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    11,660 sip trunk tls srtp робіт знайдено, ціни вказані в USD

    Що потрібно… «Розвернути» на сервері ASTERISK Налаштувати транк sip gsm +transfer Запис розмов та фіксацію номерів трансферних Адмін панель Панель доступу user Надаштування GSM шлюзів

    $576 (Avg Bid)
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    5 заявки
    Git Versioning For my System 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    Hi dears I need to create a ve...follows - I have 3 Virtual Machines viz. Dev,UAT, Prod (Live VPS in a datacenter) I need to redirect the versions to each one of them from local workstations I also need to use Trunk and advance techniques of Github. I need a detailed training program while installing and configuring Github. Thanking you in advance

    $39 (Avg Bid)
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    1 заявки
    Email Client 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...ending messages • filtering messages • handling footnote • configuration of the webmail client with email account e.g. protocols (IMAP, SMTP, POP3) and certificates (SSL, TLS) and ports (e.g. 110/ 25 etc.) This is a basic and simple description of this module, we have a detailed requirement file which we will send you after the contact on portal

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    Hi, I look for Freelancer who can implement Asterisk with my MGCP provider. Not SIP. Can somebody do? Thanks

    $261 (Avg Bid)
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    SIP register with Custom software 4 дні(-в) left
    ПІДТВЕРДЖЕНО

    hi i want to register sip to sip and pass call g729 problem is i dont want to use big pc and lots of HW so i found a program its allow to do that i want [увійдіть, щоб побачити URL] i want to run this program on a router Router Link [увійдіть, щоб побачити URL] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580

    $579 (Avg Bid)
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    Programming Bandwith.com 4 дні(-в) left
    ПІДТВЕРДЖЕНО

    Help programming sip endpoints, applications voice & sms in [увійдіть, щоб побачити URL] I need to connect my [увійдіть, щоб побачити URL] account to my PBX, be able to forward DID numbers to another number, and direct sms's messages. I would also like to complete the project described here [увійдіть, щоб побачити URL] Prior experience

    $9 / hr (Avg Bid)
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    Multichannel Phone Dialer 4 дні(-в) left
    ПІДТВЕРДЖЕНО

    In need of software or asterisk/freepbx hosted solution for "Voice Broadcast" or "Robo Dialer". - Connects to SIP Trunk - Alternate Caller ID; Area Code Caller ID campaigns with alternating numbers - Ability to play and add audio files messages that are randomized - Multichannel lines (100) to make hundred or thousands calls per minute - Choose number

    $669 (Avg Bid)
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    Trophy icon Design a Logo for a Clinic 2 дні(-в) left

    Design the logo of a orthopedic clinic. Please use your imagination but you have to include the following: - Company Name: Telge Ortopedi - The symbol of ortho...of a orthopedic clinic. Please use your imagination but you have to include the following: - Company Name: Telge Ortopedi - The symbol of orthopedic (a rope tied around the trunk of a tree)

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    Гарантований
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    I need help with setting u...setting up the Outbound Parameters of my 3CX-based SIP Trunk (in the cloud). I am able to make & receive calls using Nexmo. But with Anveo, I only succeeded one time to make a call by tweaking the Outbound Parameters in the 3CX SIP Trunk Setup and I was never able to reproduce that again. Knowledge of SIP trunking a MUST.

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    sip to Native dialer APP 3 дні(-в) left
    ПІДТВЕРДЖЕНО

    Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for this

    $510 (Avg Bid)
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    SIP 2 SIP call optimization bandwidth 3 дні(-в) left
    ПІДТВЕРДЖЕНО

    i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many peppole

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    VOIP, SIP Expert 3 дні(-в) left
    ПІДТВЕРДЖЕНО

    I’m looking for an expert of VOIP, SIP, Hack for educational purpose

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    Connect IP PBX to Twilio WebRTC 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    I want my IP PBX to be connected to Twilio WebRTC so that can be able make calls using Twilio SIP account. Calls from my PBX will come from an IP so I need someone who can make the set up for me to start using Twilio voice trunk.

    $42 (Avg Bid)
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    SIP Proxy Server install 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know

    $193 (Avg Bid)
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    skype for business on-premise installing and setting up 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    you have to install skype for business 2019 and set it up. we have multiple sip phone numbers which have to be also connected to skype for business 2019 for in and outgoing calls.

    $250 (Avg Bid)
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    i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for

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    Secure Issabel PBX (asterisk) 19 годин(-и) left
    ПІДТВЕРДЖЕНО

    I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

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    Trophy icon Création logo Startup 18 годин(-и) left

    ...companies. It's a startup centred on cloud technologies. (Voip, Sip) The name is " Caraïbes VOIP " The main color must be blue like the sky. Je démarre une startup dans la téléphonie sur IP à destinations des entreprises. Le projet est axé sur les nouvelles technologies CLOUD. (Voip, Sip) Le nom du projet est "Ca...

