Sip trunk asterisk quintum jobs

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    17,036 sip trunk asterisk quintum jobs found, pricing in USD
    Asterisk PABX Integration 6 days left
    VERIFIED

    We are needing to integrate an existing service, to receive calls, make calls and root calls.

    $512 (Avg Bid)
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    7 bids

    ...description WORK ORDER- PART 2 1. Simple non technical user UI with SSO ( Sign sign on cloud managed access ) 2. Fully Customized UI for openwrt Barrier Breaker unknown / luci Trunk 3. Should have company logo at the login page and all other pages on the top left 4. Cloud based UI should be mobile/smart device compatible 5. UI should have the dashboard

    $551 (Avg Bid)
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    Hi, My company is in the process of finding a research writer that is interest...will be gain a guaranteed $500 contract to write for the website and most likely more. All of this will be explained in detail later. In the submission piece please put an asterisk as the very first character in your submission and post title so I know you read this.

    $100 (Avg Bid)
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    0 entries
    Asterisk Expert-FreePBX 5 days left
    VERIFIED

    I need someone who can help with a command to display running calls on Inbound Route based on Toll Free DID. Need someone with expertise in Asterisk Communication Server and FreePBX GUI.

    $401 (Avg Bid)
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    9 bids

    I am looking for someone who can setup a FreePBX server on a VPS to be used by our team. > Latest FreePBX version installed on Linux (cannot be installed from the boot ISO). > Firewall rules to close unused ports. > TFTP Server installed and open with config files for Cisco 7940 and 7960 phones. > Setup of 2 lines (extensions) for me to carry internal test between the two phones. ...

    $93 (Avg Bid)
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    7 bids
    Modify JAVA file in App -- 2 5 days left
    VERIFIED

    ...needs to be modified to perform the functions below instead of sms.. starts by placing 2 calls at the same time (one using native dialer other using a SIP account) call one times out after 1 sec (SIP account call) call two placed at same time goes to voicemail (native dialer) call two leaves a prerecorded voicemail at voicemail tone then hangs up

    $109 (Avg Bid)
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    14 bids

    designing a range of modern houses to be constructed using structural insulated panels (sip) this will be an ongoing job, will create a standard range of designs for builders, and clients will have the choice of making alterations to design or layout 3d renderings would be an advantage

    $653 (Avg Bid)
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    ...The sides could be just two parallel lines from front to back, for the front of the hood a 12inch design like the enclosed picture and for the trunk two parallel lines above and a design similar to the two trunk pictures enclosed. Please get back to me by phone or email as soon as possible so that I can proceed to have the job. Sincerely, Jim Medina

    $155 (Avg Bid)
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    1 bids

    Hi, You worked on...ultimately will want to have one of the options on the IVR go to an outside number. I am not really sure how to accomplish this. Also looking at the console for the system "asterisk -rrv" there are all sorts of attempts to connect it seems. Any suggestions appreciated. Would love to work with you again if you are available.

    $75 (Avg Bid)
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    1 bids

    We are looking for an Asterisk or Elastix developer capable of customizing and integrating the soft PBX with a third party distribution. To those of you that claim to hold the skill requested we'll delivery proper documentation about the job. Thank you.

    $597 (Avg Bid)
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    16 bids
    Trophy icon New animation video 2 days left

    ...explain the distances with visual animations of trees, shrubs etc. show that the lawn gets mowed/cut to less than 100mm Distances to the trees are to the canopy edge (not the trunk) the explanation for the canopy cover percent needs to be 10% or less.(nt 15 or higher) It is required to get a (Bushfire attack level) BAL Level of BAL-29 or lowerto be

    $156 (Avg Bid)
    Featured Guaranteed Sealed
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    4 entries

    I get this error when making a call on vicidial. *********** WARNING[24182]: chan_sip.c:21126 handle_response_invite: Received response: "Forbidden" from '"80" <sip:+18xxxxxxxxxx@[url removed, login to view]>;tag=as0c9f627d' *********** On Twilio site Debugger i get this error this. ********************* Error - 32204 MESSAGE Invalid Caller

    $31 (Avg Bid)
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    I am a VoIP reseller. I am looking for a technical person, familiar with the Asterisk PBX and Linux platform, to manage a pbx in the cloud. I am thinking for FreePBX and I like RoseHosting for the VPS.

    $45 (Avg Bid)
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    15 bids

    ...I will be looking for the best design elements for the Admin Fitness Class Input Form, User Search Results & Filter layout, the Class/Host Detail Pages.. I have placed an asterisk next to those features I will be looking for. View the attached images for ideas and feel free to Message me for any questions! Key Functions for Admin - Host will

    $240 (Avg Bid)
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    0 entries

    We have blog articles/content that will need Infographics Design will be based on the content th...samples and good reviews, we will hire for multiple projects. Examples of infographic subjects include: 1. Internet Options (DSL, Cable, Fiber) 2. VoIP options (hosted versus SIP trunking) 3. SaaS solutions for SME's Thank you for checking in.

    $8 / hr (Avg Bid)
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    Hi Malay P., I would like to discuss a project with you, I can't find anyway this system won't create a whole job :| I'm looking for a SIP/RTP Proxy with Repacketization I have a system already in place which spits out 30-45ms packets, I need them 20ms or less when they head off to the provider. Can you let me know what you're thinking that would cost

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    ...Debian with postfix and Dovcot My epost clients is Microsoft outlook 2015 I work a lot with customers or elsewhere. I experience up that email my not work because SIP / network admin has sprret port 25 (SNMP) to that I must use their SMTP server to send mail. I've tried to open my server with 2 ports. 25 and 9800 it did not solve the problem

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    I have asterisk installed and I need the following: When the client calls the asterisk respond with "give your password" and then the asterisk read a table or a database (MySQL o SQL) and if the caller id (telephone number) and password are matching in a table read the status code from this table and respond to the client. If the status code is 0

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    Looking someone to build complete VoIP (Class4/5) Softswitch using Open Source platforms such as Asterisk, Open Sips, Free Switch etc with complete billing

    $407 (Avg Bid)
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    16 bids

    We are a VoIP company using both MiRTA and FreePBX (eventually just MiRTA) and we are looking for support when needed. We have lots of provisioning requests and some (but few) complicated troubleshooting requests that we would need help with. We would prefer someon that speaks English and works during US working hours.

    $35 / hr (Avg Bid)
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    11 bids