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    12,351 asterisk tls srtp proiecte găsite, la prețul de USD
    Project for Mucahit T. S-a încheiat left

    Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta

    $1000 (Avg Bid)
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    Project for Macovei M. S-a încheiat left

    Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta

    $250 (Avg Bid)
    $250 Oferta medie
    1 oferte
    Project for Ionut R. S-a încheiat left

    Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta

    $1000 (Avg Bid)
    $1000 Oferta medie
    1 oferte
    Project for slk S-a încheiat left

    Salut, am instalat goautodial care include si asterisk, am reusit sa trec peste mai multe probleme succesive pana m-am impotmolit din nou. Cand incerc sa initiez apeluri imi spune ca nu este prefixul corect. Eu cred ca este din Dialplan Entry:, m-am uitat si la logurile CLI dar nu gasesc nici o solutie. Ma poti ajuta? Am un abonament de siptrunk de

    $30 (Avg Bid)
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    Project for MikeRRR S-a încheiat left

    Salut, am instalat goautodial care include si asterisk, am reusit sa trec peste mai multe probleme succesive pana m-am impotmolit din nou. Cand incerc sa initiez apeluri imi spune ca nu este prefixul corect. Eu cred ca este din Dialplan Entry:, m-am uitat si la logurile CLI dar nu gasesc nici o solutie. Ma poti ajuta? Am un abonament de siptrunk la

    $30 (Avg Bid)
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    Project for ionut0709 S-a încheiat left

    Configurare campanii, db, si securizare asterisk, goaudiodial, vtiger.

    $150 (Avg Bid)
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    Salut poti lasa un icq jabber yahoo daca ai te rog. Am nevoie de ceva pentru asterisk................

    $1500 - $3000
    $1500 - $3000
    0 oferte
    FreePBX Project S-a încheiat left

    Salut poti lasa un icq jabber yahoo daca ai te rog. Am nevoie de ceva pentru asterisk................

    $1500 (Avg Bid)
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    NOTICE that you with you bid must deliver a detailed description of how you will solve the task and which software it includes, and why this solution will work (preferably with a link, that supports this claim). Today we have one active PBX server, which receives around 500-1000 calls per day through 200 DIDs. As we within the next few months expect to upscale our business, so we daily will recei...

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    We are looking for someone who can setup webcalls in our current system, see more here [conectează-te pentru a consulta linkul URL] (mellem 0:10-0:30) Today our agents receive the calls in Bria and either transfers them (hard or soft) to the employee or leaves a mesage, we want to be able to do this in our new system. Our agents can be either online, and if they are not, then they must be able t...

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    incoming calls on a DID. Caller gets chacked from a MsAccess database for service access and a dialtone to enter a destination number. Call must get transfered to destination and a log of the call length kept in a MsAccess database.

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    Se dispone de un servidor WebRTC Janus o Asterisk en linea. Se desea obtener una página web con un marco que realice la conexión webRTC a un destino mediante el servidor Janus o Asterisk y establezca la videoconferencia. La página web debe tener tambien un botón para llamar y otro para cortar. La página debe funcionar con sistema operativo Android (...

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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 - Has functions to call and return

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    Acord de confidenţialitate
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    Me gustaría instalar este certificado de seguridad en mi pagina [conectează-te pentru a consulta linkul URL] : [conectează-te pentru a consulta linkul 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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    ...io"; [conectează-te pentru a consulta linkul URL] = "[conectează-te pentru a consulta linkul URL]"; [conectează-te pentru a consulta linkul URL] = 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...

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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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    ...ultraschnelle CRM-System, dass wir zum Herzstück unserer Firma ausbauen wollen. Derzeit ist die Situation folgende: Wir haben auf der einen Seite eine Telefonanlage in der Cloud (Asterisk). Das klappt zuverlässig und bleibt. Auf der anderen Seite haben wir ein Cloud-CRM System. Dieses System muss auf Grund diverser technischer Schwierigkeiten durch unser eigenes

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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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    VoIP termination solution 4 zile left
    VERIFICAT

    We are termination provider we have a specific solution ...on/off feature Server C: register gsm gateways using pptp vpn and load balance outgoing traffic to gateways based on number of calls For server A we have a2billing and asterisk in mind And for server C we want opensips with gui installed. When we finish we must test call from one of our .

