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.

If you're exploring similar solutions or want to implement this in your business, I’m open to a quick discussion. Let’s evaluate your use case and identify the most efficient approach to get results.