Asterisk pbxpráce
Project Title: SIP to WhatsApp Integration Description: I am in need of a developer who can integrate SIP with WhatsApp to enable both text messaging and voice calls. I currently use Asterisk as my SIP provider and already have a WhatsApp Business Account. Ideal Skills and Experience: - Strong knowledge and experience with Asterisk and SIP integration - Proficient in WhatsApp Business API and its integration with SIP - Familiarity with handling text messaging and voice calls through WhatsApp - Ability to ensure smooth and reliable communication between SIP and WhatsApp - Experience in troubleshooting and resolving any integration-related issues - Strong problem-solving skills to address any challenges that may arise during the integration process
I am looking for a freelancer who can install Asterisk on my Ubuntu 20.04 LTS server for VoIP services. I have specific requirements for the configuration and customization of the Asterisk installation. Skills and Experience Needed: - Strong knowledge and experience in installing and configuring Asterisk on Ubuntu 20.04 LTS - Expertise in setting up VoIP services with Asterisk - Ability to understand and implement specific requirements for the Asterisk installation If you have experience with Asterisk installations and can meet these requirements, please submit your proposal.
Project Description: Kamailio and Fusion PBX Integration with Prepaid Billing I am looking for a skilled freelancer who can help me integrate Kamailio and Fusion PBX with Prepaid Billing to enhance the billing functionality of my system. The main goal of this integration is to achieve enhanced billing functionality. Specific features required in the prepaid billing system include: - Customizable rate plans to meet our unique business requirements. The integration process should be: - Fast and efficient to minimize downtime and get the system up and running quickly. Ideal Skills and Experience: - Strong knowledge and experience with Kamailio and Fusion PBX. - Proficiency in integrating systems with prepaid billing functionality. - Understanding of customizable rate ...
We are looking for a skilled developer to create a WebRTC app for our Asterix Cloud PBX system integrated with Pipedrive. Preferred Platform: - The app should be developed for the web browser platform. Specific Features: - We have a detailed list of features that we require for the app. But basic functions like incoming call, outgoing call, integration for click to dial , log call history are the minimum Timeline: - We have a timeline for the project, but it is flexible. Ideal Skills and Experience: - Proficiency in WebRTC development - Strong knowledge of PBX systems and integration with Pipedrive - Experience in developing web applications - Attention to detail and ability to follow a detailed feature list
I am seeking a freelancer who can provide one-on-one lessons in VOIP, PBX, and Asterisk. As a beginner in this field, I am looking for guidance in setting up a PBX system, configuring Asterisk, and troubleshooting VOIP issues. The ideal freelancer should have advanced knowledge in these topics and be able to provide theory support and practical configuration instructions.
...manager with experience in Asterisk PBX systems to help me set up and maintain my communication infrastructure. Specific features needed: - Call routing and forwarding to ensure efficient call management - Voicemail and greetings for professional and personalized customer interactions - Conference calling for team collaboration and remote meetings Integration with other communication tools: - Integration with other communication tools is required to streamline communication processes and ensure seamless connectivity. Ongoing maintenance and updates: - Assistance with regular maintenance and updates of the Asterisk PBX system is needed to ensure smooth and uninterrupted communication. Ideal skills and experience: - Strong knowledge and experience with ...
Hello, I recently acquired an OpenVox SWG-3032 and am looking to set up a telephony system from scratch for my office. Here are my basic requirements, more functionalities are welcome: 1. **Gateway Model**: OpenVox SWG-3032. 2. **Current Setup**: No existing telephony system. We're starting new. 3. **Intended Use & Features**: - Full integration with a CRM system (suggestions welcome). - Call recording for quality assurance. - Setup for both PCs and mobile phones in the office. - Recording and analytics for all inbound/user initiated outbound calls. - Also I want a system where everyone will know who to call today(database connectivity) (maybe a campaign where my employee go thru automated multiple back to back calls) - Outbound IVR calls for customer feedback, including detail...
hello, i have a mikrotik rb4011 router where i have a wan from isp on ether1 and a sip connect (without internet) on ether2. i need a config and correct routing on ether3 for pbx and config on ether4 for intern network and internet. you also can choose which router. maybe ccr2004 is the better one.
