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    18,599 asterisk sip tls робіт знайдено, ціни вказані в USD

    ...машини використовували спільну IP адресу з сервером pfSense для вихідного. Для вхідного трафіку відбувалося прокидання лише окремих портів на ці машини (приміром на сервер з Asterisk прокидаються порти 5060, 5061, 10000-20000 і т.д.) 3) всі віртуальні машини мають перебувають у спільній локальній мережі pfSense (з можливістю обмеження трафіку між окремими

    $125 (Avg Bid)
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    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

    $576 (Avg Bid)
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    We are developing an application to work with asterisk AMI. We need help with understanding the events emitted for different scenarios. We need consulting assistance for the solution design - no development is required at present.

    $150 (Avg Bid)
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    goautodial installation in vps 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    i want the installation of the goautodial v4 and the sip set up in my vps

    $31 (Avg Bid)
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    VOIP system implementation 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    I have a project for a VOIP system, would like to contact with a person who has experience with asterisk and GPON router

    $24 / hr (Avg Bid)
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    we want someone to make a simple change in the asterisk dial plan

    $131 (Avg Bid)
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    10 заявки
    Asterisk/Vicidial Configuration 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    We have FreePBX installed. We need a freelancer who can quickly configure it for us. - Configure 2xDID - 8 extensions - IVRs (I will provide recording) - Hunt group/Ring Group (example: tech support #1 -> call goes to all techs at once, whoever accepts) - Find Me/Follow Me - Voicemail - Music on hold - Conference call - Configure outgoing routes (Preferred outgoing DID, ability to select Out...

    $193 (Avg Bid)
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    FreePBX / Asterisk Consultant 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    we need a consult on the PBX architecture and then more specifically FreePBX FreePBX platform can stream real-time via a websocket to a cloud service for further real-time processing.

    $186 (Avg Bid)
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    Now we want to use linphone 4.1 mobile application with asterisk but chat message not working correctly. It has some problem like multiple dialogues in chat list for same number. If you can solve problem, Please make sure that you are a professional in sip and asterisk. You can try the software before of apply for this project. Just clone linphone android

    $291 (Avg Bid)
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    We need and expert with Opensips in order to configure and connect it to Asterisk PBX to configure Opensips for handling SIP messaging to SMS provider and vice versa. a complete Opensips solution to send/receive sms from peer-to-peer SMS provider. Encoding and splitting the messages by Opensips with following guideline: All software must auto start

    $22 - $185
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    ...zwingend notwendig, wäre aber wohl sinnvoll - Sicherheit auf Stand der Technik muss gewährleistet sein (Passwörter verschlüsselt/gehasht, evtl. 2-Faktor-Authentifizierung, https, tls) - Webseite muss muss möglichst Google-afin sein, bevorzugter Browser ist Chrome, dann Firefox - Es werden im Übrigen Multi-Chanel-Kanäle genutzt (Amazon, eBay,) vielleicht...

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    ...[увійдіть, щоб побачити URL] for more information I have a 1and1 Imap email account and I want to relay mail through that ideally (it must be port 587 TLS) and so I need sendmail configuring securely .So the setup has to comply with IPV6 mail sending guidelines. Alternatively we can use the servers own sendmail as a secure relay

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    Looking for someone who can change a asterisk dial plan Current setting , call connects and will play an audio message We want call to connect , asterisk will press 1 then play message

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    Dialogflow and Asterisk with UniMRCP 5 дні(-в) left
    ПІДТВЕРДЖЕНО

    I am done with asterisk and Dialogflow. I say "Who are you" and agent answer me "i am virtual agent". So i need partner to continue work with voice agents. Chalabi G if you interested - write me.

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    Project for Aneel M. 8 дні(-в) left

    Hi: i have installed Asterisk on Centos distribution on Samsung s7 edge (rooted) using Linux Deploy app . Now we have Asterisk on an Android mobile phone . As i saw many calling app , like skype or other calling app's , can give choices to user to call over SIP or internally through app or use GSM (local GSM network on mobile phone) if GSM option is

    $1500 (Avg Bid)
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    Trophy icon FaceBook Ad Image 1 день left

    ...catching phrase and shows a broken iPhone 7 or iPhone X screen. It needs to say that we fix broken iPhone screens using premium quality screens with life time warranty* (an asterisk by the "lifetime warranty). It also needs to say "iPhone 6 to iPhone 8 Plus screens from $69.95 to $99.95) Also fit in "We fix all devices". LIKE OUR PAGE FROM LIMITED

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    12 робіт

    Instalar Vicidial / Asterisk en servidor dedicado dejarlo completamente configurado: - 1 campaña - 4 extensiones - 2 truncales - llamadas entrantes y salientes El sistema PBX es para un call center

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    Adtran TA924e outbound caller id override 5 дні(-в) left
    ПІДТВЕРДЖЕНО

