Freepbx asterisk jobs

Installation of A2Billing latest and asterisk on a debian server. For future integration with linphone SDK I want someone who has past experience in configuring asterisk to work with A2Billing. Your job will be to and install A2Billing billing 2. install and configure Asterisk with cdr to be working on A2Billing I should be able to an account from A2Billing and register on a softphone. 2. Make calls using that account 3. CDR should be made in database table and I should be able to see it in A2Billing billing. You also need to tell me which configuration files were changed so that I can do this myself next time in case of server failure or new installation.

Hi, I have a new installation of freeside billing 4 and asterisk on a debian server. I want someone who has past experience in configuring asterisk to work with freeside. Your job will be to free side billing 2.Asterisk I should be able to an account from freeside and register on a softphone. 2. Make calls using that account 3. CDR should be made in database table and I should be able to see it in freeside billing. You also need to tell me which configuration files were changed so that I can do this myself next time in case of server failure or new installation. Thanks

I have a FreePBX instance and an Avaya IP Office system and am looking over time to begin moving my voice service to the FreePBX but want to do it over time as time permits. I'd like the FreePBX to act as the Primary PBX trunking calls to Avaya and being able to pass calls between systems.

...your previous CRM projects and modifying it to suit my needs. Of course, there are many open source CRM solutions available but I really want to build my own. Many do not have phone integration which is a must as it's the most difficult aspect of managing communications within a medical office. I have a number of suggested solutions that I have yet to fully explore (Zoho, Salesmate, CallRail, FreePBX, etc). I'm told that Ring Central has a strong history in the medical field and that they have the best API on the market (per an independent person who has worked with several). Furthermore, they apparently have an almost "ouf of the box" solution though some additional development may be required. But they have fees and perhaps others would be better. So...

Repair , delete zulu and many other modules

I'm lookin for of a skilled engineer to manage and deploy a SIP & Kamailio on docker compose that also requires some admin services. I need these services done on a part-time basis, but the engineer must stay available and be ready to help should any issues arise. This means it is essential that the engin...environment (not final, but wished): - Ubuntu 22.04 - dockerized - in a later task, to enable kubernetes for Kamailio and HA Good entry points: Budget? will not be disclosed. So place your best hourly rate!

Need to add Staff Freepbx SIP Extension in Laravel Website Staff Login to dial customers after the login web browser

We have FreePBX running on AWS and want to move to a fresh new install of FusionPBX on our other AWS account and need an expert to lead/ guide us, specifically: 1. Stand up the new instance of FusionPBX (current version) - with support of our sysadmins and create one tenant with our sysadmins to learn/test with 2. Document the procedure to migrate (noting configurations/SIP changes etc.) (we will configure the extensions - you will sanity check), cutover, and validation-test. 3. Support our sysadmin to perform the migration & cutover, and if needed the fallback The FreePBX configuration (for scope reference only) is attached.

We provide cloud-based VoIP services for a small hotel in the US using AWS as our cloud service provider. We have two AWS accounts, one named “Legacy_AWS”, the other “New_AWS” Legacy_AWS has a server, named Hotel_FreePBX_13, running FreePBX, configured, and providing service to the hotel. We need to migrate that server from “Legacy_AWS”, the other “New_AWS” and would like an expert to lead the process and be assisted by our two sysadmins. Overall we will make preparations for, and then: 1. Execute a pre-migration performance check on the Lakeside_Inn_FreePBX_13 running on Legacy_AWS 2. Cutover 3. Execute a post-migration performance check on the Lakeside_Inn_FreePBX_13 running on New_AWS Then if needed: 4. Fallback ...

we are looking for voip expert to setup a system that will handle call termination must work with anydesk

I'm looking for help resolving an issue I am having with Asterisk on a Linux server. I'm unsure of what caused it, but it needs to be fixed urgently. The ideal freelancer for this project should have expertise in System Administration for Linux platforms. I'm open to suggestions on how to diagnose and remedy the issue, so if you think you can help, don't hesitate to reach out and make your case. We are getting a lot of chan_sip.c:4413 __sip_autodestruct and have tried restarting the server once but without any positive results.

