Sip server tls srtpproiecte

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    9,924 sip server tls srtp proiecte găsite, la prețul de USD
    Scrie un software S-a încheiat left

    Am nevoie de cineva sa imi configureze un telefon cisco 7960 pentru serviciul de sip.

    $37 (Avg Bid)
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    Salut, Am un VPS cu FreePBX functional. Doresc sa adaug HylaFax + AvantFax si sa pot sa trimit faxuri prin iax sau sip. Am vazut ca ai facut ceva similar. 40$ fixed rate Poti sa ma ajuti?

    $23 - $194
    $23 - $194
    0 oferte

    Salut, Am un VPS cu FreePBX functional. Doresc sa adaug HylaFax + AvantFax si sa pot sa trimit faxuri prin iax sau sip. Am vazut ca ai facut ceva similar. 40$ fixed rate Poti sa ma ajuti?

    $31 (Avg Bid)
    $31 Oferta medie
    1 oferte

    Sunt interesat sa discutam pentru un proiect android, de voip (sip) asa cum este si viber sau skype... in primul rand ma intereseaza aplicatia pentru client dar si partea de server ca aplicatia sa fie functionala

    $30 - $250
    $30 - $250
    0 oferte
    Aplicatie android voip S-a încheiat left

    Sunt interesat sa discutam pentru un proiect android, de voip (sip) asa cum este si viber sau skype... in primul rand ma intereseaza aplicatia pentru client dar si partea de server ca aplicatia sa fie functionala

    $150 (Avg Bid)
    $150 Oferta medie
    1 oferte

    Ma intereseaza sa discutam mai multe legate de o aplicatie de voip pe android ca si viber , caut pe cineva care sa faca partea de client sau sa customizeze un c...intereseaza sa discutam mai multe legate de o aplicatie de voip pe android ca si viber , caut pe cineva care sa faca partea de client sau sa customizeze un client open source de sip existent

    $231 (Avg Bid)
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    1 oferte

    Ma intereseaza sa discutam mai multe legate de o aplicatie de voip pe android ca si viber , caut pe cineva care sa faca partea de client sau sa customizeze un c...intereseaza sa discutam mai multe legate de o aplicatie de voip pe android ca si viber , caut pe cineva care sa faca partea de client sau sa customizeze un client open source de sip existent

    $150 (Avg Bid)
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    1 oferte
    Aplicatie de voip (sip) S-a încheiat left

    Buna seara, as vrea sa discutam cu privire la o aplica...de voip in care utilizatorul sa se logheze pe baza de nr de tel si sa primeasca sms pt confirmare precum si prin facebook, sa aibe o parte din functiile viber, configurarea SIP sa se faca printr-un api fara ca sa trebuiasca sa faca ceva utilizatorul final pentru a configura la inceput aplicatia.

    $150 (Avg Bid)
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    Soft Phone S-a încheiat left

    ...Adevaratul OK va fi localizat, invizibil, altundeva. Telefonul se inregistreaza prin 3 parametrii: Server Username Password (campurile username si password devin hash-uri si nu pot fi vazute in clar) Telefonul functioneaza atat pe protocolul SIP cat si pe IAX2. Telefonul poate deschide oricate linii in paralel. Eventual, fiecare linie noua

    $250 - $750
    $250 - $750
    0 oferte

    You must have experience with unreal and XMP...false; [conectează-te pentru a consulta linkul 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 with normal client like pidgin with ssl

    $26 - $330
    $26 - $330
    0 oferte

    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.

    $377 (Avg Bid)
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    4 oferte

    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 4 zile left
    VERIFICAT

    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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    ...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 - Can be integrated into Laravel

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    Acord de confidenţialitate
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    ...program with contacts, task, calendar, email marketing. Extra – phone book, manual dial, auto dial. Multiple users. 2. Program has to be php friendly, located on own web/ftp server. Database is mySQL reachable form own CPanel. No any third party monthly payable api. Purchaser owns the full licence of program. Program is editable, customise-able – not

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    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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    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 [conectează-te pentru a consulta linkul URL], mas estou com erros nessa validação. Preciso de ajuda para terminar o mesmo. Quem...

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    Goautodial specific functions 2 zile left
    VERIFICAT

    ...any number in any list: show a different calling number on customer phone for each attempt. Will give you 10 calling numbers that we have 3) Set up load balancing 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

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    Digital logo 1 zi 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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    We need someone to configure our emails

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    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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    ...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 server has

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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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    1 oferte

    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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    ...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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    ...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, and encryption. This feature requires additional configuration at your web server. Instructions are

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    WebRTC / SIP expert 5 ore left
    VERIFICAT

    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. [conectează-te pentru a consulta linkul URL]

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    VOIP SIP CALLING SYSTEM S-a încheiat 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 S-a încheiat 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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    Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support

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    Looking for freelancer to configure FusionPBX with PowerPBX hosting. [conectează-te pentru a consulta linkul URL], [conectează-te pentru a consulta linkul 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 sc...

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    Trophy icon Logo design for "Livingston Depot" S-a încheiat 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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    Garantat
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    91 intrări
    Asterisk Expert S-a încheiat left

    Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support

    $17 / hr (Avg Bid)
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    6 oferte

    Help setting up initial OTRS community version on Centos 7 (VPS), configure outgoing SMTP-TLS mail. Linux Sysadmin, Perl, and previous OTRS experience required. ([conectează-te pentru a consulta linkul URL] )

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    3CX Gateway for Skype S-a încheiat 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. [conectează-te pentru a consulta linkul URL] [conectează-te pentru a consulta linkul 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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    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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    We need an implementation of the services mentioned in the subject. Thanks.

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    ejabberd configuration S-a încheiat 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 S-a încheiat left

    Have you tried using encrypted SMTP on port 465? I bet they won't detect that. [conectează-te pentru a consulta linkul URL]

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    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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    I am looking for an experienced FreeSwitch engineer who can work with latest freeswitch version with SIP Trunks to do following tasks I will be passing destination number , CLI , ring duration , max call duration , audio file , unique call id to Freeswitch (JSON API) , the dial plan should start dialing numbers and plays a voice file when the call

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    I have already built app on play store called "Loan Emi Calculator". Please review the app befo...Calculator". Please review the app before you bid, here is a store link.. [conectează-te pentru a consulta linkul URL] New features need to add 1. SIP Calculator 2. Update Emi Calculator for variable interest rates

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    FTP TLS project

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