A2billing freepbx asterisk jobs

...for an expert Flutter Developer with a strong background in developing and managing Asterisk-based call applications. The selected candidate will possess expert knowledge in creating both Android and iOS applications using Flutter, with a particular emphasis on integrating Asterisk calls. To Do: Design, develop, and manage an Asterisk-based call application using Flutter. Implement innovative, high-quality call solutions for Android and iOS platforms. Test and maintain the app to ensure maximum performance, efficiency, and responsiveness. Identify and troubleshoot application bottlenecks and bugs. Must Have - Proven experience as a Flutter Developer. Extensive understanding of and experience with Asterisk calls integration. Familiarity with Android and iOS...

I am in need of an Asterisk API Expert who can help me with call routing functionality. There are no additional features or integrations required for this project. The timeline for completion is expected to be 3-4 weeks. Ideal Skills and Experience: - Strong knowledge of Asterisk API and call routing functionality - Experience in developing and implementing IVR systems - Proficient in programming languages such as PHP and Python - Familiarity with CRM integrations and billing systems is a plus.

I am looking for an Asterisk Expert to build a new API from scratch using Python. The API should be able to perform call routing and forwarding. I need someone who can develop the API from scratch. Ideal skills and experience for this job include: - Strong knowledge and experience with Asterisk - Proficiency in Python programming language - Experience in developing APIs from scratch - Familiarity with call routing and forwarding - Understanding of VoIP technology and protocols As a client, I am open to suggestions and guidance from the freelancer. The project should be completed within a reasonable timeframe and on budget.

Hello, we need a cloud telephony solution to receive calls using an HTML/PHP script. One option for this would be to utilize an Asterisk server and connect it with PHPAGI. Can you develop a telephone system for us that allows us to make and receive calls? It should be web browser-based so that we can integrate it into our website. I want to be able to make and receive calls using a SIP number, and the whole thing should run on an Apache web server. I have a web tool where I want to add this telephone functionality. The web tool runs on PHP

I am looking for someone to help me set up a project for Asterisk, where it will play back a sound file and run a context on a channel. I have chosen a custom recording for this purpose, and I will provide the recording myself. The sound file is intended for use an Interactive Voice Response for my business. I am looking for someone with experience with Asterisk who can help me set up this project with the sound file included. A call to B , B is anwered , than how to run [play-ivr] to B? that time A and B is not disconnect channel, A can hear sound file and get DTMF form B. [play-ivr] exten => s,1,answer exten => s,n,Wait(1) exten => s,n,Background(testwavfile) exten => s,n,WaitExten(5,)

I am looking for someone to help me set up a project for Asterisk, where it will play back a sound file and run a context on a channel. I have chosen a custom recording for this purpose, and I will provide the recording myself. The sound file is intended for use an Interactive Voice Response for my business. I am looking for someone with experience with Asterisk who can help me set up this project with the sound file included. A call to B , B is anwered , than how to run [play-ivr] to B? that time A and B is not disconnect channel, A can hear sound file and get DTMF form B. [play-ivr] exten => s,1,answer exten => s,n,Wait(1) exten => s,n,Background(testwavfile) exten => s,n,WaitExten(5,)

I am looking for a skilled freelancer to help me with a WebRTC setup on an A2BILLING/FREEPBX server. I have answered a few questions to best guide the freelancer with my needs: No, I need to configure a server; both inbound and outbound calls are necessary; and I already have SIP trunks. I require someone with experience with WebRTC integrations, VoIP gateways, Linux and FreePBX/A2BILLING setup and management. The server will be used for inbound and outbound calling services, and must be secure, reliable, and scalable. The ideal candidate should have good communication skills, and be able to work in a timely manner to complete the required deliverables. The candidate will also need to provide training and support in the set up and maintenance of the WebRTC s...

I am looking for a freelancer to help me set up and test a SIP server using Asterisk. The ideal candidate should have experience working with Asterisk and be able to handle 50-100 concurrent calls. Additional features and integrations may be required, such as call recording, IVR setup, and billing integration. Overall, I am looking for someone who is knowledgeable in SIP server setup and can ensure that the server is running smoothly and efficiently.

hellow , i have instlled freepbx , chap-sccp and sccp_manager everything is working but cisco phones keep restarting when trying to upgrade i need someone has experience with that to solve the problem and connect cisco 7942 to the system

I am in need of an Asterisk dial plan for outgoing calls. The primary purpose of this plan is to distribute calls equally among multiple GSM gateways, of which I have between 4-7. Specific features required for this plan include time-based routing and call routing. It is important that calls are distributed equally among the available gateways as allowed ports. Ideal skills and experience for this project include knowledge of Asterisk, dial plan creation, and experience with multiple carrier systems.

