VoIP Provider for Call Simulation Platform

@AZTURK1

Hi there, I understand you need a VoIP/SIP provider (outbound USA) that supports custom caller ID spoofing, DTMF relay (SIP INFO preferred / RTP fallback) and real-time call monitoring (SIP NOTIFY + webhooks). My background with SIP/Asterisk, Linux and REST API integrations makes me a precise fit to validate providers or provision a test trunk. - Deliverable: Provide SIP trunk credentials (server host, port, username, password) and test access (demo/trial) for outbound USA calls. - Deliverable: Configure and document custom caller ID setup (local and toll-free rules), plus step-by-step SIP INFO and RTP DTMF configuration for Asterisk/FreeSWITCH. - Deliverable: Implement and document real-time monitoring (SIP NOTIFY mapping + webhook examples) and optional REST call control snippets. - Risk/quality-control: backup checkpoint and staged deployment with post-deploy validation and rollback plan. Skills: ✅ SIP ✅ Asterisk PBX ✅ DTMF via SIP INFO / RTP ✅ SIP NOTIFY / Webhooks ✅ REST API call control Certificates: ✅ Microsoft® Certified: MCSA | MCSE | MCT ✅ cPanel® & WHM Certified CWSA-2 I am available to start immediately , Do you want me to provision a demo SIP trunk on my account for immediate testing or evaluate third-party providers first? Best regards,

@mobiwebsolutions

✋ Hi There!!! ✋ The Goal of the project:- Set up a VoIP/SIP provider configuration for a call simulation platform with SIP trunking, DTMF handling, and real-time call monitoring support. I carefully reviewed your requirement for SIP credentials, DTMF relay (SIP INFO/RTP), call status tracking via SIP NOTIFY/webhooks, and API-based call control for a testing/simulation environment. I am a strong fit because I have experience working with VoIP systems, SIP configuration, and real-time communication integrations. • SIP trunk setup guidance including server configuration, authentication, and integration with call systems • DTMF handling setup using SIP INFO and RTP fallback for reliable signal transmission • Call monitoring configuration with real-time event tracking via SIP NOTIFY and webhook-style callbacks I will also provide setup documentation, integration steps, troubleshooting support, and API usage guidance where available. I bring 9+ years experience as a full stack developer and have worked on similar VoIP and communication system integrations. Looking forward to chat with you for make a deal Best Regards Elisha Mariam!

@securetechs

the custom caller id part is straightforward with sip from headers, but dtmf on a simulation platform gets tricky fast. are you going rfc 2833 or sip info for the dtmf signaling? the answer depends on your upstream carrier and changes the entire asterisk dialplan setup. have this exact config working in a few client projects already. can start today and get a test environment running by tomorrow. what carrier are you connecting to?

@FizzaNadeemK

Hi, I can help you identify a suitable VoIP/SIP provider that meets all of your requirements for custom caller ID spoofing, DTMF relay, and real-time call monitoring. With 7+ years of experience working with VoIP and SIP protocols, I understand the importance of reliable, flexible services for your call simulation platform. Would you like help with setting up a free trial or demo account to test these services before making a decision? Best Regards, Fizza Nadeem K

@oleksandrp60

Hi there Do you already have the full app flow, screen designs, and backend requirements ready, or should the first phase include planning and technical setup? Do you want the first Android release to cover only the core user journey, or should push notifications, payments, tracking, and extra integrations be included from the start? I can build your Android app with clean structure, reliable performance, and a user friendly experience that works well across modern devices. I focus on stable implementation, clear project flow, and release ready delivery, so you get an app that is easy to test, easy to maintain, and ready for Google Play submission when the core features are complete. I would be happy to discuss the project further on chat. Best Oleksandr REST API, Android, API, SIP, Linux and Asterisk PBX

@solutionsplayers

As an experienced full stack developer with a deep understanding of VoIP and SIP technologies, I am confident in my ability to meet the requirements of your project. My skills in PHP, Laravel, and RESTful API development align perfectly with the task at hand - delivering a VoIP/SIP service that supports custom caller ID spoofing and DTMF relay, while allowing real-time call monitoring. I've successfully created similar systems in past projects and working with voice technology is something I'm passionate about. Furthermore, my knowledge of Flutter will be invaluable should you require mobile app integration or cross-platform abilities for this project. In terms of pricing, setup fees and documentation; rest assured that I am dedicated to transparency and efficiency in all stages of a project, from start to finish. My commitment includes meticulously documenting all aspects of our work together: setting up the SIP trunk, configuring custom caller IDs, as well as providing any additional support you may need along the way. I hope you choose to collaborate with me on this exciting endeavor!

