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    18,216 asterisk sip tls ocs travaux trouvés au tarif de EUR

    Live call monitoring in Asterisk.

    €88 (Avg Bid)
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    1 offres

    I have a bunch of phones ...firmware update when they are factory reset. I have a DHCP 160 option pointing to HTTPS server for provisioning. Firmware is also on that HTTPS server but my phones connect to my SIP server correctly and ignore the firmware update task. I need help getting my phones to force firmware upgrade when they come up and provision.

    €9 - €26
    €9 - €26
    0 offres

    We're looking for a developer to join our team and help with daily tasks. Required skills: - SIP and VOIP in general - Asterisk - PHP & Codeigniter framework - SMPP - GIT Please share quick brief of your past experience related to skills listed above, your availability & time zone you're working in. Your written English must be good enough to understand

    €1991 (Avg Bid)
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    22 offres

    ...[se connecter pour voir l'URL] 1) Login Entry 2) Login Response 3) Normal Order Entry 4) Normal Order Entry Response 5) SIP Order Entry 6) SIP Order Entry Response 7) X-SIP Order Entry 8) X-SIP Order Entry Response 9) Spread Order Entry (Overnight) 10) Spread Order Entry Response (Overnight) 11) Switch Order Entry 12) Switch Order

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    Looking for help troubleshooting the setup for CartoDB on a production server with ssl/tls/ We have an Ansible script that gets everything set up based on [se connecter pour voir l'URL] but configured to run in production mode for ssh/tls/ The problem is there are various intermittent bugs that keep popping up and we need someone

    €17 / hr (Avg Bid)
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    Hello i need FreePBX + Asterisk + WebRTC websocket config on linux server.

    €210 (Avg Bid)
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    ...IP Address (IP Tables)? -Disable PHPmyAdmin from External – Disable phpmyadmin running on all ports, 8080? 80? 8043? etc -Configure SSL/TLS servers to only use TLS 1.1 or TLS 1.2 if supported.<br/>Configure SSL/TLS servers to only support cipher suites that do not use block ciphers. Apply patches if available.<br/><br/>Note that additional con...

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    Asterisk Developer 6 heures left

    We are looking for a full time asterisk developer. Job Location, Rohini East, Delhi, India

    €2518 (Avg Bid)
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    7 offres

    Application key features: * high definition audio and video calls * audio c...sharing * accounts reachable at any time, even if the app is closed (thanks to push notifications) * secure communication (encryption options) * compliant with a large number of SIP-compatible VoIP service providers allowing to reach everyone that has a "classic" phone line.

    €957 (Avg Bid)
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    7 offres

    I need a Homer Seven server and ELK servers build to support SIP capture from Asterisk services. I need someone that works with SIP

    €175 (Avg Bid)
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    1 offres
    FreePBX Setup 3 heures left

    We have 4 Digium D60 in office with a VPS that has FreePBX installed off site. There is VPN from phones to FreePBX. We are using Twilio for SIP. -Setting up extensions for all phones (4) -IVR -dial out rules -voicemail settings -inter office transfer setup -instructions on how to setup a fifth phone in the future (conference room phone) Start your

    €21 / hr (Avg Bid)
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    VTiger / sugar CRM 2 heures left
    VERIFIÉ

    Installation customised VTiger/ Sugar CRM on Asterisk Vicidial 8.1

    €313 (Avg Bid)
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    7 offres
    Set up Cisco phone to sip S'est terminé left

    Need to set up a Cisco phone to convert to SIP to use with fpbx. Only person with direct experience in it need to bid.

    €28 (Avg Bid)
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    6 offres
    voice broadcast dialer S'est terminé left

    I am loo...who has built their own voice broadcast dialer. Dialer must be able to create Press 1 campaigns and survey campaigns. Need full cdr reporting and dtmf capturing. Based on asterisk as well with admin panel to create sub users and load accounts. Again only reply if this is a platform you have built and is up and running with a demo account.

    €826 (Avg Bid)
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    ...this solution should be compatible with the Asterisk dialplan. The text that I analyze is always the same and the same words, but they appear at different intervals of time. Therefore, we must constantly analyze the voice and run of the appearance of the phrase we need to carry out the next stage of the Asterisk plan. Perhaps one consultation will be enough

    €172 (Avg Bid)
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    6 offres

    Hi We are looking for freelancer who is aware of complete integration and workflow of GrandStream HT813 with Issabel. Issues to resolve Call Progress Dial Tones to be set for Airtel (India) Call disconnection (FXO Termination) to be set Other settings Secure Connection between device and server Complete Trunk related settings in Grand Stream and Issabel (Elastix) Only bid if you have complete ...

