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    9,714 sip tls srtp windows jobs found, pricing in USD

    To provide us VoIP, SIP hacked accounts login

    $30 / hr (Avg Bid)
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    1 bids

    Add sip input tab to frontend to phone dialer.

    $136 (Avg Bid)
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    3 bids

    Dear Audi G 1-i have FreePBX system on -Virtual machine CentsOS 5.10 . 2- you can login via teamViwer on widows7 that the virtual machine on it and then you can logi...that the virtual machine on it and then you can login to cents via SSH or via VMware 3- there is web control also for OS and FreePBX 4- we just need to add new setting for new SIP line

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    Dear Amrit M 1-i have FreePBX system on -Virtual machine CentsOS 5.10 . 2- you can login via teamViwer on widows7 that the virtual machine on it and then you can login t...the virtual machine on it and then you can login to cents vis SSH or via VMware 3- there is web controal also for OS and FreePBX 4- we just need to add new setting for new SIP line

    $20 (Avg Bid)
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    Hi Sunil I saw that you have experience with SIP and VoIP, on android and iOS I need someone to build iOS and android App. If you are interested we can continue talking on chat See you

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    We need Java and React developer to complete this task. We will give 500/- per hour ...complete this task. We will give 500/- per hour Requirement: Java task: Based on my UI i need to 1)Define a project template which includes TLS certificate and 2)Update code generation service to include TLS certificates 3) writing test cases for the above 1 and 2

    $24 / hr (Avg Bid)
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    Hello There i need some help to make some config on my FreeBPX server like: 1-Remove and add new SIP line 2- there is problem to login to the control panel after i tried to make restore backup . so can you do it for me and home many time you need to finish it note: there is teamviwer access and then you can access the Unix server by web

    $50 (Avg Bid)
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    Hello There i need some help to make some config on my FreeBPX server like: 1-Remove and add new SIP line 2-make my system cluod on internet so the agent can use there extension in anywhere 3- there is problem to login to the control panel after i tried to make restore backup . so can you do it for me and home many time you need to finish it

    $50 (Avg Bid)
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    1 bids

    We need Java and React developer to complete this task. We will give 500/- per hour ...complete this task. We will give 500/- per hour Requirement: Java task: Based on my UI i need to 1)Define a project template which includes TLS certificate and 2)Update code generation service to include TLS certificates 3) writing test cases for the above 1 and 2

    $21 (Avg Bid)
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    1 bids

    ...configuration issue stopping mail from being sent from localhost. 2. Set up a single mailbox with pop3 access for a user named Admin on the VPSs domain. 3. Set up postfix to use TLS, either using a new certificate you create or using the wildcard certificate already on the server. I will give you the location of the certificate files when I accept the bid

    $96 (Avg Bid)
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    ...error_temp(resp) ftplib.error_temp: 421-Sorry, cleartext sessions and weak ciphers are not accepted on this server. 421 Please reconnect using TLS security mechanisms. --- NEW CODE I HAVE BEEN TRYING: --- It seems I can connect via TLS OK but I get an error, maybe I missed something simple.. from ftplib import FTP_TLS ftp = FTP_TLS() [login to view URL] = 2 [login to view URL](...

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    I need 7 web pages of varying length converted into a single Microsoft Word 2010 file so I can then print out the document...[login to view URL] [login to view URL] [login to view URL] [login to view URL] [login to view URL]

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    Hello, i have tinkerboard computer running xubuntu 18.04 tls. i use rdesktop to access my win10 virtual machine on server in local network. i want to connect smartcard and usb device to xubuntu and want to use it in my win10 vm. thats your tasks! i will create milestone, but i will only pay if work is finished! thanks.

    $174 (Avg Bid)
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    Skype SIP client in PHP 3 days left
    VERIFIED

    We have a few Skype Connect profiles with inbound numbers at...script should (persistent) connect to Skype servers, login and answer the calls by returning a recording (depending on the inbound number). That's it folks:-) It's all standard SIP, pretty straight forward. --> $100 FIXED (it's max 3 hours work, MAX, 30 minutes if you've done it before)

    $147 (Avg Bid)
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    Email Client 1 day left
    VERIFIED

    ...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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    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 [login to view URL] i want to run this program on a router Router Link [login to view URL] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580

    $579 (Avg Bid)
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    Programming Bandwith.com 1 hour left
    VERIFIED

    Help programming sip endpoints, applications voice & sms in [login to view URL] I need to connect my [login to view 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 [login to view URL] Prior experience

    $9 / hr (Avg Bid)
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    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

    $683 (Avg Bid)
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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.

    $152 (Avg Bid)
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    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

    $541 (Avg Bid)
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    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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    I’m looking for an expert of VOIP, SIP, Hack for educational purpose

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    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.

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    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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    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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    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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    ...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

    $335 (Avg Bid)
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    ...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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    ...(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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    required a AVAYA Call Exchange PHP scripts for SIP, call recording, call records and so on ... along with DB

    $181 (Avg Bid)
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    ...(by external means) that the ACME server challenge 'http-01' will succeed [login to view 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 [login to view URL]

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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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    ...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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    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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    ...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 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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    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

    $281 (Avg Bid)
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    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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    VOIP ASSISTANCE SIP DEVELOPPEMENT

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    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: [login to view URL] ( 12 seconds to load ) [login to view URL] ( 288 ms to

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    ...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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    ...discutere i dettagli tramite chat. A) TRUNKS INFOSTRADA Protocollo SIP username 390415541985 From Domain [login to view URL] secret GZD397XU Host [login to view URL] port 5060 Register String 390415541985@[login to view URL]:GZD397XU:390415541985@[login to view URL] B) TRUNKS MESSAGENET Protocollo SIP username 5406073184 From Domain [login to view URL] secret ajFFzpNj

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    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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