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    11,916 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

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    Build PHP SoapClient Class to consume WSDL service functions 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...a PHP class or classes to implement the connection and execution to a set of WSDL services. The finalised project: - Has valid PHP 7+ code - Connects to the service using TLS 1.2 - Authenticates using a digital certificate (X.509 Certificate Token Profile 1.1) using the WS-Security 1.0 Oasis standard - Uses SOAP 1.2 - Has functions to call and return

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    Угода про нерозголошення
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    Configure Vicidial SIP 6 дні(-в) left

    I have an installed Vicidial server on which I want to configure SIP line. Ping if you have experience with that.

    $18 / hr (Avg Bid)
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    Apple Maps API 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    Apple Maps app has a feature called PARKED CAR, we are looking to add an identical feature in the settings called CAR TRUNK, which would be used in similar fashion as parked car. See URL's below. [увійдіть, щоб побачити URL] Succesful candidate will be able to provide the above request

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    RECOLECTAR INFO Y LOGRAR LA INSTALACION DE UN CERTIFICADO SSL EV 5 дні(-в) left
    ПІДТВЕРДЖЕНО

    Me gustaría instalar este certificado de seguridad en mi pagina [увійдіть, щоб побачити URL] : [увійдіть, щоб побачити URL](ev)-ssl/?entryURL= tengo entendido que requiero ciertos requisitos en mi pagina así como reunir ciertos documentos, por lo que el proyecto incluye la recopilación de información, los

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    Cisco Support Person 5 дні(-в) left

    We have a Cisco 3800 that is setup with a E1 (PRI), we need the following 1. Setup Dial Peer or modify existing to allow calls Inbound to SIP 2. Setup Dial Peer to allow SIP to send calls through the PRI 3. Setup access List Please contact us if you are able to work with us.

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    ...io"; [увійдіть, щоб побачити URL] = "[увійдіть, щоб побачити URL]"; [увійдіть, щоб побачити URL] = false; [увійдіть, щоб побачити URL] = true; it gives the error: @ejabberd_receiver:handle_info:194 TLS error = SSL_do_handshake failed: error:14094418:SSL routines:ssl3_read_bytes:tlsv1 alert unknown ca the issue is in unreal engine our xmpp server works fine w...

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    vicidial configuration 5 дні(-в) left

    vicidial expert, to create a sip trunk with an external sip provider

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    We are looking for someone who has good experience in video chatting app and H.323, SIP. It is very urgent project, so if you don't have good experience in this field, please don't waste our time. Thank you.

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    Hello. I need someone who would install IP based PBX system and A2Billing for its provisioning. Also transport channel needs to be TLS and SRTP.

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    sip VoIP dialer with rcs futures voip call sms chat video ex...

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    I need someone to code and design payment gateway which can used similar like PayU , RazorPay or PayPal . It will have a UI for customer and Vendor payment management...transaction , USD , EURO , POUND, INR , GBP , DINAR etc. It should be compliant with PCI DSS . it should accept VISA , MasterCard ,RUPAY, UPI and AMEX. Should support SSL and TLS .

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    i need someone that can login to my computer and setup my fusion PBX should accept calls from vitelity SIP Provider, and also to make calls thru my outgoing provider.

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    Robocall server 3 дні(-в) left
    ПІДТВЕРДЖЕНО

    A center of family planning is looking for a Robocall solution to remind patient to visit doctors, etc. Informa...reminder them the date of the visit. at the end of each day, a full report should be given. The voices will be prerecorded by our team here. We will use goip gsm gateway (sip) for the calling. We have servers to host the application.

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    ...– simple dial surface, numbers, #, * etc. all via PC/laptop windows 10. NO landline or mobile phone for dialing only the PC and PC microphone + PC headset. Each user has own SIP account and should use the program separately. 6. Dial #2 – open phone book. Select a city/name/phone number from phonebook. Given list each item (each lien) is one-click call

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    I need someone who is capable of configuring a Vultr hosted Freepbx system with callcentric did and trunk. This is just a basic setup to test reliability of the configuration if it works well it will be expanded in the future. There is 1 did 1 sangoma 705s desk phone 1 Grandstream GXV3370 deskphone 1 db20n basestation with a single d10 cordless phone

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    TRAINING ASTTP, FREESWITCH 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    TRAINING ASTTP, FREESWITCH, create rates, carrier with any provider, trunk with any provider, like IDT, users, maintenance of server, security, etc

