AI Outbound Calling System Completion

@Shrutibimpl

Hello, I can resolve your Asterisk 407 SIP rejection and integrate your AI voice pipeline on your GCP Mumbai server. My plan is to inspect your PJSIP signaling logs and trace the exact Kamailio SBC headers. A 407 response without a Proxy Authenticate header usually means TeleCMI has misconfigured the IP authentication realm on their Kamailio SBC or expects specific PJSIP digest headers. I can coordinate with TeleCMI technical support to ensure your GCP static IP is properly bound to their IP authentication profile. Once outbound SIP and RTP traffic flows cleanly I can integrate the Silent Voice Kernel. I can write the ARI audio bridge connecting Deepgram Flux STT Sarvam 30B LLM and the dynamic TTS selector controlled by the VOICE_MODE environment variable for Bulbul v2 or ElevenLabs. 1) Do you have access to pjsip set logger on output logs for the malformed 407 response? 2) Is the Silent Voice Kernel using Asterisk ARI or WebSocket audio streaming to capture live RTP? 3) What is the preferred fallback behavior if the ElevenLabs or Sarvam TTS API experiences latency? Thanks, Bharat

@iimtpradeepdash

With my comprehensive knowledge of the project requisites and substantial experience in Linux and VoIP, I can skillfully troubleshoot the outbound INVITEs issue you've been grappling with. As a seasoned freelancer who has successfully delivered over 575 projects in my 7+ years of experience, I cultivate an approach that focuses on thoroughly understanding the entire system and proposing an efficient solution promptly. Moreover, I understand the importance of clean code architecture from day one to enable the long-term smooth functioning of your AI calling system. Drawing on this expertise and my comfortable experience with core technologies like Asterisk 18.10, GCP Mumbai VM, and TeleCMI SIP trunk service, I am confident in delivering a robust solution within your fixed budget and strict timeline. Lastly, I appreciate your concern about providing a complete handover along with source code and audit trails. As a reliable partner, I ensure transparent communications throughout the project duration and a thorough handover which will equip your team to later maintain the system independently if needed. Let's connect and dive deeper into your project - I’d love to provide real-time feedback on possible solutions we can implement to align your system with Silent Capital's needs.

@ashinnd84

Hello there, we are a team of senior automation, Full Stack Web and Mobile App Developers. Please, send me a message to discuss the work and finish in no time. Thanks Ashish Kumar.

@priyanshuk879

Hi, I can help improve your e-commerce website’s SEO performance and increase organic traffic with a proper long-term approach. I understand the site already has good content and clean Laravel code, so I will focus on finding ranking gaps, competitor analysis, technical SEO fixes, and improving visibility. I will audit the current setup, optimize on-page factors, improve site structure, and work on strategies that can bring steady growth. I’m looking for a long-term partnership and can put consistent effort into improving your search performance within your budget.

@shivp488

Hi, I have experience with **Asterisk, SIP/PJSIP, Ubuntu, VoIP integrations, and C++ backend development**, and I can help diagnose and complete your AI calling pipeline. I'll focus on identifying the root cause of the **407 Proxy Authentication Required** issue using SIP traces instead of repeating already-tested configurations. If the problem is carrier-side (TeleCMI/Kamailio provisioning, IP authentication, or routing), I'll coordinate with TeleCMI support and provide the exact changes required to achieve stable outbound calling. Once outbound SIP is working, I'll complete the integration with your existing Voice Kernel, Deepgram, Sarvam, and ElevenLabs, implementing a single configurable pipeline that supports both **FREE (Bulbul v2)** and **PREMIUM (ElevenLabs cloned voice)** modes via a simple configuration switch. I can deliver a fully tested solution, clear documentation, and assist with deployment and handover. Looking forward to discussing the details and getting started immediately. Best regards, Shiv

@rohank2502

Your 407 with no Proxy-Authenticate header is not an Asterisk bug. It is the SBC refusing to even challenge an INVITE it does not trust, and on GCP that is almost always the source address story. GCP VMs never bind the static IP directly. If external_signaling_address and external_media_address are not pinned in the pjsip transport, your Via and Contact carry the internal 10.x address, TeleCMI's Kamailio sees a mismatch with the whitelisted Origin IP, and rejects without a challenge. Exactly your empty 407. The flow: pjsip trace on one INVITE, confirm what TeleCMI actually sees, pin the transport, retest. If the trace proves it is their side, you get the exact provisioning line to hand TeleCMI support. Then VOICE_MODE=FREE/PREMIUM as one config switch on the same pipeline, no forked code. I run Sarvam integrations in production already. I can start the trace within the hour of SSH access. Two questions: is external_signaling_address set at all right now, and do I get TeleCMI portal access or their support thread?

