Asterisk sipproiecte
Salut, Pe lan 1 avem reteaua locala calculatoare si telefoane voip si internet Pe lan 2 avem acces la serviciul digi sip care la care sa conectat dar nu mai vedem telefoanele din retea initial vedeam telefoanele ba chiar am gacut si teste cu un modul GOIP care a functionat bine .
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.
Am nevoie de cineva sa imi configureze un telefon cisco 7960 pentru serviciul de sip.
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................
Salut, Am un VPS cu FreePBX functional. Doresc sa adaug HylaFax + AvantFax si sa pot sa trimit faxuri prin iax sau sip. Am vazut ca ai facut ceva similar. 40$ fixed rate Poti sa ma ajuti?
Salut, Am un VPS cu FreePBX functional. Doresc sa adaug HylaFax + AvantFax si sa pot sa trimit faxuri prin iax sau sip. Am vazut ca ai facut ceva similar. 40$ fixed rate Poti sa ma ajuti?
Sunt interesat sa discutam pentru un proiect android, de voip (sip) asa cum este si viber sau skype... in primul rand ma intereseaza aplicatia pentru client dar si partea de server ca aplicatia sa fie functionala
Sunt interesat sa discutam pentru un proiect android, de voip (sip) asa cum este si viber sau skype... in primul rand ma intereseaza aplicatia pentru client dar si partea de server ca aplicatia sa fie functionala
Ma intereseaza sa discutam mai multe legate de o aplicatie de voip pe android ca si viber , caut pe cineva care sa faca partea de client sau sa customizeze un client open source de sip existent
Ma intereseaza sa discutam mai multe legate de o aplicatie de voip pe android ca si viber , caut pe cineva care sa faca partea de client sau sa customizeze un client open source de sip existent
Buna seara, as vrea sa discutam cu privire la o aplicatie de voip in care utilizatorul sa se logheze pe baza de nr de tel si sa primeasca sms pt confirmare precum si prin facebook, sa aibe o parte din functiile viber, configurarea SIP sa se faca printr-un api fara ca sa trebuiasca sa faca ceva utilizatorul final pentru a configura la inceput aplicatia.
...functie de cum iti este mai usor) Fereastra in care se introduce parola are, sub campul de text, buton OK si Cancel, ambele fara functii atribuite. Adevaratul OK va fi localizat, invizibil, altundeva. Telefonul se inregistreaza prin 3 parametrii: Server Username Password (campurile username si password devin hash-uri si nu pot fi vazute in clar) Telefonul functioneaza atat pe protocolul SIP cat si pe IAX2. Telefonul poate deschide oricate linii in paralel. Eventual, fiecare linie noua deschisa, le trece pe celelalte pe Hold, in sensul ca se aude numai linia activa. Atunci cand se inchide o linie, din oricare motiv in afara de apasarea efectiva pe butonul Hang-up, telefonul incearca sa o redeschida la infinit (sau de un nr de ori care poate fi setat din men...
I'm looking for an expert who can help me set up Docker on my Linux VPS. The purpose of this setup is to host multiple web applications and a VoIP service, specifically Asterisk. Key Requirements: - Full Docker installation and configuration - Implementation of multiple Docker images for my web applications and Asterisk - Ensuring the setup is efficient and reliable for hosting purposes Ideal Skills & Experience: - Extensive experience with Docker - Prior experience with VPS and Linux systems - Knowledge of setting up and managing Asterisk - Ability to implement and manage multiple Docker images Please note, I need someone who can handle the complete Docker setup from scratch on my VPS. I am looking for a professional who can deliver a robust and scalable ...
We have a current issue with our PBX and our Ubiquiti UDM Pro. When running the PBX on our internal network with the Ubiquiti, our desk phones on that server do not work properly. We can make outgoing calls, but incoming calls do not ring on the phones in the office. This issue only occurs on our internal network with our desk phones. Interestingly, the softphone app we use works fine. However, we have 2 Yealink phones and 1 Grandstream phone, and all 3 can make outgoing calls but do not ring for incoming calls. Additionally, on the Yealink phones, in the BLFs, we can see the other phones, but they are displayed as offline; perhaps this is a clue that can help resolve the issue.
I'm seeking an Asterisk dialplan expert to assist with setting up call recording and time-based routing on my system.
I am in need of a seasoned VOIP engineer with extensive Asterisk experience. My current Asterisk system requires regular performance monitoring and troubleshooting of any arising issues. Key Responsibilities: - Diagnose and resolve issues within the Asterisk system promptly - Regularly monitor the system's performance and identify potential areas for improvement Ideal Skills and Experience: - Proven experience with Asterisk - Strong problem-solving skills - Ability to conduct thorough performance assessments Tools: - Use of Asterisk CLI commands for monitoring system performance.