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    ...com/click-to-call/ etc You don’t even need a fax machine…,Receive Fax from customer as PDF in your Email. The system will support setting up/administrate the call center SIP accounts (with hunt groups). Of course system needs configuring Asterisk for custom DID:s, call forwarding, internal calling, voicemail, ability to activate/deactivate service

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    PEAP-TLS Configuration 5 годин(-и) left

    ...Device to Access-point by using PEAP-EAP-TLS authentication. This authentication method is available in windows but not in Ubuntu/Redhat as of now. But AFAIK, Cisco confirms that Linux can connect by using mentioned method. Currently, following is the key components in picture Wired 802.1x with PEAP-EAP-TLS Authentication method. 1) Supplicant ( Embedded

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    Build A Website Закінчується left
    ПІДТВЕРДЖЕНО

    ...(GDPR & ePR compliance functionality) WP Rocket Installation and configuration (caching Functionality) WP Security Installation and configuration (Security Functionality) SSL/TLS certificate installation and configuration...

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    Hi, I deliver VoIP services to my customers and i use 90% of the time the Yealink SIP phones. The problem in 50% of the cases is that when the client is turning on their voicemail, they can't see it anyway if it is turned on yes or no. I know it is possible to use something with XML and a webserver to create a connection with the phone which can enable

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    require a AVAYA Exchange PHP Scripts Закінчився left

    required a AVAYA Call Exchange PHP scripts for SIP, call recording, call records and so on ... along with DB

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    Node.js App Errors Закінчився left

    ...(by external means) that the ACME server challenge 'http-01' will succeed [увійдіть, щоб побачити URL][tls-sni-01].loopback should be defined as function (opts, domain, token, keyAuthorization, cb) { ... } and should prove (by external means) that the ACME server challenge 'tls-sni-01' will succeed Error 2: Error: listen EACCES 0.0.0.0:80 at [увійдіть, щоб...

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    A2billing & freePBX Quick Project Закінчився left

    I have a brand new installation of a2billing & FreePBX combo and would like some guidance on testing and troubleshooting my first call. I have a user, trunk, call plan and rate card created and already uploaded some rates. I need someone very experienced with a2billing & freePBX (must prove it) and need the project completed pretty quickly (say in

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    I need someone to setup my softphone with vpn and connect to an existing hosted pbx. my sip and voip are blocked by my isp

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    I require a new firmware for my GPS+ sensor device. The device uses a sim card and sends data via TCP/IP. The requirements are for a high-security firmware (TLS 1.2, secure boot, etc). the microcontroller used is stm32f103c8t6. More details about the requirements and hardware can be shared.

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    logo developer Закінчився left

    Hi I'm looking for help to build an image for my new business. ...Elegant and simple. I was thinking something like a tree but instead of leaves tiny hearts and butterflies. with the word wedding over or below and centerpieces as the tree trunk...but this may be a little OTT. Alternatively just the words Wedding Centerpieces with a 'logo' to the side

    $140 (Avg Bid)
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    ...even runs successfully, but the request to Google's servers ultimately times out. We need to find out why this is failing. Possible reasons for the failure could include: SSL/TLS connection problems Android environment variables A bug in the gRPC source code So it is possible that there is a problem with the gRPC source code itself, and meaningful debugging

    $135 (Avg Bid)
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    Cisco voice gateway connection to SIP server and PSTN on E1 PRI

    $158 (Avg Bid)
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    VOIP, SIP, cyber security, Networks. Закінчився left

    I need someone who is expert in hacking, also expert in networks, cyber security, VOIP, Telecommunication servers.

    $52 / hr (Avg Bid)
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    sipxcom sip system setup Закінчився left

    ...3.64, but i cannot setup outside domain, i want to use sip outside network, i know there are instructions but i cannot fix the problem. I know i just miss very little thing. i will give you teamviewer to setp up my pc. You need to do: 1)help me to set up sip system using sipxcom, i need the sip can be use in internal and external network. 2) tell me

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    I need logo and letterhead designer Закінчився left

    I have just registered new firm in financial services. Stock broking, mutual fund, SIP investment advising, Various types loans & insurance services will be our products. I need logo & letterhead for this.

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    Cisco Cube PBX Expert Закінчився left

    we need some assistance with the setup of a backup VSAT Trunk and ISDN connection. we have 2 issues which we need assistance with which may be an issue with the Cube router 3rd party equipment or VSAT provider.

