Sip tlsJobs

Looking for art instructor with personality, group training experience, experience with acrylic paint - water based paint and canvas. Must be able to work with beginners in a creative environment with a smile. Opportunity to present art portfolio in the artsy village of Nyack NY. Budget is open.

Golang TLS HTTP Client for sending outward HTTP requests.

Hello, Few call center employees need to work from home, so VPN need to be set up to connect them from home to head office network to make & receive phone calls. Input data: - We use PRI line with 100 numbers, out of each around 5 need to work remotely. - PBX is Asterisk-based; also Yealink, windows server 2012, Call center agents use X-lite old version as free SIP-phone tool. If needed, there is DyDNS address of the head office, as head office IP address is dynamic. Server & PABX map is attached. Time to finish project : 2 days + 1 day testing. There is no VPN existing in head office currently. Call center laptops at home for connection with VPN have Windows.

I need TLS implemented into a golang HTTP client used to sent HTTP requests outwards. It needs to be able to work with HTTP version 1.1 and 2, be able to use custom fingerprints (can be done inside the library uTLS) and have header order. The ideal, and most likely easiest way to implement it, is by using this library: in order to choose the fingerprints and such. However, if you have a better way, you can convince me. Below was an attempt (failed attempt) of my own to show the sort of thing I need to do. (Probably best you do not go off my attempt, as it doesn't work, but it is just to show that I need to be able to send HTTP requests with it). client := &{ Transport: &{ DialTLSContext: func(ctx ,

...firewall rules so as to only allow necesary openvpn server/client communication between my computer and cell and a site-hosting VM instance as well as a torrent client located on another VM. 3)Implement best security practices like creating new Admin UI user and passwd along with other things i cant think of 4)Obtaining CA and private and public certificates and obtain a signed good quality SSL with TLS for Admin UI Anything you know of that can assist with ensuring high availability across the US for the torrent client. Anything security- *)Any and all pertinent steps in establishing secure confirmed connections to the mentioned clients. I have another project that entails this skill set. If the job is

...firewall rules so as to only allow necesary openvpn server/client communication between my computer and cell and a site-hosting VM instance as well as a torrent client located on another VM. 3)Implement best security practices like creating new Admin UI user and passwd along with other things i cant think of 4)Obtaining CA and private and public certificates and obtain a signed good quality SSL with TLS for Admin UI Anything you know of that can assist with ensuring high availability across the US for the torrent client. Anything security- *)Any and all pertinent steps in establishing secure confirmed connections to the mentioned clients. I have another project that entails this skill set. If the job is

I need an iPhone app. I already have a de...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the database (firebase) 5- the connection details to the server ( IP/ username/ password ) and the cost per min are to be stored on a server ( firebase ) 6- cost per min is to be a variant that I can change later ...

Hello, the project is based in develope a configuration into a CISCO ASR1002 to customize a SIP/SS7 signiling switch. The SS7 signiling is received in TDM/E1 using a Channelized STM1 Interface connected directly to the Operator, inside the STM1 the first 8 E1's has the SS7 link & CIC's. The system will have to convert to an ITP / Point Code for SS7 signaling as a part of the final solution to be used as a SIP SBC or Sigtran point. Components: 1x CISCO ASR1002 1x Channelized STM-1 port (Cisco 1-Port Channelized OC-3/STM-1 Shared Port Adapter), the stm1 has 8 E1's. 2x SS7 Link (OPC/DCP/ADJ Point)

Develop in Delhi xe7 an email client. Is required the Pascal code for receive the emails in imap4, and to send with smtp. No code for user interface is required. The code must support tls 1.3 authentication. In Delphi Xe7 there are components tool as Indy 10, and IP*Works, but I suppose they don't math tls 1.3 Is required the code with the final procedure with example data to show how to call smtp and imap 4 filling the following fields: Username: .... Password: ..... Incoming Server: ......... IMAP Port: 993 POP3 Port: 995 Outgoing Server: ..... SMTP Port: 465 Budget until 100 euro. Thanks Gae

...certainly be a plus, but is not a requirement. The type of jobs that we'd be expecting the freelancer to perform are: 1. Listen through the un-edited version of each new episode and annotate time-stamps for editing (e.g. delete section from 1:39 to 1:52). The positions for cuts will be obvious, as it is usually sections where the podcast host and guest discuss logistics, take a break to have a sip of coffee etc). Each episode will be roughly 60 min long. 2. Annotate the timestamps where images would be added (we'll provide the images, so it would just be necessary to find the right timestamp when that image should be shown in the video) 3. Annotate timestamps for sections that are "highlights". Those will be cut together separately for a shortened summary/h...