I am in need of a freelancer who can provide me with one-time troubleshooting assistance with my FreePBX system. I have been experiencing configuration errors and I need someone to help me resolve them. The setup is as follows: - VULTR server I have setup FreePBX 16 - I am able to make internal calls between ext 102 and 101 - I am using a GOIP GSM Gateway on my local network with a SIM card in it The problem I am having - I am unable to get GSM Gateway to link together with the FreePBX server. It is located in China, I am physically able to access it and on the same local network. I get error 403 and 401. I only have a basic knowledge of networking / VOIP so I might have made a simple error that someone will be able to spot and troubleshoot. I need live assistan...

Custom Upgrade Discord Bot that will go beyond basic read/write rules per channel based on role. I want to show a custom message on channels certain roles don't have access to and put an asterisk next to the channel so people know those are 'premium' channels. That is the functionality, I want a very basic site/connect page and management page to fill in all settings. Something similar to what you see with 90% of Discord bots, connect with discord option then all backend settings to customize headline message, form fields etc. We will make front end site, don't need that.

We need a skilled freeswitch or asterisk developer to help develop an API interface capable of initiating phone calls on freeswitch (similar to Twilio APIs for voice calls). Sample use cases are to embed the API in a CRM or webservice to enable to user click a button on an external web application, and initiate a call on the freeswitch machine and the call is then connected to a softphone or webrtc endpoint for audio.

I am looking for a freelancer who can troubleshoot my existing Debian server's login system. ...login system. The ideal candidate should have experience in server troubleshooting and be familiar with Debian OS. The project requires remote access to the server to identify and resolve the login issue. No credentials are shared, but access through anydesk session. Skills and Experience: - Experience in server troubleshooting - Familiarity with Debian OS , VMWare esxi, Quagga BGP, ISPConfig, Freepbx - Proficient in remote server access - Knowledgeable in login systems and authentication protocols Requirements: - Remote access to the server - Detailed step-by-step guidance on the troubleshooting process - Excellent communication skills to provide regular updates on the progress ...

I am looking for a freelancer who can help me set up FreePBX and connect it to VTiger. I need to setup a FreePBX to support 6 to 8 SIP phones and 5 to 7 mobile devices. The FreePBX needs to be connected to my Vtiger setup so we can track the voice and SMS communication between agents and the leads. The setup needs to be installed on a VPS. If we can use the existing VPS that has my Vtiger installation running, that would be great. I can add more memory and CPU if needed. (VPS is running on Namecheap with Ubuntu 20.04.) The intent is to copy the functions of OpenPhone (). That said, our FreePBX setup must support both voice and SMS text messaging. I currently have an account with Vitelity for my SIP trunks, but I am fine with switching to another SIP solution ...

I am looking for a freelancer to integrate my personal dot net CRM software with my Asterisk VoIP system. I do not currently have the necessary API documentation for my CRM software, but I can obtain it with some effort. The main feature I want to integrate is automated call recording. Ideal skills and experience for this project include: - Experience with integrating CRM software with VoIP systems, specifically Asterisk - Knowledge of dot net programming language - Experience with API documentation and integration - Familiarity with call recording and monitoring systems

We need asterisk developer for unified communication

I need to configure Kamailo SBC to connect multiple Microsoft Teams account in the same SBC server. - Install Kamailio - Configuration TLS certificates, it'll probably need to work with wildcard cert - Configuration RTP Proxy / RTP Engine - Routing Inbound / Outbound - Security Example MS Teams 1 <--> Kamailio SBC <--> Asterisk MS Teams 2 <--> Kamailio SBC <--> Asterisk

I am looking for a freelancer who can help me with my FreePBX Dynamic Routing project. My current setup is a Single Server. I need a developer who can set up a mySQL database to capture caller inputs in order to enable dynamic routing. I require 2-4 different routing options. The contractor will only be able to access the system through me I don't have remote access. Also, the person will need to walk me through each step of development and implementation. Ideal skills and experience for this job include: - Strong experience with FreePBX and mySQL - Knowledge of dynamic routing - Ability to create customizations and integrations when necessary - Excellent communication skills and ability to work independently

I am in need of a freelancer with strong skills and experience in PBX maintenance. Specifically, I have a digital PBX system that requires attention. The main issues th...with strong skills and experience in PBX maintenance. Specifically, I have a digital PBX system that requires attention. The main issues that I am experiencing are related to configuration problems, as well as the need to migrate to Freepbx and develop new applications. Ideally, the freelancer will be able to complete this work within 1-2 months, though there is some flexibility with the timeframe. The ideal freelancer will have experience with digital PBX systems and be able to troubleshoot and solve configuration problems. Additionally, experience with Freepbx migration and application development would ...

