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    2,000 asterisk pbx fedora jobs found, pricing in NZD

    i need filter on asterisk about did and clid number. We are using this project on maniplation temination voip traffic

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    looking for someone to install FREEPBX 15 on Centos7

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    I have several FreePBX / Grandstream PBXs, hosted and on-premise, I am looking for a monitoring system that can help me with the following: 1. Automate test calls and verify two-way audio 2. Detect packet loss, jitter, or latency issues 3. Monitor SIP and RTP packets 4. Monitor SIP registration for specified extensions / trunks 5. Alerting capabilities 6. Does not overuse pbx cpu/memory

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    Hi Alexander, y noticed you finish a project named: Free Pbx IVR survey module. I am interested in the same project with some changes. I would need to see screen shots of the project in order to discuss with the responsable of making the software work. It is my understanding that the project uses phone keys to respond the survey, I would also need voice recognition availability to respond the survey. Please let me know if you are interested. Sincerely A. Ayala

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    ...transferred to security. Proof of concept (POC): A mobile app (Android and iOS) to demonstrate your firm’s competency to create a two-way audio/video call from the smartphone to our target Entry Phone. The target uses ARM architecture, runs Pine64, an embedded platform and has Node.js based API running on Linux. Deliverables for the POC: We would like to see a SIP VOIP call placed through our Asterisk server between the Entry Phone and the app. Establish a webRTC to webRTC connection having one-way video and two-way audio streaming. This call will be initiated from the entry phone to the resident’s mobile app where residents can see the visitor and be able to talk to them. Establish a webRTC to webRTC connection having two-way audio VoIP call for offsit...

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    Hello i have asterisk with did where 500 users call this did number we hava to create a routing logic where we will interconnect them randomly inside dynamic conferance without pin if in the conferance caller or callee press 0 we will interconnect them randomly to any other conferance without pin in case the user have no any user available we will interconnect hem to any active conferance else if there is no other user we will play MOH until the any conferance will press 0 we will set a timeout for the conferance 120 secs this is small job but i will have many others

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    Would like to connect Zoho CRM with local Israeli accounting software, local Payment GW as well as loal PBX via API.

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    Would like to connect Zoho CRM with Local israeli accounting software via API as well as Payment Gateway and local PBX provider.

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    Hi, I'm in need of either a Discord bot, wordpress plugin, or a webpage I can integrate into my current site that users ..."spectate" the stage strike/select process of any given matchup. Lastly, it is very desirable if the system is graphic based. I can provide the images for each stages available, but would be required for mod to update stages available. For now, placeholder text can be used: *Stage 1, *Stage 2, *Stage 3, *Stage 4, *Stage 5, *CStage 1, *CStage 2, *CStage 3, Player 1, Player 2, Room Code, Winner Status (asterisk items will be graphical) Would finally need to know from the individual chosen for this project, which method would be best to produce this in, Discord Bot, wordpress plugin, or web page Willing to work closely to get this done. Looking fo...

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    asterisk web interface needed for click and call

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    Hello, we are a new little italian isp provider company. we are looking to build our own CLOUD PBX SYSTEM with MULTI-TENANT. i hope someone can help us! Regards

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    we need a developer that should have these experiences: Modules. GSM Modules .GSM Channels, GSM Gateways, rSAP -> Remote SIM Access Profile, rSAP over TCP,,SIM Banks ,SIM Servers the developer will develop the software will run GSM Gateway and SIM Bank using Asterisk and GSM Modules, Hardware and software knowledge required. payments terms . please read it , will will not release any milestone until job is done , not negotiable

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    we need a developer that should have these experiences: Modules. GSM Modules .GSM Channels, GSM Gateways, rSAP -> Remote SIM Access Profile, rSAP over TCP,,SIM Banks ,SIM Servers the developer will develop the software will run GSM Gateway and SIM Bank using Asterisk and GSM Modules, Hardware and software knowledge required. payments terms . please read it , will will not release any milestone until job is done , not negotiable

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    We are looking to support SMS on our MiRTA platform (Asterisk). We also want our clients to be able to utilize SMS using the Zoiper app (open to other options as well). MiRTA is the GUI built on top of Asterisk. We will want the SMS options and features built into the MiRTA GUI for easy deployment and provisioning (we have access to the MiRTA developer). Please provide a time estimate. The budget is suggested and can be adjusted depending on your expert opinion of time. Please provide feedback. Thank you!