@BrendanVerwey

✅ You don't need another promise. You need proof. One hidden challenge in your project is the outbound INVITE rejection due to the SIP 407 Proxy Authentication Required error, which can significantly hinder the system's performance if not addressed effectively. I recently completed a similar project for an AI-driven telephony system, where I successfully diagnosed and resolved complex SIP authentication issues, resulting in seamless outbound calling and improved system reliability. Understanding your goal of a two-mode AI calling system, my approach will involve a thorough diagnosis of the existing setup, focusing on the outbound SIP rejection while ensuring the architecture remains flexible for both FREE and PREMIUM modes. I prioritize practical solutions, ensuring long-term reliability through meticulous testing and quality execution. Good execution isn't obvious until you compare it to what's missing. Regards Brendan

@clintonv2

OVERWHELMED with all the AI generated proposals. Give me a few seconds to show you why I am different. I have extensive experience completing complex calling systems for service businesses. I understand the need to resolve the outbound 407 rejection and ensure seamless integration with TeleCMI and Asterisk. I would love to chat about your project, the worst that can happen is you walk away with a free consultation. Regards, Clinton.

@anshikam254

Hello, I understand that your existing AI calling system is already functional for inbound calls, and the main blocker is the outbound 407 authentication issue. I will troubleshoot using SIP traces and Asterisk logs instead of making configuration assumptions, identify whether the problem lies with Asterisk, Kamailio, or TeleCMI provisioning, and coordinate with TeleCMI if any carrier-side changes are required. Once outbound calling is stable, I will complete the end-to-end integration with your existing voice pipeline and implement a single configuration-based switch between Sarvam Bulbul v2 (FREE) and ElevenLabs cloned voice (PREMIUM) without creating separate systems. Deliverables include stable outbound SIP/RTP, configurable voice mode, end-to-end testing, debugging, and complete handover with documentation. I am committed to clear communication, systematic troubleshooting, and delivering a reliable production-ready solution. Looking forward to working with you.

@bhargav922002

Hello, I'm bharghav, with a decade of experience in Linux, Asterisk PBX, and VoIP, specializing in connecting complex communication systems. My background aligns perfectly with your AI outbound calling system. I understand you need to resolve the 407 Proxy Authentication issue and complete the two-mode AI calling system integration. I will diagnose the Asterisk outbound INVITEs, troubleshoot the Kamailio and TeleCMI integration, and ensure stable SIP and RTP. The existing Silent Voice Kernel will be connected, supporting both FREE and PREMIUM modes via a simple configuration change.

@ExactSpec

Business problem: outbound calls fail with SIP 407 (no Proxy-Authenticate) and the system must support a single pipeline with a config switch for FREE (Sarvam Bulbul v2) vs PREMIUM (ElevenLabs cloned voice). Understanding: current stack is Asterisk 18.10 on Ubuntu 22.04 in GCP Mumbai, Kamailio SBC, TeleCMI SIP trunk. Inbound works; outbound INVITEs return 407 without Proxy-Authenticate header, preventing Asterisk retry. Recommended approach: 1) obtain full SIP traces and current res_pjsip/Kamailio configs; 2) triage on-node (Asterisk/Kamailio) to reproduce 407 and identify missing Proxy-Authenticate or routing mismatch; 3) coordinate a focused carrier session with TeleCMI to surface any IP-auth or provisioning rules; 4) implement Asterisk/Kamailio routing fix and add VOICE_MODE toggle (env var) that selects TTS provider at runtime; 5) test end-to-end RTP and audio in both modes and deliver runbook and minimal rollback. Approved ExactSpec capability: ExactSpec developed customer-facing intake and platform integrations (CellularPort Buyback Platform) and has practical experience aligning product and platform priorities (Innovation36T Platform Collaboration). Delivery stages and milestones: Milestone 1 — Diagnosis & test plan: capture SIP traces, review configs, reproduce 407, carrier contact plan. Amount: 20; Percentage: 20; Delivery days: 2. Milestone 2 — Fix & VOICE_MODE implementation: implement routing/auth fix, add VOICE_MODE config switch, unit tests. Amount: 70; Percentage: 60; Delivery days: 3. Milestone 3 — End-to-end validation & handover: full test in FREE and PREMIUM modes, carrier sign-off, runbook, short support window. Amount: 41.35; Percentage: 20; Delivery days: 2. Price: INR 131.35 fixed (deliverables as above). Focused questions: 1) Can you provide SIP traces (full INVITE/407 exchange) and current pjsip + Kamailio config files? 2) Do you have written rights to use the cloned voice with ElevenLabs? Next step: if you confirm traces and voice-rights, I will schedule a 2–7 day engagement and begin Milestone 1 upon platform payment/PO and access. Deliverables: updated configs, runbook, test logs, and brief carrier coordination notes.

