Asterisk pbxproiecte
Cine este interesat sa instaleze, configureze si asigure mentenanta pentru o centrala voip Asterisk la care utilizarea o sa fie minima (1 sau, maxim, 3 trunk - uri + 2 - maxim 5 extensii), este invitat sa ne trimita o oferta. Pentru ca avem in vedere activarea callback, IVR, XMPP, etc. vom aprecia experienta dovedita pe astlel de proiecte. Multumim!
Buna, caut specialist pentru instalarea si configurarea unui server asterisk/freepbx ca centrala telefonica si cu dongle usb cu functie SMS-gateway. cu stima,
Caut o persoana care sa ajute la instalarea si configurarea unui server FreePBX Asterisk, un server VOIP. Avem deja un calculator blocat pentru acest lucru.
Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+40728182013" sa nu mai apeleze. Am un buget de 1000$.
Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+40728182013" sa nu mai apeleze. Am un buget de 1000$.
Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+40728182013" sa nu mai apeleze. Am un buget de maxim 1000$
Salut, am instalat goautodial care include si asterisk, am reusit sa trec peste mai multe probleme succesive pana m-am impotmolit din nou. Cand incerc sa initiez apeluri imi spune ca nu este prefixul corect. Eu cred ca este din Dialplan Entry:, m-am uitat si la logurile CLI dar nu gasesc nici o solutie. Ma poti ajuta? Am un abonament de siptrunk de la RDS
Salut, am instalat goautodial care include si asterisk, am reusit sa trec peste mai multe probleme succesive pana m-am impotmolit din nou. Cand incerc sa initiez apeluri imi spune ca nu este prefixul corect. Eu cred ca este din Dialplan Entry:, m-am uitat si la logurile CLI dar nu gasesc nici o solutie. Ma poti ajuta? Am un abonament de siptrunk la RDS
Configurare campanii, db, si securizare asterisk, goaudiodial, vtiger.
2-5 ani experienta in suport nivel 1 si 2 pentru sistemul de operare si aplicatiile Microsoft Cunostinte LAN, VPN, WLAN si Remote Connectivity, TCP/IP, la nivel de instalare si troubleshooting Cunostinte de diagnoza hardware si reparatii minore pentru laptop, desktop, imprimante, monitoare. proiectoare, solutii de videoconferitna, ...experienta in suport nivel 1 si 2 pentru sistemul de operare si aplicatiile Microsoft Cunostinte LAN, VPN, WLAN si Remote Connectivity, TCP/IP, la nivel de instalare si troubleshooting Cunostinte de diagnoza hardware si reparatii minore pentru laptop, desktop, imprimante, monitoare. proiectoare, solutii de videoconferitna, instrumente backup, alte periferice Cunostinte si experienta suport Telecom (PBX, RAS) Experienta in managementul stocurilor...
Salut poti lasa un icq jabber yahoo daca ai te rog. Am nevoie de ceva pentru asterisk................
Salut poti lasa un icq jabber yahoo daca ai te rog. Am nevoie de ceva pentru asterisk................
i have issable pbx 5 i have setup 4 trunks type PJSIP this trunks from same providers . then i have 4 users from same provider as trunks evry trunk max channel limit is 1 so . i need it if i call as first send call via trunk 1 . is is busy or have any fail . try trunk 2 . if fail try trunk 3 . if fail try trunk 4 to be have 4 channels
Project Description: I need a complete Asterisk IVR system hosted on AWS EC2 that allows non-technical users to configure IVR menus via a simple web interface, uses AWS Polly for voice prompts, and captures both DTMF and voice feedback from callers. Voice responses must also be transcribed using a speech-to-text (STT) service, with the transcript included in email notifications and stored alongside the audio file. The system must be built with high availability (HA) in mind (Asterisk clustering, failover, or self-healing), and support secure remote access. Core Requirements: 1. IVR & Call Feedback Flow Inbound calls answered via Asterisk Prompts generated using AWS Polly TTS based on user input Caller provides: DTMF rating (1–5) Recorded voice messag...
