Asterisk openser sipJobs
Integrate Asterisk + UniMRCP + AWS LEX
I m working in voip in Morocco from 2014 I have a system of asterisk Isabel the problem is when my client send me trafic give him an erreur 486 must be 503 In this case my client lost his traffic https://www.freelancer.com/users/l.php?url=https:%2F%%2Fusers%%3Furl%3Dhttp:%252F%%252Fpt%252FA42hBUju%26sig%3D50c67e1b3a7f27f20771fd2ee320f07b9eb302904f9e19edea3a28d328aafbdf&sig=dbffe148af4b5367f232a97a6acda81aea4364dfabcb09ccf07b413f3f1c3be9
We must have set up a new asterisk server that follows the normal asterisk guidelines. We also need a script where we provide the telephone number (DID), opening hours also agents who are queuing, then it must do the setup in configs.
Vb6 ivr asterisk dialogic programme
Sonetel & Asterisk Configuration
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Добрий день, Мені потрібно виконати інтеграцію Asterisk+Zendesk - бажано, щоб при надходженні дзвінку створювався тікет, ідентифікувався дозвонювач, і т.д. Чи є в Вас такий досвід роботи? Ми знаходимося у Києві. Бюджет та подробиці оговоримо.
...already partially developed: The system is built using Issabel and Asterisk. The call center module in Issabel is set up and functional. OpenAI APIs have been integrated for dynamic conversational responses. Current Issue: The project needs to be finalized. Specifically, we need the call center module to handle the following: Import large .csv files containing 1,000-5,000 phone numbers. Automatically call customers listed in the .csv file. Offer services and respond to their questions dynamically using OpenAI API. We are seeking a freelancer who can complete this project efficiently, ensuring the system is capable of: Handling high-volume call operations without errors. Maintaining seamless integration between Issabel, Asterisk, and OpenAI APIs. Providing clear docum...
I need a freelancer who can help me integrate the online/offline status of my easybell softphone into my website. Key Requirements: - Experience in web development and SIP integration. - Knowledge about easybell softphone is a plus. - Ability to deliver a seamless and user-friendly interface. Please note, the only status information that needs to be displayed is whether the softphone is online or offline. Could you please provide a detailed outline of the requirements needed to initiate work, key design considerations, and the step-by-step approach you intend to take in order to achieve the desired outcome?
I am seeking a SIP trunk provider account primarily for customer support. The requirements for this account include: 1️⃣ Billing based on actual call duration to ensure cost-efficiency. 2️⃣ Fixed caller ID for all calls, simplifying identification for our clients. 3️⃣ Capability to make calls to China. The ideal provider should be able to offer 24/7 customer support, assisting us in maintaining seamless communication.
More details: What type of system does your organization currently use? Hybrid solution What level of caller ID customization do you require? Full control (custom name and number) What languages do you need the technician to be proficient in? English
Looking for a SIP trunk provider. We need a technician to help apply for a local SIP trunk provider to access our system. Requirements: 1️⃣ It needs to support calling China. 2️⃣ The billing method is calculated in seconds. 3️⃣ Customizable caller ID
...as possible, we need a technician who can directly provide the source code and help us build the VOIP system and software phone, and help us connect to the SIP trunk provider. Our customers need to make calls to China, and they need millions of call minutes every month. Key Requirements: - Design and build a custom VOIP system - Integrate the system with a SIP trunk provider - Implement call routing functionality - Ensure compatibility with Linux-based softphones The ideal candidate will: - Provide the necessary source code for the VOIP system - Offer long-term support and cooperation - Have a deep understanding of VOIP systems and SIP trunk providers Experience with Linux and softphone configuration is a must. Please only apply if you can provide a custom-built ...
I require a skilled professional to help me configure two FusionPBX IPPBX systems over a LAN for effective call routing between the two systems. Key Responsibilities: - Configure the two FusionPBX systems for seamless call routing. - Implement IP-based authentication for secure connection. Ideal Skills: - Extensive experience with FusionPBX configuration. - Proficient understanding of IP-based authentication methods. - Prior experience with LAN-based PBX systems is a plus.