@ammarahmed20

As a highly skilled AI engineer with over six years of experience, my proficiencies and track record perfectly align with the out-of-the-box project you've posted. My work to date has consistently focused on combining artificial intelligence with real-world applications, specifically in voice communication and voice recognition domains—just as your project requires. A notable example of my skillset's tangible outcome that I believe will resonate with your needs is the creation of an autonomous AI agent for a medical provider who books calls intelligently into GoHighLevel. This demonstrates my understanding of complex architectures and ability to streamline intricate processes for smooth functioning. In summary, I offer not only the technical expertise in Asterisk and voice AI development necessary to resolve your current issues but also an inclusive understanding gained from working on challenging projects that involved various integrations across different industries. It all culminates in producing stable outbound SIP and RTP through Asterisk, ensuring that your system comprehensively serves its purpose as it should—in both its free and premium modes. I look forward to discussing this further or answering any questions you may have about how we can successfully complete this project together.

@shaxzodj

As an experienced and adaptable freelancer, I am uniquely qualified to complete your AI outbound calling system. My comprehensive understanding of Asterisk 18.10 and voice AI engineering positions me to effectively integrate, debug, test, and deliver a functional platform for Silent Capital. Additionally, I've worked extensively with the tools you mentioned - TeleCMI, Sarvam, and ElevenLabs - so I'm well-equipped to navigate any complications they may present. In my previous projects as a solution-driven Asterisk and voice AI engineer, I have always prioritized clear communication and timely execution. This has not only earned me satisfied clients but has also honed my problem-solving abilities. In line with your needs for FREE and PREMIUM modes 'switching' feature using the same pipeline, you'll benefit from my excellent coding skills and attention to detail that ensures seamless configuration changes.

@prathamd58

Hi — I can take this on and complete it in your 1–3 day window. Diagnosing the 407: A 407 with no Proxy-Authenticate header from Kamailio almost always means the SBC is rejecting the INVITE before it even reaches the auth challenge stage — commonly an IP-ACL/whitelist mismatch (static IP registered but not propagated to the outbound routing table), a mismatched From/P-Asserted-Identity vs. the whitelisted trunk identity, or TeleCMI's SBC expecting a specific Contact/Via format since Register is OFF. I'll pull SIP traces (pjsip set logger on, sngrep) on both legs to confirm exactly where Kamailio drops it, rather than guessing — this is usually a carrier-side provisioning gap, not an Asterisk config issue. Plan: trace → isolate root cause → coordinate the specific fix with TeleCMI (or adjust res_pjsip endpoint/identify sections if it's local) → verify stable outbound INVITE/200 OK/RTP → wire in VOICE_MODE config switch for Sarvam Bulbul v2 vs. ElevenLabs cloned voice on the same pipeline → full test + handover docs. Relevant experience: I've built and deployed Asterisk-based inbound/outbound AI calling systems as the answering layer for 3 different clients, each handling around 1000 concurrent/served customers — so I'm comfortable with SIP trunk debugging, Kamailio SBC quirks, and wiring AI conversation/TTS pipelines into production PBX infra under real load. Happy to start immediately with the SIP traces.