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    Asttp Support (Freeswitch) S'est terminé left

    ...following error: 2019-01-08 01:07:11.651234 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;maddr={local_ip_v4};transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC) 2019-01-08 01:07:16.671190 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;m

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    i need a software to remove or change SIP RTP PayLoad like if i install this software on os it have to remove or change all incoming and outgoing SIP RTP PayLoad if you can make it please let me know

    €487 (Avg Bid)
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    Mobile VoIP Dialer S'est terminé left

    Build a VoIP dialer app for Android/IOS and WM using either Linphone or Csi...Account auto provision/Signup via REST. - Account Balance Check Via Rest. - Account Topup via rest.(In App Payments) - Multiple Sip registrar options (Selectable) - User Credit display in app. - Voice/Video/Chat support - *** SIP Header Compression support! within the app.

    €260 (Avg Bid)
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    ...and can’t sent us any email. We think that this problem is coming from different elements. SPF and dns setting in our primary dns servers, lake of SSL certificate in Cgpro (TLS), CGpro wrong or incomplete relaying settings, maybe something else? Our environment : CGpro (Stalker communigate pro server) on linux [se connecter pour voir l'URL] is a secondary dns server

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    We require a UK local rate telephone number to be diverted to a UK mobile phone number. We would like to use Asterisk to create our Voip System. [se connecter pour voir l'URL]

    €238 (Avg Bid)
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    15 offres

    We need to integrate an Asterisk with Jerasoft Billing by Radius [se connecter pour voir l'URL]

    €215 (Avg Bid)
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    10 offres
    solución pbx S'est terminé left

    Tengo una centralita pbx con issabel y 2 troncales sip con gateway goip en redes difefentes pero siempre se desconectan y tengo que reiniciar el servidos pbx para que se vuelvan a restablecer

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    Design project S'est terminé left

    Sip and paint artist needed for one week event

    €113 (Avg Bid)
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    need hobby geeks S'est terminé left

    Looking for an On the Call Technician, who has some ex...6. Basic Computer Knowledge 7. Have a laptop or desktop with windows 7 or higher. 8. Photo ID 9. Paypal account for payments 10. Headset Optional but preferred. for windows sip phone, or android, iphone 11. willing to be creative, and outside the box thinker 12. Honest Contact me to Learn More

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    Hi Dear In running local FreePbx server under elastix and the server have 2 network card One card link to Sip line from local Telecome company and another network card for internet connection My local server is online now via port forward from the router and it’s working by Zoiper app very fine in computer an mobile... in local network and online

    €11 / hr (Avg Bid)
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    Hi Dear In running local FreePbx server under elastix and the server link to Sip line from local Telecome company and there is another network card for internet connection My local server is online now via port forward from the router and it’s working with Zoiper app very fine in computer an mobile... But the problem its just working with Zoiper nothing

    €22 / hr (Avg Bid)
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    1 offres

    I am looking for someone who can build a sip calling APP using a Mobile Hybrid platform such as ionic or phonegap or any other Mobile Hybrid platform. the requirement is to connect to our SIP servers and make calls through a SIP proxy. we will provide all the server details needed

    €1075 (Avg Bid)
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    30 offres

    Would like an Asterisk devoloper who can build it and integrate a CRM into it.

    €1970 (Avg Bid)
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    25 offres
    I Need Vpn Reslaer Panel S'est terminé left

    I need a ocs Vpn Panel with this all function i mention Ocs Vpn Panel Rebuild with Php 1 Vpn Accounts 2 Add Vpn 3 Take Trail 4 History 5 Servers And after setup with anyconnect vpn i have vpn script Who Is Really Know About This Then Contact us Otherwise Dont Waste Time...

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    Description. A simple PBX pure software would have to be [se connecter pour voir l'URL] software should be installed on Windows Caller calls the number to Will and can select appropriate department through IVR. The language should be stored under MP3. Customer lists to maintenance music + and waiting position. Once customer is connected with the agent and the agent wants to end the conversati...