...call starts from the Twilio SIP trunk. Under this SIP trunk, I have configured my Asterisk server. Every call begins when the user dials the Twilio number, and it is received by the Asterisk server. My Asterisk server immediately calls the Twilio SIP domain. Under the Twilio SIP domain, I have a webhook that routes the call to my worker. If my Twilio client’s workers do not respond within 30 seconds, I use Twilio Refer to send the call back to my Asterisk server, which then connects to my backup SIP. The backup SIP is a simple SIP for Linphone. Flow: Caller → Twilio SIP Trunk → Asterisk Server → Twilio SIP Domain → Worker (if available) If no response from workers...
Looking for US short duration dialer SIP traffic provider What VoIP professionals know good options for US-to-US dialer termination? Paying tip for good recommendations
...entry) and backyard, common swimming pool Design should incorporate: - Civil engineering and foundation plans. - Detailed kitchen layouts to enable kitchen manufacturers to use the plan without lengthy Q&A. The naterial shall be fire-proof, humidity-proof, and termite-proof. - Use of solar panels as a roofing material. - Preferably a steel structure with lightweight panels such as MgO, EPS, or SIP. - Everything except for furniture to be 'turnkey'. - Emphasis on durable (50+ years), yet sustainable materials. - A modular design with an option to build two dwelling units at a time. - High fire resistance and good insulation. - Prefab, but not based on containers. Deliverables: - Architectural plans - Civil plans - Bidding documents stating intentions, expectation...
...to help me recreate a UI for both Android and iOS. This app will connect to an existing FreePBX/Asterisk server for international calling and integrating with APIs for other functionalities like billing. All necessary APIs and a Figma design for the UX/UI are available. Current Status: - An Android version of this product exists, but we are redesigning and recreating it with Flutter for cross-platform compatibility. Required Features: - User registration/login - Call history tracking - Push notifications -app to phone calls (international) Visual Style: - The app should have a modern and sleek aesthetic. Ideal Skills and Experience: - Proficiency in Flutter - Experience with FreePBX/Asterisk - Familiarity with API integration - Ability to work from Figma designs - Prev...
...PBX and other commercial PBXs, utilizing SIP registration capabilities. Upon download the application should generate a unique Identifier and sync this data to the database. - Dynamic Configuration: The registration and configuration of the application should be dynamically assigned based on entries from a database tied to a unique identifier for each device and leverage GPS location for correct registration. This database would live in Azure. - User Interface: I prefer a native iOS and Android interface for the application. - Call Handling: The integration with the PBX system will primarily involve basic call handling and access to voicemail. Ideal skills and experience for the job: - Expertise in VoIP application development. - Strong knowledge of SIP and PBX systems. -...
I need a modern and sleek flyer for my bottleshop that promotes general brand awareness. The flyer should prominently feature: - Our store address and contact details - Our operational hours - Our late night cheers campaign, which offers additional discounts. I am running a Campaign named "Late Night Cheers" with the phrase "Sip More, Save More After 9!"Highlight the time slots, discounts, and urgency: 9 PM to 10 PM: "Start your night with 5% off your favorite drinks!" 10 PM to 11 PM: "End your night with 10% off every bottle!" * No further discount on already discounted products * I have setup a complete new renovated store that was previously run down , i need to attract a lot of attention to the new store and market it for the locals. ...
I'm looking for a skilled professional to integrate my on-premise PBX with SIP and WebRTC. The primary purpose of this project is to facilitate voice communication through this integration, with a specific focus on enabling real-time analytics as one of the key features. Ideal skills and experience for the job include: - Extensive knowledge and experience with SIP, WebRTC, and PBX systems - Expertise in voice communication systems - Proven track record of implementing real-time analytics in communication systems - Strong problem-solving skills to troubleshoot and resolve any potential issues during the integration process - Excellent communication skills to provide updates and explanations throughout the project. Please provide your portfolio and examples of similar ...
I'm looking to integrate JSSIP with Asterisk to provide a web-based client interface. The client interface should include call history, specifically focusing on missed calls. Key Requirements: - JSSIP and Asterisk integration: Proficiency in these technologies is crucial for this project. - Development of a Web-Based Client Interface: The primary goal of this integration is to create a user-friendly, web-based client interface. - Call History Feature: The interface should have a 'Call History' section that records missed calls. Ideal Skills: - Strong knowledge in VoIP technologies, particularly JSSIP and Asterisk. - Web development skills with a focus on creating interactive and intuitive user interfaces. - Experience with developing telecommunication...