In ASTPP version 6.0, after fail and restart of MySQL, sip profile does not start. From fs_cli, it shows the following error: [ERR] switch_core_db.c:106 SQL ERR [database disk image is malformed] [ERR] switch_core_db.c:246 SQL ERR [database disk image is malformed] [CRIT] switch_core_sqldb.c:645 Failure to connect to CORE_DB sofia_reg_default! [CRIT] sofia.c:3165 Cannot Open SQL Database [default]! The project is to correct the error, so that when the sip profile is started or reloaded from https interface, customers can connect using phone number and password and server name. Freelancer must document each step done from the command line of the server to correct the error with small description for each step, so that anyone can take the doc and follow the procedure in case...
I'm seeking an expert in AI integration to enhance our PBX (SIP/VOIP) system. This project will involve using AI for tasks such as transcription, appointment scheduling, and order processing. The AI will also need to handle modifications to appointments. Key Requirements: - Integration of AI into our current practice management systems; specifically Best Practice and Dental for Windows. - The AI's role will be multifaceted, including but not limited to: - Transcribing calls and conversations - Scheduling appointments - Processing orders - Adjusting existing appointments - The AI should also have the ability to transcribe and generate letters for my review Ideal Skills and Experience: - Extensive knowledge and experience in AI integration, specifically with PBX sy...
I need a precise conversion of a SketchUp house model into AutoCAD and Revit. This includes capturing all architectural, structural, and MEP (Mechanical, Electrical, and Plumbing) details with high detail level. Ideal Skills: - Proficiency in AutoCAD and Revit. - Experience with high-detail model conversions. - Strong und...and Plumbing) details with high detail level. Ideal Skills: - Proficiency in AutoCAD and Revit. - Experience with high-detail model conversions. - Strong understanding of architectural, structural, and MEP details. The Revit model should be prepared to LOD 200 (General elements). Please only apply if you have a strong portfolio in similar work. Will need to use revit to convert drawings for MGO sip panels. You may need to tidy up the ketchup drawing before...
...and start within 48 hours. Let’s build this! ? Scope of Work ✅ AI Voice Assistant Development Answer patient calls, handle FAQs, and book/reschedule appointments. Real-time natural language processing (NLP) with zero latency. Route emergency cases to human receptionists. ✅ Phone Number Porting & VoIP Integration Port our existing business phone number to a compatible VoIP system (e.g., Twilio, Asterisk, Plivo, or a recommended alternative). If porting takes longer than 2 weeks, a temporary call-forwarding solution must be implemented immediately to ensure the AI system is live while the porting is finalized. Ensure seamless call routing from our existing number to the AI voice assistant. Maintain call quality, compliance, and security (HIPAA-compliant). Provide call tr...
I'm seeking an experienced developer to create a Telegram chat bot that uses Asterisk. The bot should be able to make calls, capture DTMF tones, and spoof caller IDs. It should also read custom text to the receiver of the call and be capable of placing calls from a spoofed number. Key features should include: - Making calls - Capturing DTMF tones - Spoofing caller ID The bot will only run on the Telegram platform, specifically as a chat bot. The ideal candidate should have: - Extensive experience with Asterisk - Proficient in developing Telegram bots - Strong understanding of DTMF tone capture - Ability to implement caller ID spoofing - Experience with text-to-speech technologies
I need a professional to configure the open-source Linphone for both iOS and Android. This setup is to connect with my SIP server for voice call purposes. Key Requirements: - Configure Linphone to work seamlessly with my SIP server for voice calls. - Implement a push server in the configuration to enhance connectivity. - Integrate specific branding elements into the Linphone app. This includes my logo, the designated color scheme, and the app name. Ideal Skills: - Extensive experience with Linphone configuration. - Proficient in SIP server integration. - Strong background in iOS and Android app configuration. - Capable of implementing a push server. - Skilled in app branding and customization.
I need a Revit engineer to modify a BIM model for a "sandwich panel" (Structural Insulated Metal Panel), or "SIP". The panel is currently 80mm thick and needs to be updated to 100mm thick. The ideal freelancer for this project would be someone who has substantial experience working with metadata and structural data in Revit. The modifications required for this project are not solely about adjusting the panel's thickness. The successful candidate will also need to update the following attributes: - Thermal properties - Structural load capacity - Material composition At this stage, we are uncertain whether the modified panel will need to meet specific industry standards or certifications. Therefore, the ability to understand and apply relevant industry norm...
I'm looking for an expert who can create a fully functional, cloud-b...allowing us to make calls to US customers and candidates from India. Key Features: - The server should support call recording, voicemail, and call analytics. - All features should be reliable and easy to use. Technical Requirements: - The server must be built on a cloud-based architecture. - It should utilize the SIP (Session Initiation Protocol). Ideal Skills: - Extensive experience in VoIP server development. - Proficiency in cloud technologies. - Deep understanding of SIP protocol. - Ability to implement call recording, voicemail, and call analytics features. - Excellent problem-solving skills and attention to detail. Please bid if you are able to deliver a professional and reliable VoIP server...
