Sip server tls srtp jobs

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    9,825 sip server tls srtp jobs found, pricing in NZD
    IVR Product 6 days left

    ...Maintain and manage Genesys Routing, Framework and reporting. Responsible for supporting call center routing strategies and have experience with Genesys reporting, URS routing, SIP and GVP technology. Good understanding of Genesys (CTI) infrastructure. Strong working knowledge of IRD and CME. A working knowledge of Genesys Systems Architecture to create

    $11 / hr (Avg Bid)
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    Need install asterisk and sip server on virtualbox for receive calls directly to computer and if 1st line is busy automatically forward to second free line with call recording and without monthly fee

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    Asterisk Queue Setup 6 days left
    VERIFIED

    I have a production asterisk installation running on my server. I have a requirement. I want to setup a queue such that Agents and end users can use queue using their mobile phones. Lets Say, their are 3 agents Agent 1: Mobile : +91-XXXXXXXXX1 Agent 2: Mobile : +91-XXXXXXXXX2 Agent 3: Mobile : +91-XXXXXXXXX3 Lets there are 5 users who will dial

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    I looking Expert in SSL/TLS Digital Certificate /PKI API to define PKI API specs (define payload per certificate category). Regards,

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    We are using the Asterisk PBX With a Linphone SIP client in a Linux environment operating on the Olimex A20 and PINE64 and are experiencing very high echo. We understand Linphone uses the Speex Echo Canceller. We either do not know how to adjust the echo canceller or we need to substitute it for another excellent echo canceller. We would like someone

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    We need to migrate current email server from Yandex to Exabytes and set up TLS secure email communication to our external email counterpart. 4-5 email user setup. 1 month support.

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    I am looking for someone who can customise the UI on Linphone for iOS, Android, Windows and Mac in that order. We will be providing all of the re...who can customise the UI on Linphone for iOS, Android, Windows and Mac in that order. We will be providing all of the required graphics and we also have an internet facing SIP server it will register to.

    $20 / hr (Avg Bid)
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    Hi altr, I'm looking for someone to help me r...using freeswitch for wholesale scenario, When i receive 181 call is being forward sip message from supplier, it is being absorbed by freeswitch and not sent to Leg A. I don't want to spend time to figure it out, i just need you to tell me how i can make sure this SIP message is forwarded to the customer

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    Java developer for webserver 3 days left
    VERIFIED

    Looking for java server expert having experience in tcp/ip with tls which can hold socket for 60 seconds and can hold 10,000 concurrent connection and need to use pure java without third party library

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    Upgrading Cisco 7945G IP Phone to SIP Firmware

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    ...VueJS, Bootstrap 4) 2. Back-end: (Python, Django, Django Rest Framework, API Development) 3. Database: (PostgreSQL) 4. Deployment: (Ubuntu 16.04 (Xenial), Let’s Encrypt (TLS), Docker)...

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    Need SSH and TLS to work on a server. After switching to EVP functions, some ciphers and macs aren't working properly for SSH and TLS like they did with lower level OpenSSL function calls. The work would entail debugging existing code and fixing. I am a developer, but need help. Should only take a few hours.

    $61 / hr (Avg Bid)
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    Need SSH and TLS to work on a server. After switching to EVP functions, some ciphers and macs aren't working properly for SSH and TLS like they did with lower level OpenSSL function calls. The work would entail debugging existing code and fixing. I am a developer, but need help. Should only take a few hours.

    $25 / hr (Avg Bid)
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    I am providing vas on IVR by using java platform, so I need a java coder who as I need to get some changes done in coding , add new product and also create a SIP for OBD

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    ...script will be provided but you will have to update it with notes. - Qualifications: I need someone that speaks great English, has organization skills, experience with VOIP (SIP) or has a similar program already so that we can check. Duties: Negotiations with the top officials of the companies; Sale of services in the field of B2B; Requirements:

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    Hi , I am looking for someone who is able to configure ZRTP...like to know if all features work with this protocol? I read some articles said that call recording is not possible with ZRTP. Does it work with all other protocols such as, TLS, UDP and TCP? Let me know if you are interested in this kind of work and let`s discuss the duration and price.

