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    11,470 sip softphone tls srtp jobs found, pricing in AUD

    i am looking for source code for a simple working android sip / voip phone for android i dont need it to be complex in any way just stable and working

    $391 (Avg Bid)
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    2 bids

    Goal ------------------------------------------------ Tha goal of this project is to enable our operators to issue an SMS message to a potential customer linking to Google maps with instructions on how to get to our location from their current whereabouts. Background ------------------------------------------------ we got a lot of callers looking for information about directions. The address al...

    $682 (Avg Bid)
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    3 bids
    fix issue on voip client 6 days left
    VERIFIED

    i have a android sip client app but can not identify where i have to enter the sip address to send calls to i am looking for someone to debug this

    $276 (Avg Bid)
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    12 bids

    Need help with Freeswitch troubleshooting realtime configuration and various issues. Current configuration includes Debian OS, Freeswitch using LUA for subscriber configuration, SIP profile configuration and dial plan all stored in database for realtime functionality. Also python used for socket connectivity into Freeswitch for event drive functions. Must have knowledge with Freeswitch, LUA, my...

    $30 / hr (Avg Bid)
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    6 bids

    We have freepbx running in production but would like to have the server optimized. Here are few things for a start that need to be done: 1) Securing freepbx server - fail2ban, iptables etc ipv4 and ipv6 2) ensuing that secure channel is used - iax, tls etc 3) ensure that the SIP setup is optimal 4) ensure that codecs and sip settings are right 5) configuring DID for extensions 6) Ensuring that bac...

    $391 (Avg Bid)
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    6 bids

    Looking for an experienced web site developer that have experience using Yahoo small business website tools to redesign current website. We are seeking the right software developers and technologies for our website development project: Custom web app development design: Expert in MVC (CodeIgniter, Laravel, CakePHP & YII) and Open-source Custom Framework example PHP Development, or Laravel P...

    $212 (Avg Bid)
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    1 bids

    Need help with Freeswitch troubleshooting realtime configuration and various issues. Current configuration includes Debian OS, Freeswitch using LUA for subscriber configuration, SIP profile configuration and dial plan all stored in database for realtime functionality. Also python used for socket connectivity into Freeswitch for event drive functions. Must have knowledge with Freeswitch, LUA, my...

    $79 / hr (Avg Bid)
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    12 bids

    Need help with Freeswitch troubleshooting realtime configuration and various issues. Current configuration includes Debian OS, Freeswitch using LUA for subscriber configuration, SIP profile configuration and dial plan all stored in database for realtime functionality. Also python used for socket connectivity into Freeswitch for event drive functions. Must have knowledge with Freeswitch, LUA, my...

    $36 / hr (Avg Bid)
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    6 bids
    .net developer c# 5 days left
    VERIFIED

    i am looking for a .net developer with good experience with paypal api. i am facing a problem: : Invalid HTTP response: The request was aborted: Could not create SSL/TLS secure channel. that i would like a second developer to look at. impotent: developer needs to have good speaking English level. i communicate directly to the developer. experience with mvc core thank you Ori

    $11 / hr (Avg Bid)
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    21 bids

    Hello, This is VoIP/SIP Mobile Dialer (Android) Mobile Application Project. ============ In the application, the following functions are required. - Register / Login - Call History - Contact - Dialpad - Rate Information - More Tab ============ The "Register" / "Login" functions are the following. "Register" process ....(like viber) - step 1 >> just re...

    $1819 (Avg Bid)
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    1 bids
    Android REST API Client 4 days left
    VERIFIED

    I need an android Studio application that do following things: a basic button press send a post request as follows: Host = [login to view URL] Header = "Content-Type", "application/x-www-form-urlencoded" Body = grant_type=password&username=siemens&password=ATLAS5005$ need to ignore all certificate errors SSL/TLS the response have to be printed in text view.

    $155 (Avg Bid)
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    need to configure the Goip-1 GSM Gateway for dialing in to my sip phone. so when i m travelling abroad. i can still call and receive in my homeland country on mobile networks.

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    Develop based on serverless Google Cloud platform simple SIP client that: when receive inbound call send POST request to spicified WEB API with sending called ID and when send sall cancelation (withount hangup) so SIP line. It's need for SIP PBX that initiate call to many SIP phones for log inbound calls.