...Linphone (). Linphone es un cliente SIP o softphone para iOS, Android, MacOS, Linux y Windows. Actualmente Linphone es OpenSource, por lo que el trabajo es realizar una aplicación OpenSource a partir del código fuente de Linphone con algunos cambios que paso a indicar: 1) Hay que modificar los logotipos, nombre de la app, logo en instalador, colores y textos. Es decir, cambios en la marca y colores actuales por los de nuestra marca 2) Eliminar ciertas funciones y pantallas. Actualmente Linphone permite agregar una cuenta SIP externa y crear o agregar una cuenta “linphone” gratis. Se deberá eliminar esta última función y dejar únicamente agregar una cuenta SIP externa. Las pantallas de registrar y hacer login ...
...issues within my current network setup. My infrastructure includes Ubiquiti and Cisco equipment, and I need help with optimization, IP management, and security. Below are the key details of my network: Current Setup: • Router: Ubiquiti UDM Pro • Switches: 5 Ubiquiti switches, 3 Cisco switches • Access Points: 9 Ubiquiti APs • VoIP System: UCM6308 with 10 Grandstream IP phones and 5 Grandstream SIP speakers • Gateway: Yeastar TG400 • Servers: 2 servers running ERP • Client Devices: 50 computers (static IPs) and various Wi-Fi devices • Subnet: • DNS Issue: Devices with static IPs can only use the internet when manually setting DNS to 8.8.8.8. Problems to Address: 1. IP Conflict Resolution: Prevent multiple devices from using...
Posiadamy autorską aplikację IVR ASTERISK i potrzebujemy jej rozbudowę o dodatkowe funkcjonalności
As a registered mutual fund distributor, I'm seeking a digital marketing executive who can post ads across all social media platforms (Facebook, Instagram, LinkedIn) and help generate leads. The goal is to secure 50 account openings, each with a minimum monthly SIP of 1000rs. Key Responsibilities: - Posting ads on the aforementioned platforms and generating leads - Each successful lead will be rewarded with a commission of 100rs Skills & Experience: - Proven experience in digital marketing and lead generation - Proficient in using Facebook, Instagram, and LinkedIn for business promotion - Excellent communication skills, able to convey the mutual fund's investment guarantee to potential clients If a lead is generated, I will personally speak to them if they require...
...Flask/FastAPI for the server side). • JavaScript (Node.js for platform integration). • Knowledge of REST API for integration of third-party services. 4. Machine learning: • Basic understanding of model training, especially in the area of speech processing. • Working with TensorFlow, PyTorch libraries. 5. Integration with telephone systems: • Experience with automatic calling services: • Asterisk, SIP, VoIP. • Possibility of integration of CRM systems. 6. Experience in creating communication scripts: • Development of dialogue chains. • Creation of “human-like” communication. Positions to look for: • NLP engineer / AI engineer. • Developer of voice bots. • Developer of dialogue systems...
...for banking product data and financial services. 2. Platform 2 (Stock Market Basics, Trading Strategies, Risk Management): Identify and negotiate with stock market data providers, educational resources, and risk management tools. Find opportunities for brokerage partnerships or data integrations for trading tools. 3. Platform 3 (Mutual Funds, SIP, Portfolio Management, Tax-Saving Investments): Research and negotiate with mutual fund data providers, SIP calculators, and portfolio management services. Establish partnerships with tax-saving investment platforms and fund comparison services. Key Responsibilities: Research & Identify Collaborations: You will research and identify the necessary APIs and collaborations for each platform, ensuring they align with our goals ...
Project Overview We aim to develop a mo...Telecommunication Routing Configure routes to enable phone calls from our server to partner companies that accept calls for termination. Utilize SIP trunking and integrate with GoIP equipment. Infrastructure Recommendations Provide expert advice on the type of servers required for the application. Recommend necessary services to ensure optimal performance and reliability. Communication & Collaboration Daily communication with our team for updates and collaboration to ensure timely delivery. Required Skills VoIP technology. Magnus Billing installation and integration. SIP trunking configuration. IP telephony expertise. GoIP equipment setup. Asterisk and PBX configuration. Mobile app development (Android and iOS). Server ...
