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    18,369 asterisk sip tls srtp jobb har hittats, med prissättning USD

    We have setup bitrix24 onpremise and we have asterisk-vici installed. We want to use SIP connector. Any one with experience in setting up onpremise Bitrix24 and telephony with SIP connector can respond.

    $654 (Avg Bid)
    $654 Snittbud
    1 bud

    Hi All I have a SIP client on my network, its a Intercom. It registers to my FreePBX/Asterisk solution and makes calls to my mobile phone fine. I want to intergrate Skype because Amazon Echo's can answer/make calls using Skype. The intention is to receive the Intercom call on my home network to an Echo Dot, which I believe will work when the FreePBX

    $21 - $177
    $21 - $177
    0 bud

    Hi Kingasterisk T., We have a problem with our asterisk configuration. as a brief when call directed to extension queue after ivr. there is silence and no result. caller cant reach to extension. as example we get this kind of logs. [2019-02-14 08:56:15] WARNING[7089] chan_sip.c: Autodestruct on dialog '223205511-3759123182-18707@TcomNextoneSBC4-DRC

    $100 (Avg Bid)
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    1 bud

    I'm new to FusionPBX/FreeSwitch. Can't set up the same thing I've set up in minutes in Asterisk. Need remote assistance/general consulting or just a setup example in accordance with my infrastructure conditions. My FusionPBX is a VM that has only one LAN NIC. It uses pfSense VM as a gateway. pfSense VM has both LAN and WAN links with a public IP on

    $27 / hr (Avg Bid)
    $27 / hr Snittbud
    7 bud

    I Need help with a vicidial server that is not leaving vvoicemail message and is dropping calls in a press 1 campaign

    $208 (Avg Bid)
    $208 Snittbud
    6 bud

    We need an expert in OpenSIPS that can change the SIP response of 200 ok to something else or delay it full details will be given later.

    $328 (Avg Bid)
    $328 Snittbud
    4 bud

    We have developed a voice bot that understands vernacular languages of India. We are looking for a developer who can help us integrate a user call with our voice b...asks a question - e.g. what are the current offers? The bot responds: Buy goods worth Rs 5000 and get cash back worth Rs 1000. Please get in touch immediately if you know Asterisk well.

    $2779 (Avg Bid)
    $2779 Snittbud
    7 bud

    We have developed a voice bot that understands vernacular languages of India. We are looking for a developer who can help us integrate a user call with our voice b...asks a question - e.g. what are the current offers? The bot responds: Buy goods worth Rs 5000 and get cash back worth Rs 1000. Please get in touch immediately if you know Asterisk well.

    $2892 (Avg Bid)
    $2892 Snittbud
    6 bud

    ...allow only Checked In guests to sign in using surname and Room number. once signed in also allow guest to connect to the Hotel WIFI also within the APP we will need a build SIP/VOIP on the App which will interface with hotel PABX and the app will become the guest extension as long as his signed in and for the duration of his stay. This app will need

    $2471 (Avg Bid)
    $2471 Snittbud
    136 bud

    Authenticated Origin Pulls Authenticated Origin Pulls allow you to cryptographically verify that requests to your origin server have come from Cloudflare using a TLS client certificate. This prevents clients from sending requests directly to your origin, bypassing security measures provided by Cloudflare, such as IP and Web Application Firewalls, logging

    $22 (Avg Bid)
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    2 bud
    AMD detection in Asterisk 5 dagar left
    VERIFIERAD

    I want someone expert in asterisk who can fix my AMD detection in Asterisk, I'm using Vicibox and running Press 1 Survey Campaign.

    $54 / hr (Avg Bid)
    $54 / hr Snittbud
    7 bud
    WebRTC / SIP expert 5 dagar left
    VERIFIERAD

    The client is using astrix server. They will give us a vpn connection to their network And we need to setup WebRTC client to talk to astrix server. We need to use open source functionality working which it can't currently get a function to work within a type. [Logga in för att visa URL] So that they can make a 1 way audio call which is having issues with local and remote media streams to...

    $21 / hr (Avg Bid)
    $21 / hr Snittbud
    23 bud

    Want to start a VOIP/calling card solution ,need to install the necessary software (asterix,freepbx,a2billing etc).Advice the best solution

    $293 (Avg Bid)
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    13 bud

    im looking for someone to make landing page simple nice looking landing page have form and need image

    $29 / hr (Avg Bid)
    $29 / hr Snittbud
    6 bud
    Help with TLS Traffic 4 dagar left
    VERIFIERAD

    Ive got a sever that sends tls traffic to a ip address every time it starts up. I need help understanding the traffic being sent and received. the TLS certificate files are on the server. Once the traffic is understood ill be looking to develop a protocol to replace the remote ip and answer the messages locally on the server.

