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The Future of GenAI, Cybersecurity, and VoIP: What You Need to Know

Configuring BSNL SIP Trunk (SIP PRI) on Asterisk with OpenVPN | A Complete PJSIP Guide (2026 Edition)

  A practical, production-ready guide to configuring BSNL SIP PRI (IMS SIP Trunk) on modern Asterisk (PJSIP) with OpenVPN. Covers authentication, routing, user_eq_phone, scoring logic, debugging, and real-world carrier behavior. BSNL now delivers SIP PRI over FTTH fiber using IMS-based SIP signaling routed through VPN (OpenVPN or SoftEther). Key realities: SIP Proxy is reachable only via VPN SIP authentication may use: IMS expects proper user=phone format RTP may traverse different subnets If your PBX doesn’t support OpenVPN natively, you must deploy a router or gateway device that does. 🛠 Step 1 – Configure OpenVPN BSNL provides: VPN Server Primary IP VPN Server Secondary IP Virtual IP (Client) Gateway Mask After connecting: ip addr show tun0 ip route ping 10.x.x.x # SIP Proxy Ensure: VPN interface is UP Route to SIP proxy goes through VPN SIP proxy is reachable 🧩 Step 2 – Configure BSNL SIP Trunk (PJSIP – NOT chan_sip) Modern Asterisk uses PJSIP , not sip.conf. Below is a ...

SIP Trunk Registered but No Calls?


 

The Hidden Asterisk + PJSIP + IMS Trap (BSNL Case Study)

Why a perfectly registered SIP trunk can still make ZERO calls — and how to fix it


Most SIP engineers panic when they see this:

REGISTER → 401 → 200 OK
Status: Registered

…and then nothing happens.

No outbound calls. No INVITE. No SIP logs. No errors.

I recently debugged a BSNL IMS SIP trunk on Asterisk 20 (PJSIP) that worked flawlessly on another PBX — yet refused to place calls here.

👉 Credentials were correct 👉 Registration was successful 👉 Network & NAT were fine

Still: no calls left the system

This article explains why this happens, how to diagnose it, and how to fix it permanently.


🔍 The Illusion of “Trunk Working”

In Asterisk, registration success does NOT mean call routing success.

You can have:

✅ pjsip show registrations → Registered ❌ Zero outbound SIP traffic

Because registration ≠ dialing.


🧠 The Real Root Cause (90% of Such Issues)

Your dialplan never dials the trunk

In modern PBX architectures (especially API-driven ones), outbound calls often flow like this:

Extension
 → Outbound policy
   → Stasis (Node.js / API)
     → (nothing originates a trunk call)

So Asterisk happily waits… while the trunk sits idle.

No INVITE is ever generated.


🚫 The Silent Killer: Stasis Without Originate

In my case, the outbound dialplan ended with:

Stasis(clixxo,outbound,${ORIG_EXTEN},${DIALED_NUM})
Hangup()

That looks “advanced” but unless your Stasis app explicitly originates a call, the trunk will never be used.

This is why:

  • pjsip logger shows nothing
  • Provider sees no traffic
  • Engineers blame SIP credentials 😅


✅ Two Correct Ways to Fix It

Option 1 — Classic Asterisk Dialplan (Fastest Fix)

If you want Asterisk to control calls:

Dial(PJSIP/${DIALED_NUM}@tr_BSNL,60)

This immediately generates:

INVITE sip:number@proxy

Perfect for stable production PBXs.


Option 2 — API-Controlled PBX (Modern Architecture)

If you’re using Node.js / Stasis / ARI, then your app must originate the call:

ari.channels.originate({
  endpoint: `PJSIP/${number}@tr_BSNL`,
  callerId: '911202210100',
  app: 'pbx'
});

Without this, registration is meaningless.


📡 BSNL IMS-Specific Lessons (Very Important)

BSNL (and most IMS providers) are strict:

❌ Public DNS won’t resolve IMS domains

*.ims.bsnl.in often requires:

  • Private DNS
  • VPN DNS
  • Or direct proxy IP

❌ +91 breaks many IMS trunks

Use:

911202210100

NOT:

+911202210100

❌ Realm in trunk auth

IMS trunks usually do NOT want a local PBX realm


🔥 Golden Rule of SIP Debugging

If auth works on another PBX but not on yours, 90% of the time it’s NOT credentials it’s headers or dialplan flow.

Compare:

  • REGISTER headers
  • INVITE headers
  • From / Contact / Request-URI

Byte-for-byte.


🎯 Key Takeaways for PBX Architects

  • Registration ≠ routing
  • Stasis ≠ magic
  • No Dial() or originate → no call
  • IMS trunks are unforgiving
  • Asterisk does exactly what you tell it nothing more


🚀 Why This Matters

Modern PBXs are no longer just dialplans. They’re distributed telecom systems with:

  • APIs
  • Web dashboards
  • Policy engines
  • Real-time analytics

If you don’t understand where SIP actually leaves the box, you’ll lose days debugging ghosts.


💬 Final Thought

If you’re building:

  • Multi-tenant PBX platforms
  • IMS-based SIP integrations
  • Node.js + Asterisk systems

👉 Design outbound call flow explicitly don’t assume it.

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