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    ...outbound calls IP/SIP add provider for inbound calls IP/SIP Dialer where can select phone book survey wait time provider caller id for campaigns and run DID NUMBERS UPLOAD FOR INCOMING CALLS ROUTE THEN WITH IVR /CALL CENTER IP SIP/ REROUTE OPTION Also Can create SIP account for receive incoming calls agents so when did routed to sip agent calls

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    ...outbound calls IP/SIP add provider for inbound calls IP/SIP Dialer where can select phone book survey wait time provider caller id for campaigns and run DID NUMBERS UPLOAD FOR INCOMING CALLS ROUTE THEN WITH IVR /CALL CENTER IP SIP/ REROUTE OPTION Also Can create SIP account for receive incoming calls agents so when did routed to sip agent calls

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    Hello, We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through whatsapp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows/Android, or by using the

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    Hi Tran H., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I saw you were awarded to this project "SIP to Whatsapp gateway" did you finish it?

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    Want to build a google data studio dashboard displaying phone extension statuses, sip provider statuses, and outages for digium phone system.

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    Install VPN and setup a server connection Install OpenSIPS with graphical user interface install codecs connect GSM gateways to server setup OpenSIPS billing module install fail2ban and configure configure sip connections with clients and perform test calls

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    ...back end provider (SIP or otherwise) and server details.(ie. Centos with WHM and cpanel running Asterisk) I already have some hosting options in mind and I prefer Centos with WHM and cpanel, running various services to accomodate the VOIP server and the website. Basically, we need to figure out what voip server and back end sip or trunk providers

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    Integrate Issabel-Asterisk with Twillio Elastic Sip Trunking to records Calls from Twillio (through Flex)

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    Skype Connect has the SIP trunk feature to use Skype as a SIP trunk of PBX. I tried to configure Skype connect with FreePBX but couldn`t make it work. If someone can do that and already configured such setup, I`m ready to pay.

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    ...FreePbx installed from iso on a bluehost server. I can register my sip phone, I can make inbound and outbound calls. The only problem is there is one-way audio on the call. If I call from my Windows Sip phone to my cell phone -- My cell can hear me, but I cannot hear the cell. If the cell phone calls my Windows Sip phone -- The cell can hear me, but

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    I am looking for provider who provide me sip gateway for Indian operator but with open caller id and API. My concern is to send broadcast pre recorded voice.

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    ...CLEAR ABOUT THE REQUIREMENTS: 1. There is a fully created backend api that will send JSON Response calls through a TLS encrypted backend supporting openssl. You will be tasked with creating an SSL connection at domain to create secure TLS connection to app using SSL Pinning. 2. The API will route all API calls through the TOR Onion Routing Service Next-Gen

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    ...CLEAR ABOUT THE REQUIREMENTS: 1. There is a fully created backend api that will send JSON Response calls through a TLS encrypted backend supporting openssl. You will be tasked with creating an SSL connection at domain to create secure TLS connection to app using SSL Pinning. 2. The API will route all API calls through the TOR Onion Routing Service Next-Gen

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    Kamailio sip proxy server need with custom functions

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    ...CLEAR ABOUT THE REQUIREMENTS: 1. There is a fully created backend api that will send JSON Response calls through a TLS encrypted backend supporting openssl. You will be tasked with creating an SSL connection at domain to create secure TLS connection to app using SSL Pinning. 2. The API will route all API calls through the TOR Onion Routing Service Next-Gen

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    I need an application that runs on Android phones (4.0 or above) and can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any requirements

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    Asterisk PBX application to perform the following. To setup a SIP call between two servers using a specific codec. Have one side playback audio files. These servers will be connected over a WANem emulated link to induce packet loss/jitter/delay etc and produce degraded audio. Have the other record the degraded audio and store it .

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    Build a VoIP dialer app for Android/IOS and WM using either Linphone or Csi...Account auto provision/Signup via REST. - Account Balance Check Via Rest. - Account Topup via rest.(In App Payments) - Multiple Sip registrar options (Selectable) - User Credit display in app. - Voice/Video/Chat support - *** SIP Header Compression support! within the app.