I need a webrtc to sip gateway to be implemented so I can connected some webrtc softphones (asterisk webrtc softphones on our Odoo CRM) to twilio sip domain

I am looking for an experienced Asterisk API developer to create an Asterisk Webhook API. The developer should have extensive expertise with Asterisk and be able to develop the API using an open source programming language. The API must include functionality for call control, call recording, and voicemail management. We are open to suggestions for the programming language and frameworks used to create the API, however, it must be versatile and reliable. We expect comprehensive documentation for the API as well as ongoing support to ensure it functions properly. Thank you for your time and we look forward to your proposal.

I have a freepbx server which is adding the sip servers ip to the block list with "reject-with icmp-port-unreachable" I have already added this to the whitelist but it keeps adding it back to the blacklist within a few mins. Require someone to remotely login by Anydesk and diagnose fix this for me. I have access by SSH and GUI

Installation and configuratoin of Asterisk/FreePBX on Ubuntu server Integration and configuration of softphone client on Raspberry Pi, this client will be command line based for easy integration with 3rd party API

I am looking for a person who can install and configure a SIP client(preferably PJSIP or any other) on my Raspberry Pi board, then install a Asterisk Server on a Linux computer that is on the same nextwork. After installation, configure both to talk to a VoIP SIP phone which is on the same network. Also confgigure the PJSIP client in listen only mode

Hello, I am looking for an experienced freelancer to create a virtual phone call center using Asterisk and outbound and inbound calling services. Our organization needs a reliable, efficient and cost-effective solution to provide customer support services. The ideal candidate should have extensive experience with these technologies and must demonstrate that capability through the workload provided. Examples of previous work will be an asset. We will need them to provide a detailed proposal of your plan for the project. This system must support outbound and inbound call services, ensuring that our customers always have a friendly voice ready to answer their queries. We need a reliable and easy to use system that can handle a variety of inbound and outbound calling features such as:...

I am seeking a FreeSWITCH or Asterisk expert to help me set up a new enterprise. The ideal freelancer should have prior experience in this area. The project requires call routing and forwarding features to be implemented. Other functionalities such as IVR and voicemail, as well as conference calling and recording may also be required. Please provide your experience in this field in your application. Automation will be required for this as well. I basically want people to go to my website, sign up, and get started with processing calls. Thank you.

I am in need of an experienced freelancer who can fix the poor audio quality issue in my FreePBX phone system. The connection being used is VoIP. Ideal Skills and Experience: - Strong knowledge of FreePBX phone system - Experience in fixing poor audio quality issues in VoIP connections - Knowledge of various audio codecs and their compatibility with FreePBX - Excellent problem-solving skills and attention to detail

...am seeking an expert for the installation and operation of a VoIP system using Asterisk and A2Billing. The work will involve the following: Full installation and configuration of Asterisk and FreePBX on an AWS Ubuntu server. Full installation and configuration of A2Billing for managing customer accounts, billing, and prepaid services. Configuration of GSM gateways to work with Asterisk / FreePBX. Configuration of a mobile phone application to work with the system (such as Zoiper, Linphone, or Bria). Ensuring that customers can easily download the app and set up their accounts. Ensuring that the system is secure and regularly updated. I am looking for someone with prior experience working with Asterisk and A2Billing, as well...

...for a skilled freelancer to help me set up and integrate Asterisk PBX with our VoIP phone system and a CRM platform of my choice. Phone System: My current phone system is VoIP. CRM Platform: Although I do not have a preferred platform, I am looking for a freelancer who is experienced with integrating Asterisk PBX with different CRM platforms. Please let me know which platforms you are experienced with and your recommendations for my business. Project Timeline: I do not have a strict deadline for this project, so I am looking for a freelancer who is able to work on it with flexibility and deliver quality work. Ideal Skills and Experience: - Expertise in Asterisk PBX setup and configuration - Experience in integrating Asterisk PBX with CRM platforms suc...

I am looking for a freelancer to set up an asterisk PBX and integrate it with our CRM, which is EspoCRM. Our company is small, with less than 50 employees. Our main goal for this integration is to improve our customer service. Ideal skills and experience for this job include: - Experience with asterisk PBX setup and integration with CRM systems - Familiarity with EspoCRM - Knowledge of customer service best practices - Strong communication skills to ensure smooth integration and training of employees.