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    Simple softphone SIP client app for android and iphone 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    I need a simple softphone that allow to SIP register on a freepbx server. With just the dialer, not accessing phone contacts and all the features that most softphones comes. It must run in the background when closed so it can receive calls. Guess is just that, hope your offers! Thanks

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    animate butterfly 3d 1 день left
    ПІДТВЕРДЖЕНО

    $50 budget, 3 sec butterfly fly character our model fly in, land on trunk of elephant, stay on trunk as he talks or hover over trunk, fly up and away not to side like sample, we need 3200 x 1800 transparent background so we can just drop into premier i will send reference links to download video clips attached video is a sample there is one issue with

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    Configurar OpenVPN 19 годин(-и) left

    Necessito de configurar um servidor OpenVPN com c...mim. Até ao momento já tenho um servidor OCSP a funcionar e o servidor OpenVPN a funcionar parcialmente. Necessito que o certificado seja validado no OCSP, atualmente utilizo tls-verify [увійдіть, щоб побачити URL], mas estou com erros nessa validação. Preciso de ajuda para terminar o mesmo. Quem poder por favor...

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    Goautodial specific functions 13 годин(-и) left
    ПІДТВЕРДЖЕНО

    Set up ...on 3 trunks (sip provider). When dialing on lists, send one call to one trunk, next to the second trunk and next to third trunk and so on... 4) Failover : after dialing a number on a trunk, if bad system status (congestion, channel unavailable...) then dial right away to trunk 2 and if still bad system status, then dial right away to trunk...

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    Digital logo 10 годин(-и) left

    I'd like to create a logo for my website and work as an author and speaker on personal development. My initial idea is a play on my initials (SIP) and I've attached a sketch of versions that I have used. Would like someone to play around with that and come up with something professional/digital.

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    need help with vps email SSL/TLS 8 годин(-и) left

    We need someone to configure our emails

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    BITRIX 24 and SIP connector 5 годин(-и) left

    We have setup bitrix24 onpremise and we have asterisk-vici installed. We want to use SIP connector. Any one with experience in setting up onpremise Bitrix24 and telephony with SIP connector can respond.

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    Hi All I have a SIP client on my network, its a Intercom. It registers to my FreePBX/Asterisk solution and makes calls to my mobile phone fine. I want to intergrate Skype because Amazon Echo's can answer/make calls using Skype. The intention is to receive the Intercom call on my home network to an Echo Dot, which I believe will work when the FreePBX

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    ...kind of logs. [2019-02-14 08:56:15] WARNING[7089] chan_sip.c: Autodestruct on dialog '223205511-3759123182-18707@TcomNextoneSBC4-DRC.mydomain.com-onnet_1-b2b_1' with owner SIP/telsam_ip-00003486 in place (Method: BYE). Rescheduling destruction for 10000 ms. if you interested in I can share you more information. and budget is negotiable fixed or hourly

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    OpenSIPS Expert Needed Закінчився left

    We need an expert in OpenSIPS that can change the SIP response of 200 ok to something else or delay it full details will be given later.

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    Hospitality Mobile App Закінчився left

    ...allow only Checked In guests to sign in using surname and Room number. once signed in also allow guest to connect to the Hotel WIFI also within the APP we will need a build SIP/VOIP on the App which will interface with hotel PABX and the app will become the guest extension as long as his signed in and for the duration of his stay. This app will need

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    Cloudflare Authenticated Origin Pulls Закінчився left

    Authenticated Origin Pulls Authenticated Origin Pulls allow you to cryptographically verify that requests to your origin server have come from Cloudflare using a TLS client certificate. This prevents clients from sending requests directly to your origin, bypassing security measures provided by Cloudflare, such as IP and Web Application Firewalls, logging

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    WebRTC / SIP expert Закінчився left

    The client is using astrix server. They will give us a vpn connection to their network And we need to setup WebRTC client to talk to astrix server. We need to use open source functionality working which it can't currently get a function to work within a type. [увійдіть, щоб побачити URL] So that they can make a 1 way audio call which is having issues with local and remote media streams to wor...