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    ...IP (means the speed of scanning increases by n number of threads per IP) - added custom port selection feature - option to select between multiple scan types (Regular, SSL or TLS also with default or custom ports) - added auto scan with configuration file for better management. - added tokens - added other smtp codes which were reported missing - now

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    .../ linux server administrator to help at anytime - Have a virus/malware scan on entire server - Can you increase the number of connections to mysql ? what is the current number ? - Do you have any « General Data Protection Regulation » skills ? if yes can you send us a report regarding security ? - Check why we still can’t connect using TLS to FTP -

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    ...Modify HTML to include "Points" to the number of points being awarded to the question. - Write a script that will install CTFd on Ubuntu using the fastest DB, web server, load balancer and TLS certs. CTFd recommends some of the technologies to use and which ones are the fastest. You will need to let me know which ones you will use and why. - Deliver an

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    Dear all, We have a urgent Vicidial system to se...information and agent disposition data into one outlook. That means initial upload of list information must converge with output of agent disposition to create one report 6) To create sip trunk for testing 7) To create sample lead list 8) Testing and commissioning 9) Agent and supervisor training

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    Its an cafe i want a logo for it

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    ...suggestions on security implementations, please feel free to let us know. For this project, we need 3 things to be completed - 1. Help generate & install a Commercially Signed TLS Certificate on all 3 of our servers (Dev, Stage, Production). 2. Setup App & API version control (ex: v1 of the app's API would be located /api/v1 and v2 of the app's api would

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    SIP dialer for Android and IOS. Should support most Android and IOS versions including IOS version 12. Features: Runs behind NAT Support Bluetooth Echo Cancellation Push notification Call history display Account balance display Address book integration Support UDP and TCP protocols Support users to Login or Signup Supported languages: English, Arabic

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    ...with other features) to successfully deploy phone identity certificates to establish the identity of a client or server during a TLS session. Phone will also need to be added to Windows Active Directory as a computer account. Windows Server, AD forest, PBX/Communications Platform and IP phones have already been deployed. Deliverables in the project

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    Need an Asterisk/FreePBX guru to show me how to setup xlite and Microsip sip phones to register with FreePBX extension.

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    ...basic IVR. Only one level. 2. Setup of 2 profiles of users. a. Will be able to make local calls only b. Will be able to make local and external calls 3. Configuration of 2 SIP and DID services 4. Configuration of Conference rooms (phone bridges) 5. Configuration of VOIP VLANat the IPPBX and Phone level (network switch is already configured) 6. Configuration

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    i need someone to create and IVR for me using Asterisk that will interact with mysql database for callers details and readout details form database using TTS (Hindi ...done it using PHP api but never got time to finish this (i don't have the codes now) there are 3 levels of menu and it must use 3G datacards to receive make calls not just sip or fxo

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    Hello. I am looking for someone that can build a system that will deliver voicemail messages to external numbers without ringing the phone...This is also knowen as VoiceCasting. It's important that you understand that the phone numbers are NOT phone numbers or extensions within the PBX. Must have vast knowledge of SIP and PBX with FreePBX platform..

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    ...(subscription) - News area - Editable and deactivable % of bonus money if user deposit more of X money - Mobile Site (Responsive Design) - Secure Login; Built-in HTTPS (SSL/TLS) Support - Monitor Orders & Deposits Receipts - F.A.Q / TOS Page ADMIN PANEL - API support for smm panels - Manage and pay the affiliates + sub-affiliates - Set fav icon and

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    Help setting up initial OTRS community version on Centos, configure outgoing SMTP-TLS mail. Sysadmin previous OTRS experience required.

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    I want voice engineer to do IP Telephony Cisco call manager installation and expressway server ,so that sip phone register over internet.

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    Hello, I have an article with about 50 pages. It talks about VOIP and TOR networks. I need to rephrase this article so it can not be recognized or even a...plagiarism checkers. All the article is in English and rephrasing should be in English too, I guess a technical background is needed. Specially talking about protocols Like RTP, SIP,UDP and VoIP.

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