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    We have an android app that makes video calls from our system to the users mobile phone. It currently supports up to android 7.0 We need someone who is proficient in Asterisk and Linphone Client We would require the app with upgraded compatibility with android 8.0 and 9.0 phones.

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    Asterisk AMD 2 zile left

    Need AMD configured in dialplan for asterisk

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    I want to make real use of Asterisk Background detection , want a very small simple dialplan and configurations to Use BackgroundDetect of Asterisk

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    ...an open source bounty. There are some who have gotten this to work with other providers: [conectează-te pentru a consulta linkul URL] There are a variety of ways to interact with Twilio's service. The goal is to be able to tie FreePBX to it in a way that doesn't break upgrades to core, etc...Functionality

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    php mysql schedule creation 1 zi left
    VERIFICAT

    I have a server running with mysql and php, I am using it for asterisk based server for voice broadcast. I need help to resolve some issue and enhance its features. The out look will be like attached picture. Details will be discussed during chat.

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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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    Hi Aqs Y., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I am looking for some auto dialer based on asterisk

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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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    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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    Hi Kingasterisk T., We have a problem with our asterisk configuration. as a brief when call directed to extension queue after ivr. there is silence and no result. caller cant reach to extension. as example we get this kind of logs. [2019-02-14 08:56:15] WARNING[7089] chan_sip.c: Autodestruct on dialog '223205511-3759123182-18707@TcomNextoneSBC4-DRC

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    I'm new to FusionPBX/FreeSwitch. Can't set up the same thing I've set up in minutes in Asterisk. Need remote assistance/general consulting or just a setup example in accordance with my infrastructure conditions. My FusionPBX is a VM that has only one LAN NIC. It uses pfSense VM as a gateway. pfSense VM has both LAN and WAN links with a public IP on

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    I Need help with a vicidial server that is not leaving vvoicemail message and is dropping calls in a press 1 campaign

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    We have developed a voice bot that understands vernacular languages of India. We are looking for a developer who can help us integrate a user call with our voice b...asks a question - e.g. what are the current offers? The bot responds: Buy goods worth Rs 5000 and get cash back worth Rs 1000. Please get in touch immediately if you know Asterisk well.

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    9 oferte

    We have developed a voice bot that understands vernacular languages of India. We are looking for a developer who can help us integrate a user call with our voice b...asks a question - e.g. what are the current offers? The bot responds: Buy goods worth Rs 5000 and get cash back worth Rs 1000. Please get in touch immediately if you know Asterisk well.

    $2912 (Avg Bid)
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    10 oferte

    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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    AMD detection in Asterisk S-a încheiat left

    I want someone expert in asterisk who can fix my AMD detection in Asterisk, I'm using Vicibox and running Press 1 Survey Campaign.

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    im looking for someone to make landing page simple nice looking landing page have form and need image

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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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    We have a call centre system using Asterisk. With Asterisk only supporting the following call distribution. We need to tweak the queue behaviour to reset agent idle time in two event, after agent has hung up an incoming call and after agent end pause. If you base on the ring stretagy of least recent, we can achieve the agent idle time reset on the incoming

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    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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    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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    Asterisk Problem S-a încheiat left

    hello i need to stream live calls from my asterisk more or so like chansky but i need to stream in demand and to wav files so i can play it in brower i am useing freepbx

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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 are looking for an Asterisk expert who has experience installing Kamailio or similar software to be used as an outbound proxy for NAT Traversal front ending multiple Asterisk/FreePBX servers. Some PBXs are having load spikes due to frequent device registration/authentication and NAT traversal functionality, so we're looking to install a solution

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    Add ADI input to Asterisk frontend. Thank you!

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    Need to develop end to end solution of call center.

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    freepbx FOP2 S-a încheiat left

    i need help fop2 its installed on a standalone (separate) server than Asterisk Freepbx i can get the recordings to work.

    $30 - $250
    $30 - $250
    0 oferte
    freepbx fop2 S-a încheiat left

    i need help fop2 its installed on a standalone (separate) server than Asterisk Freepbx i can get the recordings to work,

    $43 (Avg Bid)
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