop, someone give me laptop smaller as original size (crazy!) or compression too much and resolution bad (crazy more!), so simple task and people dont understand. This task is for 30 mins work

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    ...as the upstream webserver. Nginx handles SSL connections, caching (simple cache) and on-the-fly zlib(-ng) compression upon delivery of files to the browser. We want to outsource development of an Nginx module that alters how caching & compression is handled in Nginx. Currently Nginx reads the uncached files from Apache (thus not from the file system)

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    Featured NDA
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    Personal assistant on monthly basis 150-250$ Solid bandwidth required. English grammar mastery a must. 40 hour week needed with time tracking. Clear audio voice needed. No bad microphone or low speed internet.

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    ...and the the PHP file are attached. We might also consider this to be done client side using something like XSLT. Notice the XML compresses down to a few hundred K with a compression program such as gzip. Your first quote is expected to be your final quote, I will not accept anyone changing their bids and asking such questions as "What is your budget

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    I want to add 2 custom header fields to the OpenFlow table. 1 buffer threshold (e.g. 80%) and 1 link bandwidth threshold (also in percent). An incoming flow is matched against that row only if the current link bandwidth or buffer usage is below their thresholds.

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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 3 days left
    VERIFIED

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

    $525 (Avg Bid)
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    ...addition, name , address, LC ID, location, Link type, media give, hardware details, ISP provider, etc. ii) Fault raising should have Sr. No , Location Name, LC ID , Link Type, Bandwidth, down type , up time, Failure remarks etc 3) Final it should give the output in excel sheet for every month end, with calculation of failure rate %. It should also have a

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    Secure Issabel PBX (asterisk) 2 days left
    VERIFIED

    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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    ...to work Compression - output size is reduced from the original size Unicode - Support any system language Multiple Saving Formats - .exe .scr .com .pif .bat Arguments - Support Shell arguments Icon Creator - Creates a random pixels icon Assembly Cloner - Cloned Assembly is a Perfect match File Binder - Supports any file format + Compression Anti Memory

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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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    clean up cpanel files 1 day left
    VERIFIED

    We have a WordPress site on a vps with Hostgator. It's become bloated over time we need someone to clear out any unnecessary files to stop excess bandwidth and clear any speed issues

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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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    ...+ MAGENTO work. My DEDICATED Server has been tested, NO HARDWARE ISSUES. All modules and gzip compression are enabled on server side. Server Details: Processor E3-1230 V2 @ 3.30GHz | Ram 32 GiB Disk 4 TB | Operating System Citrix Xenserver 6.2 Bandwidth 1gbit: 20TB Traffic If this is not the best server, suggest a server. TASKS: 1. Optimize mySQL

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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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    The resultant 'lossy' file means that quality can never be recovered. The JPG format is often used in digital camera memory cards. The JPG file is a great format as it of...camera memory cards. The JPG file is a great format as it often manages to compress files to 1/10 of the size of the original file which is especially good for saving on bandwidth.

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    ...indicators to be written are around 80. See sample list below ATR Average Trend ADX Alligator AMA Accumulation And Distribution Line Band Squeeze Bollinger Bands Bollinger Bandwidth CAGR Candlestick Momentum Index-CMI Chikou Span Chaikin Volatility Commodity Channel Index CMO Coppock Disparity Index Dochain Channel Efficiency Efficiency Ratio Force Index

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    ...The tap target <a href="tel:3767007">376 7007</a> and 4 others are close to other tap targets. Speed Reduce server response time Enable compression Compressing resources with gzip or deflate Enable compression for the following resources to reduce their transfer size by 160.1KiB (73% reduction). i.e. Compressing [login to view URL]

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    ...LIVE Push-To-Talk. - No-voice-when-receiving mode; when receiving live Push-To-Talk recording other users cannot use the Push-To-Talk button. - Fast delivery; using low-bandwidth live Push-To-Talk recordings / communication. - Speaker & Playback; sender and receiver must have as look-and-feel and usability the following options; 1) All send (live)

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    ...LIVE Push-To-Talk. - No-voice-when-receiving mode; when receiving live Push-To-Talk recording other users cannot use the Push-To-Talk button. - Fast delivery; using low-bandwidth live Push-To-Talk recordings / communication. - Speaker & Playback; sender and receiver must have as look-and-feel and usability the following options; 1) All send (live)

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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 4 days 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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    Design an LTE module in NS-3 simulator where we need to analyze resource depletion attack and defense mechanism and ddos attack and defense mechan...and defense mechanism and ddos attack and defense mechanism. We also need to analyze the energy utilization of the mobile device. Resource depletion mechanism parameters - bandwidth, latency, throughput.

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    now i have one like when a calls come to server [login to view URL] [macro-answer] exten => s,1,AGI([login to view URL],${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 c...

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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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    ...clients will have their own permissions and access. Main things: Maybe same this [url removed, login to view] Users can view there online time and bandwidth usage. Ability to limit bandwidth usage per product/plan. Ability to limit download/upload speed. Automatic account creation through WHMCS. Automatic Suspension/Unsuspension.

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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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