    $2027 (Avg Bid)
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    12 bud

    Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support

    $13 / hr (Avg Bid)
    $13 / hr Snittbud
    8 bud

    Looking for freelancer to configure FusionPBX with PowerPBX hosting. [Logga in för att visa URL], [Logga in för att visa URL], Debian ...Plans, PBX Extensions, I/O Trunks, Voicemail, DISA, IVR & local number display on the go, Call Forwarding, Call waiting, Linphone setup and instructions, call recording, setup sip trunks. for 4 - 6 lines. PHP or LUA scripting, VOIP, interconn...

    $462 (Avg Bid)
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    We have a call centre system using Asterisk. With Asterisk only supporting the following call distribution. We need to tweak the queue behaviour to reset agent idle time in two event, after agent has hung up an incoming call and after agent end pause. If you base on the ring stretagy of least recent, we can achieve the agent idle time reset on the incoming

    $455 (Avg Bid)
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    Trophy icon Logo design for "Livingston Depot" 2 dagar left

    "Livingston Depot" is the Name we need a logo for.. - "Sip & Shop Station" is the Tag Line This is a boutique cafe that is located in Livingston, Tennessee. This shop offers local artists, hand crafted items for sale, with a small cafe offering beverages, and lunch items. The keyword inspirations are elegance, down home, boutique, station, depot

    $40 (Avg Bid)
    Garanterad
    $40
    93 bidrag
    Asterisk Expert 2 dagar left

    Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support

    $17 / hr (Avg Bid)
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    5 bud

    Help setting up initial OTRS community version on Centos 7 (VPS), configure outgoing SMTP-TLS mail. Linux Sysadmin, Perl, and previous OTRS experience required. ([Logga in för att visa URL] )

    $66 (Avg Bid)
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    1 bud

    I need a call with multi skype user via SIP. and i saw 3cx gateway for skype program. this product is no longer available for download nor maintained. [Logga in för att visa URL] [Logga in för att visa URL] The 3cx gateway for skype is running only windows 7. and needed skype old version

    $29 (Avg Bid)
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    hello i need to stream live calls from my asterisk more or so like chansky but i need to stream in demand and to wav files so i can play it in brower i am useing freepbx

    $464 (Avg Bid)
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    currently got tls 1.1 and need to upgrade to tls 1.2 in websphere server

    $28 (Avg Bid)
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    must be connect multi Skype user to a SIP-PBX. and call via skype to pstn or mobile number. multi skype user can running on asterisk machine. extensions are calling via each skype-account

    $30 (Avg Bid)
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    1 bud

    We are looking for an Asterisk expert who has experience installing Kamailio or similar software to be used as an outbound proxy for NAT Traversal front ending multiple Asterisk/FreePBX servers. Some PBXs are having load spikes due to frequent device registration/authentication and NAT traversal functionality, so we're looking to install a solution

    $557 (Avg Bid)
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    9 bud

    We need an implementation of the services mentioned in the subject. Thanks.

    $1495 (Avg Bid)
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    7 bud
    ejabberd configuration 8 timmar left
    VERIFIERAD

    I need help you configure my ejabberd server with sylkserver for translate jingle/sip between xmpp client and SIP proxy and/or PBX. Goal is that XMPP client can make a call through the PBX to the outside work (as a regular UA) and that the XMPP client can be called via the PBX.

    $22 / hr (Avg Bid)
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    2 bud

    Add ADI input to Asterisk frontend. Thank you!

    $153 (Avg Bid)
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    3 bud

    We have a multi-tenant PBX system and need a simple SIP phone provisioning tool written in Javascript Node.JS, using Angular or React, running on a MEAN/MERN stack. This will not be integrated with the PBX, however our B/OSS system will add, suspend, and delete tenants and extensions to this new system you will create via a REST API. A commercial

    $198 (Avg Bid)
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    5 bud

    Need to develop end to end solution of call center.

    $1643 (Avg Bid)
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    14 bud
    freepbx FOP2 Avslutades left

    i need help fop2 its installed on a standalone (separate) server than Asterisk Freepbx i can get the recordings to work.

