Configure SIP Server Connection

Job ID: 39933820

Budget: ₹600 – ₹1,500 INR

I’m looking for someone who can quickly connect my SIP-enabled PBX to a small dedicated server so we can place and receive voice calls. The setup only needs to support roughly 11–50 concurrent calls, so a straightforward configuration will do.

What I need:
• Install or configure a lean SIP platform on my server—Asterisk, FreeSWITCH, Kamailio or whatever you know best.
• Tie in my carrier trunk, enable registration, and verify two-way audio for inbound and outbound calls.
• Handle NAT, firewall rules, and basic codec tuning so audio quality is clean and consistent.
• Supply a short change log or cheat sheet outlining the commands and files you touched.

I can provide immediate SSH access and test credentials. The job is time-critical (ASAP), so please respond only if you’re ready to start right away and deliver a working setup with minimal fuss.
Related categories: Linux Asterisk PBX Cisco VoIP SIP Firewall