I'm looking for a skilled professional to integrate Asterix with my SuiteCRM system. The main purpose of this integration is to automate call logging. Everything needed and the process of doing this is given on In simple term, its asterlink setup thats what I am looking at Astrix 13.23 and suitcrm 8.5 is ready. calls from LAN via sip dialer goes to astrix and then dials out via GSM gateway and this setup is perfectly working Key features of this integration should include: - Recording calls - Logging call duration and time - Tracking call outcomes I would like all call logs to be accessible through a centralized dashboard within SuiteCRM. Ideal skills for this job include: - Extensive experience with Asterix and SuiteCRM
...Implement a PHP-based web form that triggers outbound calls to users, utilizing CallFluent’s AI via Twilio. Enable bulk call initiation for sales campaigns from a list of phone numbers (CSV or DB). Ensure smooth SIP trunking between Dinstar and Twilio/CallFluent. Optimize call routing, logging, and reporting. Technical Requirements: Experience with VoIP and SIP (Asterisk, FreePBX, or similar platforms). Hands-on experience with Twilio API (voice calls, webhooks, and outbound call triggers). Understanding of Dinstar GSM Gateways and SIP trunking configurations. CallFluent AI integration expertise (or similar AI-based telephony platforms). PHP Development skills to implement web forms and initiate calls. F...
...have knowledge of Twilio, SIP, and Asterisk. We are trying to achieve the following use case: A caller dials a Twilio number. After some time or under certain conditions, Twilio forwards the call to an Asterisk server. Asterisk receives the call from Twilio and immediately connects the caller to sip:test (essentially bridging the caller directly to the SIP address). Once the call is connected between the caller and sip:linphone, Twilio should no longer be involved in the call (to avoid duration charges). The desired flow is: Caller → Twilio → Asterisk → sip linphone At the end, the connection should be: Caller → sip inphone We understand that we will need to pay Twilio for the incoming and outgoing legs un...
I'm facing issues with my Asterisk 22 dialplan. The problems are primarily with extension dialing. Specifically, call forwarding for the extensions is not functioning as it should. Key Requirements: - Diagnose and correct the issues within the dialplan - Ensure all extensions are correctly set up for call forwarding Ideal Skills and Experience: - Extensive experience with Asterisk, particularly version 22 - Strong understanding of dialplans and call routing - Excellent troubleshooting skills
...captures the essence of North, South, East, and West, harmonizing them into a single, refreshing experience. A Tapestry of Traditions From the vibrant Holi celebrations in the North to the rhythmic beats of Onam in the South, from the exuberant Durga Puja festivities in the East to the serene Ganesh Chaturthi celebrations in the West, Anoohya Botanicals embodies the spirit of unity in diversity. Each sip is a reminder of the rich tapestry of traditions that make India so unique. Colors that Dance Just as colors add vibrancy to India's festivals, they also bring life to our beverage. The rich, natural hues of Anoohya Botanicals are a visual treat, reflecting the colorful soul of India. Each bottle is a celebration of colors, mirroring the vivacity of India's cultural...
...manage their own SIP trunks, configure custom Caller IDs and prefixes, and track minute usage. The system must provide billing for users and reconcile minutes with providers. This will be a scalable, secure, and feature-rich platform capable of handling thousands of users and trunks. --- Project Goals 1. User Management: Support 1,000+ users, each managing up to 50 trunks. 2. Custom Configurations: Enable users to set custom prefixes and Caller IDs for each trunk. 3. Billing System: Track and bill users for minutes used and reconcile minutes with the provider. 4. Monitoring: Provide real-time metrics for active calls and system performance. 5. Scalability: Handle thousands of concurrent calls and scale with increasing user base. 6. Security: Ensure secure SIP...
