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    9,566 sip tls jobs found, pricing in USD

    ...In general we are interested in the level that implies a lower number of transactions, which in principle is the most basic and is the one that fits with the startup models. TLS 1.2 certificate for communications encryption (required for PCI certification) Components: - Web control panel (custom Backend). Hosted by Amazon AWS - iOS Mobile App (Not yet

    $670 (Avg Bid)
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    10 bids
    Country Sales Manager UK 4 days left
    VERIFIED

    ...end of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. We are looking for a

    $4158 (Avg Bid)
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    7 bids

    ...end of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. When you apply make sure

    $4206 (Avg Bid)
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    7 bids
    LinPhone Development Project 4 days left
    VERIFIED

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

    $2897 (Avg Bid)
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    ...his SIP server and we need to create separate account for each gateway . and calls should send to specific termination . call should pass with sip , g729 or g723. in client end we want to setup asterisk module which can convert calls from sip to sip and pass the calls to gateway . 01. asterisk or SBO server. which receive calls from many sip server

    $723 (Avg Bid)
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    I have an Asterisk server for private use already running, and would like to acid...server for private use already running, and would like to acidify a Trunk using an FXO VoIP Gateway, for this is necessary to create a sip trunk in Asterisk and I do not know how to do. In my attempts, I can even connect SIP between them, but I can not complete calls.

    $96 (Avg Bid)
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    9 bids

    I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.

    $9 / hr (Avg Bid)
    Local
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    11 bids
    Freepbx and Asterisk settings 3 days left
    VERIFIED

    I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be

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    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

    $1126 (Avg Bid)
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    22 bids

    ...nr1: comes through normal ISDN to IP PBX Set nr2: comes through incoming Vitual Voip DID. All the numbers of set nr 2 work but i have 1 nr that does not work as i get error sip 503. For security reasons , you will be operating through Teamviewer on my computer which is connected to the remote site of my client where the IP PBX is. I also need to check

    $45 (Avg Bid)
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    1 bids

    Looking for someone to pass asterisk logs to, to determine why some calls are dropped or why calls are not routing properly.

    $52 / hr (Avg Bid)
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    19 bids

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

    $836 (Avg Bid)
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    10 bids
    VOIP yate call pass 1 day left
    VERIFIED

    i need to install yate on openwrt and pass calls server to my gateway we pass call useing sip to sip if you can make it please bid

    $200 (Avg Bid)
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    1 bids

    Android sip cellular gateway we are looking for an expert in developing mobile app to develop an app that will expect voice calls using usip server and dial out local number using the mobile network

    $602 (Avg Bid)
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    49 bids

    Hi, I need a wordpress plugin which will allow the users to place a call on the website. I will be integrating a sip gateway for same. Users can place free calls with some restrictions like 1 minutes, or a 10 seconds ad after every 1 minute. Paying users can place unlimited calls until their credits are exhausted I wish to achieve a website like https://ievaphone

    $123 (Avg Bid)
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    Actionscript 3.0 SIP 19 hours left
    VERIFIED

    1. REGISTER + SUBSCRIBE to SIP server, with authentication 2. Accept INVITES 3. play inbound SIP packets and convert microphone to outbound sip That's it. If you've done it before, it's 1 hour work. Fixed payment $100. We need Actionscript in AIR.

    $75 (Avg Bid)
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    2 bids
    PHP SIP client 17 hours left
    VERIFIED

    PHP SIP client: 1. REGISTER + SUBSCRIBE to SIP server, WITH authentication 2. Accept INVITEs to send only, NO recv (so no traffic inbound) 3. Return WAV file to caller; depending on called number either file 1 or file 2 4. BYE to disconnect, that's it. - FIXED $40 for working code (do NOT ASK FOR MORE) - It's not even 30 minutes work if you've done

    PHP
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    12 bids

    ...However, here is the problems that I need your help with to get it to resolved. Problem 1: SMTP Banner Check Reverse DNS does not match SMTP Banner SMTP TLS Warning - Does not support TLS. More Info SMTP Transaction Time 15.598 seconds - Not good! on Transaction Time Problem 2: There are alot of warnings in the Exim Configuration Manager

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    VOIP Project 14 hours left

    Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined

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    14 bids

    Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets

    $34 / hr (Avg Bid)
    Local
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    3 bids

    I am looking for a SNOM phone guru that can create a dial plan for me with the following...strip the '"9". A long distance call would be 91XXXXXXXXXX. We do not want to use time outs for anything but the international calling. We require the complete string for SIP account 1, 2, & 3, Delivery in a text file. PBX: 3CX Professional Phone: SNOM D765

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    i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [login to view URL] or yate or besip [login to view URL]

    $216 (Avg Bid)
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    add sip trunk to elastix i have sip trunk from STC

    $18 / hr (Avg Bid)
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    ...A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registered! I need to get all Registered, and the call is go through SIP...

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    ...tools needed to make the box a complete antispam gateway. Installing proper mail server components for multi-domain future use. - Nginx with SSL (let's encrypt) - Postfix with TLS - Redis for caching - Dovecot with Sieve With SSL, nginx, and all the standard tools that we need. Then we should setup a test subdomain so system can fully route and process

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    Looking to buy ready to use Call Centre CRM. Get back to me with demo. Complete documented installation and setup guide will also be required. I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Call Monitoring &bul...

