Asterisk PJSIP Tips and Tricks

Multiple Devices per Extension

To ring multiple devices on a single extension, increase max_contacts on the AOR and use PJSIP_DIAL_CONTACTS in your dialplan.

pjsip.conf

[6001_Aors]
type = aor
max_contacts = 2
qualify_frequency = 50

extensions.conf

; Old (rings only the last registered device):
exten => _6XXX,1,Dial(PJSIP/${EXTEN})

; New (rings ALL registered devices simultaneously):
exten => _6XXX,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})})

PJSIP_DIAL_CONTACTS expands to a comma-separated list of all registered contacts for that endpoint, so every device rings.

Useful CLI Commands

These Asterisk CLI commands are essential for debugging PJSIP trunk and endpoint issues.

Registration & Trunk Status

# Check trunk registration status
asterisk -rx 'pjsip show registrations'

# Show all configured endpoints
asterisk -rx 'pjsip show endpoints'

# Detailed view of a specific endpoint
asterisk -rx 'pjsip show endpoint Simtex_Endpoint'

# Show AOR contact bindings (which devices are registered)
asterisk -rx 'pjsip show aors'

# Check a specific AOR's contacts
asterisk -rx 'pjsip show aor 6001_Aors'

Live Debugging

# Watch SIP traffic in real time (signalling only)
asterisk -rx 'pjsip set logger on'

# Turn it off when done (very verbose)
asterisk -rx 'pjsip set logger off'

# Show active channels / calls
asterisk -rx 'core show channels'

# Show active calls with detail
asterisk -rx 'core show calls'

# Reload PJSIP config without restart
asterisk -rx 'pjsip reload'

Common Troubleshooting

Trunk won't register

# Check registration status and error
asterisk -rx 'pjsip show registrations'

# Look for auth failures in the log
asterisk -rx 'pjsip set logger on'
# Then trigger a re-register:
asterisk -rx 'pjsip send unregister Simtex'
asterisk -rx 'pjsip send register Simtex'

Common causes: wrong credentials, firewall blocking outbound SIP, DNS resolution failure.

One-way audio

Usually a NAT issue. Verify your transport config includes:

[Transport-TCP]
type = transport
protocol = tcp
bind = 0.0.0.0
local_net = 192.168.0.0/16
external_media_address = YOUR.PUBLIC.IP
external_signaling_address = YOUR.PUBLIC.IP

No audio at all

# Check RTP port range is open
asterisk -rx 'rtp show settings'

# Verify codec negotiation
asterisk -rx 'pjsip show endpoint Simtex_Endpoint' | grep allow

Simtex Portal — Live Endpoint Monitoring

In addition to Asterisk CLI tools, the Simtex Customer Portal provides real-time visibility into your SIP registrations without needing SSH access to your PBX.

Subscriptions Page

Navigate to Subscriptions in the portal to see:

• Endpoint status indicators — green (registered), red (unregistered), grey (unknown) for every SIP endpoint on your account
• Endpoint filtering — search and filter endpoints to quickly find specific devices
• Registration data — view the registered contact URI, user agent, and last registration timestamp

SIP Events Timeline

Click into an endpoint to access the SIP Events Timeline — a chronological view of registration and unregistration events. Useful for:

• Confirming whether a phone has registered recently
• Diagnosing intermittent registration drops
• Verifying failover behaviour after network changes
• Correlating phone issues with network events

Caller ID Tips

Set outbound caller ID per trunk

[Simtex_Endpoint]
type = endpoint
from_user = 214XXXXXXX
send_pai = yes
send_rpid = yes
trust_id_inbound = yes

Override caller ID per call in dialplan

; Set caller ID before dialling out
exten => _0.,1,Set(CALLERID(num)=61894883344)
 same => n,Dial(PJSIP/${EXTEN}@Simtex_Endpoint)

Related

PJSIP Configuration on Asterisk — full trunk setup guide