I'm looking for experienced mentoring services to help setup and integrate an existing SBC system with Asterisk. I need assistance for 1-3 people with setting up the system, configuring SIP trunking, and ensuring that the integration with Asterisk is working properly. With the right combination of assistance, knowledge, and experience, I'm confident that this project will be a success.

I'm in need of a skilled engineer to manage and deploy a SIP & Kamailio & Asterisk solution on-premise that also requires some admin services. I need these services done on a part-time basis, but the engineer must stay available and be ready to help should any issues arise. This means it is essential that the engineer is experienced with the SIP & Kamailio & Asterisk solution and be comfortable dealing with any issues that may arise. The engineer must also have experience in deployment on docker & docker-compose and later on administration of this system. This position will require some extra work every week, but I'm ready to discuss the amount of work needed with the chosen expert in order to get the best possible results. We usually di...

Hello! I am looking for a freelancer to help me set up a VoIP system using Asterisk and a Skype trunk specifically for external calls. This setup is for my personal use, not for an existing or new business venture. The setup should be able to handle a high volume of external calls with clear voice and quality. I expect a complete solution in terms of setup and installation of all the VoIP components to ensure that I am connected and ready to place external calls. I am essentially looking for someone who can configure Asterisk and the Skype trunk with good quality, so that I can get started making external calls. Thank you!

Need to integrate Asterisk PBX into Zendesk. Using Asterisk AMI and POST API into Zendesk. When a call comes in on a specific ques ticket must be created in Zendesk using caller ID as a requester (if the customer number is not in Zendesk).

Setup FreePBX on Hetzner with instructions and make two phones talk to each other.

Need the following 4 customizations 1. After queue call completion it will dial another extension for survey 2. After call hangup agent will be paused automatically 3. Get Dial and Hangup event from agent outbound calls via call to queue 5. Soundboard , so that agents can play pre recorded files at any time of the conversation

I have issues with the production server so I am looking someone to fix it asap. I have 2 different applications assigned to the the same carrier and carrier transfers calls simultaneously to both the ip, but right now as the call reaches fusion its disconnecting the call not allowing it to get transferred to asterisk server.

I have a linux server running asterisk running Chan_sip which only support UDP, and can not be upgraded. The asterisk Server communicates to Phones and SIP TRUNK Providers on UDP port 5060 using the LAN Public IP X.X.X.X interface. I need to have some new phones connect to the Server over TCP Port 5062 I need to have some new phones connect to the Server over TLS Port 5089 We will need to have kamalio or Opensips centos 7 to run on the same server as asterisk, and should allow the following: 1) The Proxy should communicate with the asterisk server on UDP SIP port 5060. 2) Listen for incoming SIP TCP traffic on 5062 on the LAN IP X.X.X.X. and proxy this SIP TCP Traffic to and from the Asterisk server on UDP Port 5060 using the interface. 3) ALL T...

This is for a Asterisk and Zoho CRM integration. We need to receive and make calls on Zoho. Also be able to record, transfer, pop up a window with caller contact on Zoho.

API SDK module required with Asterisk using PHP and MySQLwith subscription and credit billing model for users to enable them to add their own SIP trunk and route their own sip accounts and use API through any of the SIP trunk they have to process their IVR instructions on given webhooks. Basic MySQL structure to start with. a) users table [api_key (unique), loginid (unique), pwd, trunk_id , user_type (admin/user) ] b) trunks table [trunk_id, username, password, ip/address , port, protocol, transport_type, did_numbers, status] and other fields and table that is required. Experience Required : Asterisk AGI PHP MySQL Reference API SDK documentation Calls will be made through the voip server using the rest API tht needs to be developed in PHP below is the sample how we will

Hi Pablo B., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I'm installing Kommo CRM in my company and we currently use voip offered by local telephone company. I'd like to configure our voip numbers on asterisk and then Integrate with kommo CRM. Can you do it?