El sistema debe estar en php/bootstrap la idea es poder ingresar usuario sip (voip) en una plataforma web y que automaticamente se creen en asterisk (estoy usando VitalPBX). Debe tener: Creacion de usuario Modificacion de usuarios Eliminacion de usuarios lista de usuarios activos y no activos eso para comenzar luego podemos ir añadiendo mas funcionalidades

Searching for the valuable software house who has exp on developing iOS and Android Mobile application on SIP protocols with the help of Admin panel and Must have experience on asterisk or elastix. Deep Project details will be provided over the PM.

The app is 2 pages (+ login page) Page o...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the database (firebase) 5- the connection details to the server ( IP/ username/ password ) and the cost per min are to be stored on a server ( firebase ) 6- cost per min is to be a variant that I can change later ...

I would like to get the SIP trunk configuration for goTo and Nextiva sip provider

I have a node js backend server and i want to run mqtt broker it on server with TLS layer Please read over job description carefully and bid Look forward to hear from you. Thank you

A list of apartments and their owners on a screen. By pressing a phone button, you will call that person. If the person wants to open the door, he presses *9 or something like that. All on a web page. There are JavaScript sip clients. The phone part is by asterisk installed on the same raspberry pie using google voice.

...page, whereby the sub-brands will sit under the motherbrand. The objectives are : 1/ to promote the brand awareness 2/ to increase market share & promote sales 3/ to have more engagement with people The new branding will focus on: - Having one singular brand to promote all the products - Tone of voice will be more human then informative, 1st person conversations - Setting a new positioning: A sip of Life [in french: Gouter a la Vie] What we imperatively need in the offer: - creative ideas for the year 2021, ex. an idea for Water Day / Environment Day etc. - improvements from the actual Facebook page [Vital Eau de Source] - a roadmap calendar, which can be referenced by all stakeholders - a good management tool which will suit our agency, the client as well as the community...

Need to Upgrade Freepbx from 13 to current and install: SIP Poe Mullion Video Keypad Intercom 8039 8063 SIP Interface Module (IP Relay controller) for Algo door control only. 8063

Необходим человек который оперативно сможет рассказать как и что настраивать в VOS3000. В первую очередь необходимо установить SIP Header Diversion для поставщика услуг

For a one-off assignment, we are looking for a confirmed Python develop...and technical developments of scrapers able to retrieve precise data on the web under Python - Storage of crawled data in csv files - Staff training on the use and maintenance of these tools in order to be able to periodically update the data and extend the scraping to new data Skills : - Language: Python (Mandatory) / Shell - HTTP protocol and related technologies (http version, Cookies, SSL / TLS, HTML, Javascript) - Environment: Windows Planned start date : As soon as possible. This project includes : 1. the full transmission of the code (with comments) and of intellectual property to us 2. the training of one person in our team in order that she can use and modify the code properly. 3. A confidentiality ...

we need an android expert to develop an app that when activated it will 1 to . rout the audio (speaker and Mic ) of a sip client to jack 3.5 headset 2. rout the audio (speaker and Mic ) of the cellular gsm calling to USB headset 3 . this audio routing need simultaneous ( in the same time you can hear and talk both conversation ) the budget is 250 dollars

white paper about the social issue investigated for SIP. The purpose of the SIP white paper is the following: (1) capture the main essence and challenges inherent in a prevailing social issue and (2) propose an actionable recommendation to address the challenges contained in (1).

I am looking for a developer to write a code for IOS/Android push notification for VitalPBX SIP services to work with Zoiper or Linphone Project Cost US$ 100.00

...to clean your Lists automatically Automatic Bounce Handling Unlimited A/B Split Testing Unlimited Auto Responders Unlimited Lists Unlimited Contacts No suspension for your list Input Email Template ------------------------------ BENEFITS: Send approximately 200k-2M mails monthly per IP Clean IPs Guaranteed on startup IP warm-up guides Exceptional Inboxing rate Send secure emails with standard SSL/TLS certificate for each SMTP servers Clean your list from hard bounces by Port25 bounce handler Upload your 1M list in few seconds into your Email server ----------------------------------------------------------------------------------------------------- Give us your offer and if your offer is good, you will get the job. And when we mean the job we do not mean only for the 1st server ...