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    VOIP SIP CALLING SYSTEM Закінчився left

    Want to start a VOIP/calling card solution ,need to install the necessary software (asterix,freepbx,a2billing etc).Advice the best solution

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    Help with TLS Traffic Закінчився left

    Ive got a sever that sends tls traffic to a ip address every time it starts up. I need help understanding the traffic being sent and received. the TLS certificate files are on the server. Once the traffic is understood ill be looking to develop a protocol to replace the remote ip and answer the messages locally on the server.

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    Trophy icon Build Me a LOGO Закінчився left

    Starting a tea company in Australia Using sri lankan tea The brand name being Tea'S'Me A Play on words being Tea is me and tease me Using a elephant trunk as the "s" in tea"s"me, or even as a a flow under or throughout being a nod to sri lanka Would like a fresh looking logo you would like to see on packaging, websites, stalls and so on

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    freeswtich Закінчився left

    Freeswitch + ASTPP interface Full installation and configuration Trunk Setup Few Rates Configuration Test calls create user dialplan, prefix routing

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    Our company needs an Asterisk Expert Закінчився left

    ...SSL, Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future: We need to have a setup, where we are more

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    Configure FusionPBX & Softphones Закінчився left

    Looking for freelancer to configure FusionPBX with PowerPBX hosting. [увійдіть, щоб побачити URL], [увійдіть, щоб побачити URL], Debian ...Plans, PBX Extensions, I/O Trunks, Voicemail, DISA, IVR & local number display on the go, Call Forwarding, Call waiting, Linphone setup and instructions, call recording, setup sip trunks. for 4 - 6 lines. PHP or LUA scripting, VOIP, interconnect

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    Trophy icon Logo design for "Livingston Depot" Закінчився left

    "Livingston Depot" is the Name we need a logo for.. - "Sip & Shop Station" is the Tag Line This is a boutique cafe that is located in Livingston, Tennessee. This shop offers local artists, hand crafted items for sale, with a small cafe offering beverages, and lunch items. The keyword inspirations are elegance, down home, boutique, station, depot

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    Гарантований
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    Asterisk Expert Закінчився left

    ... Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future: We need to have a setup, where we

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    Help setting up initial OTRS community version on Centos 7 (VPS), configure outgoing SMTP-TLS mail. Linux Sysadmin, Perl, and previous OTRS experience required. ([увійдіть, щоб побачити URL] )

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    3CX Gateway for Skype Закінчився left

    I need a call with multi skype user via SIP. and i saw 3cx gateway for skype program. this product is no longer available for download nor maintained. [увійдіть, щоб побачити URL] [увійдіть, щоб побачити URL] The 3cx gateway for skype is running only windows 7. and needed skype old version

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    currently got tls 1.1 and need to upgrade to tls 1.2 in websphere server

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    Needed multiskype account via voip Закінчився left

    must be connect multi Skype user to a SIP-PBX. and call via skype to pstn or mobile number. multi skype user can running on asterisk machine. extensions are calling via each skype-account

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    ...that extends the distance from the trunk to the end of the branch) that will be mounted on the trunk off an artificial Christmas tree approximately 3 inches in diameter, the light rail will have multiple lights on it extending from the trunk of the tree out ward parallel to the floor, the lights will extend from the trunk with the ability to be hooked on

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    We need an implementation of the services mentioned in the subject. Thanks.

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    ejabberd configuration Закінчився left

    I need help you configure my ejabberd server with sylkserver for translate jingle/sip between xmpp client and SIP proxy and/or PBX. Goal is that XMPP client can make a call through the PBX to the outside work (as a regular UA) and that the XMPP client can be called via the PBX.

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    We have a multi-tenant PBX system and need a simple SIP phone provisioning tool written in Javascript Node.JS, using Angular or React, running on a MEAN/MERN stack. This will not be integrated with the PBX, however our B/OSS system will add, suspend, and delete tenants and extensions to this new system you will create via a REST API. A commercial

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    Project for Nelson112 Закінчився left

    Have you tried using encrypted SMTP on port 465? I bet they won't detect that. [увійдіть, щоб побачити URL]

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    VoIP server for Termination Закінчився left

    Setup a VoIP server for termination the requirements are: 1. SIP connections to gsm gateways through VPN is required 2. Billing software 3. Load balance traffic to gateways 4. setup connections with 3 clients

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    ...Financial Planning ;Invest online;Testimonials ;Blogs; Careers ;Contact us With A live Chat options & watsapp messenger Few Algorithms need to be used –like calculator for – sip;retirement;marriage;child education etc...

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