I have 2 asterisk servers A and C A is running on Centos 7 and has a static IP C is running on Freepbx (raspberry Pi 4) and has no access from the outside. We also have B which is running OpenVPN and has a static IP current configuration is A -> B > C -> outside Trunk We need the following to be done 1. A SIP remote extension to log in to A 2. A passes the registeration via B (OpenVPN) and register as an IAX2 extension in C 3. SIP remote extension made the call, A will translate the call from SIP to IAX2 Another thing, you would need to access using Anydesk via my PC to access both A and C due to VPN. Thanks

I need someone who experience in Linux and script development knowledge for Asterisk /freepbx patching. I will integrate existing asterisk PBX to Kommo CRM. The Asterisk/freepbx V18 PBX and kommo are ready, but don't know how to run the script development as below link:

I am looking for a freelancer who can help me build a cloud PBX telephony software that is supported by Twilio or Asterisk, using the SIP protocol. This telephony software will not need to be integrated with an existing management platform.

I need an expert in installing FreePBX system on our server. I have specific requirements and I want these requirements to be fulfilled before setup. If you think you are the right fit, please let me know – I am looking for someone to start this project as soon as possible. Thank you for your time!

We are having major issues with our vicidial. Channels not connecting and calls being burned and reps sitting waiting. We are losing money and employees daily and need someone to help us get dialer running smooth asap.

We are looking for someone who can integrate Asterisk in our Laravel system. For each incoming call, a pop up should open with the details matching in our existing database. For outgoing calls we should click the phone icon and call should be redirect to the customer number. You should have prior experience in Asterisk /FreePBX integration with Zadarma or Plivo.

Hello, We have server on which is installed FreePBX. We want to buy Yeastar TG100 GSM and Cisco SPA502G. We need to configure this 2 devices so when call is made to GSM SIM cart that is in Yeastar the call to be trough VOIP to Cisco

I'm looking for someone experienced in setting up a SIP trunking configuration with the Freepbx 16 platform and Twilio. Specifically, I'm seeking assistance setting up an IP-based Twilio SIP trunk configuration. Knowledgeable in settings and settings optimization is a must. Additionally, I'm not looking to setup Voicemail at this time so Voicemail setup experience is not required.

Dear Freelancers, We have a video bridge jitsi installed. We need to achieve the following: Need to be able to login from a raspberry pi with a calling of url. The url call will happen if someone dial an extension on an asterisk pbx. Once the extension dialled from a SIP phone asterisk call a url which makes the raspberry online in the jitsi meeting room. From this time the raspberry is opened the meeting roon and participants can log in to the room. The raspberry pi need to work with a Konftel conference system . So the Konftel will be connected with USB to the rasperry and then the raspberry need to use its camera and microphone in the meeting.

Looking for someone who can configure auto dialer for freepbx We already have it working but now it dials the customer and when that is picked up, it dials the agents. It should start dialing the agent the moment it starts ringing on the other end

i have laravel app that is managing user and apis . now i want someone who help me to in aws configuration for Laravel and SIP server asterisk ( user mapping with device)

On our IncrediblePBX vps : add contacts db source and make connection with growl to get customized notification via Growl send-to source based on SIP line called, and caller details if found in local phonebook mysql db. this module had to be compatible with latest freepbx 16 version. - Freepbx Asterisk distribution : - superfecta module to work with : fixed price : 100 $ - ASAP within 1 day.

I'd like the following to happen. A user registered to asterisk will dial the extension 1001. When they dial 1001, they will enter a conference room Another user registered to asterisk will then be dialled and forcibly put into that conference I like this to be done in Node.js, using Asterisk ARI to orchestrate as much of the above as possible. I'd also like the freelancer to share with me instructions on how to deploy asterisk, their config on asterisk and the node.js code too, so that I can recreate their experience locally. Be great for this to be done within the next 24 hours!

The project is a design and implementation of a doorbell using raspberry pi4 with python using asterisk (or another open source), zoneMinder (using ip camera), with 2 android mobile phone, at deferent flour. I need only the CODE of the asterisk (or another open source like: freepbx or fusionpbx), in orderto call and listen rasberry pi4 and android mobile phone, (flour number 1 or flour number 2). I also need a intercome between the 2 flour from each mobile android phone, at deferend flour.

