VICIdial SIP Troubleshooting: Fix One-Way Audio, 503s, and Registration Failures

You’re staring at the Asterisk CLI. Calls connect but nobody can hear anything. Or worse — they hear you but you can’t hear them. Welcome to SIP troubleshooting.

This isn’t a theoretical guide. Every problem here came from a real VICIdial system we fixed. Packet captures included.

The Debugging Toolkit

Before touching any config, get your tools ready:

# Live SIP packet capture — the single most useful command
sngrep -d eth0 port 5060

# If sngrep isn't installed
tcpdump -i eth0 -n -s 0 port 5060 -w /tmp/sip-capture.pcap

# Check current trunk registration
asterisk -rx "sip show peers"

# Active channels with codec info
asterisk -rx "core show channels verbose"

# RTP debug — shows media flow in real time
asterisk -rx "rtp set debug on"

sngrep is the real MVP here. It shows the full SIP ladder diagram — INVITE, 100 Trying, 180 Ringing, 200 OK, ACK, BYE — in a visual flow. Install it: yum install sngrep or apt install sngrep.

Problem 1: One-Way Audio

This is the most common SIP issue in VICIdial deployments, and it’s almost always NAT.

Symptoms:

• Agent hears the customer, customer hears silence (or vice versa)
• Works fine on internal calls, breaks on outbound through the carrier
• Intermittent — some calls fine, others one-way

Root cause: The SDP body in the INVITE contains a private IP (10.x, 172.16.x, 192.168.x) instead of the public IP. The carrier sends RTP media to the private IP, which is unreachable.

Check it:

# Look at the SDP in sngrep — find the line starting with "c="
# BAD: c=IN IP4 10.0.0.50
# GOOD: c=IN IP4 203.0.113.50

# Also check the "o=" line for the same issue

Fix in sip.conf:

[general]
externaddr=203.0.113.50       ; Your public IP
localnet=10.0.0.0/8           ; Your private network
localnet=172.16.0.0/12
localnet=192.168.0.0/16
nat=force_rport,comedia        ; Force symmetric RTP

If your public IP is dynamic, use externhost instead:

externhost=sip.yourdomain.com
externrefresh=60

The other cause: Firewall blocking RTP ports. VICIdial uses UDP ports 10000-20000 for RTP media by default.

# Verify RTP ports are open
iptables -L -n | grep -E "100[0-9]{2}|1[0-9]{4}|20000"

# If using firewalld
firewall-cmd --list-ports | grep udp

Open them:

iptables -A INPUT -p udp --dport 10000:20000 -j ACCEPT

Problem 2: 503 Service Unavailable

Symptoms:

• Calls fail immediately with 503
• sip show peers shows the trunk as “UNREACHABLE” or “UNKNOWN”
• Happens suddenly after working fine

Common causes:

1. Carrier is down or overloaded. Check their status page. This is more common than you’d think.

2. You hit the channel limit. Most SIP trunks have a concurrent call limit. Check:

# Count active channels on a specific trunk
asterisk -rx "core show channels" | grep -c "SIP/your-trunk"

3. Registration expired. If your trunk requires registration:

# Check registration status
asterisk -rx "sip show registry"

# Force re-register
asterisk -rx "sip reload"

4. DNS failure. If the trunk uses a hostname, DNS resolution might have failed:

# Test DNS resolution
dig sip.carrier.com

# Check if Asterisk cached a stale IP
asterisk -rx "sip show peer your-trunk"

Quick fix — implement failover trunks in your VICIdial campaign dial plan:

# In Admin > System Settings > Dialplan
# Primary trunk
exten => _NXXNXXXXXX,1,Dial(SIP/${EXTEN}@primary-trunk,60,tT)
# Failover if primary returns 503
exten => _NXXNXXXXXX,2,Dial(SIP/${EXTEN}@backup-trunk,60,tT)

Problem 3: Registration Failures

Symptoms:

• sip show registry shows “Rejected” or “Request Sent” that never resolves
• New trunk just won’t connect

Debug it step by step:

# Enable SIP debug
asterisk -rx "sip set debug on"

# Watch the CLI — you'll see the full REGISTER exchange
# Look for:
# - 401 Unauthorized (credentials wrong)
# - 403 Forbidden (IP not whitelisted)
# - 407 Proxy Authentication Required (wrong auth type)

401 Unauthorized fix:

Triple-check your credentials. The most common mistake is whitespace in the password field in sip.conf:

; WRONG — trailing space after password
secret=MyP@ssw0rd

; RIGHT
secret=MyP@ssw0rd

Also check username vs defaultuser. Some carriers want the account number in defaultuser, not username:

[carrier-trunk]
type=peer
host=sip.carrier.com
defaultuser=12345          ; Account number goes here
secret=MyP@ssw0rd
fromuser=12345             ; Some carriers need this too
fromdomain=sip.carrier.com ; And this

403 Forbidden fix:

Your carrier needs to whitelist your IP. If you recently changed servers or IPs, contact them. Some carriers auto-whitelist on first successful registration, others require manual provisioning.

Problem 4: Oualysis Timeouts

If your carrier is returning audio but VICIdial’s answering machine detection or call routing takes too long, you’ll see calls drop during the analysis phase.

# Check oualysis timing in the Asterisk CLI
# Look for "AMD" entries with timing data
asterisk -rx "core show channels verbose" | grep AMD

This one is deep enough to warrant its own post. See our AMD configuration guide for the full breakdown.

Problem 5: Codec Mismatch

Symptoms:

• Calls connect but audio sounds garbled, robotic, or has artifacts
• “No compatible codecs” in Asterisk CLI

Check what codecs your carrier supports:

asterisk -rx "sip show peer your-trunk" | grep -A5 "Codecs"

Fix in sip.conf:

[carrier-trunk]
; Allow the codecs your carrier supports, in order of preference
allow=!all
allow=ulaw        ; G.711 ulaw — best quality, 87 kbps
allow=alaw        ; G.711 alaw — same quality, EU standard
allow=g729        ; G.729 — compressed, 8 kbps, needs license

G.711 (ulaw) is the safe default for domestic US/Canada calling. Only use g729 if bandwidth is a real constraint — it adds latency and requires purchasing codec licenses for Asterisk.

The Nuclear Option: Start From Scratch

If nothing above fixes it, sometimes the fastest path is a clean SIP trace from scratch:

# 1. Clear all state
asterisk -rx "sip reload"

# 2. Start fresh capture
sngrep -d eth0 port 5060

# 3. Make one test call and watch the entire flow

# 4. Look for the FIRST thing that goes wrong
#    - INVITE sent but no response? Firewall/routing.
#    - 100 Trying received but no 180? Carrier routing issue.
#    - 200 OK received but no audio? NAT/RTP issue.
#    - Audio works for 30 seconds then dies? Session timer.

When to Call for Help

If you’ve been staring at packet captures for more than 2 hours and the problem isn’t obvious, it’s usually something environmental — an ISP middlebox doing SIP ALG, a firewall with stateful UDP tracking, or a carrier-side misconfiguration.

We’ve fixed hundreds of these at ViciStack. The pattern is usually the same: 80% of SIP problems are NAT or firewall, 15% are credentials/registration, and 5% are actually weird. The weird ones are where things get interesting.

Need help debugging SIP issues on your VICIdial system? That’s literally what we do — vicistack.com can usually pinpoint the problem in under 30 minutes.