...performance in bullish markets. o IT: A safe sector for long-term growth. o Pharma: Defensive play during economic downturns. 3. Thematic ETFs: o EV, ESG, or Green Energy ETFs for emerging trends. 4. Debt ETFs: o Bharat Bond ETF 2030 or 2031 for predictable returns. 5. Commodity ETFs: o Gold ETFs: Hedge during inflationary times. ________________________________________ Step 3: Timing and Execution 1. SIP for Consistency: o Invest systematically on a fixed date every month. o Suggested amount: Start with 10-20% of monthly income. 2. Bulk Buys During Corrections: o Invest additional amounts when markets fall by 10-20%. o Example: During significant dips in Nifty/Sensex. 3. Trailing Stop-Loss for Exit: o Sell ETF units if they drop by 10-15% from peak value. _______________________...
...encountering challenges with integrating Twilio dialing and LiveKit SIP for adding phone participants to the conference. I’m looking for a skilled developer with expertise in LiveKit and Twilio SIP integration to help troubleshoot and resolve this issue. Beyond this initial task, I am open to discussing a long-term partnership for future enhancements and features. Key Tasks: Analyze the current implementation of LiveKit and Twilio SIP integration. Identify the root cause of the failure when phone participants attempt to join. Implement a robust solution to enable seamless phone integration into the conference platform. Requirements: - Strong experience with Node.js and Express.js. - Expertise in LiveKit and Twilio SIP technologies. - Prac...
...a dedicated SIP trunk with the capacity for 32 simultaneous calls. Each FreePBX system is responsible for managing 40 registered softphones, meaning a total of 2000 softphones are connected across the network. The FreePBX-trunk pairing is fixed, and each FreePBX is bound to its SIP trunk exclusively. All devices (FreePBX systems and softphones) are on the same local network, so connectivity between them is not an issue. If one of the SIP trunks fails, the 40 softphones associated with the corresponding FreePBX system are rendered unusable, even though there is ample capacity available in other trunks across the network. Goals We want to introduce a centralized, intelligent routing solution to optimize call routing and improve network resilience. Specifically: A cen...
I'm seeking a skilled professional to configure an Asterisk PBX system tailored for customer support. The main feature we require is call recording. There's no need for integration with any other software or systems at this point. Key Requirements: - Extensive experience with Asterisk PBX - Knowledgeable in setting up call recording features - Able to deliver a standalone, non-integrated system - Websocket integration for 2 way calls Your expertise will help us enhance our customer support communication and monitoring. Please bid if you can fulfill these requirements.
...developed in Visual Studio 6.0 (C/C#) on Windows XP. This system currently processes SIP-based phone calls through softphones, connects to a SQL Server backend, and includes a middleware component in C and C# (.NET). Key Areas of Focus: - Enhancing the system features - Improving the SIP-based phone call processing - Upgrading the integration capabilities with other systems The ideal candidate should have: - Strong expertise in legacy Windows environments - Deep knowledge in telephony protocols - Proficiency in .NET technologies - Proven experience in modernizing and integrating older codebases I value your ability to maintain the current system stability while pushing for these enhancements. Experience with SIP call processing is a plus. Your role will be pivota...
...Development of a FreePBX-Based Phone System Integrated with Call Flow Design Project Description: I am looking for a professional with expertise in FreePBX to design and implement a complete VoIP phone system that integrates with a Call Flow Design interface for easy management of call flows. Requirements: Installation and Configuration of FreePBX: Basic system setup. Addition and configuration of SIP trunks. Configuration of extensions and call groups. Integration with Call Flow Design: Implementation of a graphical interface that allows intuitive creation and management of call flows (e.g., drag-and-drop functionality). Ensure that the created flows are immediately applicable and functional on FreePBX. Additional Features: Implementation of an auto-attendant (IVR) c...
I'm looking for a skilled Android app developer to create an application for my company. Key Features: - The app will display the current time and company name at the top center of the screen. - Users will log in using a personal number and password, both entered into text boxes. The password will be concealed with an asterisk. - The app will include a button to submit the login details, labelled "Anmelden". - There will be a copyright notice at the bottom of the screen, aligned to the right and in a font size smaller than 16. CSS: Background: #000 Font Color #fff Button Color: orange Authentication: - The app will utilize an NFC system for authentication, along with a clear username and password scheme. - Users will authenticate using a text-based password or NF...
I'm seeking a professional to configure our on-premise 3CX PBX with SIP Trunks. This will connect to Twilio and subsequently to VAPI.ai. The primary functionality required from this setup is voice bot integration through VAPI.ai. Key Responsibilities: - Integration of an existing voice agent into the 3CX configuration. - Setting up SIP Trunks to connect to Twilio, and then to VAPI.ai. Ideal Candidate: - Extensive experience with 3CX PBX configuration. - Proven track record with Twilio and integrations. - Skilled in voice bot technology and configuration. - Familiarity with integrating pre-existing scripts into a new system. - Excellent problem-solving skills to troubleshoot any potential issues during the setup. Job will be considered finished when: - A call in the 3...