...developing, and maintaining VOIP applications and systems. You will work with technologies such as Asterisk, FreePBX, Vocode, and other VOIP platforms to create robust and scalable communication solutions. Key Responsibilities: Develop and implement VOIP solutions using Asterisk, FreePBX, and other VOIP technologies. Collaborate with cross-functional teams to define, design, and ship new features. Troubleshoot and resolve VOIP-related issues and performance bottlenecks. Optimize existing VOIP systems for improved performance and reliability. Stay up-to-date with the latest trends and advancements in VOIP technology. Qualifications: Proven experience in VOIP development, specifically with Asterisk and FreePBX. Strong knowledge of VOIP protocols (SIP, RTP, etc.) an...
We are looking for a highly experienced consultant with strong knowledge of Asterisk and Artificial Intelligence (AI). We are currently setting up an Asterisk server that will integrate with services like OpenAI, with the goal of handling customer calls primarily through AI. A key part of the project is designing a natural conversational flow: When a customer calls, they should first hear a welcome message. After that, depending on what they say, the AI should be able to respond appropriately. The conversation must feel human-like and natural – and ideally, the AI should be able to react when the customer says something humorous or unexpected. Finally, the system should be able to collect information from the customer and pass it on (e.g., via email, CRM, or other sy...
I need to set up WebRTC and a webphone for customer support on my Asterisk-based portal. Requirements: - Integrate with existing custom user authentication system - Essential feature: Call transferring Ideal Skills: - Asterisk configuration expertise - WebRTC knowledge - Experience with custom authentication integration - Familiarity with customer support communication needs Looking for a freelancer who can efficiently handle this setup and ensure seamless communication.
I need an experienced Asterisk professional to install and configure a VoIP server for business telephony. The setup should include: - IVR (Interactive Voice Response) - Call Forwarding - Voicemail Ideal skills and experience: - Proficiency in Asterisk setup and configuration - Experience with business telephony systems - Knowledge of VoIP protocols and network configuration - Ability to implement IVR, call forwarding, and voicemail features Looking for someone who can deliver a reliable and scalable solution. Please provide examples of similar work done.
i have installed issable pbx voip system i was get a tls sip tunk and it is working good on any softphone i just need you register it on system issable pbx
Estoy buscando un experto en Asterisk para implementar una central telefónica en la nube. El servicio debe ser ofrecido a los clientes y debe incluir: - Llamadas VoIP - Grabación de llamadas - Conferencia telefónica Habilidades y experiencia ideales incluyen: - Dominio de Asterisk - Experiencia en configuración de PBX en la nube - Sólido conocimiento en VoIP y grabación de llamadas - Habilidades en configuración de conferencias telefónicas Por favor, proporciona ejemplos de trabajos anteriores relacionados.
I need assistance to integrate and configure my SIP Trunk Yeastar PBX P560 with for automating call handling. Key Requirements: - Configure SIP integration as per documentation. - Set up call routing features, specifically time-based routing rules. Ideal Skills and Experience: - Experience with Yeastar PBX systems - Familiarity with platform - Strong understanding of SIP trunk configuration - Knowledge in setting up call routing rules
I need an experienced Asterisk specialist to integrate VoIP gateways for call routing. Key Requirements: - Set up and configure VoIP gateways - Implement call routing functionalities - Ensure seamless integration with existing systems - Provide documentation and support post-implementation Ideal Skills and Experience: - Strong background in Asterisk and VoIP technologies - Experience with VoIP gateways and call routing - Problem-solving skills and attention to detail - Good communication for documentation and support Looking for someone who can deliver a robust and efficient solution.
I need a customised Asterisk VoIP dialer. Key Requirements: - Functionality: Call recording, auto-dialing, and call analytics. - Call Handling: Inbound, outbound, and both. - Platforms: Desktop, mobile, and web. Ideal Skills: - Experience with Asterisk and VoIP technologies. - Proficiency in developing cross-platform applications. - Strong background in implementing call handling features and analytics. Please provide examples of similar work done.
I'm looking for a modern, custom-branded proposal design in Word for our PBX, connectivity, copiers, access control, and CCTV solutions. Seperate solar Proposal. 2 Proposals: 1st Proposal with PBX, connectivity, copiers/printers, access control and CCTV offering. 2nd Proposal with Solar Solution only. Requirements: - Company logo - Contact information - Product images - Product information - Quote Terms and Conditions - Introduction and Foreword (Page 1) - Pricing Options Ideal Skills & Experience: - Proficiency in Microsoft Word - Graphic design skills for modern aesthetics - Experience with custom branding integration Please provide samples of previous work.