Hi there, I have Twilio and VitalPBX (Asterisk server). I need the following flows to be created: 1- An IVR flow via Twilio studio • 1 for Sale • 2 for Technical Support • 3 for Order follow up - After any of the above extensions are dialled, the call needs to be forwarded to the corresponding ring group of Asterisk extensions. If the dialler doesn't dial any extention, the call will be automatically forwarded after a cetrain time to operator ring groups. - At any time a caller can request a call-back by dialling a number. - Extensions can define a number (e.g. their mobile number) and have all outgoing and incoming calls originated/terminated to/from their mobile phone 2- After a call is finished, the call transcript and call summary (by calling chat...

We need to develop a SIP to WAPP gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through WAPP to complete the call to the called party number. The development ...correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project, we'll select the one, triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to WAPP gateway 2)WAPP gateway converts the sip/iax signal to WAPP protocol 3) the termination number carried from the origination header will be checked by the Asterisk gateway , if the number is used by WAPP and if the number is online, the call will terminated on WAPP!. 4) if the number is not used in WAPP it sends 503 error and rerouted to ot...

I am looking for a person who can install and configure a SIP client(preferably PJSIP or any other) on my Raspberry Pi board, then install a Asterisk Server on a Linux computer that is on the same nextwork. After installation, configure both to talk to a VoIP SIP phone which is on the same network. Also confgigure the PJSIP client in listen only mode

We are seeking an experienced SIPROXD server configuration specialist to assist with the setup and configuration of our SIPROXD server. Our server is hosted at IP address 4.4.4.4, and the SIP port is set to 7090. Requirements: In-depth knowledge of SIPROXD server configuration and setup Proficiency in SIP protocols and routing Experience with SIP servers like Asterisk or other SIP engines Ability to test and troubleshoot configurations Familiarity with virtual machine environments (VMs) Strong communication skills and ability to work collaboratively Job Responsibilities: Configure SIPROXD server to handle incoming calls from IP address 5.5.5.5 Route incoming calls from 5.5.5.5 to a separate SIP server at IP address Ensure proper routing

I am looking for a freelancer to create or sell an open source CentOS based IVR application that will have advanced IVR functions. The ideal candidate should have experience with PHP programming language. The IVR application should be able to handle more than 50 users interacting with it. The following features are requir...looking for a freelancer to create or sell an open source CentOS based IVR application that will have advanced IVR functions. The ideal candidate should have experience with PHP programming language. The IVR application should be able to handle more than 50 users interacting with it. The following features are required: - Advanced IVR functions - Customized IVR functions - Basic IVR functions - Asterisk Based - Should Also Install Application And Give Installati...

Integration of Asterisk FreePBX into Perfex CRM Hello, We are seeking a skilled freelancer to assist us in integrating Asterisk FreePBX into our Perfex CRM system. Our goal is to streamline communication processes and enhance our customer relationship management capabilities. Please note that we have some additional requirements to add to the initial project description. Here is the complete list of requirements: 1. Expertise in Asterisk FreePBX: You should have a strong understanding of Asterisk FreePBX and its features, such as call routing, voicemail, IVR, call recording, and reporting. 2. Proficiency in Perfex CRM: Familiarity with Perfex CRM or similar CRM systems is essential. You should have experience working with APIs and...

I am looking for an experienced freelancer to develop an IVR system in Asterisk for my customer service needs. The project will involve creating IVR prompts based on a pre-existing script provided by me. The system should have 6-10 menu options for callers to choose from. Ideal skills and experience for the job include: - Strong understanding of Asterisk and IVR systems - Experience in creating IVR prompts and scripts - Ability to customize IVR menus based on specific business needs - Familiarity with integrating IVR systems with other business systems and applications If you have the required skills and experience, please submit your proposal with relevant examples of past work. I look forward to working with you.

hi i have issabel pbx and i want to integrate it woth zoho crm to send and recive calls from zoho crm

Building an Asterisk VoIP Server ubuntu

We are looking a telecom engineer to configuration and manage a powersmpp sms platform (need for sms transit) and goip devices on a part time basis. need know softwares powersmpp, asterisk, easyphone. thank you

Call Forwarding Portal with CC and Call Limit I am looking for a freelancer who can create a web-based portal for call forwarding with simultaneous ringing. The ideal candidate should have experience in developing call forwarding portals and be proficient in Asterisk. Features: - Simultaneous ringing - Call forwarding with CC and call limit - Customizable settings - CRM integration with Asterisk - Analytics and reporting Skills: - Proficient in Asterisk - Experience in developing call forwarding portals If you have the necessary skills and experience, please submit your proposal. Thank you.