    $334 (Avg Bid)
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    We are looking for someone with knowledge of VoiceMail DROP or Ringless Voicemail to build backend code and provide us API which will accept JSON object with NumberTO, ToName, VoiceMessageFile, SenderNumber, SenderName, and system should call NumberTO and without ringing the phone should drop VoiceMessageFile in the recipient's voice mailbox. We use FreePBX V14 hosted on AWS as our VOIP serv...

    $33 / hr (Avg Bid)
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    Looking for a logo that reads "Sip & Bitch" This will be going on wine glasses. I will need the logo in Ai. and fully vectorized. I have attached a wine glass jpeg

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    27 entries

    Hi! I need the same solution! Are you ready to start this project? What about price and time? https://www.freelancer.com/projects/mobile-phone/whatsapp-sip-gateway/

    $168 (Avg Bid)
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    1 bids

    Hi! I need the same solution! Are you ready to start this project? What about price and time? https://www.freelancer.com/projects/mobile-phone/whatsapp-sip-gateway/

    $168 (Avg Bid)
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    1 bids

    Hi! I need the same solution! Are you ready to start this project? What about price and time? https://www.freelancer.com/projects/mobile-phone/whatsapp-sip-gateway/

    $168 (Avg Bid)
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    1 bids

    Hi! I need the same solution! Are you ready to start this project? What about price and time? https://www.freelancer.com/projects/mobile-phone/whatsapp-sip-gateway/

    $168 (Avg Bid)
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    1 bids
    Setup Vicidial 2 days left
    VERIFIED

    We have vicidial and a carrier just need someone to help setup everything and integrate softphone.

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    14 bids

    User Interface Module • Couple of pages to collect data from customers o Pages should be designed to adapt to the device screen o Client-side validations on data entry pages, supported by server-side validation • Data Validation page o Shows consolidated data collected by data collectors. o Options to partially approve/reject data. • Admin page/Dashboard to see aggregated data o Tab...

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    Hi There, I am looking for an expert Logo designer in Creative Purpose. I want a Logo with the business card, Letterhead etc.. It should not be copied or similar to any other company. Name - ChaiKhatti Let me explain to you the name Chai means Tea and Khatti means few people gossip in one place. Tagline - Sip it up! A true stress buster! I am looking for some concept-oriented for the Logo. We c...

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    We use Asterisk PBX along with FreePBX using SIP Protocol. We would like a developer to integrate Whats App calls to our PBX System. Either directly by configuring the Gateway on the PBX itself or as a trunk which is piped from an Android Phone (in this case the phone will be ON and connected on WiFi) to receive and make calls and pipe it as a SIP trunk to our PBX. Requirements: Inbound and Outbo...

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    I have a multiserver vicidial cluster setup on AWS which encounters a handful of issues once upwards of 50+ simultaneous users are logged in and autodialing 100+ lines. In fact, the odd part is this is the only point at which the system begins to demonstrate the error (the reason i'm making this post). Our current configuration is the following: 1 - DB instance 1 - Primary calling server A (w...

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    I have a multiserver vicidial cluster setup on AWS which encounters a handful of issues once upwards of 50+ simultaneous users are logged in and autodialing 100+ lines. In fact, the odd part is this is the only point at which the system begins to demonstrate the error (the reason i'm making this post). Our current configuration is the following: 1 - DB instance 1 - Primary calling server A ...

    $866 (Avg Bid)
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    5 bids

    We need to install a WebRTC softphone por our callcenter system.

    $21 / hr (Avg Bid)
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    7 bids
    SIP Simple application 4 hours left
    VERIFIED

    requirement to create the simple SIP simple application in anroid for sending the message to the other client. we have SIP server

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    Bid if you are familiar with softphone and web app

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    I need an app for Android. I already have the design, I just need you to believe it. always on top. h264. peer to peer lan/wifi, reguster, phone book, customs logo, keypad for dtmf dial on call, mute, history, sos, phone, video WiFi, ethernet, 3G, 4G. and push for service with asterisk, phone book, blf list.

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    Need to assist in setting up and documenting our BURP suite professional to run a thorough security test. Including: OS command injection, SQL injection, Python code injection, Strict transport security enforced, TLS cookie with secure flag set, Cross-domain Referer leakage, Cookie without HttpOnly flag set, Email addresses disclosed, and Cacheable HTTPS response. Must be able to communicate throu...

