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    9,587 windows sip client 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

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    C++ TLS implementation 6 дні(-в) left

    Have a little project related with cryptography TLS. Need to implement 2 files. Nonce counter and ascon128. Framework is built

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    Web development 6 дні(-в) left

    Have a little project related with cryptography TLS. Need to implement 2 files. Nonce counter and ascon128. Framework is built

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    ...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

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    Country Sales Manager UK 3 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...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

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    Sales Representatives for SaaS in the UK 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...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

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    LinPhone Development Project 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...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 --> ready in four weeks

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    Low bandwidth voip solution by asterisk. 2 дні(-в) left
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    well , we need basically bandwith optimization. in client end they have termination gateways, each gateway can carry 30 simultaneous calls. and all gateway under DHCP network . we take a server which run in static IP . all client should send calls to server IP from his SIP server and we need to create separate account for each gateway . and calls should

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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.

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    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.

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    Місцевий
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    Freepbx and Asterisk settings 1 день left
    ПІДТВЕРДЖЕНО

    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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    Integración pagos online de Redeban con woocommerce 18 годин(-и) left
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    Necesito integrar el metodo de pagos de redeban multicolor con woocommerce de wordpress. s necesario que al realizar la implementación hagan uso del protocolo TLS 1.2 a nivel de comunicación. Al igual que si se realizará el consumo del servicio a través de SOAPUI se realicen los siguientes pasos: Buscar donde este el SOAPUI. Buscar el archivo SoapUI-5

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    Personalisation LinPhone Security & Design 17 годин(-и) left
    ПІДТВЕРДЖЕНО

    ...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 --> ready in four weeks

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    Project for Anton F. 3 дні(-в) left

    ...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 that the Set nr2 of numbers show the right DIDs when they call external

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

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    21 заявки

    ...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 --> ready in four weeks

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    VOIP yate call pass Закінчився left

    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

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    app development Закінчився left

    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

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    Wordpress Plugin for Phone Call Закінчився left

    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

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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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    Actionscript 3.0 SIP Закінчився left

    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.

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    PHP SIP client Закінчився left

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

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    Місцевий
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    Create SNOM Phone Dial Plan Закінчився left

    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 [увійдіть, щоб побачити URL] or yate or besip [увійдіть, щоб побачити URL]

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    add sip trunk to elastix Закінчився left

    add sip trunk to elastix i have sip trunk from STC

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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

    $32 (Avg Bid)
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    3 заявки

    ...Music On Transfer • Overhead Paging • Remote Call Pickup • Remote Office Support • Roaming Extensions • Route by Caller ID • SMS Messaging • Spell / Say • TAPI - Dialling from Windows applications • Text-to-Speech • VoIP Gateways • Voicemail • Visual Indicator for Message Waiting • Voicemail to email • Web Voicemail In...

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    Independent dialer system Закінчився left

    we wish to have someone connect and configure our [увійдіть, щоб побачити 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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    SIP面板中找到100家从事房屋建筑的公司 如果事情不明确,请写信给我,我会尽力解释。 在中文里,我通过谷歌翻译写作。 我的母语是俄语。

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    HTTPS Client - Delphi 5 Закінчився left

    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

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    Обраний Терміновий Угода про нерозголошення
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    voip语音网关模块 Закінчився left

    具备sip协议,支持30路以上fxs,可以实现点对点通信,需要提供相关技术支持

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    Rebranding of Linphone Закінчився left

    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

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    Ringless Voicemail Закінчився left

    ...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 [увійдіть, щоб побачити URL] [увійдіть, щоб побачити URL] [увійдіть, щоб побачити URL] https://www

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    Tech Blogger -Telecommunications Закінчився left

    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

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    Full Actionscript 3.0 SIP-RTMP bridge Закінчився left

    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.

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    Design project Закінчився left

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

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    webrtc asterisk Закінчився left

    Necesitamos hacer un softphone basado en WEBrtc, que funciones desde todos los navegadores compatibles con esta tecnología, se conectan por sip a nuestro servidor asterisk

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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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    ...help out on a server-client setup within C# winforms. Preferred with [увійдіть, щоб побачити URL] The basics: 1. The server must be able to handle multiple clients. 2. All clients must be dealt with in a different thread. 3. Reading from and writing to the client must be done async

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    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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    Actionscript 3.0 SIP client Закінчився left

    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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    Voicemail Drop Windows Application Закінчився left

    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 [увійдіть, щоб побачити URL] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.

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    android sip gsm gateway Закінчився left

    we are looking for en expert to develop a app that will act as sip gsm gateway i am including here a [увійдіть, щоб побачити URL] to a software that was develop for window [увійдіть, щоб побачити 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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    I need SMS Gateway Закінчився left

    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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