I am looking for a VoIP developer who can create a VoIP application for my project. The ideal candidate should have experience in developing VoIP applications with the following functionalities: - Call routing and forwarding - Call recording and monitoring - Voice recognition and transcription I have no specific requirements or preferences for the programming language to be used, so the developer can choose the most suitable language for the project. The VoIP application is expected to be used by: - Less than 100 users - 100-500 users - More than 500 users If you have experience in VoIP development and can fulfill the above requirements, please submit your proposal.
I am looking for a freelancer who can help me with a project involving an asterisk based system like free pbx. The specific features I need for the system include call forwarding, voicemail, and IVR (Interactive Voice Response). I have no specific requirements or preferences for the system, and I am open to suggestions from the freelancer. The deadline for this project is within a week. Ideal Skills and Experience: - Strong knowledge and experience with asterisk based systems - Proficiency in configuring call forwarding, voicemail, and IVR - Ability to suggest and implement best practices for the system - Excellent problem-solving skills and attention to detail - Timely delivery of project within the specified deadline
WordPress module. I need a module that will serve as an employee database added by companies. Due to GDPR, the entry in the database should look like this: 1. Name and surname (only the name should be shown on the website and the surname, the first letter and the rest under the asterisk are not visible). Here we choose a female/male avatar. 2. ID number consisting of 11 digits on the website we see only the first 2 and last 2 digits, the rest under an asterisk 3. City 4. The job position he held 5. Duration of employment 6. Rating on a scale of 1-10 stars 7. Employer's opinion - a place where we can write a subjective opinion. 8. Reason for terminating the contract: by the employee, by the employer, by mutual consent of the parties. The module requires a search engine w...
I am looking for a freelancer who can help me with a project that involves using FREEPBX and asterisk ari to place outbound calls out of a sip trunk using pjsip. The purpose of the outbound calls is for customer service. I do not have any existing infrastructure to support this project, so it needs to be built from scratch. For project updates, I prefer communication through email. Skills and Experience Required: - Strong knowledge and experience with FREEPBX, asterisk ari, and pjsip - Previous experience with setting up outbound calls and sip trunks - Excellent problem-solving skills - Ability to work independently and meet project deadlines
Project Description: Sip Trunk user with PBX support I am looking for a skilled professional who can help me with the remote installation of a PBX system. The ideal candidate should have experience working with Asterisk PBX. Requirements: - Familiarity with Asterisk PBX system - Ability to remotely install and configure the PBX system - Proficiency in setting up new features for the PBX system - Experience in configuring voicemail services - Knowledge of call forwarding and interactive voice response (IVR) setup Skills and Experience: - Previous experience in installing and configuring PBX systems - Strong knowledge of Asterisk PBX system - Ability to troubleshoot and resolve any issues that may arise during t...
3CX Fixes and Help I am looking for a freelancer who can assist me with resolving configuration issues with my 3CX system. Skills and Experience: - Familiarity with the 3CX system at an intermediate level - Experience in troubleshooting and resolving configuration issues - Knowledge of VoIP and PBX systems - Ability to integrate 3CX with other systems, if required This is a one-time fix project, and I am seeking someone who can provide prompt and efficient assistance in resolving the configuration issues. This could
I am looking for a freelancer who can integrate our Odoo community with our Asterisk-Issabel Phone Center. The specific feature that we want to integrate is the ability to be able to make calls straight from Odoo. We are open to suggestions for the preferred method of integration. The timeline for this project is immediate, with a deadline of 1-2 weeks. Ideal skills and experience for this job include: - Strong knowledge and experience with Odoo community and Asterisk-Issabel Phone Center - Proficiency in CRM integration - Familiarity with different integration methods, such as Direct API and Web Services
i want to implement a simple architecture. at this moment my asterisk server works fine with endpoints and it can initiate calls in both directions. now i want to use a proxy server in between endpoints and my server . so im using dsiprouter with domain name of and its installed successfully. now there is a problem. i connect my sip users to dsiprouter. and dsiprouter to asterisk as pass thru. when i call any number with sip endpoint there is no problem and the call reaches the dinstar gateway and dinstar call the sim number(ip2tel). but the opposite side of call when sim user call the gateway and gateway route the call through asterisk. when asterisk send request to dsiprouter the request cannot find the endpoint i tried (realm-outbound proxy and etc) but i di...
