Asterisk PBX, like any other PBX, is a complicated subject that is best handled by experts. If you are a pro in this field, then you should bid on the many jobs at Freelancer.com.

Asterisk PBX (private branch exchange) is implementation software. Created by Mark Spencer in 1999, the software simply allows connected telephones to make calls to each other and also to connect to other services. The name is based on the symbol asterisk, (*). For Asterisk PBX to function as it should, the configurations must be on point, which is why this should be done by an expert.

Asterisk PBX is a topic that needs skill and if you are an expert in this, then you should earn money through what you know best. There are thousands of jobs posted on Freelancer.com related to Asterisk PBX and if you at a pro in this particular field, then Freelancer.com will offer you a chance to work on projects you understand. The site attracts some of the best-paying clients and offers an easy-to-use platform, where freelancers can browse and bid on jobs they are interested in. You can simply start your career in Asterisk PBX at Freelancer.com today.

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    18 仕事が見つかりました。次の価格: USD

    Searching personal assistant to support ongoing project for small Voip company. You must have Web Development and VoIP Development skills. Make the bid with your skills inside, please. Only for people who want to work on long-term project with partial hiring.

    $508 (Avg Bid)
    $508 平均入札額
    12 入札

    It's a full time opportunity. Need someone who could do cold calling as well as offer customer support.

    $7 / hr (Avg Bid)
    $7 / hr 平均入札額
    6 入札

    Hi, I am looking to start VOIp business I need my own network and also billing platform Below are competitor websites I need system like that Inbound call forwarding service And voip incoming and outgoing I require billing portals to be good one where customer can see real time everything and foraward calls to another number I need someone experienced in industry I like my system to be setup on cloud azure or aws.. Asterisk and freeswitch Competitor websites Vonage Ttnc Ring Centrial Tamar voip

    $1660 (Avg Bid)
    $1660 平均入札額
    12 入札

    PLEASE READ THIS FIRST BEFORE BIDDING!!! IF YOU HAVE NOT DONE THIS EXACT JOB BEFORE, PLEASE DO NOT BID.. Hi, Have you integrated a kolmisoft softswitch into a calling card/VoIP Website using API with before? Where users can signup, login, topup account? Do you already have a site that I can look at?

    $163 (Avg Bid)
    $163 平均入札額
    3 入札
    Linphone Kolmisoft integration 5 日 left
    認証完了

    I have the source code for Linphone which is configured for A2Billing. I need to configure the Linphone to connect from A2billing to Kolmisoft. Or if you have a better softphone solution for Kolmisoft, then lets talk about it. I need to be able to: -Signup via the softphone (Using SMS) -Log in -Make calls -Topup -Wake from sleep

    $256 (Avg Bid)
    $256 平均入札額
    5 入札
    Install SIP 5 日 left
    認証完了

    I need to install SIP, for example from Nexmo or vonage

    $137 (Avg Bid)
    $137 平均入札額
    7 入札
    Avaya IP office with ASBCE -- 2 5 日 left
    認証完了

    i Have Avaya Server edation R11.1.2( VM) with Remote Worker and Sip trunk connected to ASBCE R 8.1.2(ASBCE CORE PORTWELL CAD-0230) it was working fine but now when im traying to call from remote workerto outside or local ext its givin me call failed the user not found , some times if i close the Avaya IX work place and open it again its working , in ASBCE i got two error 401 unuthorized , and 404 not found .

    $30 (Avg Bid)
    $30 平均入札額
    2 入札
    Avaya IP office with ASBCE 5 日 left
    認証完了

    i Have Avaya Server edation R11.1.2( VM) with Remote Worker and Sip trunk connected to ASBCE R 8.1.2(ASBCE CORE PORTWELL CAD-0230) it was working fine but now when im traying to call from remote workerto outside or local ext its givin me call failed the user not found , some times if i close the Avaya IX work place and open it again its working , in ASBCE i got two error 401 unuthorized , and 404 not found .

    $25 (Avg Bid)
    $25 平均入札額
    1 入札
    install nexmo /vonage voice api 4 日 left
    認証完了

    install nexmo /vonage voice api

    $190 (Avg Bid)
    $190 平均入札額
    1 入札
    Kannel fix 3 日 left
    認証完了

    My kannel after second http message goes down and stop sending and reject messages only experts please Message was rejected SMSC response was: 0x7f2344001620

    $25 (Avg Bid)
    $25 平均入札額
    2 入札

    I need a call tree flowchart with call forwarding, major voice recognition and minimal dtmf. Would prefer process to be visible and changeable in the flowchart itself. Caller calls in and says name of individual or department and is transferred to external phone number

    $21 / hr (Avg Bid)
    $21 / hr 平均入札額
    5 入札

    Install DSIPROUTER and KAMAILIO on EC2 instance (which will be provided) and configure LCR.

    $114 (Avg Bid)
    $114 平均入札額
    6 入札

    I'm looking for a developer for make custom reports of 3CX Phone System ( Web Dashboard ). I can send more details when request.

    $541 (Avg Bid)
    $541 平均入札額
    6 入札
    PHP Laravel Integration with VOIP 1 日 left
    認証完了

    Hi, I am looking for VOIP, FreePBX expert who must have experience with PHP laravel. I will share more details through chat,

    $30 - $250
    シール NDA
    $30 - $250
    12 入札
    install a caller to phone system 1 日 left
    認証完了

    I would like the caller to be charged 0.90 cents per call for the incoming call and to be credited to us and he has the opportunity to leave a message for the callback, which is set up as a voice mail by email with us for several phone numbers. can you do that?

    $11 / hr (Avg Bid)
    $11 / hr 平均入札額
    7 入札

    Twilio Expert - I want to integrate API for Voice call, SMS, WhatsApp in our CRM 2 You need to guide only, as we doing our self. So you need to have good knowledge and experience about API and costing and how those all work. We need Call, WhatsApp SMS all three API to integrate to our CRM. Thanks!

    $14 / hr (Avg Bid)
    $14 / hr 平均入札額
    1 入札
    PBX recording system 23 時間 left

    Voice recording system to IP PBX based on CTI signaling. IP PBX already exist (Cisco CUCM). Don't need frontend, only backend with API's.

    $44100 (Avg Bid)
    $44100 平均入札額
    10 入札

    Step 1 :- A server ------> Invite --->> Bserver ------->> Invite ------->> PSTN Step 2:- A server <------ 100 trying <<--- Bserver <<------- 100 trying <<------- PSTN Step 3:- A server <------ 183 session Progress <<--- Bserver <<------- 183 session Progress <<------- PSTN Step 4:- A server <------ 183 session Progress <<--- Bserver <<------- 180 Ringing <<------- PSTN A Server >>> Installed Asterisk with Freepbx B Server >>> Installed Asterisk with Freepbx PSTN (SIP) attached with Server B An extension 123 created on Server B On Server A created a SIP Trunk with configuration of extension 123 that was created on Server B Also connected server B with the server A ...

    $960 (Avg Bid)
    $960 平均入札額
    5 入札