Hi, Freepbx server up & running fine on AWS 13.51.***.*** Skyline GSM VOIP gateway 192.168.***.** Configure require to communicate FeePBX with Skyline by using anydesk & Add Sip Add Trunk Add Inbound- info available in excel Add Outbond- info available in excel Quick completion require

Hi, Freepbx server up & running fine Skyline gsm voip configuration already working on cloudfonica but i need to use now freepbx Need to configure skyline with freepbx by using anydesk Sip Trunk Inbound info available in excel Outbond info available in excel Quick completion require

Caanot update fwconsole ma update userman An exception occurred while executing 'ALTER TABLE userman_password_reminde r ADD PRIMARY KEY (id)': SQLSTATE[42S02]: Base table or view not found: 1146 Table ' _password_reminder' doesn't exist

Hi, Freepbx server up & running fine Skyline gsm voip configuration already working on cloudfonica but i need to use now freepbx Need to configure skyline with freepbx by using anydesk Sip Trunk Inbound info available Outbond info available Quick completion require

We use Google workspace at work, so all' our contacts are stored on Google contacts. We use freepbx as our pbx and I would like to be able ti have both connected to see the name of Who Is calling via freepbx querying g contacts.

We are looking for an experienced VoIP developer who can design Windows and MAC desktop VoIP applications using our Hosted PBX API. The application will have to be tightly integrated with our asterisk-based PBX and our custom API. Supported functionality will include: Voice calling via SRTP Searchable Call history with access to call recordings and call notes SMS and MMS messaging Read-only access to favorites and BLF keys Read/Write access to personal contacts Visual Voicemail Do not Disturb Call Forwarding We prefer a web application running installable with an Electron wrapper on the client's workstations but are willing to entertain other options.

Hi Need to design a sip based extension portal and forwarding portal. Looking forward to hear from you

On our IncrediblePBX vps : add contacts db source and make connection with growl to get customized notification via Growl send-to source based on SIP line called, and caller details if found in local phonebook mysql db. - Freepbx Asterisk distribution : - superfecta module to work with : price : 50€ fixed - ASAP within 1 day.

Callback widget - Asterisk Hello! I want a price for coding Asterisk to work with a callback function. Simply explained, we will publish a form on a website where customers can request to be called within a minute. When the customer has entered his phone number, it must call one or more pre-entered numbers in the Asterisk telephone. When one of the people being called answers the phone, a voice recording should be played: "A customer wants to talk to you, click 1 to confirm that the call should be connected" When the called person has clicked on the button to confirm that they want to talk to the customer who wants to be called, Asterisk must call the customer who wanted to be called (on the phone number that the customer specified in the form). ...

I need a voice bot for accounting office in greek language. 1. The client say who agent want to connect on 20-30 key words the call go to specific agent or group

We have an open spot for a professional interested in joining an amazing project that takes responsibility of the Design and implementation of end-to-end 911 Call processing! If you are an experienced VoIP developer with strong expertise in C, C++, Asterisk, and some background with JavaScript, this is a great opportunity for you! Candidates should have the skills and experience to quickly integrate the development team and provide design direction. About us: SofTech Consulting is a visionary tech consultancy focused on transformative cloud development. Leveraging 10+ years of high-level cloud architecture experience, the latest technologies, and an extensive partner network that includes Red Hat, Azure, and other top software providers, our specialized team of advisors helps ...

...structuring: ___ DB structuring & Maintenance: ___ API (creation): ______ API (integration): _____ Hostgator Server Set Up: ___ Hostgator Server Security: ____ Hostgator Server Maintenance: ____ Linode Server Set Up: _____ Linode Server Security: _____ Linode Server Maintenance: _____ Managing Linode access with SSH keys NOT with passwords: _____ Laravel:_____ Vue JS:____ React:____ Github: _____ Asterisk: ______ Trouble Shooting: ____ PCI Compliances______ Please add anything else you would like us to know about your skills and be honest so that we know what projects are best for you as we have many + What is the best per hour rate you can give: ...

seems freepbx not connect to AMI but seems config file is ok, something wrong with this server, and need fix start by log. some days ago server work perfectly, need help with this.

The project is to create a service running on Debian operating system. The service should connect Asterisk PBX, open a socket on port 7891, and send the output of the following Asterisk events: - Inbound Calls: Call waiting in the queue, Call Ringing, Call Answered, Call Hangup, - Outbound Calls: Call Waiting, Call Answered, Call Hangup. The output of each event should be in one line only (Not sending few outputs for the same event). The project includes also a client that should run on a pc, connect the remote server on port 7891 and receive the server outputs.