    $155 (Avg Bid)
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    1 bud
    freepbx fop2 Avslutades left

    i need help fop2 its installed on a standalone (separate) server than Asterisk Freepbx i can get the recordings to work,

    $43 (Avg Bid)
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    3 bud

    Have you tried using encrypted SMTP on port 465? I bet they won't detect that. [Logga in för att visa URL]

    $250 (Avg Bid)
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    Setup a VoIP server for termination the requirements are: 1. SIP connections to gsm gateways through VPN is required 2. Billing software 3. Load balance traffic to gateways 4. setup connections with 3 clients

    $238 (Avg Bid)
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    ...Financial Planning ;Invest online;Testimonials ;Blogs; Careers ;Contact us With A live Chat options & watsapp messenger Few Algorithms need to be used –like calculator for – sip;retirement;marriage;child education etc...

    $178 (Avg Bid)
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    I am looking for an experienced FreeSwitch engineer who can work with latest freeswitch version with SIP Trunks to do following tasks I will be passing destination number , CLI , ring duration , max call duration , audio file , unique call id to Freeswitch (JSON API) , the dial plan should start dialing numbers and plays a voice file when the call

    $1272 (Avg Bid)
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    5 bud

    I have already built app on play store called "Loan Emi Calculator". Please review the app befo...Calculator". Please review the app before you bid, here is a store link.. [Logga in för att visa URL] New features need to add 1. SIP Calculator 2. Update Emi Calculator for variable interest rates

    $158 (Avg Bid)
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    33 bud

    I want someone expert in ASTERISK and freeswitch and ios to send calls over bluetooth but to use whatsapp or viber instead of gsm dialing to be smart enough to tell asterisk use whatsapp when dialing (send calls direct to whatsapp) android or iphone whatever they can fix

    $2826 (Avg Bid)
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    21 bud
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    I have an existing Shoretel system with a PRI switch. Using the SETU VTEP device I want to replace my existing PRIs with SIP trunking by using the SETU VTEP as a gateway. I need help programming the device to integrate with my existing Shoretel system.

    $200 (Avg Bid)
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    1 bud

    Trabalho há mais de 25 anos em telecomunicações, estou procurando desenvolver um sistema de interfonia baseado em asterisk com acesso através de app (soft phone)

    $442 (Avg Bid)
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    Current software versions: FreePBX: [Logga in för att visa URL] Asterisk: 13.11.2 Project Description : . After a call, our Call Center Agents will transfer a customer call to an extension where the survey should be executed. Survey will consist of 2 fixed questions. We need to be able to define the Survey number and Name, then record the questions and select valid

    $573 (Avg Bid)
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    1) Set up Asterisk. You will get a machine with Linux OS of your choice. 2) Install UniMRCP and UniMRCP Kaldi Plugin. Configure SIP line for inbound calls. We will provide the URI to configure in the plugin. 3) Configure an Inbound call plan that does the following for inbound calls: (a) Prompt the caller to punch in a 4 digit code. (b) Post

    $318 (Avg Bid)
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    Hi, The goal is to establish an asterisk server and freepbx for the purpose of voip termination in GSM network. The freelancer has to do the following: - Install asterisk server and configure it to operate with Huawei usb dongles (modems) having prepaid simcards to connect to the GSM network. - Be able to attach more huawei usb modems to the server

    $551 (Avg Bid)
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    convert cisco ip phone to sip for working in yeastar s100

    $125 (Avg Bid)
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    Need to Configure 2TATA Sip Trunk line on my freepbx via remote desktop like TeamViewer. FreePBX hardware is assembled and installed freepbx.

    $18 / hr (Avg Bid)
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    Sip client for Raspberry pi3 We need to make sip client for raspberry pi3 on parking system machine, Software must have source code in python and auto start with raspberry linux SIP client must call just one number that we insert in code, over GPIO inputs trigger sip client start call. (Gpio inputs must configurable in code) Sip client must have

    $245 (Avg Bid)
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    Hello, We are looking for someone to deploy a FusionPBX (Freeswitch) for us from scratch we even need everything to be migrated to the new PBX from our Asterisk.

    $206 (Avg Bid)
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    4 bud

    Hi, I'm looking for someone to connect a twilio number that we currently have to a sip endpoint thats already configured for voice. We are using the X-lite Voip client and would like to get SMS functionality enabled on it.

    $39 (Avg Bid)
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    5 bud