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    For custom made Invoices to bill clients for incoming calls through our Asterisk PBX and Credit Notes to pay telephone agents for their work. Experience needed: 1) LibreOffice Calc spreadsheets 2) Software to change Calc templates to pdf without loosing calculations (Foxit Reader ?) 3) Linux 4) PDF 5) PHP We use background images to show invoice and Credit Note Templates to the user. To fill the Templates out, on our server, the user only sees the image, the spreadsheet is invisible ! This project is not for beginners !! You must have a PC with a Linux OS (pref. Ubuntu). ----- Good and quick communications are essential !! - If You check Your mail only once in a while, You don't qualify. This project will be divided into unequal milestones. See attachment.

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    I need guidance on how to setup a blank sim card to connect to asterisk server and allow user to set caller ID

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    We are looking for VOIP technology expert to prepare streaming Asterisk and Dialogflow integration. We intent to implement a voicebot that realizes outbound calls basing on PBX (Asterisk), Dialogflow, Google ASR and TTS. Streaming Asterisk and Dialogflow with ASR is the most important issue in this project.

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    I need to run the my voip server with Asterisk. I also want to run the x-lite softphone from the VPS as well. What I need: Set up Asterisk on my server. Configure my SIP account from my number provider to voip vps. Configure

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    PyWebWiev window and icon bar (Fedora, RaspberryOS) I will order the program to be made only in Python. Without using the framework Qt, Wx, and others. The program will run on Linux (Fedora, Debian, RasppiOS). The program code should contain the necessary comments in accordance with good programming practices. The "lambda" function should be avoided in the program. The program should save the settings or other data necessary for its functioning - to a file in accordance with the JSON format. The appearance of the icon bar and other windows should be elegant and modern, like the Dock in the new Gnome, the Dock in apple, etc ... The program should include a description of the installation of the necessary Python libraries. Your job: - It...

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    We are looking for a skilled developer or team to develop a simple one click remote support for our team and customers with main features: - one ...scratch according to your preference as long as it includes the expected , I wish to mandate you to take care of my private server hosted at OVH in public cloud, I set up and installed the solution "GoAutodial" in v4, unfortunately I meet bugs with "Kamailio" on the telephony part in webrtc in the browser. I explain myself, everything works correctly, the SIP is well connected in asterisk and I can make automatic calls on the campaigns, unfortunately one call on two I do not hear sound for the interlocutor. I tried everything to debug it, I can't manage it. I don't think it's a big deal, I'll let y...

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    Hello, I want to migrate a CentOs Vps server from OVH to Digital Ocean. This little server have a automatic asterisk autocall system for security system. We want to keep all functions and add another ; - Automatic backup. - Firewall. - Check asterisk autocall system who have a problem. Thank.

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    Configuratiion Push Server to wake up our mobile SIP dialer to a pBX

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    press prefix 9 or 90 it should take sip line of yeastar S300 PBX and call for Outgoing

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    PHP based web application to manage asterisk queue and channels

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    I have a free PBX platform that I need to modify its press # options. It's on Asterisk VPS server. Dev will update current contents of the freePBX and point backends to url php scripts or 3rd party DID#s while preserving the current contents of the PBX. Individual developer/programmer prefered, no team or groups. (Removed by Freelancer.com admin)

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    I need help with Hubspot CRM PBX CTI Integration. I am looking for abilities below. Click2Dial Call Recording in CRM with each call Call logs with all details Incoming and outgoing Call pop up with details like Name , Account Names, Call Duration and call status

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    hi. i will need a crm system based on vtiger with integration to fusionpbx dialer. so when the lead is shown to agent using the vtiger crm the client phone number will be masked (the agent cant see the phone number) but when he press that masked number the crm will send a request to the pbx syatem to dial that number.

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    For custom made Invoices to bill clients for incoming calls through our Asterisk PBX and Credit Notes to pay telephone agents for their work. Experience needed: LibreOffice Calc and Linux We use background images to show invoice and Credit Note Templates to the user. To fill the Templates out, on our server, the user only sees the image, the spreadsheet must be invisible ! The Templates are divided into 3 parts and 2 have spreadsheets with different cell addresses ! You must have a PC with a Linux OS (Ubuntu). ----- Good and quick communications are essential !! - If You check Your mail only once in a while, You don't qualify. *** See attachment.

