Post-Installation Configuration Guide This guide assumes:
- Asterisk 22 LTS is already installed
- OS: Linux (Ubuntu / Debian / Rocky / Alma / CentOS)
- You are using PJSIP only (required in 22+)
1️⃣ Directory Structure Overview (Know This First)
Important Asterisk paths:
/etc/asterisk/ → Configuration files
/var/lib/asterisk/ → Sounds, MOH, database
/var/log/asterisk/ → Logs
/var/spool/asterisk/ → Voicemail, queues, recordings
/usr/lib/asterisk/ → Modules
2️⃣ Core Configuration Files (What Each One Does)
| File | Purpose |
|---|---|
asterisk.conf | Main system paths & options |
modules.conf | Load/unload modules |
logger.conf | Logging |
pjsip.conf | SIP endpoints (phones, trunks) |
extensions.conf | Dialplan (call logic) |
queues.conf | Call queues |
voicemail.conf | Voicemail |
features.conf | Call features |
musiconhold.conf | MOH |
http.conf | HTTP / REST |
rtp.conf | Media |
3️⃣ asterisk.conf (Core Engine)
📄 /etc/asterisk/asterisk.conf
[directories]
astetcdir => /etc/asterisk
astvarlibdir => /var/lib/asterisk
astlogdir => /var/log/asterisk
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astmoddir => /usr/lib/asterisk/modules
[options]
documentation_language = en_US
verbose = 3
debug = 0
✅ Usually minimal changes needed.
4️⃣ modules.conf (Critical for Asterisk 22)
📄 /etc/asterisk/modules.conf
[modules]
autoload=yes
noload => chan_sip.so
noload => chan_iax2.so
🚫 chan_sip must NOT be loaded (removed in 22).
5️⃣ logger.conf (Logging & Debugging)
📄 /etc/asterisk/logger.conf
[general]
dateformat=%F %T
[logfiles]
console => notice,warning,error
messages => notice,warning,error
security => security
full => debug,verbose,notice,warning,error
📍 Logs live in:
/var/log/asterisk/
6️⃣ pjsip.conf (MOST IMPORTANT FILE)
6.1 Global Settings
[global]
type=global
user_agent=Asterisk22PBX
6.2 Transport (UDP Example)
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
6.3 Endpoint Template (Best Practice)
[endpoint-template]
type=endpoint
context=from-internal
disallow=all
allow=ulaw,alaw
direct_media=no
6.4 Authentication Template
[auth-template]
type=auth
auth_type=userpass
6.5 AOR Template
[aor-template]
type=aor
max_contacts=1
remove_existing=yes
6.6 Create an Extension (Phone 1001)
[1001]
type=endpoint
endpoint_template=endpoint-template
auth=1001-auth
aors=1001-aor
[1001-auth]
type=auth
auth_type=userpass
username=1001
password=StrongPassword123
[1001-aor]
type=aor
max_contacts=1
📞 Register SIP phone using:
- Username:
1001 - Password:
StrongPassword123 - Server: PBX IP
7️⃣ extensions.conf (Dialplan – Call Logic)
📄 /etc/asterisk/extensions.conf
7.1 Internal Calls
[from-internal]
exten => _1XXX,1,NoOp(Internal Call)
same => n,Dial(PJSIP/${EXTEN},20)
same => n,Voicemail(${EXTEN}@default)
same => n,Hangup()
7.2 Outbound Calls
exten => _9X.,1,NoOp(Outbound Call)
same => n,Set(CALLERID(num)=1001)
same => n,Dial(PJSIP/${EXTEN:1}@siptrunk)
same => n,Hangup()
8️⃣ voicemail.conf
📄 /etc/asterisk/voicemail.conf
[default]
1001 => 1234,User One,user1@example.com
1002 => 1234,User Two,user2@example.com
Check voicemail:
*97
9️⃣ queues.conf (Call Center Example)
📄 /etc/asterisk/queues.conf
[support]
musicclass=default
strategy=ringall
timeout=15
retry=5
maxlen=10
member => PJSIP/1001
member => PJSIP/1002
Dialplan:
exten => 600,1,Queue(support)
🔟 musiconhold.conf
📄 /etc/asterisk/musiconhold.conf
[default]
mode=files
directory=/var/lib/asterisk/moh
1️⃣1️⃣ rtp.conf (Audio Issues Fix)
📄 /etc/asterisk/rtp.conf
[general]
rtpstart=10000
rtpend=20000
🔥 Open these ports in firewall.
1️⃣2️⃣ http.conf (Optional – APIs)
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
Used for:
- ARI
- WebRTC
- REST control
1️⃣3️⃣ Reload & Verify
asterisk -rvvv
core reload
pjsip reload
dialplan reload
Check:
pjsip show endpoints
core show channels
1️⃣4️⃣ Security Checklist (VERY IMPORTANT)
✅ Firewall:
5060/UDP (SIP)
10000-20000/UDP (RTP)
✅ Fail2Ban
✅ Strong SIP passwords
✅ No anonymous calls
✅ Never expose AMI publicly
1️⃣5️⃣ Common Post-Install Tests
| Test | Command |
|---|---|
| SIP Registered | pjsip show endpoints |
| Calls Active | core show channels |
| Audio | rtp set debug on |
| SIP Debug | pjsip set logger on |
| Logs | /var/log/asterisk/full |
✅ Final Best Practices
- Use templates in
pjsip.conf - Reload instead of restart
- Backup
/etc/asterisk - Keep Asterisk updated
- Document your dialplan