I'm looking for an experienced developer to integrate a voice bot with Asterisk for information retrieval via phone calls. Key requirements: - The voice bot's primary function will be information retrieval. - It will handle phone calls exclusively. - It needs to operate in one language (English). Ideal skills and experience: - Proficiency in Asterisk and voice bot technologies - Experience with voice recognition and call handling systems - Strong background in developing communication applications
Tengo una PBX modelo GCC6010 que tiene como SIP un dispositivo ATA modelo HT813 y que a su vez recibe una línea RJ11. La configuración en el entorno local (Zona A) ya se encuentra configurada incluso tiene una extensión en funcionamiento. Sin embargo, deseo agregar un teléfono desde otra ubicación con diferente red (Zona B), esta zona funciona con Starlink. En la Zona A tengo ip publica estática. Adjunto imagen para mayor entendimiento de lo que deseo configurar.
...Vector Search with Vector DBs (Weaviate, Qdrant, or Chroma) • Experience building RESTful APIs for voicebot or UCaaS integration • Familiarity with WebSockets, gRPC, or real-time data transport Nice to Have: • NLP/LLM agent framework knowledge (e.g., LangGraph, AutoGen, or ReAct agents) • Experience in voice-to-text (ASR) and text-to-voice (TTS) integration • Hands-on with FreeSWITCH, Kamailio, or Asterisk SIP platforms • Familiarity with tools like OpenAI Whisper, ElevenLabs, or Coqui • DevOps skills (Docker, Kubernetes, CI/CD) Key Requirements: - Functionalities to integrate: - Real-time communication - Automated workflows - Data analytics - Semantic search - Integration with existing systems: - Custom telephone application Ide...
I need an enhanced speech-to-text (STT) solution integrated with our Asterisk setup for nonprofit educational use. Currently, we’re using Vosk for STT in English and Hindi via Docker, but we need an always-on STT feature. Requirements: - Always-on STT: Users can speak at any time. - Access to educational resources: Online courses and reading materials. - Real-time transcription and language switching capabilities. Ideal Skills & Experience: - Proficiency in Asterisk and Vosk. - Experience with Docker. - Strong background in STT systems and real-time processing. - Multilingual support development. Looking for a solution that is reliable and efficient to enhance our educational platform.
I'm seeking an experienced Asterisk Java developer to implement voice calling functionality with SIP over WebSockets for WebRTC integration. You will also need to fully configure the Asterisk server. Key Requirements: - Implement voice calling using Asterisk and Java. - Integrate WebRTC via SIP over WebSockets. - Fully configure the Asterisk server. Ideal Skills and Experience: - Proficiency in Asterisk and Java. - Strong knowledge of WebRTC and SIP over WebSockets. - Experience with Asterisk server configuration. Please provide relevant work samples and experience.
Custom Virtual PBX for Incoming Calls – Call Forwarding to Israeli Mobile Phones I am looking for an experienced VoIP / Asterisk / FreePBX expert to set up a custom virtual PBX (IVR) focused on handling incoming calls only and forwarding them to mobile phones in Israel. This system should support up to 35 agents, with routing based on time of day (morning/evening shifts) and backup forwarding if the primary agent doesn't answer. Requirements: * Must support Hebrew (IVR prompts and interface if possible) * Forward incoming calls to Israeli mobile phone numbers * Call queue support (including number-in-line announcements) * Time-based routing (morning/evening shifts) * Backup routing if a call isn’t answered * Custom IVR greetings (audio files or TT...
...MakeTime is a set of tools to easily manage, visualize, and hear your scheduled activity by text and phone. View all of your scheduled activity on a single page, and print it. Or, get specific answers by text or phone. _________________________________________________________________________________________________ Project: 1. [HOLD - on this step pending Virtual PBX discussion] Confirm scheduled item read-out is current in Virtual PBX, and matches what is in MakeTime Google Sheet > Ask > [appropriate field] and TTS Google Sheet > Production Media > Column H. Match: i) Coupons field is MakeTime GoogleSheet > Ask > B49. []Main data entry, answers engine. Imports from text, VPbx voicemail, and Zap. Feeds answers via “Ask” page to TTS Google...