...Narration/Caption: “Infused with the power of nature and tradition, every sip tells a story.” Scene 4: Approaching the Statue of Liberty • Visuals: The yacht approaches the Statue of Liberty, now towering in full view. The camera zooms in as the statue’s torch glimmers faintly. • Focus: A hand reaches out, lifting the Namah coffee mug from the table. Scene 5: The Sip That Ignites • Visuals: The person (a silhouette to keep it universal) takes a deliberate sip of the coffee. As they lower the mug, the torch of the Statue of Liberty suddenly bursts into a vibrant, radiant glow, symbolizing strength and brilliance. • Sound Effect: A warm, resonant hum as the torch’s brightness intensifies. • Narration/Caption: &ld...
...knowledge of Twilio, SIP, and Asterisk. We are trying to achieve the following use case: A caller dials a Twilio number. After some time or under certain conditions, Twilio forwards the call to an Asterisk server. Asterisk receives the call from Twilio and immediately connects the caller to sip:test (essentially bridging the caller directly to the SIP address). Once the call is connected between the caller and sip:linphone, Twilio should no longer be involved in the call (to avoid duration charges). The desired flow is: Caller → Twilio → Asterisk → sip linphone At the end, the connection should be: Caller → sip inphone We understand that we will need to pay Twilio for the incoming and outgoing ...
...connected to an Asterisk instance on my Linux home server. This project involves integrating the intercom to call Linphone accounts and executing specific actions based on inputs from the software's numeric keypad during a call. Key Tasks: - Configure the intercom to call Linphone accounts. - Implement a system where pressing digits on the software numeric keypad triggers pre-defined bash or Python scripts. (The primary action is to open the gate when a certain digit is pressed by my own API) I have accounts on Linphone and my Asterisk instance is set up. Importantly, I want to keep my Asterisk ports secure and not exposed to the internet. Therefore, the Asterisk should function as a client relative to Linphone. Ideal candidates should have: - Extens...
...Modification 4: Add the following data fields for note uploads (seller instructions): University Name* Lecture Notes for* (Subject Name) Course Name* Course Code Semester Teaching Professor* Year of Study* How to Prepare for This Subject* Price Type (Fixed/Negotiable) Price Upload Images of Notes Your Name Mobile Number Agree to Terms and Conditions Fields marked with an asterisk () are mandatory. 4. Modification 5: Add a WhatsApp button for direct communication between buyers and sellers of notes, similar to the textbook feature. 5. Modification 6: Add a login/registration requirement for students to upload or access notes. ________________________________________ Feature 3: University Events Management • Current Functionality: Students can uploa...
I have a small issue to address. My app is a VoIP iOS application that uses PJSIP to make SIP calls. However, after some updates were made, outgoing calls are working fine, but incoming calls are not functioning properly. When I accept an incoming call, there is no audio. Requirements: Strong experience in iOS development (Swift/Objective-C). Familiarity with pjsip or similar VoIP libraries (SIP, RTP, etc.). Experience with push notifications (PushKit) for VoIP calls. Ability to troubleshoot issues related to audio streams, network switching, and SIP protocols. Experience working with background tasks for VoIP apps. Skills: iOS Development (Swift/Objective-C) VoIP Development (pjsip, SIP) Networking (NAT traversal, network switching) PushKit (for VoIP push n...
Actualmente en mi empresa utilizamos una terminal de conmutación telefónica las líneas telefónicas externas son provistas por un IPS. Debido a que la terminal de conmutación PBX ya no puede soportar mayor cantidad de extensiones me veré forzado a modernizar a tecnología VOIP quizá utilizando Asterisk en un servidor rocky linux 9 que se encuentra de forma física. Además necesitare un archivo de los comandos utilizados a manera de instructivo ya que se necesita replicar lo mismo en otro edificio.
...professional who can set up an Asterisk server for me. This server is intended to manage both alarm-with-voice and alarm-without-voice functionalities. It will facilitate communication between clients through custom XML message exchanges via Asterisk server. Key Requirements: 1. Asterisk Server Configuration: The server should be configured to handle both types of alarms. 2. Real-Time Client Registration: Implement a system for instantaneous registration of clients from a MySQL database. This system should include an authentication mechanism and client status monitoring. 3. Testing Guidance: Provide instructions on how to assess the server's performance using SipSoftphone. Ideal Candidate: The perfect fit for this project is someone who possesses extensive kn...