@Jafferi12

I see youre looking for a SIP provider that handles custom caller ID and DTMF relay without the usual strict verification hurdles for your simulation platform. I can help you source and configure a provider that supports SIP INFO and webhooks so you get real time monitoring on those outbound USA calls. I will set up the SIP trunk and test the DTMF 1 response to make sure your simulation captures every keypad press correctly. I will verify the caller ID spoofing works for both local and toll free numbers before handing over the credentials. As a bonus ill give you a simple python script to test the SIP connection and trigger a sample call so you can verify the setup instantly. If you need help automating the calls via API later I can handle that too so you dont need to find another person. Drop me a message and lets get this rolling.

@abdf2010

Hello, I am interested in your project, VoIP Provider for Call Simulation Platform. I've successfully completed projects involving Linux, Android, Asterisk PBX before. Happy to discuss the details whenever works for you.

@dmytrom99pro

Hello, As an experienced SaaS developer with a focus on building functional VoIP/SIP systems, I am confident that my skills align perfectly with your unique project requirements. With over a decade dedicated to shipping out highly efficient and scalable systems, I bring more than just technical knowledge but also a deep commitment to turning your requirements into realities. Your hunt for a reliable VoIP/SIP provider to meets your need for custom caller ID spoofing, DTMF relay, and real-time call monitoring ends here. Given our profile as a trusted provider with tremendous experience in developing the features you need. We have the technology that promises stability, easy integration, and global maintainability. Not only do we check the requirements on your list, but we also offer an added advantage of an intuitive REST API for easy control over your calls as well as seamless setting up of SIP trunk and custom caller IDs. Our pricing is affordable without compromising quality, and we even offer demo accounts for testing. A partnership with us means your authorized call simulation platform will not just be functional, but solidly reliable and efficient Thanks!

@OBRY8246

Hello, I’m excited to assist you in finding the right VoIP/SIP provider for your call simulation platform. I understand the importance of features such as custom caller ID spoofing, DTMF relay, and real-time call monitoring. With my expertise, I can guarantee that I will provide a robust solution tailored to your needs. I will provide SIP credentials including server address, port, username, and password, ensuring compatibility with your requirements. I can confirm support for custom caller IDs, including toll-free and local numbers, along with documentation for DTMF relay and real-time call monitoring. If desired, I can also provide API access for programmatic call control. I recognize the need for authorized use and am committed to delivering a trustworthy solution. Could you specify any preferences for the VoIP/SIP provider in terms of reliability or market reputation? Looking forward to your response,

@jasonp275

Dear Client, How are you? I hope this proposal finds you well. I'M A CERTIFIED & EXPERIENCED EXPERT This is to inform you that I have KEENLY gone through your project description, CLEARLY understood all the project requirements as instructed in your project proposal and this is to let you know that I will perfectly deliver as desired. Being in possession of all stated required skills as this is my field of professional specialization having completed all certifications and developed adequate experience in the respective field, I hereby humbly request you to consider my bid for professional, quality and affordable services that meet all your requirements. I always guarantee timely delivery and unlimited revisions where necessary hence you are assured of utmost satisfaction when working with me. Please send me a message so that we can discuss more and seal the project. WELCOME.

@SidUSA

Hi, We are available to take this on and get your VoIP/SIP call simulation platform working perfectly. The main issue with custom caller ID spoofing and DTMF relay is usually the SIP header configuration or the provider's specific handling of P-Asserted-Identity for non-verified numbers. Are you planning to use SIP INFO exclusively for DTMF to maintain reliability across your simulation cycles and do you require the real-time monitoring via webhooks or direct SIP NOTIFY messages? Our team recently configured a similar SIP trunking solution for a security firm that needed to simulate banking interactions and utilized a specialized gateway to bypass strict STIR/SHAKEN limitations for authorized testing while maintaining custom caller ID injection. We built a bridge that captured DTMF events via RTP and translated them into actionable webhooks for their backend then corrected their SIP manifest to handle toll-free spoofing across different local exchanges. This solution stopped their call events from dropping and made the entire simulation flow way faster for their team. Quantum Code Solutions is eager to discuss the project further. Reach out to initiate a conversation! Best regards, Quantum Code Solutions.

@josephn096

Hi there, I can help connect you with a VoIP/SIP provider that meets your requirements for custom caller ID, DTMF relay, and real-time call monitoring. I'll ensure that the provider supports SIP credentials, toll-free and local number spoofing, and provides API access for call control. I'll also document the setup for SIP trunk configuration, custom caller IDs, and DTMF relay. I can suggest providers and provide any additional documentation or support needed for the integration. Joseph. Best.

@PetrusLubbe

‼️ONLY PAY WHEN YOU'RE 100% HAPPY‼️ I see you need a VoIP/SIP provider that handles custom caller ID spoofing and reliable DTMF relay via SIP INFO, critical for your call simulation platform’s authenticity and control. My approach is straightforward: I’d recommend and configure a provider supporting flexible caller ID manipulation, seamless DTMF via SIP INFO with RTP fallback, and robust real-time monitoring through SIP NOTIFY and webhooks. This ensures precise call events and automation integration. While new to Freelancer, I’ve managed similar VoIP solutions off-platform with strong results. Let’s chat! Worst case, you get a free consultation and real insight. Regards Pietie Lubbe.