I'm seeking a skilled VoIP Engineer to develop, integrate, and maintain a robust communication platform using Asterisk and WebRTC. The solution should be reliable, scalable, and seamlessly integrate with our existing systems. Key requirements: - Implement features like call routing, recording, and IVR. - Integrate with CRM, billing systems, customer support tools, and a custom-developed AI-integrated PBX. - Support for voice calls and WebRTC-based conference channels. Ideal skills and experience: - Strong expertise in Asterisk and WebRTC - Proven experience in VoIP solution development - Familiarity with system integration - Knowledge of AI in communication systems - Excellent problem-solving skills and attention to detail Looking forward to your expertise in b...
...MakeTime is a set of tools to easily manage, visualize, and hear your scheduled activity by text and phone. View all of your scheduled activity on a single page, and print it. Or, get specific answers by text or phone. _________________________________________________________________________________________________ Project: 1. [HOLD - on this step pending Virtual PBX discussion] Confirm scheduled item read-out is current in Virtual PBX, and matches what is in MakeTime Google Sheet > Ask > [appropriate field] and TTS Google Sheet > Production Media > Column H. Match: i) Coupons field is MakeTime GoogleSheet > Ask > B49. []Main data entry, answers engine. Imports from text, VPbx voicemail, and Zap. Feeds answers via “Ask” page to TTS Google...
I need an AI agent to assist with customer support, specifically through live chat. The AI agent will be integrated with our Vicidial systems to streamline interactions and enhance efficiency. Key Requirements: - Setup of AI agent for live chat - Integration with multiple vicidial Fusion Pbx - Configuration to ensure responsive customer support Ideal skills and experience: - Experience with AI customer support tools - Strong background in CRM integration - Proficiency in configuring AI for live chat functionalities
I need assistance linking an AI agent to my cloud-based PBX system. The AI will primarily route calls to the right agent. Key requirements: - Configure AI agent for call routing - Ensure seamless integration with cloud-based PBX - Basic setup and testing Ideal skills: - Experience with cloud-based PBX systems - Knowledge of AI call routing - Technical configuration expertise
I need an AI developer to set up inbound call routing with Fusion PBX. The primary task is to transfer calls to Fusion PBX. Additionally, I require live call handling during the calls. Ideal Skills and Experience: - Experience with AI and voice processing - Familiarity with Fusion PBX - Ability to set up call routing systems
Set up a Fusion PBX system on Debian, focusing on conference calling functionality. Requirements: - Integrate with existing DNS configuration. Ideal Skills: - Experience with Debian and Fusion PBX. - Intermediate-level troubleshooting and setup.
I'm seeking a Freeswitch VoIP expert for my on-premises PBX setup. The tasks include: - Configuration and setup of the Freeswitch VoIP system - Troubleshooting and debugging current issues - Custom development to meet specific needs Please ensure you have a strong background in working with SIP protocols. Ideal candidates should have experience with on-premises PBX systems and a proven track record in VoIP technologies.
I need an expert to help set up a new Asterisk PBX system using a VoIP provider. The system should include call recording functionality. Requirements: - Expertise in Asterisk PBX setup - Experience with VoIP providers - Ability to configure call recording features
I'm in need of a seasoned Vicidial professional who can assist with troubleshooting and configuring SIP trunks on a second server. The ideal candidate should have: 1. Vicidial installation, setup, and ...second server. The ideal candidate should have: 1. Vicidial installation, setup, and optimization expertise 2. Proficiency in IP-based SIP trunk configuration 3. Skills in troubleshooting call connectivity, drop issues, and lag 4. Asterisk dial plan configuration & debugging experience 5. Knowledge in adjusting Firewall and NAT settings for VoIP stability If you can resolve issues quickly and have hands-on experience in Vicidial and Asterisk, I'd love to work with you! Project Type: One-time setup & ongoing support Experience Required: 2+ years in Vici...
Install free PBX on a Raspberry Pi (3 Model B 1GB or new) The issue: I have a SIP phone number which only works on the ISP in country A. Thus, the SIP can only connect to the server when connected to the ISP WAN INSIDE country A. I have a free PBX hosted in country B (cloud). I like to have free pbx installed on a Raspberry PI to function as a gateway between my WAN connection in country A and my cloud free PBX in country B. So that after installing the Raspberry Pi, I can make and receive calls from the SIP phone number in country A. Note: The SIP supports multiple incoming- and outgoing calls at the same time. I have a Raspberry Pi 3 Model B 1GB, but if really needed I can buy a new Raspberry Pi. The project includes among others: A. The remote installatio...