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    Bid if you are familiar with softphone and android

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    We have an open source C# project that runs as a telegram mtproxy server: [login to view URL] What we need is to add support for the new version of mtproxy server: Fake TLS mode. Here is the official source code for Fake TLS mtproxy in C++: [login to view URL]

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    Hi Boris, you worked on an Adtran router for me 4 years ago. I'm having issues with the SIP registrations not working anymore. Would like to see if you can fix it.

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    Hey all, I am looking to work with a talented devops engineers to help me with connecting to the Elastic cloud operator in kubernetes, i have already managed and configured Kibana / Logstash however running into some issues with a legacy codebase which you can see here - [login to view URL] The successful candidate will be experienced in managing and configuring kubernetes deployments and speci...

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    I have a new install with fusionpbx. I'm new to it and I dont understand much.. I have used freepbx for years with no issues so I'm not a new person to the sip ordeal. I'm looking for someone to hep me out configure my fusionpbx with a gateway using flowroute and receiving and placing a call for a test to make sure it works.

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    Am opening new “Tea cafe “ looking designer , Cafe name : “Tea’s Me” and tag line is “Sip,Eat & Repeat “

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    The purpose of the project is to build an application that will run on augmented reality glasses. The app will provide a video call so a field operator can receive help by a remote web operator that can see what the glasses webcam see. Actors: - on the field operator: wear AR glasses and receive help - web operator: sits in front of a web page and gives help Use cases: a) The glasses operator...

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    Our website is built on ROR and hosted on AWS. AWS has sent attached notification to upgrade the SSL/TLS certificate. Looking for expert to perform this maintenance task. Please revert with hours required to complete the task.

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    We need to have someone configure a Cisco router as voice gateway to convert SIP to ISDN for inbound and ISDN to SIP for outbound calls. End-user has a Alcatle ISDN PABX which is not SIP capable. So we need this gateway configured for the conversion. A block diagram is attached but all details will be provided once the project is accepted.

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    We have an Android VoIP sip soft-phone dialer which keep crashing. Need to repair and if good can also make an Iphone dialer for us as well. Please give price only to fix the Android version. Please only apply if you have made Android Sip phones before. Also please send sample.

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    Hi , we have developed an HTTP web app in HTML/Apache/SQL It is currently built around HTTP and is not secure , some customers want it LOCALLY not cloud secured with TLS/SSL, I have a full pen test result and we need to meet the requirements. I need a price, I can provide our EXE for now so you can see the installed product..

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    we have a Product like Zoom which currently doesnt have a phone dial in for video and webinars. all audio(speaking and listening is throufh the device ports. Iam looking to do SIP Interconnect so that the users can hear and speak using a phone line also as needed. Let me know

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    Hi Abha M How are you? We are interested if you can help us to migrate Our 3dcart store [login to view URL] to woo-commerce development store [login to view URL] We tried self migration service but seems it’s not what we are looking for as we have extra requirements if that can be fulfilled we will be happy to work with you We are not technically expert to do a full migration and just w...

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    We have database server on aws, our special script check every caller IDs which comes in and goes out with some APIs to identify if caller is a spammer or not. Opensips serves as a proxy for IP authentication and SIP registration and once Opensips receive the calls and match everything as per configuration it sends call to freeswitch for further processing. Now, in database we have table which st...

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    We have a dsiprouter with kamailio installed to provide sip trunks to our private network. The endpoints we have freepbx, some new, some old versions of asterisk. We have run into an issue where calls are prematurely dropped for no reason around 10 minutes. Maybe a configuration error of sorts. Server 1 - Dsiprouter, kamailio, CentOS 7.6 Server 2 - FreePBX 12, Asterisk 13.21, CentOS 6.8 Server 3 -...

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    Hi Andre, looks like you have the skill set I’m looking for, see you know DNS bind, TLS, VPN and transparent proxies. Could we have a chat so I can tell you more about my project, I would also need to send you a nondisclosure agreement. Look forward to hearing from you. Regards Shane

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    sip over vpn issues. no voice from either end. freepbx/asertik. All is working well excpt what seems to be rtp.

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    We have a PBX running on Asterisk and we have around 60 extensions configured. We use this in order to do remote voice alerts in softphones running on servers and some Axis IP speakers using SIP. We have many Axis speakers working and for this we had forwarded the necessary ports for them to work and we use the public IP address. All of the extensions are outside of the network our server is run...

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    I need someone professional who can build desktop application using visual basic Application description: Voip sip Dailer

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