I am looking for a freelancer who can integrate the Yeastar PBX API with CRM. The specific feature I want to integrate is Caller ID integration to create new contact or pop up on Deskpro each incoming calls The deadline for this integration is within a week. I would like the integration to be bidirectional. Ideal Skills and Experience: - Experience with Yeastar PBX API integration - Proficiency in Deskpro CRM - Knowledge of Caller ID integration - Ability to complete the project within a week
I am looking for a freelancer who can help me with the installation of VitalPBX 4 (based on asterisk) connected with a Twilio number. OS is Debian. Additionally, I require complete configuration of this addon: (it has a .sh script auto-installation) After all is done, YOU MUST GUIDE ME to make the first succesful call with the addon.
Experiencia relevante y demostrable en Asterisk y Telefonía IP instalación Implementación Configuración Mantenimiento Soporte a usuarios
Experiencia relevante y demostrable en Asterisk y Telefonía IP instalación Implementación Configuración Mantenimiento Soporte a usuarios
Project Description: I am looking for a skilled developer who can create a WhatsApp to SIP Gateway using Asterisk. The main function of this project is to enable call forwarding between WhatsApp and SIP. I require the gateway to be built on an open-source platform. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the destination's WhatsApp number. Skills and Experience Needed: - Strong experience with Asterisk and VoIP systems - Proficiency in working with both WhatsApp and SIP protocols - Knowledge of call forwarding and routing techniques - Familiarity with open-source platforms for building gateways - Ability to troubleshoot and debug any issues that may arise during the development process
...Registered users will have the option to fill in the following information on a dedicated website: Domain Page URL for displaying the message (default: entry page of the site, the first opened page) (marked with an asterisk, meaning all pages of the domain, zero implies none) The message will contain the words "Payment with DCP Cash is available on this site" (modifiable) and any additional information site owners may want to add, such as a 10% discount for DCP payments. Wallet owner's name Wallet identification The address where the payment window will be displayed (marked with an asterisk, meaning all pages of the domain) Payment message text LocalStorage path for the payment amount LocalStorage path for currency (USD, DCP, or STAS) The address to which...
I am looking for an expert freelancer to integrate SIP services with med...freelancer to integrate SIP services with mediasoup for my project. The purpose of this integration is to enhance the video calling capabilities on my application. Requirements for the project: - Experience: The freelancer should have expert level experience in SIP video integration with mediasoup This Project may require expertise in VoIP technologies along with WebRTC, Asterisk, kamailio etc. - Tools and Technologies: NodeJS is preferred, however to get the project integration we are open to suggestions and do not have any specific tools or technologies in mind. If you have the required expertise and can provide guidance on the best tools and technologies to use, please bid on this project.
I have vicidial and asterisk installed, I need the following and the necessary configurations in vicidial for this to work. 1.- Create 17 trunks that are in the local network, [15 goip + 2 skyline gateway] 2.- Generate a way to call certain trunks together, easily, hopefully with just a few clicks: Example 1: call trunk 8, number 103 and play the dtmf tone of * Example 2: call trunk 9, number 103 and play the dtmf tone of * Example 3: call trunk 10, number 103 and play the dtmf tone of * Example 4: call trunk 16, number 103 and play the dtmf tone of * Example 5: call trunk 17, number 103 and play the dtmf tone of * Example 6: call all trunks 1, 2, 3, 4, 5, 6, 7, 11, 12, 13, 14 and 15 together, to number 103 and play the dtmf...
I am looking for a freelancer who can configure Asterisk freePBX with Linphone client for ZRTP encryption to enable secure VoIP communication. Here are the requirements for the project: - The primary purpose of the configuration is to enable post quantum encryption for secure VoIP communication. - The specific version of Asterisk and Linphone is not specified, any version will do. - There will be no requirement for ongoing maintenance or updates after the initial configuration. Ideal skills and experience for the job: - Strong knowledge and experience in configuring Asterisk freePBX and Linphone. - Familiarity with ZRTP encryption for secure VoIP communication. - Ability to work independently and deliver the project within the specified timeline. If you have the ne...
previously this solution is at Elastix PBX, now we want to move it to issabel pbx. telemarketing environment using a prefix XX to call out and will pickup random DID from database tables. there have a php trigger during outbound call and track this last called ID (customer) is below to which agent. if customer call back will return back to correct agent. I have the PHP script and database structure but I don't know how to implement at issabel pbx.