...configure an Asterisk server for me. This server will handle both alarm-with-voice and alarm-without-voice, where clients send and receive XML-type messages with each other. The server needs to support real-time registration of clients from a local MySQL database, as I expect to serve a large number of clients. Key Requirements: 1. Configure the Asterisk server for both types of alarms. 2. Implement a feature for real-time registration of clients from a local MySQL database. 3. Provide guidance on how to test the server using SipSoftphone. The ideal candidate should have extensive experience with Asterisk server configuration and a strong understanding of MySQL. Skills in network configuration and VoIP technologies will be highly appreciated. Please note, I'...
I need an expert to set up Telnyx SIP telephony via Ringotel primarily for customer support. This setup should facilitate both voice calls and text messages. Key Requirements: - Expertise in Telnyx SIP telephony and Ringotel - Proven track record in setting up telephony systems for customer support - Ability to configure the system for voice calls and text messaging The system should be capable of handling calls and texts seamlessly. Ideal candidates for this project should possess exceptional skills in telephony systems setup and have prior experience working on similar projects.
I need a professional to set up my Asterisk server. The primary purpose is to connect it with a Postgres database and an Android Linphone On-Board Unit (OBU). Key Tasks: - Setting up an Asterisk server - Connecting Postgres with the Asterisk Server and Android Linphone OBU - Configuring Linphone to work seamlessly with Asterisk and Postgres The ideal freelancer should have: - Extensive experience in Asterisk server setup - Proficient in integrating Asterisk with Postgres - Familiar with Linphone - Knowledge of Android platform integration - Skills in VoIP technology
I'm in search of a seasoned Asterisk & FreePBX developer to create a comprehensive call center CRM. This web-based interface system should prioritize Direct Inward Dialing and incorporate essential call handling features such as Call Barge, Call Whisper, and Call Mute. Additional functionalities should encompass Call Queues, Call Transfer, Call Pause, Lead Upload, Recording, and Call Scheduling. Key Features: - Direct Inward Dialing as the top priority - Web-based interface for user accessibility - Critical call handling features: Call Barge, Call Whisper, Call Mute - Other functionalities: Call Queues, Call Transfer, Call Pause, Lead Upload, Recording, Call Scheduling Ideal candidates should have: - Extensive experience with Asterisk & FreePBX - Strong web dev...
...FreePBX to configure a TATA SIP trunk for both inbound and outbound calling. The project involves the following tasks: - SIP Trunk Configuration: Set up the TATA SIP trunk on the FreePBX system to facilitate both incoming and outgoing calls. - DID Configuration: I have existing DID numbers that need to be linked with this SIP trunk. Your task will be to properly configure these DIDs for seamless operation. - Call Routing: There are no specific call routing rules for inbound calls. All incoming calls will follow the default routing. Ideal skills for this project include a deep understanding of FreePBX, SIP trunk configuration, and DID management. Prior experience working with TATA SIP trunks will be a plus. The ultimate goal is to have a full...
I'm seeking a seasoned professional with deep expertise in Linphone and SIP server specifically tailored for a VoIP service provider. Linphone is already integrated, need some help with the issues - Ideal Skills and Experience: - Proven experience with Linphone and SIP server setup for a VoIP service provider. - Proficiency in software customization and branding. - Extensive knowledge of iOS and Android platform compatibility. - Strong troubleshooting skills and attention to detail.
About Us: We’re a premium coffee brand celebrating the art of South Indian Filter Coffee, refreshing Cold Brews, and rich, Pure Coffee Blends. Tradition meets innovation in every sip, and we want our packaging to reflect that magic. The Role: We’re looking for a talented Package Designer to create sleek, functional, and eye-catching designs for our coffee products. Your work will help bring our blends to life on the shelf and in customers’ hands. What You’ll Do: Design standout packaging for filter coffee, cold brew, and coffee blends (bags, bottles, boxes, labels, etc.) Balance heritage-inspired aesthetics with modern, clean design Deliver dielines, mock-ups, and print-ready files Collaborate with our team to refine and finalize designs What You Bring: Po...