Real Estate Company seeks positive, accountable, skilled, motivated, self-driven individual with experience in sales. Great voice on the SIP phone crystal clear microphone/equipment and high speed reliable connection for making calls and must love speaking to people on the phone, silent working environment for professional calls. Sales experience, preferably real estate related, but will train as necessary for knowledge and understanding of contracts preferred but not required. " Must be skilled and sound good on the phone with some understanding of the US residential real estate market " Our payout two types, you can choose anyone. A. Hourly Based Payout. B. Commission Based Payout. ( A. ) Payment terms per hour : As long as you stay connected to a call, but not for how...

I have Dell R630 5TR Raid-5, 64G RAM housing Ubuntu-16.4. it is integrated with my SIP Soft switch. We use the Datto backup service that allows us to install a Datto Agent on the Ubuntu servers and the Datto agent software performs hourly backups. Three other U16.4 servers back up just fine. According to Datto support, they have seen this before where during the FTP transfer from the Ubuntu machine to the Datto server, data becomes fragmented. They claim the fsck needs to be run on the partition. I have attempted to do just that, however, partitions need to be dismounted and few other preemptive steps need to be taken. We are a small business, Mitel phone system distributor hosting our own VOIP platform I don't have any Linux-trained techs available so I would rather pay a kno...

HI, I have a IP phone yealink - T21P-E2. I need to know its SIP credentials, mainly the PW. I have local access to the device and can give teamviewer or anydesk to retrive the info.

I need you to develop some software for me. I would like this software to be developed for Windows using .NET. I would like to make window program to get the delivery order and connect to delivery company with API Using the Winform and make sure Must smoothly printing out the receipt description from order section on a POS This is Ko...order section on a POS This is Korean word required and I can translated it for you. Once we get the order from client and make it make it sounds for notification we gonna provide the voice for that * we have a delivery app and panel but don't have a this product. So make sure need to integrate well with AWS server as well For the security things we need to cover by window xp so using the upper tls 1.0 Q what is your name A please answer...

communication for esp32 with gprs modem simcom sim800. we provide tls 1.2 support for sim800

we need an expert in Android to create a small script that will rout audio cell phone call to USB audio and in the same time to rout sip Client (softphone) audio to Head jack the the cell phone and the sip need to work in the same time our budget is 250 dollars

I would like a multi-class SIP tutorial, at first with the most basic concepts and increasing the level with each class, begining with a basic level and finishing with more advanced concepts, to learn and understand completely how working SIP networks. The minim total duration of the videocourse with all of classes is 30 minutes.

I have to connect an 3CX pbx to an T-System SIP-pure Trunk (sip-trunk based with 3-way authentification). I want to use kamailio as sbc - but i have no experience with kamailio sbc-config. I need someone to write an to archive the following: - register to T-System Trunk (3-way auth) - connect to 3CX (sip-trunk) - rtp engine to convert g711 - t.38 and vice versa - 3cx should have the option to use re-invite and replaces - T-System doesn't support them directly wfr Michael

We need a bulk mail software, it must be fully configurable.. - Mails per time unit - Mail per account - Configure several accounts for sending - Authetntication with TLS, SSL, etc -Stadistics

Simulate SIP vulnerabilities that include DDoS, DoS, password guessing, Man in the Middle attack, and SQL injection using Omnet++ in VoIP architecture. Take a video of each step of the process. The video should have no voice but should be clear in terms of pictures. Also, include how to mitigate each attack mentioned above in the simulation and capture it in the video.

i need the numbers in twilio to be sip accounts and to create an ivr for incoming calls.

...10:46AM. If this endpoint is important to your application, please try and fix the issue. If you do not need the webhook endpoint you can delete it from your Stripe webhook settings. We will stop sending notifications to your webhook by January 14, 2021 around 10:46AM. Here is the summary of errors we have received when sending you webhooks: * 167 requests had a TLS error, you can get a detailed analysis on your host's TLS configuration at: :443&hideResults=on. You need to return anything between a HTTP 200 to 299 for the webhook to be considered successfully delivered. For more details on the errors, you can find your events page here: You can also always review the history of webhooks and

Hello, I have installed an Openvox UC 501 Asterisk Based IPPBX at my home. I have a BSNL FTTH Connection with a Landline Number. The landline is currently working through a POTS port on my Syrotech ONT Modem. I want the Landline (VoIP) to work as an SIP trunk on my IPPBX. I live in Raipur, Chhattisgarh, India. Thanks.