    I am working on configuring Adtran TA924e voice gateway to connect to a SIP trunk provider over the internet and have a few analog FXS stations call out and all show the same caller id (phone number). I have configured everything with the trunk, it registers, can call in and out, but cannot figure out how to override the caller id for all FSX ports

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    ...minimum of 20 calls at the same time, but preferably many more than that. - The calls can be terminated (sent to) a US phone number +1585-XXX-XXXX or to an IP address using SIP. We would be receiving up to 1,000 calls per day on this number. Each call would average less than 1 minute in length. We need at least 5 telephone service providers'

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    Sécuriser XIVO serveur 5 дні(-в) left
    ПІДТВЕРДЖЕНО

    Hello, It's a project for: 1- Asterisk/Xivo ipbx security 2 - Check on an Elastix ipbx, fix address book access, fix phone configuration bugs. See you soon Bonjour, Il s'agit de vérifier , analyser et sécuriser un IPBX Xivo et Elastix Cordialement

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    I have a personalized website based on PHP and ...use PHPMailer to send emails and notifications to my customers. The problem was that all my emails are bounced and need to configure PHPMailer to send the emails vía SMTP with TLS through our server. Thanks in advance! Include in your bid the comment "mailer", the offers without this will be ignored.

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    freepbx project , asterisk , phone system -- 3 4 дні(-в) left
    ПІДТВЕРДЖЕНО

    I tried to change the password on my FreePBX account however I'm getting a FAtal ERROR, do not know why and I can't log in anymore. i need to fix this.

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    ...collar hiring platform. As part of automation, we do robo-calling over several stages of the recruitment process. To enable this, we seek experts with experience in deploying Asterisk ([увійдіть, щоб побачити URL]) on the cloud with GSM / Mobile phones in India. The project is to be done in two parts. Part 1 - Training + Architecting (10 - 15 days) Based on the

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    ...a Cloud Hosting Service. We currently have about 50 clients using this system and have recently come across some items that we’re looking for a developer experienced with Asterisk-based systems, particularly with Issabel, that we can work with to improve upon our PBX solution. The primary items we’re looking for help with at this point in time are:

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    We are seeking someone who is Very Experienced in assisting us getting our VoIP Telco off the ground. We have wholesale access to SIP Trunks/Channels and DID's We need someone who is experienced in Softswitches, Cloud PBX, Website to PBX integration so customers can sign up online to our plans, self manage their own PBX. The system creates an PBX

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    Voip Whitelist Filter 3 дні(-в) left

    We need to design a voip asterisk call whitelist filter Principle of work: 1. Receive all calls comming from IP=.... if CLD (the called number) has a Prefix=..., and remove the Prefix. 2. Compare the CLD with the whitelist tables. 3. If CLD belong to whitelist table, then send the call back to IP=.... with Prefix=... 4. If CLD do not belong to whitelist

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    SYSADMIN LINUX AND WINDOWS - RADIUS SERVER WIN2019 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    Need build a Server Radius in windows server 2019 that work with mikroti...already have in freeradius and working fine ) who will hire for this project can work long term with us for maintenance of IT cloud infrastructure, other needs is: - VoIP asterisk freepbx - mysql / postgreSQL - WEBSERVER experience - Windows Linux better if Italian Language

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    Fix asterisk agi script 2 дні(-в) left

    Hi I need this agi script changed to an Asterisk Dial plan with timeouts added. See [увійдіть, щоб побачити URL]

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    Requirement of Goautodial and Asterisk expert for installation , inbound configuration & Support. Need to Configure DID in Ingroups

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    Build a PBX with Twilio 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    I am currently using Freepbx on the cloud as my VOIP service. I am wanting to build applications on T...Freepbx on the cloud as my VOIP service. I am wanting to build applications on Twilio Programmable Voice platform that will essentially behave as a PBX in which I can connect my SIP endpoints directly to. Please provide a demo of a similar project!

    $450 (Avg Bid)
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    I am currently using Freepbx on the cloud as my VOIP service. I am wanting to build applications on ...Freepbx on the cloud as my VOIP service. I am wanting to build applications on Twilio Programmable Voice platform that will essentially behave as a PBX in which I can connect my SIP endpoints directly to. Please provide a demo of a similar project!

    $250 - $750
    Місцевий
    $250 - $750
    0 заявки
    Interface requirement for Reporting - IVR/VOIP 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    Require asterisk development skills and php to complete front and back end. Interface must be functional with live traffic.

