Asterisk Troubleshooting: SIP Debugging Intro (Latest Versions)

SIP debugging is one of the most important skills for anyone working with Asterisk (modern PJSIP-based systems).

Many real telecom failures are not caused by dialplan logic, but by SIP signaling errors, authentication failures, codec negotiation issues, or NAT problems.

This article introduces a clear, practical approach to debugging SIP step-by-step, so problems can be diagnosed quickly instead of guessed.

What SIP Debugging Actually Means

SIP debugging means reading and understanding:

• SIP messages exchanged during registration or calls
• Authentication challenges (401 / 403)
• Codec negotiation inside SDP
• Call routing and response codes

Once you can read SIP logs, most Asterisk problems become predictable.

First Rule of SIP Troubleshooting

Always determine:

• Did the SIP request reach Asterisk?
• Did Asterisk match it to an endpoint?
• Did the call reach the dialplan?
• Did RTP audio start flowing?

These four checkpoints isolate nearly every issue.

Essential Debug Commands in Modern Asterisk (PJSIP)

asterisk -rvvv
pjsip set logger on
pjsip show endpoints
pjsip show registrations
pjsip show contacts

These commands reveal:

• Incoming/outgoing SIP messages
• Registration status
• Endpoint matching
• Contact IP/port information

Understanding SIP Response Codes Quickly

• 100 Trying → request received
• 180 Ringing → destination ringing
• 200 OK → call answered
• 401 Unauthorized → auth challenge
• 403 Forbidden → auth rejected
• 404 Not Found → endpoint/dialplan missing
• 486 Busy Here → extension busy
• 488 Not Acceptable Here → codec mismatch
• 603 Decline → call rejected

Memorizing common SIP codes speeds up troubleshooting dramatically.

Debugging Registration Problems

If phones or trunks cannot register:

• Check credentials in auth section
• Confirm correct server IP/port
• Verify firewall is open

Useful command:

pjsip show registrations

Look for: Registered / Rejected / Timeout.

Debugging Inbound Call Failures

If provider calls never reach dialplan:

• Enable SIP logger and confirm INVITE arrives
• Check identify IP match in PJSIP
• Verify endpoint context routing
• Confirm firewall allows SIP port

Missing identify rule is one of the most common causes.

Debugging Outbound Call Failures

• Watch INVITE sent to provider
• Check response code (403, 404, 488, etc.)
• Verify caller ID format
• Confirm trunk authentication

When SIP Works but Audio Fails

This means signaling is correct but RTP is broken.

• NAT misconfiguration
• Firewall blocking RTP port range
• Wrong external_media_address

SIP debugging alone is not enough — RTP analysis may be required.

Reading SIP Logs Without Panic

SIP logs look complex, but focus only on:

• INVITE → did it arrive?
• Response code → what failed?
• SDP → which codec/IP/port chosen?
• BYE → who ended the call?

Ignore extra headers until needed.

Common Beginner Debugging Mistakes

• Changing dialplan without checking SIP logs
• Ignoring firewall/NAT issues
• Confusing codec problems with SIP failures
• Using old sip set debug instead of PJSIP logger

Simple SIP Troubleshooting Flow (Real-World Method)

1. Enable SIP logger
2. Check registration status
3. Confirm INVITE arrives
4. Check response code
5. Verify RTP audio

Following this order prevents random guessing.

Key Takeaway

SIP debugging is about reading call signaling step-by-step, not blindly changing configuration.

By mastering:

• PJSIP logger output
• SIP response codes
• Endpoint/identify matching
• RTP vs SIP separation

you gain the ability to diagnose nearly any Asterisk call failure quickly and confidently.