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    9,920 sip tls srtp windows jobber funnet, priser i USD

    i need someone that can login to my computer and setup my fusion PBX should accept calls from vitelity SIP Provider, and also to make calls thru my outgoing provider.

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    Robocall server 6 dager left
    VERIFISERT

    A center of family planning is looking for a Robocall solution to remind patient to visit doctors, etc. Informa...reminder them the date of the visit. at the end of each day, a full report should be given. The voices will be prerecorded by our team here. We will use goip gsm gateway (sip) for the calling. We have servers to host the application.

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    ...a PHP class or classes to implement the connection and execution to a set of WSDL services. The finalised project: - Has valid PHP 7+ code - Connects to the service using TLS 1.2 - Authenticates using a digital certificate (X.509 Certificate Token Profile 1.1) using the WS-Security 1.0 Oasis standard - Uses SOAP 1.2 - Can be integrated into Laravel

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    Taushetspliktsavtale
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    ...number in excel/CSV/mysql or any other format. 5. Dial #1 – simple dial surface, numbers, #, * etc. all via PC/laptop windows 10. NO landline or mobile phone for dialing only the PC and PC microphone + PC headset. Each user has own SIP account and should use the program separately. 6. Dial #2 – open phone book. Select a city/name/phone number from phonebook

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    I need a simple softphone that allow to SIP register on a freepbx server. With just the dialer, not accessing phone contacts and all the features that most softphones comes. It must run in the background when closed so it can receive calls. Guess is just that, hope your offers! Thanks

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    Necessito de configurar um servidor OpenVPN com c...mim. Até ao momento já tenho um servidor OCSP a funcionar e o servidor OpenVPN a funcionar parcialmente. Necessito que o certificado seja validado no OCSP, atualmente utilizo tls-verify [logg inn for å se URL], mas estou com erros nessa validação. Preciso de ajuda para terminar o mesmo. Quem poder por favo...

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    Goautodial specific functions 3 dager left
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    ...any number in any list: show a different calling number on customer phone for each attempt. Will give you 10 calling numbers that we have 3) Set up load balancing on 3 trunks (sip provider). When dialing on lists, send one call to one trunk, next to the second trunk and next to third trunk and so on... 4) Failover : after dialing a number on a trunk, if

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    ...to launch our new product to the world Naturally flavoured Coffee beans. Imagine its early morning your somewhere between being asleep and being awake and you have that first sip of the freshest, best quality, real Arabica coffee with a hint of rich Caribbean vanilla rum and tropical spices or smooth almond marzipan. Ahhh absolute haven, aromatherapy

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    Garantert Forseglet Topp Konkurranse
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    Digital logo 3 dager left

    I'd like to create a logo for my website and work as an author and speaker on personal development. My initial idea is a play on my initials (SIP) and I've attached a sketch of versions that I have used. Would like someone to play around with that and come up with something professional/digital.

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    We need someone to configure our emails

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    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.

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    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

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    ...kind of logs. [2019-02-14 08:56:15] WARNING[7089] chan_sip.c: Autodestruct on dialog '223205511-3759123182-18707@TcomNextoneSBC4-DRC.mydomain.com-onnet_1-b2b_1' with owner SIP/telsam_ip-00003486 in place (Method: BYE). Rescheduling destruction for 10000 ms. if you interested in I can share you more information. and budget is negotiable fixed or hourly

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    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.

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    ...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

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    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

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    WebRTC / SIP expert 1 dag left
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    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. [logg inn for å se URL] So that they can make a 1 way audio call which is having issues with local and remote media streams to wo...

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

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    Help with TLS Traffic 1 dag left
    VERIFISERT

    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.

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    ...SSL, Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future: We need to have a setup, where we are more

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    Looking for freelancer to configure FusionPBX with PowerPBX hosting. [logg inn for å se URL], [logg inn for å se 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, interconnect

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    Trophy icon Logo design for "Livingston Depot" Avsluttet 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

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    Garantert
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    Asterisk Expert Avsluttet left

    ... Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future: We need to have a setup, where we

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    Help setting up initial OTRS community version on Centos 7 (VPS), configure outgoing SMTP-TLS mail. Linux Sysadmin, Perl, and previous OTRS experience required. ([logg inn for å se URL] )

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    ...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. [logg inn for å se URL] [logg inn for å se URL] The 3cx gateway for skype is running only windows 7. and needed skype old version.

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    currently got tls 1.1 and need to upgrade to tls 1.2 in websphere server

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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

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

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    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.

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    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

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    Have you tried using encrypted SMTP on port 465? I bet they won't detect that. [logg inn for å se URL]

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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

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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...

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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

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    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.. [logg inn for å se URL] New features need to add 1. SIP Calculator 2. Update Emi Calculator for variable interest rates

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    FTP TLS project

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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.

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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

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    convert cisco ip phone to sip for working in yeastar s100

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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.

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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

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    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.

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    SIP CLIENT FOR RASPBERRY PI3 this software to be developed for Linux using Python. 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 the code, over GPIO inputs we

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    Looking for someone knowledgeable on freepbX and mobile application development to create a VOIP application for customers service . The reference app is named as "smart line" You can check the app's features . I need the exactly same features with some new designs .

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    Forseglet
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    I am using the ESP8266 to connect to a HTTPS service (AWS). I want to replace the SSL functions with the esp TLS (esp-tls) from the SDK. Thw whole code is based on the ESP nonOS SDK.

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    looking for someone with good zimbra knowledge, my zimbra server is only allowing internal relaying and when relaying from external it gives: Anonymous TLS connection established from instead of the username to authenticate

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    Early decay in babies teeth is a big problem and is called early childhood caries...Brush the child's teeth morning and night 4) Bacteria from your mouth can be transmitted to your baby's mouth so don't share spoons etc 5) After a feed clear the mouth with a sip of water 6) don't feed after that 3) don't let the baby breast feed all night or on demand

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    Set up SylkServer Avsluttet left

    Set up sylkserver to 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.

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    Add the ability to enter sip trunk credentials to asterisk

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    I am looking for a senior architect/developer to design and develop gsm telecom solutions based on ss7 or diameter or sip communication protocols. The scope will be discussed later, the pay will be very good

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