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    1) Move some WordPress projects from Ubuntu 16.04 LTS to 18.04 LTS + l...Win/Android/IOS-Clients (Certificates/Keys installed on AsusWRT-Merlin 384.7_2) 3) Create full documentation, how to replicate installation of above mentioned (esp. if new OpenSSL with TLS 1.3 is released). - Please provide recommendations projects for OpenVPN on freelancer.com

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    WE need to connect a cisco 7206VXR with VIC adapters for voice connection to the PSTN and to a Mizu SIP Server for users authentication and billing using sip/h323 protocol. We already done a major part of the setup and configuration of both the cisco and the softswitch but calls are still failing to authenticate on the cisco gateway to the pstn network

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    Fix SMTP server Закінчився left

    I need the TLS support of my SMTP server fixed. I use PowerMTA as my mail transfer agent and I manage the servers activities with mumara. I tried adding <domain *> use-starttls yes require-starttls no </domain> but it doesnt work, I need help.

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    I have a system running Centos 7 that has a bonded network interface (4 1Gb interfaces) connected to a switch with a trunk. This trunk has one or more VLAN's exposed. I am able to configure a VLAN interface to connect and pass traffic to the switch. I am also using Virtualizor ([увійдіть, щоб побачити URL]) to mange VPS systems on this host. Virtualizor

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    Project for V Y. -- 2 Закінчився left

    please setup tls ftp (secure ftp) as i dont want to give people root lol this is for the itinvestigator project

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    VOIP ASSISTANCE Закінчився left

    VOIP ASSISTANCE SIP DEVELOPPEMENT

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    Hunter becomes the hunted 2 ! Закінчився left

    I want to create a design for a tshirt. The design showing an elephant holding a poacher high in the air in it's trunk. with the poacher's gun flying out of his hand. Perhaps another poacher in the picture running away. The design could be in the form of a silhouette or not. Thank you !!

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    Trophy icon JT&T Brand 2 дні(-в) left

    I’m looking for a logo for business cards and letterhead I’m looking for some flyers so that I can advertise VoIP and SIP services the Name of the company is JT&T Communications we offer best in class SIP and VoIP. And phone system analysis design and implementation

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    Install freepbx on CentOS machine Закінчився left

    1 - We need to configure Asterisk FREP...documentation from the work with detailed description with step by step of how to configurate it. 3 - The developer will register some of our SIP channels into the freepbx. We currently have both sip clients: - SIP Phone - SIP on PC 4 - We need someone who is an expert on it, because we need it asap. Thanks.

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    Hello, When there are more than 250 users online on my server, the "TLS handshake connection" takes up to 30 seconds to connect making my sites very slow. This only happens with https websites, the http sites load very fast. See live example here: [увійдіть, щоб побачити URL] ( 12 seconds to load ) [увійдіть, щоб побачити URL] ( 288 ms to

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    Fix open ports for smtp server Закінчився left

    ...server. I have got a problem with connecting to smtp server as I am getting an error "authentication failure [SMTP: STARTTLS failed (code: 220, response: 2.0.0 Ready to start TLS)]" when I am trying to connect from php. I want you to have a look on my server to see what the trouble is and why I cant connect to the server. You must use teamviewer to control

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    Project for Prostitov A. Закінчився left

    ...discutere i dettagli tramite chat. A) TRUNKS INFOSTRADA Protocollo SIP username 390415541985 From Domain [увійдіть, щоб побачити URL] secret GZD397XU Host [увійдіть, щоб побачити URL] port 5060 Register String 390415541985@[увійдіть, щоб побачити URL]:GZD397XU:390415541985@[увійдіть, щоб побачити URL] B) TRUNKS MESSAGENET Protocollo SIP username 5406073184 From Domain [увійдіт...

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    IAX SIP VOIP Softphone Закінчився left

    This project is only for those freelancer that has existing similar app to amend for us. Do you have IAX SIP VOIP Softphone system and apps? We would like to do similar system like www . zoiper . com Zoiper is a phone solution perfectly fit for end users, service providers, call centers or any business willing to benefit from VoIP communications

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    HSMWorks-sip panels/ Panels Закінчився left

    ...competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Our models are prepared in Solidworks and require converting using HSMWorks software. We are looking for someone who can optimize our cutting routines and prepare new

    $2413 (Avg Bid)
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    Windows PDF Print Server Закінчився left

    ...physical printer installed on the Administrator Workstation the Interface is running on. The Administrator Interface will then upload the PDF/A file to a PHP Web Server over HTTPS TLS 1.3 connection and send the unique 8 digit security code as a POST request. The web page will return a 1 for success that the PDF/A document and unique 8 digit security code

    $1103 (Avg Bid)
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    QT based SIP Client with Dynamic CRM Закінчився left

    Lightweight SIP Client with Dynamic CRM which would display the customer information. CRM Field and information would display dynamically and action should collect and send to server Database. Required Features: SIP Call with transfer/conference/hold/Mute facility SIP messenger. Dynamic CRM-Dynamic field header and value display for each call according

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