Dear FreeLancer, i have fresh installed 3CX, i need to configure the Sip trunk IP Authentications based, outbound/inbound calls.

Hello sir how are you . i want high skill VPN Application Developer For Free Internet . for android app . i will provide you admin panel and website . This is here protocol . all protocol working automatic . proxy, DNS, HTTP2, SSH, SSL, TLS, wireguard, webrtc, icmp, ikev2, sstp, ipsec Note : Please watch this video fast and check this video

I have a FreePBX instance and need assistance cleaning up the configuration, tightening security, and troubleshooting SIP connections. Please do not bid on this project unless you have in-depth FreePBX experience.

Ich würde gerne eine App erstellen lassen welche also SIP Account dient - also mit der man telefonieren kann. Ziel soll es sein dass die App sich über einen QR Code selber konfiguriert mit den notwendigen SIP Einstellungen.

...roller shutter) I will build 2 doors, one is a single door to a narrow corridor in a section of the garage that will go forward shortly and then turn left into the SIP building which is adjoined to the garage. The remaining gap in the garage opening will be large workshop doors to give access to the garage. The large workshop doors will have aviation themed artwork on them. The corridor door will have company name/aviation themed images too. The inside of the corridor will be mocked up to look like a jetway bridge like: At the end of the corridor left turn will bring you into the SIP building which will contain a small aircraft cabin section similar to leading to a cockpit like https://images.app.goo.

Dear Freelancer, i would like to configure my FreePBX with Sip lines IP based Authentications,

We are looking for a simple E-Mail System for Linux & Windows. Multiple Client connect to web interface ( PHP ). Pick the job id. Message ID and just send it. Just like MailWizz. Since with PHP you can set the email as string. The client only need to connect to server and use EHLO, MAIL FROM,RCPT TO,DATA. Also it will be able to connect via tls.... Language: PHP, C/C++ Windows Visual Studio. Linux via Gcc. Please contact me for more information about this project. Thank's

...(its a server directory not a directly addressable wordpress page, therefore the plugin doesnt see it, could just put an nginx pass on it but prefer a more elegant solution as long as its quick) * integrate woocommerce with a printable site such as cafepress ? (optional) * site is a properly made child theme of twentyseventeen. * configuration of etc/nginx/sites-available config to handle current TLS standards and to redirect to https all the time. I am a sysadmin myself im just probably not as good as you and i havent got the time :) for some reason ive managed to redirect everything to and it wipes off the www everytime. i am a music producer i sell large to reasonable sized wav files. might sell some t-shirts or mugs off some print site, but i want it to interact with

Fazer configuração de um tronco SIP TLS no asterisk

Linhome is Belledonne's latest open source SIP SDK esspecially developed for use with SIP video doorintercom systems deployed in multiple dwelling units. Info about obtaining opensource SDK/code and required coding skills your can read here We want some minor changes in the current Linhome APP that Belledonne have published themselves in Appstore and Playstore - search for Linhome: 1) Replace Linhome logo/icon with our preferred logo/icon 2) change color scheme from orange now towards blue 3) include language support xml for automated usage of English/French/German/Dutch/Marrocan/Turkish based on the phones prime language The APP will be compiled and delivered to me running in both Ios as well as Android. We will publish the supplied

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Sip proxy sends the call to kamailio, Kamailio route the call to dispatcher1. As soon as kamailio receive second sip message (after "100 trying) from dispacher1 member sip proxy, If sip message equals to 180 or 183 from dispatcher1, Kamailio will reroute the same call to dispatcher2 Only kamailio configuration support needed. All other dispatcher or sip proxy configurations have already done by our side.

Our company has been operating for 30 years, we have had 2 logo changes in that time and are looking for a fresh new look. Planet Telecommunications core business is Telephone systems, install, service and we are also a Business grade SIP and internet service provider. Our focus for future is based on cloud based communications and carriage. You will find more detail of our business from our website including the current logo. We are looking for a fresh new leading edge logo to take us forward for the next decade, we would like the logo to be be PlanetComms instead of Planet Telecommunciations. Planet Comms can be spaced and capitilised anyway you wish. Our current colours are on our website. Looking forward to seeing some ideas.