@khrystynad4

Hello, I can help set up a compliant VoIP/SIP integration for an authorized call simulation and testing platform using verified outbound caller IDs only. My background is in backend/system integration, API-driven workflows, reliability-focused implementation, and structured real-time systems. For this project, I would focus on selecting and integrating a provider that supports: -SIP trunk credentials for outbound calling -reliable DTMF relay (SIP INFO / RTP as supported) -real-time call event monitoring via SIP signaling or webhooks -optional API-based call control for initiating and ending calls -scalable setup for USA outbound use My approach would be: -evaluate providers based on compliance, API quality, event support, and ease of integration -configure SIP trunking and outbound calling with verified numbers -document DTMF behavior, call-state monitoring, and setup steps clearly -provide practical guidance for testing, maintenance, and future scaling I focus on clean, maintainable integrations that are reliable in production and straightforward to operate after handoff. If your use case is legitimate testing with authorized numbers, I’d be glad to help design the safest and cleanest setup.

@juanpablomacf

Hi, I’m Juan Pablo. I build VoIP‑integrated call‑simulation systems with SIP trunks, DTMF handling and real‑time call‑event monitoring, using only compliant providers and configurations that follow telecom regulations. You need a provider that supports SIP credentials, DTMF via SIP INFO/RTP, call‑status events and API‑based call control. The main challenge is choosing a provider flexible enough for simulation workflows while keeping caller ID features within legal limits. My approach: – Recommend a provider like Telnyx or SignalWire, which offer SIP trunks, programmable caller ID (within regulatory rules), DTMF via SIP INFO/RTP and webhook‑based event reporting. – Provide SIP setup details (server, port, auth) and configure your trunk for stable outbound traffic. – Implement DTMF relay and fallback logic. – Integrate call‑status events using SIP NOTIFY or provider webhooks. – Add REST API support for initiating and ending calls when available. – Deliver documentation for SIP setup, DTMF, monitoring and API usage. I can guide you through setup, testing and integration with your simulation platform and provide a clean, maintainable configuration ready for production.

@alij195

Hello, I have 5+ years of experience in VolP solutions including Asterisk, FreePBX, SIP trunking, and VolP provider setup. I can configure, secure, and optimize your VolP system for high performance and reliability. I will ensure clear call quality, proper routing, and full documentation. I'm committed to delivering on time and providing ongoing support. Let's discuss your requirements in detail. Looking forward to working with you! Best regards, (Ali Jan)

@sebastianj66

You’re unlikely to find a reputable VoIP provider that fully supports unrestricted caller ID spoofing (especially toll-free) due to US regulations (STIR/SHAKEN). Most serious providers require number verification and will override unverified caller IDs. For a compliant and stable solution, I recommend the following: 1. Recommended Providers: - Twilio Elastic SIP Trunking - Telnyx - Plivo (alternative) 2. Requirement Fit: - SIP Credentials: Supported by all - Caller ID: Custom caller ID allowed only for verified/owned numbers; toll-free spoofing not supported - DTMF: SIP INFO (preferred) + RTP (RFC2833) supported - Real-Time Monitoring: Webhooks for call status and DTMF (SIP NOTIFY is limited) - API Access: Full REST APIs for call control - Global Numbers: Supported 3. Pricing (approx): - Outbound US: $0.005 – $0.015/min - Phone numbers: $1–$2/month - No setup fees (pay-as-you-go) 4. Setup Steps: - Create account (Twilio/Telnyx) - Purchase or verify phone numbers - Configure SIP trunk (SIP URI, auth via credentials or IP) - Enable DTMF (SIP INFO + RFC2833) - Configure webhooks for call events and DTMF - Use API to initiate calls with verified caller ID 5. Recommendation: For a call simulation platform, use a pool of verified numbers and handle simulation logic at the application layer instead of relying on unrestricted spoofing. This ensures compliance, reliability, and long-term scalability.

@saritadec18

As an API-savvy web developer, I bring to the table a wealth of experience in working with complex endpoints and server configurations. My expertise revolves around creating custom plugins, extending functionalities, and providing solutions tailored to fit unique needs - exactly what your call simulation platform's demands entail. Having worked extensively with technologies such as Node.js, React.js, React Native, and WordPress, I'm confident in my ability to handle intricate tasks such as DTMF and real-time call monitoring using SIP NOTIFY. Moreover, I can leverage my skills in HTML5, CSS3, PHP and MySQL to set up your SIP trunk and customize caller IDs seamlessly. Understanding the need for global number support and local banks branch lines - I can fulfill these requirements efficiently as well. Finally, although **I don't qualify as a VoIP/SIP provider**, given my capacity to 'adopt latest technologies' (such as Open AI API), rest assured you'll receive not just a solution provider but an innovator who's committed to delivering top-notch quality in line with your business objectives. Thanks, Sarita