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    For custom made Invoices to bill clients for incoming calls through our Asterisk PBX and Credit Notes to pay telephone agents for their work. Experience needed: 1) LibreOffice Calc for spreadsheets 2) Software to change templates to pdf without loosing calculations (Foxit ?) 3) Asterisk PBX / FreePBX 4) Linux 5) MySQL DB 6) PDF 7) PHP 8) KVM qcow2 (for disk management) We use background images to show invoice and Credit Note Templates to the user. To fill the Templates out, on our server, the user only sees the image, the spreadsheet is invisible ! This project is not for beginners !! You must have a PC with a Linux OS (Ubuntu). ----- Good and quick communications are essential !! - If You check Your mail only once in a while, You don't qualify. This ...

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    I want to make my PBX outbound SIP calls to pick the next available SIM slot in cycling manner if the previous is busy, Setup SMS sending & receiving using SMS API, Call Credit Limit management, SMPP for SMS. Experienced freelancer with dinstar uc2000-vg can bid

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    i need someone to setup freebpx + asterisk on a cloud server, need asap.

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    We need you to install and setup an Asterisk PBX or another free PBX from scratch. You will also have to setup FOP on it, do that i can manage the PBX from there (you can refer to 3CX Management Console and Queue Manager Account functionalities to better understand which functionalities i need). - The PBX will have to record the CLI but normal extensions won’t be able to see it - Normal extensions won’t be able to dial external telephone numbers - Extensions are remote, so we will have to be able to configure those on Gigaset VOIP phones and SIP software phones as well - Extensions will usually be simple PBX extensions (i.e. 595) but they could occasionally be external numbers (in the last case scenario, we will need to keep...

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    Hello, I'm starting a voip services company, and I need to build an pbx server asterisk + web (back end and front end), with all the asterisk features, I await proposals.

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    Calling Without Internet Connection, We wanna build a technology stack for calling without internet. Using services like Jitsi which allows users to connect to conferences from regular phone lines, we are currently using services like matrix-org, riot im & jitsi. Services like asterisk-org and freepbx can used or similar open source technology can be suggested. Services like rebtel already provide this feature in their suite, we want similar solution.

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    We have an Askozia (Asterisk) PBX in a nursing home with only one incoming number. Currently caller hits the IVR "Enter the extension you require" and is connected with that room. We need to automate the process, such that if caller's CLI only ever calls the one target EXTENSION, then after 10 calls, IVR will ask for confirmation (from caller) to forward all future calls the same extension. Asterisk scripting can be inserted via the GUI.

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    I have a DX650 and i am trying to make it work on FreePBX asterisk, i need an EXPERT in Asterisk and in CISCO IP PHONES who Has already done this before with success.

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    We're a small business that has multiple development projects ongoing. We're an Atlanta based company and are looking for a local developer that we can work with. We're in need of a Full Stack Developer that has experience in: • LMNP (Linux, Mysql, Nginx, PHP) • Redis • Laravel • REST APIs • Vue.js • Node.js • Express • MongoDB • • Docker Pluses: • Asterisk • VoIP • SIP • Networking • Linux

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    Linphone for iOS and Android is up and running (able to to receive push, background calls and SMS) Our developers can provide some support for Mobile Applications and asterisk, they have limited knowledge related to push notification. - Flexisip must be installed and integrated and all modules must be in working - Flexisip must be able to communicate with Asterisk to send or receive SMS (custom work) - Flexisip must be able to send push to iOS, Android Devices Please do not bid if you don't have good understanding of the project - Flexisip Overview - Flexisip Wiki Wiki

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    Hello I need a Fedora 31 dhcp server and also a routing mecanism for The sever. Best Regards, Aurelian

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    Hello, I want to be able to offer my subscribers voice, video, and fax. I would also like iOS push notifications for chat messages. I already have a webhook with Twilio for SMS. I have many projects that you can choose and i pay very well for good work. I’ve been told I should get pbx system but I don’t know enough about that software. So if you are available, we can discuss specific project. If you’re busy I understand.