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    ...page html site. Contact form has 3 variables (name, email and message) All three should be formated to a descently formatted email and sent via smtp with authentication and tls. The site also has a subscribe option for integration with Mailchimp (haven't tested that yet but it looks straight forward to enable as per theme documentation.) Also, I

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    I need a website which is based on wordpress for hiring people. The website have to be using SSL/TLS certificate. People should be create account on my page, then confirm their mail through the own mailserver. Then they can write with me through ticket system and upload their documents (e.g. CV, photo etc...) I should be can edit the member within

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    Asterisk developer S'est terminé left

    I have work related to asterisk programming for our product

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    Proofread a chapter in TLS S'est terminé left

    I have written a book about the thin prediabetic for the general public. I have tried to make it accessible, but need help in making it flow and generally accessible. Because a proofreader has no idea of my writing ability it is difficult to judge how long it will take. I have already put the text through [se connecter pour voir l'URL] to wring out the obvious problems. I would like to put ou...

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    3CX configuration sip trunks S'est terminé left

    i need to configure 3cx and add sip trunks

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    10 offres

    I want to add new SIP domain in skype for business and make it working.

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    VoIP SIP Intercom Module S'est terminé left

    ...Speex (narrow/HD/Ultra HD), SILK, L16 PCM (narrow/HD/Ultra HD/CD) Data Flash Management (Multi-Partition File System, Advanced Firmware Management, Bootloader, Secure Area) SIP Fusion TCP/IPv4/IPv6 DHCP Client Extensible Command Line Interface Web Server with Extensible Configuration Web App and Configuration and Control Web Services VoIP Call Manager

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    Art History Essay S'est terminé left

    ...Today our fans want to ask you a few questions about some of your paintings and the media you have used. Daumier: Well of course. Daumier picks up his water and takes a small sip. Presenter: The first painting our viewers are interested in, is your print Gargantua. Daumier clears his throat. Daumier: Ah yes, one of my finest caricatures. Presenter:

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    We use Asterisk & FreePBX for our call center. Need to get actual hold time into our cdr db. We use Polycom 450/550 phones. Need to log any time a client is put on hold. (When the Customer Service presses that button). There has to be a way to track this via log files, etc. Since music is being played to the client. Client Calls in > Queue > Customer

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    ...general familiarity with backend frameworks likes Rails, Django, Phalcon, YII, etc. Fluency in SQL Databases, NoSQL Databases, Cloud Storage APIs, etc. Fluency in TCP, XML, SIP and other web protocols & patterns. Experience with building secure web applications, troubleshooting, well scalable applications, etc. Extensive experience with API architecture

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    ...passwd test Step 3: Fix thie 2 bugs: 1) An error occurs when connecting to the server with "Active" mode and SSL or TLS encrypted connection Example: Configure FileZilla Ftp Client with: - Transfer mode: Active - Encryption: SSL or TLS => Error detected: Unable to connect to the server (with a login and password) with this configuration. # Active

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    Dears Developers We are ruing Sip service under FreePBX platform And we use [se connecter pour voir l'URL] service to keep our server online using router port forward . So we need app with SIP protocol support to make viop calls with our Labled & Customization with Full Branding Mobile Dialer support IOS iPhone and android mobile. Mobile Dialer FEATURES 1-User friendly

    €588 (Avg Bid)
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    Asterisk Zoho Integration S'est terminé left

    Need to integrat Zoho with Astersik with Features like Click to To Call and Contact Alert for incoming Calls both on phone and PC.

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    8 offres

    I need you to integrate my CRM software with asterisk voip system. The CRM is ASP.net and you should know about asterisk phone systems integration

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    ...hire new people for the NOC, so ongoing training is important to us. Here are some suggestions as to what the training should cover. We want to focus on Voip and SIP. Why Voip SIP - protocol structure, format, Commands, how to read it, Error codes, interpreting the error codes. Under Voip introduction to Voip Why use Voip over Analog How

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    the scenario; - I have sip phones registered to my ip pbx - when a sip phone try to call a mobile/landline - my pbx will route the call to "skype gateway" - skype gateway will use a skype account to connect the call transparently sip phone -> ippbx -> skypegateway -> gsm/landline inbound calls to be handled with ip pbx the project should be a windows

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    I am looking for a linux server administator for the following: 1. Install Virtualmin on CENTOS 7, install Let's Encrypt - Free SSL/TLS Certificates on the server, install MYSQL server. 2. To fix the problem of an old server which refused to restart, it is possible "there is a problem with the boot file". The server is now in a rescue mode. This old

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    Integración con Asterisk S'est terminé left

    integrar Asterisk con otras tecnologías

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    Intercom SIP APP Android S'est terminé left

    ...Android APP that works as a SIP Intercom... A big botton on the main screen the show (CLIC TO CALL ) when clic it will make a VOIP (SIP) call to an espesifi extencion, also if I call the extension it needs to autoanswer. on the config menu I need to be Avaliable to change this values Sip server IP: Sip Port: Sip User name: Sip Password: Number to c...

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