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    15,441 linux sip asterisk jobs found, pricing in PHP
    Asterisk Expert-FreePBX 6 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.

    ₱20454 (Avg Bid)
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    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

    ₱4424 (Avg Bid)
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    7 bids
    Modify JAVA file in App -- 2 6 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

    ₱5701 (Avg Bid)
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    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

    ₱13680 (Avg Bid)
    Local
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    3 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.

    ₱3861 (Avg Bid)
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    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.

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    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

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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.

    ₱2523 (Avg Bid)
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    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.

    ₱412 / 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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    I have my own epost server hosted on linux 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

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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

    ₱6745 (Avg Bid)
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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

    ₱20385 (Avg Bid)
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    14 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.

    ₱1853 / hr (Avg Bid)
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    ...on a monthly retainer basis. I already have the server running and your first job would be to secure it / put in place everything required to prevent BRUTEFORCE attack of my SIP, SSH and other usual attacks on those type of servers. Once this is done, I will pay you monthly to do maintenance on my server and from time to time a few adjustments.

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    Edit my pdf 1 day left
    VERIFIED

    ...“American” colours? This means use the stars and stripes if you want to say the “US” like “USa”. - Maybe it´s necessary to explain the 350.000 english spoken people (with an asterisk); I think you mean the englisch spoken ones who are not German citizens. But does this know the one who is the recipient of the document? - I would change “W...

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    I need to install and setup asterisk to make calls via zoiper using voxbeam Centos or any softphone, with ability to change caller id via editing .conf file

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    I'm looking for someone to build a call center in Twilio as well as SIP endpoint configuration in [url removed, login to view]

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    Needed: Experienced PHP programmer for VoIP needed, knowledgeable in telecom programming particularly GoIP sim bank gateways, GoIP API programming, Asterisk A2Billing and FreePBX, and Cisco. Description: Create an API to control the actions of a GoIP sim bank gateway by Call Progress Tones. GoIP sim gateway is located in a third world

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    Hola, Buscamos un tecnico de 3CX para solucionar disintos problemas con las configuraciones. Es importante que esté en España para el horario laboral. Gracias,

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