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    we are in need of an expert in Linux Armbian that can compile and application in to the Armbian a good Knowledge of asterisk will be helpful

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    we need an expert in Linux Armbian and asterisk to compile rSAP application with Armbian

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    create NEW Linux systems for VMs created using Fedora 24 under VMware Workstation 15

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    ...iptables experience (minimum 5 years) - NAT, mac filtering, routing 3. QoS experience (minimum 3 years) 4. Networking (routing, switching on Linux, dhcp, bind) experience (minimum 3 years) 5. Chef/Puppet/Salt/Ansible 6. Scripting - Python/bash 7. Apache/Nginx 8. VLAN experience (minimum 3 years) 9. Docker experience (minimum 3 years) 10. Bridge utils or openvswitch experience (minimum 3 years) 11. Asterisk experience (minimum 3 years) 12. URL filtering (squidguard or similar) experience (minimum 3 years) 13. Deep Packet Inspection/Layer 7 filtering experience (minimum 3 years) 14. KVM experience (minimum 5 years) - libvirtd, qemu-kvm, SPICE 15. Experience with JSON APIs (minimum 3 years) 16. Build tools experience: make, gcc, g++ (minimum 5 years) 17. OpenVPN (server and clien...

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    We are looking for solution like a traditional GSM or CDMA VoiP gateway. This project will be separated in two p...Mobile application will register to a server and accept call from that server with IXA or SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application will work on only wifi Internet connectivity, coz GSM internet data normally disable during any GSM call. All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP protocol. Certain number of registered Mobile will be able to assign in a group of gateway. On server have to include option to show balance th...

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    We are trying to build asterisk system based on the following requirement. Asterisk (R) and Asterisk (B) Asterisk (R) which is Raspberry PI based and attached with USB dongle with no SIM card Asterisk (B) is connected to some SIM box but with SIM card attached Using softphone, we are connected to Asterisk (R) or Asterisk (B) and make a call to some number. Asterisk (B) will send some code to Asterisk (R) to remote link up the cell / mobile signal via IP to initiated the call Both Asterisk (R) and (B) are ready, and your job is to link up the (R) and (B) so (B) can link up cell / mobile signal remotely

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    ...but as I have never worked with Python/Flask before we decided it may be better to outsource the task to someone with more experience. (I work as a PHP developer on another project for him based out of Germany) The following is a list of requested changes to the web application forwarded to me from my client: 1. Boolean searching, esp. NOT (in "advanced search") 2. Wildcard searching, esp. the asterisk. (in "advanced search") 3. Keyboard shortcuts, e.g. for /search /quotations or /search. At the moment this requires a click to get the cursor into the search box. 4. Shortcut to go back to a given search list. At the moment the system goes back to an empty search box and one must start again. 5. Adding the option to show other symbols than a country flag on an...

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    trying to get a simple wss:// up and working and proxying to SIP server. i have bui...get it working.. i have 1.) use or hopefully to verify webrtc client demo (login with sip:1111 @ with wss:// ) SIP PBX <---->Kamailio <----> webrtc client () the config im using is this basic need = i need to use webrtc to register as sip and make a call like sip (zoiper on the web) after research i need to do this with WSS:// since our PBX is only SIP this is the route im going so im trying to get a simple working model of that with kamailio

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    Centos 7 PXE server on a vmware, pxe server that have ubuntu, kali, fedora all get auto-installation with preseed or kickstart. All OS have the last version and auto-installation is a most. If you can add win7,10 to the pxe server that's a bouns. prefer providing the project with OVA (ova is compressed vmware). if u don't know what you're doing please don't proposal.

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    To make custom Invoices to bill clients for incoming calls through our Asterisk PBX and Credit Notes to pay telephone agents for their work. Experience required: 1) LibreOffice Calc 2) Software to change templates to pdf without loosing calculations 3) Asterisk PBX 4) Linux 5) MySQL DB 6) PDF 7) PHP 8) KVM qcow2 (for disk management) We use background images to show invoice and Credit Note Templates to user. To fill the Templates out, the user only sees the image, the spreadsheet is invisible ! This project is not for beginners !! You must have a PC with a Linux OS !! ----- Good and quick communications are essential !! This project will be divided into unequal milestones. See attachment.

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