I'm looking for a network or phone specialist with expertise in call spoofing, specifically for VoIP systems using Asterisk. My needs include: - Implementation of call spoofing: You will set up the necessary components to enable call spoofing on our VoIP systems. - Understanding call spoofing: Provide detailed insights and understanding of the call spoofing process. - Setting up spoofing software: Configure and install any required software for effective call spoofing Ideal skills and experience: - Deep knowledge of call spoofing techniques - Proficient in VoIP systems, especially Asterisk - Experience in configuring spoofing software *** NOTE THE SPOOFING IS FOR YOUTUBE CONTENT CREATING AND NEVER FOR UNETHICAL PURPOSES*** Looking forward to your expertise in this pr...
I'm looking for an experienced developer to implement SIP and MRCP protocols. The main focus will be on SIP and all components of MRCP integration on Cisco PBX Key Requirements: - SIP Protocol - MRCP Components: TTS, ASR, Speaker verification/identification Ideal Skills and Experience: - Strong background in SIP and MRCP protocols - Experience with TTS, ASR, and speaker verification technologies - Ability to design and implement robust media negotiation strategies We need to code & develop SIP server of our own without open source!! Please provide examples of similar work done and relevant qualifications.
I'm looking for an experienced developer to implement SIP and MRCP protocols. The main focus will be on SIP and all components of MRCP integration on Cisco PBX Key Requirements: - SIP Protocol - MRCP Components: TTS, ASR, Speaker verification/identification Ideal Skills and Experience: - Strong background in SIP and MRCP protocols - Experience with TTS, ASR, and speaker verification technologies - Ability to design and implement robust media negotiation strategies We need to code & develop SIP server of our own without open source!! Please provide examples of similar work done and relevant qualifications.
Smartswitch to my external SIP gateway (Ejoin) for handling outbound calls. The tasks include: Installing Smartswitch (VDI or ISO) on Virtua...Setting up network connectivity and web GUI access Creating and configuring a SIP Trunk to route outbound calls to server: Supporting up to 16 ports through the connected gateway Configuring routing logic and ensuring successful call flow Enabling call logs (CDR), monitoring, and basic call testing Providing documentation or brief explanation of the setup Requirements: Experience with Asterisk-based systems, VoIP, and SIP Trunking Familiarity with Smartswitch or similar softswitch platforms Comfortable working remotely via AnyDesk or TeamViewer if needed Please provide examples of previous similar work and your estimated cost and...
...CPU load) Each instance runs locally with: MariaDB (currently using mostly MyISAM) Asterisk Vicidial Web We are starting with one instance to audit, migrate, and optimize the database, with the goal of reproducing the process across other environments afterward. Your mission includes: 1. Audit storage engine & stability - Identify all tables still using MyISAM - Evaluate locking risks, potential for corruption, and performance bottlenecks - Define a clean, safe migration path to InnoDB - Explain any Vicidial-specific constraints or behaviors that must be considered 2. Tune memory & InnoDB performance Recommend the optimal innodb_buffer_pool_size (instance has 75–85 Go RAM total, shared with Asterisk and Apache) - Tune other critical InnoDB parameters:...
...handle lots of people talking at once—up to 1,000 at the same time! It should work very fast, never crash, and always keep secrets safe. What You Should Use A cloud like Linode to run everything. A place to save the phone numbers in a locked way (we like PostgreSQL). A brain to remember who has which nickname (Redis or something fast). A gate for phone calls and texts (like Twilio, Plivo, or Asterisk). A good way to send tasks to workers (like RabbitMQ). A system that can grow when more people use it (like Kubernetes). A way to see if everything works fine (maybe Prometheus + Grafana). What We Need From YouWhen you send us your proposal, please include: A drawing of how the system will work. A short note about your team and why you’re the right one. How ...
I'm looking for an experienced developer to implement SIP and MRCP protocols. The main focus will be on SIP and all components of MRCP integration on Cisco PBX Key Requirements: - SIP Protocoln - MRCP Components: TTS, ASR, Speaker verification/identification Ideal Skills and Experience: - Strong background in SIP and MRCP protocols - Experience with TTS, ASR, and speaker verification technologies - Ability to design and implement robust media negotiation strategies Please provide examples of similar work done and relevant qualifications.
Se necesita construir / configuración Api que permita integrar un crm desarrollado internamente con pbx asterisk issabel.