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    we wish to have someone connect and configure our [login to view URL] to our SIP and Trunk

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    I have a trouble with Android 4.4 or less , Android 5 to up is not affected the app it's ok, I require support to fixed by teamviewer , and i have only a couple hour to finish. I use Eclipse, and the trouble is with SSL & KSOAP2.

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    It must compile on Delphi 5. Must be compatible with TLS V1.0 (If you can make it compatible with TLS V1.1/1.2 even better) It must run on windows 2000 or earlier. It must not depend on WinHTTP. No External dependencies. You can use any open source project as long it does not depend on external dlls or files. The final executable must not be larger

    $211 (Avg Bid)
    Featured Urgent NDA
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    I need Linphone rebranded with our Company info, logo, colors and only allow the setup of SIP accounts over TLS. Removal of options to create a linphone account. This must be done on: IOS, Android, Windows and OSX The result must be packages ready to deploy through app stores. and source code must be handed over to us, as Linphone is open source

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    ...and the VPN connects successfully, however when trying to connect through our DMZ / Virtual IP external to the internet we receieve a TLS handshake error or TLS Error: cannot locate HMAC in incoming packet Fatal TLS error (check_tls_errors_co), restarting if you think you have the expertise to help us out please let post a reply of how much you would

    $16 / hr (Avg Bid)
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    ...without ringing the phone. App will need to have standard features such as dashboards to see send data, user data, cost from carrier data. Must be able to connect to any SIP provider. Ringless Voicemail examples [login to view URL] [login to view URL] [login to view URL] https://www

    $614 (Avg Bid)
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    Looking for a Tech Blogger to cover all the topics around VoIP, SIP trunking, DIDs, different countries legislations and Etc. Candidate needs to be familiar with the industry and love tech-savvy content Please when apply, send your samples

    $27 / hr (Avg Bid)
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    We need to bridge standard SIP calls to/from our iOs/Android app written in Adobe AIR Actionscript. In other words: handle the RTP media part of SIP to/from spk/mic. If you don't know by now what is required, PLEASE DO NOT RESPOND! Fixed payment $200.

    $200 (Avg Bid)
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    1 bids

    I’m throwing a sip and paint party. I’m looking for a artist to guide the class, to do a painting

    $83 (Avg Bid)
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    I am unable to get calls from PSTN to freeswitch working. Calls from a SIP user into the switch (over the gateway) work. Calls between extensions and outbound calls work. I'm loading dialplan, configuration and directory with xml_curl (I had used a lua xml_handler script). I can confirm there is no problem on the carrier end...same carrier works with

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    ...be closed). The username and password should be in memory of the server on POC (just a simple dictionary, for instance. This will be done in a better way later on) 5. SSL / TLS must be used. 6. Transfer of binary array streams. 7. Preferred to be able to use different languages to connect to the server as well, if easy to implement. 8. All server code

    $287 (Avg Bid)
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    19 bids

    I have multiple SSIS packages that are failing due to "Could not create SSL/TLS secure channel." It seems to be related to registry authorisation issues as in the event viewer I am seeing the following error message under Event ID 10016 "The application-specific permission settings do not grant Local Activation permission for the COM Server application

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    I need a SIP client COMPLETELY written in ActionScript, so NO external libraries or other dependencies. It should be able to connect with a SIP server, ACCEPT calls only (so don't worry about dialing and invites) and handle that 2-way phone call (mic/speaker). That's it, nothing specific! If you know what you're doing, you don't need anything else

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    I require a voicemail d...csv list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as [login to view URL] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.

    $491 (Avg Bid)
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    we are looking for en expert to develop a app that will act as sip gsm gateway i am including here a [login to view URL] to a software that was develop for window [login to view URL] 2, the actual software for window s i will drop it to you once we hire you 3. a suggestion

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    Hi, I need SMS gateway, Without using any third party services like twilio etc. I need my own gateway. Freelancer must have experience with Voip/PBX/SIP etc

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    We are looking person with specific experties SIP protcol PHP DontNet Also goo din graphics

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    Build a front-end with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar ...with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar PHP framework, manipulate the Asterisk PBX from the interface (ivr, Sip/iax, did, ami/agi, voicemail, routes etc.)

    $5559 (Avg Bid)
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    ...SMS Gateway service that utilizes either PBX, SMPP, or VoIP - the platform must be able to run stand-alone meaning it can not use any 3rd party services such as a 3rd party sip provider, a 3rd party SMS service such as Twilio or Plivo. It must not need SIM Cards or modems. It must be able to create VoIP numbers (on it's own) ex; not using Google Voice

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    ...JSON format from the client side. The input is sanitized and checked then inserted to its appropriate variables then passed to an email client. The email client must use GMAIL. TLS. also API authentication code plus password. You can test using your own account as we will not supply credentials. Once the code is complete, we will place our creds. The code

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    ...v2/v3/v5/ i want to run 32calls you can connect server to local a VPN and MASQUERADE connections or you can install any SIP PBX on local route like sofia-sip free-switch and register server to local if you want you can install a SIP Signal service on openwrt and pass call to voip device also you can bridge network and assin a server IP on local Voip

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    ...also getting a second error message in Event viewer. This is the SSIS error: [Download error] Error: [login to view URL]: The request was aborted: Could not create SSL/TLS secure channel. at [login to view URL](Uri address, String fileName) at [login to view URL]() This is the Event viewer error:

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    ...(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 -

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