    $199 (Avg Bid)
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    OPENVPN OVER SSH/TLS TUNNEL FOR ANDROID 1 день left
    ПІДТВЕРДЖЕНО

    There are many openvpn apps in github for example, easyvpn or ics-openvpn You would have to build demo app that can connect openvpn through ssh or tls tunnel. we can provide server to be installed openvpn+tunneling proxy you can make demo app and you should provide full source with document. App should allow vpn for itself, otherwise you don't exlcude

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    Setup Sangoma PCI card for Asterisk 9 годин(-и) left
    ПІДТВЕРДЖЕНО

    Hello, Need to setup Asterisk server, with Sangoma PCI Card for asterisk voip server Carrier Siptrunk TDM with Audio Codec E1-PRI Best regards, CP

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    Website Development 8 годин(-и) left

    ...-Page Copy rights- Brochure design Content writing : For this website you have to write best in class. An example of sources will be on line you can go through the general asterisk PBX features and FAQ. Website should be look up in Google analytic and Generic Search Engine Optimization. With the best keyword search to be available in the first page

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    Phone system support Закінчився left

    Need a programmer familiar with asterisk, Pfsense, php, databases and able to help with assorted tasks and keeping things organized.

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    Ubuntu SSL certificate installation Закінчився left

    The Ubuntu server 16.04.2 has a SSL certificate already but a new intermediate certificate needs to be replaced. ...Ubuntu server 16.04.2 has a SSL certificate already but a new intermediate certificate needs to be replaced. Also a .pfx and private key need to be copied to a folder to get TLS working. Please provide documentation of steps taken

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    Configure Twilio Elastic SIP Trunking with FreePBX. Required to Share Screen Virtual and Troubleshoot together: I have followed all the steps in this link and my Aterisk is still not working: [увійдіть, щоб побачити URL] I think it could be a PHP5 error or maybe something I'm not

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    Hi, Asterisk: Store key pressed on External Database for both Call Back and Surveyed the agent as in the attached figure. we have Elastix 2.4, and we want the following: First: on the IVR there will be an option of call back so when caller press this option its store the caller id + Date + Time on the external MySQL database then play thank you message

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    asterisk calls drop randomry Закінчився left

    i m using asterisk to do voice trading. My acd is very low = 2-3 MIN I ssupect call drop random, but i don"t know how to solve

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    I need a custom windows desktop application for browser pop-up for Asterisk phone system. - the application will sit in window's system tray - Connect to AMI on Asterisk server & wait for events - "On call answer by my extension", do the following: --- Play audio --- Animate application icon to get the attention of the user --- Launch a new browser

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    Identity for Teleport café Закінчився left

    ...serves as a prelude to the trip. The main idea is to convey the Teleport in the shape of the takeaway coffee cup, which brings you to different part of the world with every new sip. I’d like to implement a very simple and understandable graphics, but, at the same, enrich it with interesting and attractive details....

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    Asterisk + PBX Admin Закінчився left

    I need a programmer with experience programming Asterisk and FreePBX and MUST know how to setup PBX to work with Twilio, [увійдіть, щоб побачити URL] and www.signalwire.com. I need assistance setting up / securing a PBX and then ongoing administration. I need a trunk built like this Host name – typically in the formation of a domain name. For example: [увійдіть, щоб побачити URL]

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    Asterisk on Andriod as Call app Закінчився left

    Hi: i have installed Asterisk on Centos distribution on Samsung s7 edge (rooted) using Linux Deploy app . Now we have Asterisk on an Android mobile phone . As i saw many calling app , like skype or other calling app's , can give choices to user to call over SIP or internally through app or use GSM (local GSM network on mobile phone) if GSM option is

    $2683 (Avg Bid)
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    i have a dialer installed but i dont know how to set up the sip trunk, so help me to set up the sip trunk and make a test call

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    Zoho's phonebridge plugin supports up to Asterisk 1.4 With modern asterisk versions (11 and up) it's not working at all do to (apparently) java problem. And the property in the phonebridge adapter seems like it is not being set properly. Zoho's support says that those values: dialed channel, linked channel, dialing channel are NULL instead of being

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    1 заявки

    We have fred module for WHMCS, all was working well until yertserday i updated certificate, it stopped working, it give ERROR: Error connecting to tls://[увійдіть, щоб побачити URL]: (code 0)

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    I am looking for an asterisk specialist for realizing this special project. A (SIP-)telephone should be placed in a small museum. Visitors can call up and listen to various audio announcements by entering a telephone number (two-digit? three-digit? tbd). My idea is to use a raspberry Pi 3 B+ and RasPBX on it. The installation is not connected to the

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    Build me an VOIP server Закінчився left

    Need a voip server where I can call on SIP clients App

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    Fix Phone System Закінчився left

    Knowledge base: Asterisk/FreePBX w. WebRTC, Apache Web Server, NgInx reverse proxy, MySQL DB, building Web Sites, Ubuntu OS. -- All is well documented. You should have a computer with late Ubuntu and a usable Teamviewer installed. This is not for beginners ! -- If You think You can find everything on the Internet, forget it and let's not waste each

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