Connect FreePBX with DIDWW and Extensions, it was working but today i receive a 401 authentication failed in didwww Skills and Experience Needed: - Experience with FreePBX and Asterisk - Proficiency in setting up and configuring SIP trunks - Knowledge of connecting and configuring extensions in FreePBX Project Details: - I am using FreePBX as my PBX system and I need to connect it with DIDWW. - I prefer to use a SIP trunk for the DIDWW connection. - I have 1-5 extensions that need to be connected. - The freelancer should have experience in setting up and configuring SIP trunks in FreePBX. - They should also have knowledge of connecting and configuring extensions in FreePBX. - The project involves ensuring a seamless connection between FreePBX, DIDWW, and the extensions. - P...
Connect FreePBX with DIDWW and Extensions, it was working but today i receive a 401 authentication failed in didwww Skills and Experience Needed: - Experience with FreePBX and Asterisk - Proficiency in setting up and configuring SIP trunks - Knowledge of connecting and configuring extensions in FreePBX Project Details: - I am using FreePBX as my PBX system and I need to connect it with DIDWW. - I prefer to use a SIP trunk for the DIDWW connection. - I have 1-5 extensions that need to be connected. - The freelancer should have experience in setting up and configuring SIP trunks in FreePBX. - They should also have knowledge of connecting and configuring extensions in FreePBX. - The project involves ensuring a seamless connection between FreePBX, DIDWW, and the extensions. - P...
...to assist me with setting up my issabel PBX system for incoming and outgoing call recording. Requirements: - Experience with issabel PBX system - Knowledge of Asterisk, FreePBX, and Elastix - Ability to configure the PBX system for call recording - Familiarity with telemarketing purposes for PBX setup Tasks: - Set up the issabel PBX system for incoming and outgoing call recording - agent call out with display random cli, if customer missed this call and return a call will back to particular agent. - Configure the necessary settings for seamless call record - Ensure compatibility with our existing hardware and software - Provide guidance and recommendations for optimal telemarketing functionality If you have the required skills and experienc...
The project involves: - Upgrade FusionPBX 4.4.10 to 5.1 - Upgrade Debian to 11 or 12 - Ensuring all existing functionalities are maintained during the upgrade process If this is done well, I have 2 more servers with same OS and PBX that it will need to be done to. Ready to do right now.
**Job Title:** Kamailio & Asterisk VoIP Solutions Architect **Job Description:** We are seeking an experienced Kamailio & Asterisk VoIP Solutions Architect to design and implement a comprehensive telecommunication solution that integrates WebRTC, SIP users, and robust VoIP services. The ideal candidate will have extensive knowledge in setting up and configuring multi-tenant VoIP systems, ensuring high availability, and developing secure and scalable APIs for system interactions. **Key Responsibilities:** - Design and deploy a multi-tenant VoIP solution using Kamailio for SIP routing and trunk configurations, and Asterisk for media processing. - Implement WebRTC functionality for browser-based communication, alongside traditional SIP user capabilities. - Con...
...freelancer who can help me integrate my VoIP Asterisk phone system with Google Sheets. I want to track caller ID and call details in real-time. I've attached an example of that is required, the phone system is Yeastar P-Series I will provide the Asterisk manager interface details Skills and experience needed: - Proficiency in Asterisk and VoIP systems - Experience with Google Sheets API - Knowledge of real-time data integration - Strong problem-solving skills to troubleshoot any potential issues Project requirements: - Connect the VoIP Asterisk phone system to Google Sheets - Set up a live CDR feed to track caller ID and call details - Ensure real-time updates of the data in Google Sheets Please note that there are 1-5 phone lines connected to the Vo...