Based on our conversation the following configuration changes are to be made to your FreePBX system: Create extensions for softphone and Cisco SPA8000. Configure extensions on applicable devices. Create ring group for inbound calls, and reconfigure routing to first use ring group then go to existing IVR. Diagnose and remediate issues with email and configure Fax line.

I would like a multi-class Asterisk tutorial, at first with the most basic concepts (from zero) and increasing the level with each class, starting to beginners and finishing with a professional mastering of the subject, to learn and understand completely how to use Asterisk and become proficient of the subject. The minimum total duration of the course with all classes must at least 8 hours. I need good quality on the recording and the edition of the course. Also, the content has to be original, to be able to use it without restrictions. We will sign a contract of copyrighting cession before ending the project.

I require an extremely knowledgeable Python developer with very good experience with Google API usage (TTS/STT/NLP), OpenAI API usage, FreePBX usage (ideally PJSUA/PJSUA2 to integrate the Python script with an extension on our FreePBX server). This will be a long term project with consistent compensation. Do not waste my company's time if your experience level is not superb. This is an enterprise level application and there will be requirements to sign an NDA prior to work. The selected developer will be utilized for potentially 6 months+. We will have an in depth discussion regarding the project.

Hello, i use this software from github. My settings are: Outgoing calls use extension 333 which forces calls to "optional des...contact list, script start calls and connects to the IVR.) The problem is: Have current settings for outgoing calls via TRUNK (DongleX). I want to be able to call from another extension, which is instaled in freepbx and it works normally. $callFile = "Channel: Dongle/dongle1/$numbern"; $callFile .= "CallerID: $caller_idn"; $callFile .= "Context: callblastern"; $callFile .= "Extension: 333n"; Task: Set outgoing connection via another TRUNK (not Dongle). (Missing - correct outgoing "context" and setting in FreePBX) I am looking for someone with experience. freepbx 15, a...

Job Description: Have installed the latest version of FreePBX on new hardware and Cisco SPA8000 . Requiring a talented Freelancer for this system to: Connect to Yealink T28P x 8 Connect to 2 physical PSTN phone lines 1 Fax line via Cisco SPA8000 Route calls from Cisco SPA8000 to FreePBX A separate inbound queue for each phone line Outbound line selection based on grouping Configure outbound email notification! Headset programming Set up / Configure Sangoma Softphone App Any other minor tweaks. Immediate start 5/1/2023 10:00am EST! Skills: Asterisk PBX, Linux, PHP, VoIP

I am looking for a freelancer who can assist me with a Ubuntu Asterisk SIP problem. My current setup involves Ubuntu 20.04 and Asterisk 18. I am experiencing a general issue with outgoing calls being masked, as the number in Zoiper is not being recognized. Ideal Skills and Experience: - Strong knowledge of Ubuntu and Asterisk - Experience with SIP configurations and troubleshooting - Familiarity with Zoiper and other SIP clients - Ability to identify and resolve connectivity issues - Experience with implementing advanced features such as call recording and IVR systems Overall, I am seeking a skilled freelancer who can help me identify and resolve the issue with my outgoing calls, and ensure that my SIP configuration is optimized for call quality and reliability.

Hi there, I'm looking for an experienced freelancer to help me with integrating an existing VoIP platform with an existing CRM. Voice over internet protocol(VoIP) is an ever-evolving technology that can greatly improve the way businesses communicate both externally and internally. I will be using FreePBX/Asterisk as the VoIP platform, and amoCRM/Kommo as my Customer Relationship Management (CRM) system. I need the following integration: VoIP Integration to allow real-time communication via telephone, Sales Pipeline Integration to track sales leads in my CRM,Click2Call, Incoming Call Card, Creating a Contact card, Smart Forwarding, Call Results card, Call Logging, Incoming Leads, Call List, Built-in call feature (WebRTC). All of which has examples on

I have 8 FXO and only accessiable via SIP. They are BT1, BT2, BT3, BT4, BT5, BT6, BT7 and BT8 I need asterisk to check for call counter before making a call. Only least / lowest number of the BT port counter can be dial out. Below is my logic. Everytime when asterisk receive a dial request 1. asterisk will check which BTs has the least number of call 2. once known, asterisk will dial the call. for example BT1 3. Regradless of the status of call, (ANSWERED, BUSY. UNAVAILABLE), asterisk will add BT1+1 The idea is to evenly distribute the call between all FXOs. You will need to have anydesk and access to my PC to work as asterisk is only available locally. (LAN)