I need a freelancer who can troubleshoot and repair my FREEPBX system as I am unable to make outbound calls. Although there are no specific error messages, there are unusual system be...and repair my FREEPBX system as I am unable to make outbound calls. Although there are no specific error messages, there are unusual system behaviors. I have not made any recent changes or updates to the system. Skills and experience required for this project include: - Expertise in FREEPBX system troubleshooting and repair - Knowledge of VoIP protocols and configurations - Familiarity with Asterisk and Linux operating systems - Ability to diagnose and resolve issues with outbound calls - Strong problem-solving and communication skills The freelancer must be available to start working on the proj...
...Registered users will have the option to fill in the following information on a dedicated website: Domain Page URL for displaying the message (default: entry page of the site, the first opened page) (marked with an asterisk, meaning all pages of the domain, zero implies none) The message will contain the words "Payment with DCP Cash is available on this site" (modifiable) and any additional information site owners may want to add, such as a 10% discount for DCP payments. Wallet owner's name Wallet identification The address where the payment window will be displayed (marked with an asterisk, meaning all pages of the domain) Payment message text LocalStorage path for the payment amount LocalStorage path for currency (USD, DCP, or STAS) The address to which t...
Necesito instalar vicidial + 2 troncales en red local y 2 usuarios para llamar. Deben quedar conectadas las 2 troncales sip y poder generar llamadas las cuentas sip. Plazo de entrega 24 horas I need to install vicidial + 2 trunks in the local network and 2 users to call. The 2 SIP trunks must be connected and the SIP accounts must be able to generate calls. Delivery time 24 hours
Linux / Asterisk server with 8 port sim gateway outbound call center. (partially inbound) agents with ip phone/gsm phones Agents must able to hold/CB/transfer call to supervisor and other functions with key buttons only. other all standard features. daily excel report for whatsapp pls quote.
Looking for a FreePBX Expert that can help configure my instance. Also I have this extension for my CRM which I need assistant with connecting to my instance and configuring.
I am looking for a freelancer who can install a VOIP server PBX with Ejointech hardware. The requirements for this project are as follows: Operating System: - The VOIP server should be installed on a Linux operating system. VOIP Software: - The specific VOIP software that should be used for the PBX installation is Asterisk. Features and Functions: - The VOIP server should have the capability of call forwarding. - No other specific features or functions are required at this time. Ideal Skills and Experience: - Experience in installing and configuring VOIP servers. - Proficiency in working with Linux operating systems. - Familiarity with Asterisk VOIP software. Please provide your relevant experience and any certifications you may have in this area. Additiona...
I am looking for a freelancer who can help me set up FreePbx/Asterisk and provision 4 SIP phones for my VoIP phone system. Current Phone System Setup: VoIP phone system Software Installation: I already have the FreePbx and Asterisk software installed. SIP Phones: I am using Cisco SIP phones. 2xCP-6851-3PCC Phones 1xSPA-303 Phone 1xGigaset C530A Skills and Experience: - Experience with setting up FreePbx and Asterisk software - Knowledge of provisioning SIP phones, specifically Cisco phones - Familiarity with VoIP phone systems and configurations I NEED AN EXPERT IN CISCO PHONE.
Tenemos una planta Asterisk que ya está funcionando correctamente, pero necesitamos registrar teléfonos Cisco de las series 7800 y 8800 en ella. Este proyecto se divide en dos etapas, y estamos buscando a un profesional con las habilidades adecuadas para llevarlo a cabo. Etapa 1: Registrar 5 teléfonos Cisco 7841. Registrar 1 teléfono Cisco 8855.
Hola, Tenemos una planta Asterisk que ya está funcionando correctamente, pero necesitamos registrar teléfonos Cisco de las series 7800 y 8800 en ella. Este proyecto se divide en dos etapas, y estamos buscando a un profesional con las habilidades adecuadas para llevarlo a cabo. Etapa 1: Registrar 5 teléfonos Cisco 7841. Registrar 1 teléfono Cisco 8855. Etapa 2: (Para más adelante) Registrar 10 teléfonos Cisco 7841. Registrar 2 teléfonos Cisco 8855. Requisitos: Experiencia en configuración de sistemas Asterisk. Conocimiento de teléfonos Cisco de las series 7800 y 8800. Capacidad para acceder de forma remota a nuestra planta y al servidor TFTP. presupuesto son 60 dolares postualrse solo persona con experiencia y...
Hola, Tenemos una planta Asterisk que ya está funcionando correctamente, pero necesitamos registrar teléfonos Cisco de las series 7800 y 8800 en ella. Este proyecto se divide en dos etapas, y estamos buscando a un profesional con las habilidades adecuadas para llevarlo a cabo. Etapa 1: Registrar 5 teléfonos Cisco 7841. Registrar 1 teléfono Cisco 8855. Etapa 2: (Para más adelante) Registrar 10 teléfonos Cisco 7841. Registrar 2 teléfonos Cisco 8855. Requisitos: Experiencia en configuración de sistemas Asterisk. Conocimiento de teléfonos Cisco de las series 7800 y 8800. Capacidad para acceder de forma remota a nuestra planta y al servidor TFTP. presupuesto son 60 dolares postualrse solo persona con experiencia y...
Hola, Tenemos una planta Asterisk que ya está funcionando correctamente, pero necesitamos registrar teléfonos Cisco de las series 7800 y 8800 en ella. Este proyecto se divide en dos etapas, y estamos buscando a un profesional con las habilidades adecuadas para llevarlo a cabo. Etapa 1: Registrar 5 teléfonos Cisco 7841. Registrar 1 teléfono Cisco 8855. Etapa 2: (Para más adelante) Registrar 10 teléfonos Cisco 7841. Registrar 2 teléfonos Cisco 8855. Requisitos: Experiencia en configuración de sistemas Asterisk. Conocimiento de teléfonos Cisco de las series 7800 y 8800. Capacidad para acceder de forma remota a nuestra planta y al servidor TFTP. presupuesto son 60 dolares postualrse solo persona con experiencia y...
Hola, Tenemos una planta Asterisk que ya está funcionando correctamente, pero necesitamos registrar teléfonos Cisco de las series 7800 y 8800 en ella. Este proyecto se divide en dos etapas, y estamos buscando a un profesional con las habilidades adecuadas para llevarlo a cabo. Etapa 1: Registrar 5 teléfonos Cisco 7841. Registrar 1 teléfono Cisco 8855. Etapa 2: (Para más adelante) Registrar 10 teléfonos Cisco 7841. Registrar 2 teléfonos Cisco 8855. Requisitos: Experiencia en configuración de sistemas Asterisk. Conocimiento de teléfonos Cisco de las series 7800 y 8800. Capacidad para acceder de forma remota a nuestra planta y al servidor TFTP. presupuesto son 60 dolares postualrse solo persona con experiencia y...
Magnus Asterisk Inbound and Outbound DID Setup Skills and Experience Required: - Proficiency in Asterisk setup and configuration - Experience in setting up both inbound and outbound calling - Familiarity with DID providers and integration - Ability to customize Asterisk setup at a basic level Project Description: We are looking for a freelancer who can help us set up an Asterisk system with both inbound and outbound calling capabilities. We already have a DID provider in place and require basic customization for our Asterisk setup. Tasks: - Configure Asterisk for inbound and outbound calling - Integrate our existing DID provider with the Asterisk system - Customize the setup at a basic level to meet our requirements If you have experienc...
configurar telefonos ip cisco en entornos asterisk sin necesidad CEM CUM. instalacion de firmware y carga de configuraciones mediante tftp
Fusion PBX installation integrated with PSTN SIP trunk, Webphone(SIPML5 or any other for Audio, Video and conference), Webchat and Screensharing configured with Webrtc enabled. on GCP ubuntu22VM Recording Enabled
I am looking for a freelancer who can assist me with setting up a freepbx server. I already have all the necessary hardware and software in place. I need the freelancer to configure asterisk to meet my business needs and migrate my existing PBX to Alibaba Cloud. The ideal candidate should have experience in freepbx, asterisk, and Alibaba Cloud. The project should be completed within a week.
hi i have run voip example on esp32 lyraT kit and used local sip server(minisip) then it is working fine for call , but i have hosted the asterisk sip server on goolge cloud (the asterisk is working fine as i tested by calling using mobile apps. ) but when esp32 connects with this asterisk server whenever i call from mobile app upon pressing play button it says " no body is available to attend your call" .
